| 00:00.38 | mishehu | jbot: A Mog is half man, half dog. A mog is his own best friend. The most famous Mog is, of course, Barf. |
| 00:00.39 | jbot | okay, mishehu |
| 00:00.39 | KalD|Work | how do I strip the last digit in a number if it is a # ? |
| 00:00.43 | *** join/#asterisk RoyK (~roy@83.80-203-29.nextgentel.com) |
| 00:00.54 | Beirdo | heheh |
| 00:00.58 | Nugget | Cows are cool. |
| 00:01.10 | JerJer[mobile] | show application striplsd |
| 00:01.13 | mishehu | "Who are you?" 'I'm Barf' "Not in here, this is a Mercedes" |
| 00:01.23 | KalD|Work | JerJer[mobile], doh! forgot about lsd =) |
| 00:01.30 | KalD|Work | erm StripLsd =) |
| 00:01.34 | mishehu | JerJer[mobile]: strip the lsd? don't you want to just lick the stamps soaked in it? |
| 00:01.40 | Nugget | exten => woo,1,DropLSD(IAX/JerJer) |
| 00:02.04 | jontow | exten => woo,2,Goto(default,woo,1) |
| 00:02.20 | SleepyCow | I know about the dogcow |
| 00:02.24 | RoyK | ~lart JerJer[mobile] |
| 00:02.25 | SleepyCow | I owned a Mac 512e |
| 00:02.32 | SleepyCow | i ate hypercard |
| 00:02.42 | *** join/#asterisk syslod (~yurplsl@65.114.15.70) |
| 00:02.49 | SleepyCow | but can someone tell me if a 4port for fxs card will give me improved audio quality |
| 00:02.57 | SleepyCow | or should i go with the cheaper ethernet bridges |
| 00:03.00 | ManxPower | SleepyCow, improved 0ver WHAT? |
| 00:03.03 | marlowe | lol |
| 00:03.04 | SleepyCow | (where quality is essential) |
| 00:03.11 | *** join/#asterisk MikeJ[Jayden] (~ircatjerr@pcp02795302pcs.roylok01.mi.comcast.net) |
| 00:03.16 | Nugget | I remember when hypercard came out, I thought it was the dumbest thing I'd ever seen. Now, 20 years later, I look at the web and realize what a myopic fucktard I was. |
| 00:03.19 | SleepyCow | you know, the boxes, with ethernet port and rj11 poer |
| 00:04.01 | jontow | sleepycow; i think the device wouldn't matter that much as long as its decent quality.. vs. the codec you use? i don't have enough experience in the area though |
| 00:04.09 | opus___ | is asterisk boot camp worth it? |
| 00:04.23 | marlowe | Go to my asterisk boot camp |
| 00:04.34 | ManxPower | Nugget, For a long time I thought the Web was a "passing fad" I even had a Yahoo like search engine, but stoped maintaining it since I knew the web would never get anywhere. |
| 00:04.38 | marlowe | Shit, I'll take money to teach asterisk. |
| 00:04.53 | opus___ | how do you make money now |
| 00:05.11 | marlowe | I own a small ISP |
| 00:05.17 | SleepyCow | Anyone know if the sipura hardphones have a voicemail waiting idicator? |
| 00:05.19 | opus___ | oh. i use a fancy printer |
| 00:05.33 | marlowe | nice, I like the legal way. |
| 00:05.45 | opus___ | j/k |
| 00:06.43 | opus___ | so has anyone taken apart an iaxy before and figured out whats in it? |
| 00:07.16 | ManxPower | opus___, No. I find that buring $100 bills is much less hassle. |
| 00:07.36 | *** join/#asterisk adker (~adker@67-136-216-215.dsl1.glv.ny.frontiernet.net) |
| 00:07.49 | opus___ | can asterisk register as a mgcp client? |
| 00:08.46 | opus___ | Can Asterisk register as a MGCP client on a remote MGCP service?: |
| 00:08.46 | opus___ | The MGCP implementation in Asterisk is a CallAgent, not a UserAgent. |
| 00:08.52 | opus___ | Ohhhh that makes a lot of sense. |
| 00:08.53 | opus___ | (???) |
| 00:09.36 | *** join/#asterisk abatista (~Ariel@dsl-20-177.cofs.net) |
| 00:10.13 | ManxPower | . |
| 00:10.23 | *** join/#asterisk Goshen (~Goshen@70-57-80-147.slkc.qwest.net) |
| 00:10.35 | syslod | Anyone else having problems with PRI in cvs HEAD? |
| 00:12.32 | ariel_ | Hello everyone. |
| 00:12.47 | syslod | Sup |
| 00:13.02 | ariel_ | syslod, I don't use head at all. |
| 00:13.29 | syslod | What is the tag U have werking now? |
| 00:14.10 | ScythelX | hello all I'm setting up voicemail to read messages from the database in blob format - I have the table setup now do I edit extconfig.conf to point asterisk to start using the database to store its information? odbcstorage => odbc,mysql1,voicemessages isnt working |
| 00:14.25 | fishhead | so how many death threats have you guys got from telco's ? |
| 00:14.26 | fishhead | :P |
| 00:14.38 | Rez | fishhead: don't troll. |
| 00:14.51 | fishhead | no I was serious |
| 00:15.05 | fishhead | rez btw did you manage to find a webpal ? |
| 00:15.09 | fishhead | err juicebox |
| 00:15.13 | fishhead | sorry wrong embedded linux box |
| 00:15.16 | fishhead | :P |
| 00:16.01 | fishhead | rez sometimes I wish I was joking about stuff like that |
| 00:16.02 | Rez | haven't found one for cheap |
| 00:16.05 | *** join/#asterisk santiago (~santiago@63.245.86.95) |
| 00:16.10 | fishhead | well it's walmart only |
| 00:16.19 | fishhead | that they are cheap but the rumor is the toy is a dude |
| 00:16.21 | fishhead | dud |
| 00:16.27 | fishhead | so it should shoon be cheap everywhere |
| 00:16.31 | syslod | Anyone know what this is? zt_pri_error: PRI: Call Reference |
| 00:16.31 | syslod | <PROTECTED> |
| 00:16.42 | *** join/#asterisk lethargic (~lethargic@ool-18bee80e.dyn.optonline.net) |
| 00:17.19 | KalD|Work | GotoIf(${EXTEN}=911?10:100) and I dial 911 always puts me to 100... is my = wrong? |
| 00:17.37 | fishhead | keep up the good work folks :) |
| 00:18.49 | fishhead | rez as always you are welcome to idle in #pixterdev it gets very lively at times |
| 00:18.50 | fishhead | later |
| 00:19.03 | fishhead | that is the juicebox chan |
| 00:19.48 | Darwin35 | ok back |
| 00:19.52 | Darwin35 | with dinner in hand |
| 00:22.53 | *** join/#asterisk FaithX (~Faithful@203.152.246.178) |
| 00:22.54 | iceyp | how do i get asterisk to monitor a peer? i.e. NuFone 66.225.202.72 (S) 255.255.255.255 4569 Unmonitored |
| 00:23.08 | iceyp | I want to know the latency like it shows for users connected |
| 00:23.18 | ManxPower | KalD|Work, you need a space around the = |
| 00:23.29 | *** join/#asterisk cbachman (~chatzilla@victory.ece.northwestern.edu) |
| 00:24.39 | *** join/#asterisk HeppyCat (~unknown@cpe-24-164-217-41.jam.res.rr.com) |
| 00:24.53 | HeppyCat | good evening |
| 00:25.35 | ManxPower | iceyp, qualify=yes |
| 00:27.17 | Lethargicclown | would it be better to use the cd that automatically installs *, or compile it on an existing linix installation |
| 00:27.18 | Lethargicclown | ? |
| 00:27.28 | Lethargicclown | linux that is |
| 00:27.56 | KalD|Work | ManxPower, Read(ACCOUNT,,0,,,) |
| 00:27.57 | KalD|Work | <PROTECTED> |
| 00:28.49 | Darwin35 | why did they add Bri as a main part of * |
| 00:29.01 | Darwin35 | unless you have the card you should not neeed it |
| 00:29.34 | iceyp | thanks ManxPower |
| 00:30.40 | Darwin35 | BRi is for wich cards ? |
| 00:31.05 | *** part/#asterisk dca[laptop] (~dca[lapto@sta-207-174-139-178.rockynet.com) |
| 00:32.59 | ManxPower | KalD|Work, Read(ACCOUNT) |
| 00:33.12 | fishhead | "water babies" by william orbit is freaking awesome ambient tune, I would love to dream to this one |
| 00:33.24 | Nugget | Strange Cargo III is a kickass album. |
| 00:33.55 | Nugget | his other stuff is good, but not as good as. |
| 00:34.03 | Nugget | (as strange cargo iii, rather) |
| 00:34.07 | ManxPower | Darwin35, BRI is ISDN BRI is ISDN 2B+D |
| 00:34.52 | Darwin35 | ok hmm wy are they making it a main part now in the ports tree |
| 00:34.56 | Darwin35 | makes no sense |
| 00:35.00 | ManxPower | Darwin35, BRI is not part of the official Asterisk source. |
| 00:35.35 | ManxPower | How people fuck with it for "ports" is anyone's guess. |
| 00:39.37 | opus___ | whats BRI |
| 00:40.00 | sivana | ~bri |
| 00:40.01 | jbot | methinks bri is the Basic Rate Interface , an ISDN access interface type composed of two B-channels each at 64 kbps and one D-channel at 16kbps (2B+D). |
| 00:40.11 | *** join/#asterisk soundguy (~soundguy@zeus.soundguy.id.au) |
| 00:40.12 | *** join/#asterisk zignig (~simon@203.217.15.10) |
| 00:40.14 | opus___ | oh .. duh |
| 00:40.36 | opus___ | in the future everyone will have high speed isdn connections and will use video phones |
| 00:40.58 | opus___ | and this will be brought to you by at&t |
| 00:40.59 | ManxPower | opus___, You time-traveled from the mid/late 1980's????? |
| 00:41.03 | sivana | in the future, I'll have a neural link to my mainframe :P |
| 00:41.21 | sivana | the size of my wrist watch |
| 00:41.26 | opus___ | haha |
| 00:42.03 | opus___ | yeahn, and only john mnemonic can do double space ( 8 gigs) |
| 00:43.38 | Darwin35 | grrr 1.0.6 needs some major issues |
| 00:43.55 | fishhead | it needs issues? |
| 00:44.00 | fishhead | what are you a liberal democrat? |
| 00:44.05 | Darwin35 | has |
| 00:44.07 | Darwin35 | sorry |
| 00:44.14 | HeppyCat | so, ive been looking all over the asterisk and digium sites, but can't find the hardware requirements |
| 00:44.24 | HeppyCat | all ive found is pbx specific stuff |
| 00:44.30 | HeppyCat | what kind of machine does this need to run? |
| 00:44.30 | zignig | fishhead: heh |
| 00:44.55 | ManxPower | HeppyCat, That's because Asterisk's hardware requirements vary depending on what you want to do with it. |
| 00:44.57 | zignig | fishhead: or you could be Rep. and need _less_ issues. ;P |
| 00:45.01 | fishhead | heheh |
| 00:45.05 | opus___ | pentium III 800mhz 512mb ram |
| 00:45.13 | Darwin35 | grr ok I am closer |
| 00:45.19 | opus___ | its under there small business basic plan |
| 00:45.21 | Darwin35 | but this bri is not linking |
| 00:45.27 | Darwin35 | patching |
| 00:45.50 | zignig | has anyone else tried destar ( config system ) ? |
| 00:46.04 | *** join/#asterisk syslod (~yurplsl@65.114.15.70) |
| 00:46.42 | Darwin35 | ok is bri going to be migrated into the base |
| 00:48.38 | opus___ | hmmm, when one installs mysql and tries to login it will ask for a password by passing -p |
| 00:48.55 | opus___ | curious, does anyone know this password |
| 00:48.57 | ManxPower | Darwin35, You realize you are the only person in the channel that actually cares about *BSD ports, right? |
| 00:49.16 | ManxPower | opus___, default for Mandrake is an empty password |
| 00:49.25 | Darwin35 | no there are a few of us |
| 00:49.39 | ManxPower | Why are they not in #asterisk-bsd? |
| 00:49.42 | opus___ | manx - somebody decided to change that and forgot to tell red hat |
| 00:49.46 | *** join/#asterisk Kumbang (~ecvs@167.205.24.4) |
| 00:49.48 | Darwin35 | the channel died |
| 00:50.28 | Darwin35 | bsd users join #asterisk-bsd |
| 00:51.31 | Primer | </crickets> |
| 00:51.33 | HeppyCat | is there an #asterisk-sparc ? |
| 00:53.40 | HeppyCat | haha |
| 00:53.43 | KalD|Work | we need an asterisk-toaster channel |
| 00:54.14 | techie | #asterisk-nintendo |
| 00:54.23 | KalD|Work | ohhh.. that'd be good! |
| 00:54.30 | Silik0n | anyone using audiocodes hardware FXO box? |
| 00:54.32 | ManxPower | .join #asterisk-stoners |
| 00:54.33 | ManxPower | oops |
| 00:54.58 | ManxPower | ~google site:lists.digium.com audiocodes |
| 00:55.03 | KalD|Work | ManxPower, damn... I'm the only one there |
| 00:55.18 | Silik0n | yeah I'm looking at that now |
| 01:02.05 | harryvv | my ata is trying to register with * after i did a password change. Now its is not able to register. Got this in sip debug but dont have a nat between my system and the ata. I have firewall disabled. (no NAT) to 192.168.10.20:5060 |
| 01:03.07 | harryvv | keeps flashing the ata registration contents every so many seconds across the screen. |
| 01:04.41 | *** join/#asterisk hermie (~nick@24.236.167.53.bay.mi.chartermi.net) |
| 01:06.55 | ManxPower | WHOO!! WHOO!!! http://www.sipura.com/Documents/SPA841AdminGuide0105-DRAFT.pdf |
| 01:06.57 | harryvv | Strange. I put nat from no to yes now its registering. I guess it has something to do with the firewall in the server thats causing a issue even though its off. |
| 01:07.10 | opus___ | whats that |
| 01:07.40 | harryvv | manx, I dont need to see that it was a previosly working system. |
| 01:08.15 | ManxPower | harryvv, That was for anyone that cared, not for you. |
| 01:08.22 | harryvv | okay |
| 01:09.02 | harryvv | Do you know what issue in fedc3 would block this registration? fw is off |
| 01:09.41 | *** join/#asterisk _mwoodj_ (~MWoodJ@hyper-eye.digium.sponsor.pdpc) |
| 01:11.39 | ManxPower | harryvv, I still don't see a paste of the registration failure (NOT sip debug) |
| 01:12.39 | *** join/#asterisk Damin_Mobile (~pocketirc@70.sub-70-214-10.myvzw.com) |
| 01:13.17 | ManxPower | harryvv, If all you did was change the password, then there's your problem. Maybe the device didn't get the updated password applied, maybe there was a caps lock problem, maybe sip.conf didn't get reloaded. |
| 01:14.22 | *** join/#asterisk Nickle (~nospam@bdsl.66.14.154.88.gte.net) |
| 01:14.29 | opus___ | is app_dbodbc.c built in to cvs head? |
| 01:14.35 | *** join/#asterisk soulz- (~soulz@host-137-132-45-121.imcb.nus.edu.sg) |
| 01:14.42 | soulz- | hello all |
| 01:14.53 | *** join/#asterisk TheBear (~brif8@lazyjtrainingcenter.com) |
| 01:15.04 | soulz- | does the current cvs have any bugs CVS-HEAD-03/07/05-17:14:42 |
| 01:15.05 | soulz- | ? |
| 01:15.22 | soulz- | Unable to create channel of type 'SIP' (cause 3) ?? |
| 01:15.59 | TheBear | ok I installed asterisk using gentoo emerge I have version 1.0.5 when I try to start asterisk -vvvgc it fails with "Ouch ... error while writing audio data: : Broken pipe" just after skinny.so |
| 01:16.08 | *** join/#asterisk aminorex (~tony@c66.191.69.132.dul.mn.charter.com) |
| 01:16.16 | soulz- | anyone please? |
| 01:16.17 | harryvv | that was not the problem. I changed operating systems when my original astrisk failed two night ago and now have fedc3 running. I did a reload a sip reload reloaded the config on the ata web address and untill i changed the nat from no to yet in the sip.conf it would not register. Soon as I did that it was instant. Both the ata and the * have a common switch between them. |
| 01:16.47 | ManxPower | "The mailing list archives have a VERY long memory. Centuries from now people with extra large brains will see your message in the mailing list archives and spend a day trying to figure out why their Gotos in Asterisk are not working. " |
| 01:17.22 | ManxPower | soulz-, That means Asterisk cannot communicate with your SIP device. Maybe it's not registering. |
| 01:17.37 | harryvv | Anyway it works now :) |
| 01:17.53 | *** join/#asterisk ScaredyCat (~ScaredyCa@j25065.upc-j.chello.nl) |
| 01:18.03 | opus___ | nat=yes in general does magical things that don't make sense to me |
| 01:18.31 | opus___ | if ( broken ) fix++; |
| 01:19.32 | soulz- | manxpower: owl*CLI> sip show registry |
| 01:19.32 | soulz- | Host Username Refresh State |
| 01:19.32 | soulz- | sip.broadvoice.com:5060 2122020832@s 3584 Registered |
| 01:19.38 | soulz- | oops sorry dude |
| 01:19.45 | ManxPower | opus___, Correct. Don't use nat=yes unless your SIP client is behind NAT and your NAT router is not SIP-Aware. |
| 01:19.52 | ManxPower | soulz-, it's "sip show peers" |
| 01:20.07 | ManxPower | "sip show registry" shows what ASTERSIK is registered to |
| 01:20.31 | soulz- | ahh |
| 01:20.32 | soulz- | broadvoice/2122020832 147.135.8.128 255.255.255.255 5060 UNREACHABLE |
| 01:20.37 | soulz- | but its registered |
| 01:20.44 | soulz- | <-- confused |
| 01:20.57 | ManxPower | soulz-, See that unreachable? That means "it's not responding, check your firewall" |
| 01:21.21 | TheBear | ok I took out the musiconhold.conf file (temp) now I still get the failure, but without the error message just [chan_features.so] => (Feature Proxy Channel) Segmentation fault (core dumped) |
| 01:21.21 | ManxPower | soulz-, But your ASTERISK server is behind NAT, so you need externip and localnet set, but I think you need nat=no. |
| 01:21.32 | *** join/#asterisk yxa (~void@203.118.40.42) |
| 01:21.56 | soulz- | ok manxpower, will give that a shot, my asterisk server is on public ip |
| 01:22.09 | ManxPower | soulz-, then you don't have to worry about those options. |
| 01:22.17 | ManxPower | you just need to make sure your firewall is working. |
| 01:22.39 | opus___ | it'd be pretty nice to have a sip_nat_test.sh program |
| 01:22.59 | TheBear | could anyone advice me why ? |
| 01:23.09 | soulz- | i have no firewall |
| 01:23.10 | HeppyCat | its da bear |
| 01:23.10 | ManxPower | TheBear, File a bug report. |
| 01:23.13 | soulz- | emmm |
| 01:23.29 | ManxPower | soulz-, then you have something SERIOUSLY wrong since broadvoice is not responding to SIP packets |
| 01:23.57 | soulz- | manxpower: emm, okay dude, i will investigate further |
| 01:24.03 | TheBear | ManxPower: where |
| 01:24.25 | ManxPower | TheBear, bugs.digium.com, but I would post to the mailing list first to confirm it's actually a bug |
| 01:24.45 | ManxPower | soulz-, traceoute 147.135.8.128 |
| 01:25.17 | soulz- | its okay |
| 01:25.25 | soulz- | 200ms |
| 01:25.37 | soulz- | using mtr |
| 01:25.42 | ManxPower | mtr? |
| 01:25.45 | opus___ | is 200ms to much? |
| 01:25.50 | soulz- | matt traceroute |
| 01:25.52 | ManxPower | opus___, not for this problem |
| 01:26.01 | opus___ | matt's traceroute is awesome |
| 01:26.04 | soulz- | nope, i am in sg |
| 01:26.12 | soulz- | so 200ms is quite normal |
| 01:26.17 | opus___ | really |
| 01:26.28 | opus___ | whats the maximum allowable latency in general? |
| 01:26.28 | TheBear | I'll also try the CVS version, perhaps it's a gentoo ebuild issue |
| 01:26.35 | ManxPower | 200ms is enough that users might notice it. |
| 01:26.40 | HeppyCat | TheBear: fancy seeing you here |
| 01:26.42 | opus___ | oh |
| 01:26.49 | ManxPower | TheBear, You can't report bugs unless you use official Digium source code. |
| 01:27.10 | ManxPower | TheBear, Use cvs co -r v1-0 asterisk asterisk-sounds libpri zaptel |
| 01:27.23 | *** join/#asterisk Godsey (lanny@207-229-102-125.cortland.com) |
| 01:28.17 | soulz- | weird, i can trace, i have tried all their proxies |
| 01:28.21 | tzanger | what do the dev conferences use ? |
| 01:28.28 | tzanger | meetme, meetme2, app_conference, what |
| 01:28.38 | soulz- | iax2 show peers |
| 01:28.42 | soulz- | works |
| 01:29.31 | TheBear | ManxPower: ok |
| 01:29.38 | ManxPower | soulz-, Well are you connecting via SIP or IAX2? |
| 01:29.38 | TheBear | HeppCat: do I know U |
| 01:29.44 | opus___ | If I install mysql, odbc and mysql-odbc, then this sentence is correct? : |
| 01:29.49 | opus___ | When you start to compile Asterisk, the Makefile inside res/ should detect if you have ODBC properly installed and if so compile the res_config_odbc.so module for you. |
| 01:30.05 | opus___ | i just do a make clean && make and I should ihave a res_config_odbc module? |
| 01:30.23 | soulz- | manxpower: via sip, bv doesn't support iax yet |
| 01:30.30 | soulz- | well atleast not that i know off |
| 01:30.49 | TheBear | using kernel 2.6 but NOT udev do I still build zaptel with "make linux26" ? |
| 01:30.50 | opus___ | soulz which peer are you currently using? |
| 01:30.55 | ariel_ | Hello everyone hope you all are doing fine. |
| 01:31.04 | soulz- | i am trying all |
| 01:31.09 | opus___ | some don't work |
| 01:31.09 | soulz- | mia, lax, chi, dca |
| 01:31.11 | HeppyCat | hurmm....guess not |
| 01:31.16 | opus___ | sip.broadvoice.com has always worked for me however |
| 01:31.29 | opus___ | instead of proxy.mia... etc etc try just sip.broadvoice.com |
| 01:31.30 | soulz- | i am rebooting my machine to see if that makes a diff |
| 01:31.45 | opus___ | hey, that fixed windows98 |
| 01:32.27 | soulz- | opus: what is ur register statement for bv? |
| 01:32.33 | soulz- | in sip.conf? |
| 01:32.35 | Godsey | I'm going to order AG-168VE |
| 01:32.43 | harryvv | manx well now its not registering again. |
| 01:32.49 | Godsey | I'm so behind in getting asterisk running properly at home |
| 01:32.57 | *** join/#asterisk mesi (~player@dsl-082-083-179-254.arcor-ip.net) |
| 01:32.58 | *** join/#asterisk brc__ (~brian@brc.base.supporter.pdpc) |
| 01:33.09 | *** part/#asterisk zignig (~simon@203.217.15.10) |
| 01:33.32 | Godsey | I'm derelict in my chatting here too :) |
| 01:33.35 | Godsey | just sold our house |
| 01:33.58 | opus___ | register => 5037150059@sip.broadvoice.com:secret::5037150059@sip.broadvoice.com |
| 01:34.17 | opus___ | stuff:secret:stuff even |
| 01:34.21 | soulz- | ok let me try the same thing |
| 01:34.26 | harryvv | http://pastebin.ca/6989 |
| 01:34.32 | soulz- | but its funny as its registering |
| 01:35.15 | soulz- | opus : is there :: two of these |
| 01:35.18 | soulz- | after secret? |
| 01:35.19 | opus___ | no |
| 01:35.25 | opus___ | my typo |
| 01:35.30 | soulz- | okay thanks |
| 01:35.36 | soulz- | i have the same |
| 01:37.56 | opus___ | that sucks |
| 01:38.04 | opus___ | whats your [sip.broadvoice.com] |
| 01:38.07 | Silik0n | this audiocodes box is pissing me off |
| 01:38.48 | *** join/#asterisk kulp (~kulp@generic-net216-173.mtc.net) |
| 01:39.54 | mesi | When using one of the modem groups, e.g. g1, instead of ttyI1, one of all the lines in the group will be used, right? |
| 01:40.46 | harryvv | sip registration occures across port 5060? |
| 01:40.59 | Silik0n | yes |
| 01:41.03 | harryvv | mmm |
| 01:41.09 | Silik0n | 5060 UDP |
| 01:41.27 | Silik0n | anyone here using audiocodes gear? |
| 01:41.45 | *** join/#asterisk JohnJacob (~JohnJacob@pcp0011542342pcs.mainf01.in.comcast.net) |
| 01:42.05 | harryvv | reload |
| 01:42.14 | *** part/#asterisk Kumbang (~ecvs@167.205.24.4) |
| 01:44.07 | ManxPower | ~google site:lists.digium.com audiocodes |
| 01:44.09 | *** join/#asterisk kks (~kks@203.115.210.253) |
| 01:44.10 | ManxPower | ~google site:lists.digium.com audiocode |
| 01:44.30 | soulz- | manxpower: can u have a look at this |
| 01:44.31 | soulz- | http://pastebin.ca/6990 |
| 01:44.52 | soulz- | the sip debugs to see where the problem could be |
| 01:45.59 | *** join/#asterisk miguellinux (~miguellin@200.47.223.190) |
| 01:46.29 | ManxPower | soulz-, No. Sorry. |
| 01:46.30 | Silik0n | gee thanks ManxPower |
| 01:46.38 | Silik0n | i already looked there and its useless as it lies |
| 01:46.50 | harryvv | sip show registry shows blank. |
| 01:49.18 | *** join/#asterisk FaithX (~Faithful@202.136.108.110) |
| 01:49.34 | *** join/#asterisk RRM (~ryan@stan.inter-intelli.com) |
| 01:49.59 | *** join/#asterisk wdatkinson (~icechat5@pcp986542pcs.northw01.in.comcast.net) |
| 01:54.03 | wdatkinson | Looking for some assistance getting an Audiocodes MP108 to work with Asterisk. I know how to config the MP, and I've had it running with another IP PBX, just missing something with Asterisk. |
| 01:54.22 | opus___ | how do I make sure that res_odbc was built in my asterisk source directory? |
| 01:54.37 | ManxPower | opus___, um, look for res_odbc.so? |
| 01:54.50 | *** join/#asterisk JohnJacob (~JohnJacob@pcp0011542342pcs.mainf01.in.comcast.net) |
| 01:54.52 | opus___ | hmm, doesn't seem to be built |
| 01:55.07 | opus___ | is there a way to force it. wait i'll read the docs i think i saw a makefile change i need to do |
| 01:55.22 | opus___ | :( no ./configure --enable-dumb-dumb-mode=yes |
| 01:57.12 | wdatkinson | Or perhaps someone can provide a few pointers on configuring my SIPTrunk. The MP I'm using is and FXO unit. |
| 01:57.27 | Silik0n | wdat |
| 01:57.38 | Silik0n | wdatkinson hey check your pm's |
| 01:57.57 | wdatkinson | Silik, I'm there now... missed it! |
| 01:58.31 | *** join/#asterisk Tr0j4N (b3nz3r@pcp08761169pcs.mtlrel01.nj.comcast.net) |
| 01:59.15 | jontow | wow.. asterisk-stable wasn't so stable |
| 01:59.21 | jontow | it just made my freebsd gateway deadlock |
| 01:59.37 | jontow | smart me.. shoulda left it at HEAD from 11/23/2004 |
| 01:59.37 | jontow | :( |
| 02:00.26 | opus___ | hmmm res_odbc not building |
| 02:01.24 | opus___ | <PROTECTED> |
| 02:01.24 | opus___ | [root@u2 asterisk-addons]# echo $? |
| 02:01.28 | opus___ | 1 |
| 02:02.35 | RRM | man I hate AudioCodes |
| 02:03.33 | harryvv | very very odd! I changed the nat from yes to no now the ata embeded web site said its registered yet sip show registry shows nothing is registered. That does show when clients need to register with asterisk? |
| 02:05.37 | Mavvie | try sip show peers |
| 02:05.40 | slePP | anyone feel like doing some translation work for me? :> |
| 02:05.57 | *** join/#asterisk Trionnis (buffy@12-203-113-15.client.insightBB.com) |
| 02:06.35 | *** join/#asterisk soulz- (~soulz@host-137-132-45-118.imcb.nus.edu.sg) |
| 02:06.38 | soulz- | back |
| 02:06.38 | opus___ | <PROTECTED> |
| 02:06.38 | opus___ | <PROTECTED> |
| 02:06.38 | opus___ | <PROTECTED> |
| 02:06.39 | *** join/#asterisk brc__ (~brian@brc.base.supporter.pdpc) |
| 02:06.45 | opus___ | ^^^^^ is that the odbc config linker? |
| 02:06.50 | *** mode/#asterisk [+o brc__] by brc_ |
| 02:13.27 | *** join/#asterisk wrzf (~chris@cvg-165-103-153.cinci.rr.com) |
| 02:14.38 | wrzf | Good evening Gents and ladies |
| 02:15.06 | *** join/#asterisk JohnJacob (~JohnJacob@pcp0011542342pcs.mainf01.in.comcast.net) |
| 02:15.41 | Trionnis | evening, yes... good? ehhh.... debateable |
| 02:15.50 | ChrisF0rd | yeah I hear ya |
| 02:16.10 | ChrisF0rd | Seen PCM round |
| 02:16.23 | Trionnis | use ~seen |
| 02:16.31 | Trionnis | I don't recall his complete nick |
| 02:16.33 | ChrisF0rd | ~seen PCM |
| 02:16.34 | jbot | pcm <~pcm@user-69-73-0-22.knology.net> was last seen on IRC in channel #asterisk, 4d 7h 23m 52s ago, saying: 'where ?'. |
| 02:16.52 | Trionnis | oh, I thought you were talking about someone else, apparently |
| 02:16.59 | ChrisF0rd | no that was ity |
| 02:17.01 | ChrisF0rd | it |
| 02:17.03 | Trionnis | ok |
| 02:17.06 | ChrisF0rd | He dissapears like that |
| 02:17.16 | ChrisF0rd | 4d 7h 23m 52s |
| 02:17.22 | ChrisF0rd | since last logon |
| 02:17.26 | *** join/#asterisk tresvon (~drew@adsl-67-38-22-141.dsl.sfldmi.ameritech.net) |
| 02:17.27 | ManxPower | harryvv, "sip show registry" shows things ASTERISK is registered to. |
| 02:17.39 | ManxPower | "sip show peers" will show devices that are registered with asterisk |
| 02:18.17 | tresvon | Anyone know how to get Asterisk to work with FC3? |
| 02:19.21 | ChrisF0rd | I have it running on FC3 * ver 1.0.6 |
| 02:19.35 | ChrisF0rd | I just installed it |
| 02:19.54 | opus___ | data_mysql.c: In function `mysql_vm_change_password': |
| 02:19.54 | opus___ | data_mysql.c:1706: `MYSQL' undeclared (first use in this function) |
| 02:19.57 | ChrisF0rd | but if you have the Xwindows system installed wioth it you will have problems getting it to work |
| 02:20.01 | opus___ | ast_data fucks shit up |
| 02:20.10 | tresvon | i installed it but can't find it. |
| 02:20.24 | opus___ | updatedb && locate asterisk |
| 02:20.25 | Trionnis | ? |
| 02:20.45 | Trionnis | can't find what? |
| 02:20.45 | MikeJ[Jayden] | anyone using broadvoice? |
| 02:20.48 | ChrisF0rd | type /usr/sbin/asterisk -vvvvvgc |
| 02:20.49 | Tr0j4N | tresvon: http://voip-info.org/wiki-Asterisk+Fedora+Core+3 |
| 02:20.51 | Trionnis | I am mike |
| 02:20.55 | iceyp | When linking 2 PBX's together, I choose peer, do i need to register to each? |
| 02:21.00 | MikeJ[Jayden] | how's it working for you? |
| 02:21.00 | soulz- | mike: i am using bv |
| 02:21.08 | soulz- | and having problems |
| 02:21.10 | Tr0j4N | tresvon: I used those instructions and had it up and running in 2 hours |
| 02:21.12 | Trionnis | a couple issues, but nothing like Vonage |
| 02:21.12 | channan | MikeJ- I used bv since Dec 2004 |
| 02:21.12 | tresvon | Thanks Tr... |
| 02:21.17 | Tr0j4N | np |
| 02:21.33 | tresvon | 2hrs? Do you know if it works with Vonage? |
| 02:21.43 | Trionnis | nope |
| 02:21.45 | Trionnis | well |
| 02:21.47 | Trionnis | kinda |
| 02:22.01 | Trionnis | softphone account only, and even then it's tempramental |
| 02:22.03 | Tr0j4N | I know it works with analog phones which is all I'm interested in at the moment |
| 02:22.14 | harryvv | manx okay like iax.cc |
| 02:22.32 | Tr0j4N | short answer, no |
| 02:22.43 | channan | working great so far. can't complain for $20 :) |
| 02:22.48 | Trionnis | yeah, what he said |
| 02:22.50 | Trionnis | :) |
| 02:22.54 | tresvon | :-) |
| 02:23.01 | MikeJ[Jayden] | what kinda issues? I hear issues w/ dtmf passing |
| 02:23.01 | channan | the only thing is I've tried to reach customer service at night and never got thru. |
| 02:23.11 | harryvv | ManxPower, so like my spa1000 info should be registered in sip show registry. |
| 02:23.18 | tresvon | Cool. I have a sft phone acct with hrd phone too. |
| 02:23.26 | harryvv | disregard |
| 02:23.30 | iceyp | Mar 8 15:23:21 NOTICE[73071]: chan_iax2.c:5833 socket_read: Registration of 'ape' rejected: Registration Refused |
| 02:23.31 | harryvv | yea sip show peers |
| 02:23.47 | harryvv | its registered in there and right now its working. |
| 02:24.37 | marlowe | <PROTECTED> |
| 02:24.37 | marlowe | <PROTECTED> |
| 02:24.37 | marlowe | <PROTECTED> |
| 02:24.37 | marlowe | <PROTECTED> |
| 02:24.37 | marlowe | <PROTECTED> |
| 02:24.37 | marlowe | <PROTECTED> |
| 02:24.39 | marlowe | <PROTECTED> |
| 02:24.41 | marlowe | <PROTECTED> |
| 02:24.43 | marlowe | <PROTECTED> |
| 02:24.45 | marlowe | o sit |
| 02:24.47 | marlowe | sorry wrong window |
| 02:24.49 | marlowe | lol |
| 02:25.09 | *** join/#asterisk syslod (~yurplsl@65.114.0.198) |
| 02:25.18 | syslod | Hello |
| 02:25.19 | Trionnis | ~lart marlowe |
| 02:25.23 | marlowe | ! |
| 02:25.24 | channan | * needs brake pads for sure LOL |
| 02:25.25 | Trionnis | => |
| 02:25.45 | marlowe | Brakes are too much |
| 02:25.49 | marlowe | to think theres a kit for 2,500 |
| 02:25.50 | marlowe | sheesh |
| 02:26.05 | syslod | Seems to be problem with PRI and cvs HEAD. Anyone else having problems? |
| 02:26.23 | Trionnis | I have no problems with head whatsoever... I enjoy it quite a bit |
| 02:26.34 | Trionnis | 3 |
| 02:26.35 | Trionnis | 2 |
| 02:26.36 | Trionnis | 1 |
| 02:26.41 | *** part/#asterisk tresvon (~drew@adsl-67-38-22-141.dsl.sfldmi.ameritech.net) |
| 02:26.43 | marlowe | i like head too |
| 02:27.02 | marlowe | That what you were waiting for? |
| 02:27.07 | Trionnis | yeah |
| 02:27.09 | Trionnis | pretty much |
| 02:27.34 | Trionnis | the whole "angelina jolie" effect (ala red v blue) |
| 02:27.39 | marlowe | Speaking of such.. I gotta get ready for tonight |
| 02:27.46 | Trionnis | hah |
| 02:27.50 | harryvv | anyone seen a message of failed to write frame ? |
| 02:28.08 | cbachman | I've finally compiled asterisk for mipsel/WRT, but am getting a segfault when attempting to call in with SIP. Anyone around have any hints/tips/suggestions? |
| 02:28.23 | Trionnis | harryvv: are you sure your zaptel is ok? that's really starting to sound like it could be a hardware problem |
| 02:28.48 | harryvv | Tri, yea its okay ive not had issues with it in the past |
| 02:29.08 | syslod | harryvv: U using HEAD or recent version? |
| 02:29.25 | harryvv | cvs recent |
| 02:29.35 | harryvv | like 2 days ago. |
| 02:29.35 | Trionnis | hey, I installed Fedora on a laptop the other day that picked that *exact* time for the battery to go bad and cause me to spend 3 days "troubleshooting" ACPI |
| 02:29.38 | opus___ | cbachman - cross compile 'strace' and find out where it crashes |
| 02:29.41 | Trionnis | so shit happens, ya know? |
| 02:29.42 | Trionnis | :) |
| 02:30.01 | harryvv | it comes up periodically |
| 02:30.09 | mesi | 1800555tell just rocks ;-) |
| 02:30.10 | MikeJ[Jayden] | can you set outbound callid on broadvoice? |
| 02:30.19 | Trionnis | not that I've found mike |
| 02:30.19 | harryvv | it even happened on the other asterisk box untill the hard drive failed. |
| 02:30.21 | opus___ | cbachman - which compiler set did you use |
| 02:30.27 | Trionnis | if you can, I'd sure like to know how :) |
| 02:30.49 | syslod | harryvv: I see that now too. As well as I can't dial NXX-XXXX it forces NPA-NXX-XXXX and gives an PRI error if you try. |
| 02:30.53 | cbachman | opus___ this is on a 4M flash, space is very tight. I'm not sure I'm loading all the modules necessary. At the moment I'm just trying to get * to hangup as a test. |
| 02:31.25 | opus___ | mesi - what does it run on |
| 02:31.46 | cbachman | opus__ I'm using openwrt, so the same compiler set they use for their firmware. If this works, I'll have a nice easy way to compile it |
| 02:32.19 | opus___ | cbachman - you can nfs mount all the tools you want |
| 02:32.33 | opus___ | i don't think an openwrt can transcode fast enough to be a pbx |
| 02:32.43 | opus___ | although it would be cool if it did:) |
| 02:32.49 | opus___ | openwrt |
| 02:32.54 | opus___ | i ment wrt54g |
| 02:34.39 | cbachman | opus___, show translations is 82 from ulaw->gsm, 21 gsm->ulaw I'm more just looking at using it to shift sip->iax2 |
| 02:35.21 | opus___ | cool |
| 02:38.05 | opus___ | ARM processors don't have floating point registers, i think |
| 02:38.09 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
| 02:38.34 | opus___ | my apple newton has a 20mhz arm processor with 2megs of ram... :) |
| 02:39.09 | JunK-C | what a machine! |
| 02:39.10 | JunK-C | :P |
| 02:39.21 | *** join/#asterisk t3t (~t3t@galley.pangalacticgargleblaster.com) |
| 02:39.27 | opus___ | yeah, now if i can port pearpc to it that'd rock |
| 02:40.06 | Sedorox | think a wrt54g makes a good asterisk box if there is no codec translation? |
| 02:40.16 | epoch | haha |
| 02:40.39 | Sedorox | like as a local endpoint.. persay |
| 02:40.52 | Nugget | "persay" is not a word. |
| 02:40.55 | *** join/#asterisk Moc_ (~Moc@207.253.44.130) |
| 02:40.59 | Nugget | even if it were, it wouldn't belong there. |
| 02:41.00 | Sedorox | lol |
| 02:41.19 | Moc_ | anyone ever saw the problem that when your dialling an extention, it doesn't ring ? |
| 02:41.28 | ManxPower | Moc, Actually, yes. |
| 02:41.38 | Trionnis | noooo! get outta here! |
| 02:41.42 | ManxPower | Moc, you want to show me your's first, or me show you mine first? |
| 02:41.43 | Trionnis | you've seen that?? |
| 02:41.48 | Trionnis | ;) |
| 02:41.55 | ManxPower | the config |
| 02:42.35 | Moc_ | urm ;) you want to see my macro ? |
| 02:43.41 | ManxPower | Moc, I have a SIPura SPA-2000 when calling a Polycom IP 500 FROM LINE 2 on the SPA, the far end phone doesn't ring. Each phone is registered to it's local Asterisk server and the two Asterisk servers talk voa IAX. |
| 02:43.47 | BoRiS | Moc!!!! |
| 02:43.47 | Moc_ | ManxPower: http://pastebin.ca/6992 |
| 02:44.04 | Moc_ | hi boris |
| 02:44.08 | JunK-C | ~seen siftach |
| 02:44.10 | jbot | siftach <~siftach@80.74.108.206> was last seen on IRC in channel #asterisk, 130d 11h 59m 34s ago, saying: 'It seems the problem is in zapata.conf/zaptel.conf'. |
| 02:44.10 | BoRiS | What's up? |
| 02:44.16 | Moc_ | ManxPower, actually, my call is already answered |
| 02:44.48 | Moc_ | and I get the problem using 1 provider, but I dont get it from the other provider |
| 02:45.59 | *** join/#asterisk setjmp (setjmp@ip-wv-66-190-126-051.charterwv.net) |
| 02:47.31 | JunK-C | ~seen paradise |
| 02:47.32 | jbot | paradise <~paradise@n219079205023.netvigator.com> was last seen on IRC in channel #debian, 21d 10h 11m 33s ago, saying: 'takatumi: in xchat'. |
| 02:47.33 | ManxPower | Moc, in my universe the phones don't normally ring unless the phone is idle. |
| 02:50.02 | *** join/#asterisk myconid (~myconid@12.177.133.243) |
| 02:50.05 | ManxPower | blah |
| 02:50.16 | myconid | CVS head wont compile :( |
| 02:50.28 | JunK-C | myconid: which erros? |
| 02:50.29 | myconid | ../include/asterisk/channel.h:290: error: field `varshead' has incomplete type |
| 02:50.40 | myconid | repulling it and cleanly building it just to check |
| 02:51.13 | ManxPower | . |
| 02:52.23 | JunK-C | odd. |
| 02:52.28 | ManxPower | Hmm? |
| 02:53.46 | myconid | I did gmake.. |
| 02:53.49 | myconid | make seems to be working better. |
| 02:53.52 | myconid | whats the difference? |
| 02:54.23 | myconid | did it again |
| 02:54.43 | myconid | vtwireless.com/asterisk.txt |
| 02:55.28 | myconid | <PROTECTED> |
| 02:56.29 | *** join/#asterisk soundguy (~soundguy@zeus.soundguy.id.au) |
| 02:57.43 | rvhi | try to compile astman, but got this |
| 02:57.49 | rvhi | ../include/asterisk/lock.h:302: error: `PTHREAD_MUTEX_RECURSIVE' undeclared (first use in this funct |
| 02:57.58 | Trepalium | Ow. Always remember to remove Milliwatt() test lines from extensions.conf before dialing the number and putting the phone close to your ear. |
| 02:57.59 | rvhi | what might be wrong? |
| 02:58.26 | ManxPower | No animal should ever jump on the dining room furniture unless absolutely certain he can hold his own in conversation. -- Fran Lebowitz |
| 03:01.50 | BoRiS | rvhi: Compile in the root asterisk directory |
| 03:06.21 | *** join/#asterisk atmel (~vlad@ip68-4-101-199.oc.oc.cox.net) |
| 03:11.53 | *** join/#asterisk Nukemizer (~Nuke@66.237.85.58.ptr.us.xo.net) |
| 03:11.57 | *** join/#asterisk WilliamK (~wkeller@c-24-0-130-177.client.comcast.net) |
| 03:14.49 | syslod | Anyone have cvs HEAD and a PRI card? I'll like to know if others are getting same errors. |
| 03:15.31 | JunK-C | syslod: i do. |
| 03:15.40 | *** join/#asterisk sp-ed (~dk@dhcp-66-212-222-189.myeastern.com) |
| 03:15.50 | myconid | JunK-C doesnt get any errors |
| 03:15.53 | myconid | I think hes a liar :) |
| 03:16.20 | syslod | JunK-C: Having any problems dialing NXX-XXXX? |
| 03:16.32 | opus___ | <PROTECTED> |
| 03:16.32 | JunK-C | syslod: nope. |
| 03:16.45 | opus___ | any mysql enthusiast who can help? |
| 03:16.52 | syslod | JunK-C: What zaptel card are u using? |
| 03:16.55 | opus___ | <PROTECTED> |
| 03:17.05 | *** join/#asterisk yxa (~void@203.118.40.42) |
| 03:17.07 | JunK-C | TE410P and T400P |
| 03:17.08 | sp-ed | hi i just installed asterisk @ home... and signed up for a broadvoice byod account, but I am having problems setting it up, i think i am having problems with the dlaiplan (extentions.conf) |
| 03:17.20 | Trepalium | opus___: Get the devel package for mysqlclient. |
| 03:17.52 | sp-ed | i've been reading stuff on voip-info.org, and looking in the extentions.conf, but i don't understand what i am supposed to be doing or how this works. |
| 03:17.54 | syslod | JunK-C: I hadn't changed anything in my configs but after updating I now get --Mar 7 22:17:03 WARNING[3871]: chan_zap.c:7414 zt_pri_error: PRI: Call Reference |
| 03:17.54 | mikegrb | sp-ed: try asterisk@home irc channel, asterisk@home works differently |
| 03:17.54 | syslod | <PROTECTED> |
| 03:18.00 | opus___ | trepalium - i have them installed, i'm going to add the library directory to ld.so.conf |
| 03:18.15 | JunK-C | <PROTECTED> |
| 03:18.17 | mikegrb | sp-ed: or maybe they have thier own mailing list |
| 03:18.28 | syslod | If I dial the whole NPA-NXX-XXXX it works. |
| 03:18.38 | opus___ | Still no good... :( |
| 03:18.41 | JunK-C | syslod: isnt blocked??? |
| 03:18.54 | syslod | blocked? |
| 03:19.25 | JunK-C | by ur telco |
| 03:19.34 | opus___ | Oh, i think i fixed it |
| 03:19.35 | sp-ed | i was wondering if anyone could give me a hand, or point me to some reading meterial other than voip-info or something maybe a little more that i can understand |
| 03:19.36 | Trepalium | opus___: Is there a libmysqlclient.a file in that directory? |
| 03:19.40 | *** join/#asterisk Wi_Fi (~OUT@c-24-127-12-85.we.client2.attbi.com) |
| 03:19.55 | opus___ | trepalium - i needed to add -L/usr/lib/mysql to the makefile |
| 03:20.08 | Wi_Fi | hey guys |
| 03:20.16 | Wi_Fi | are these clones |
| 03:20.21 | Wi_Fi | http://search.ebay.com/Digium-Wildcard-X100P-FXO-card_W0QQfkrZ1QQfromZR8 |
| 03:20.35 | JunK-C | X100P is deprecated. |
| 03:20.44 | Trepalium | Clones of a card that is no longer sold by digium. |
| 03:20.52 | mikegrb | Wi_Fi: yes |
| 03:21.23 | opus___ | Awww shit |
| 03:21.24 | Wi_Fi | they are saying Digium OEM |
| 03:21.43 | Trepalium | They're cheap winmodems that happen to be compatible with the X100P. |
| 03:21.55 | mikegrb | Wi_Fi: they lie |
| 03:22.18 | Wi_Fi | and that is exactly what i dont want |
| 03:22.19 | Wi_Fi | k |
| 03:22.21 | Wi_Fi | thnx guys |
| 03:22.22 | Trepalium | They work alright as long as you're using them on a north american PSTN. |
| 03:23.18 | topping | so i could get one of these, plug it in to the phone and get reasonable functionality? |
| 03:23.38 | topping | (pstn phone line) |
| 03:24.05 | topping | sounds cool for a toy |
| 03:25.16 | syslod | Junk-C: I am the telco. I tested with other systems and it works. Its something in that particular * box. |
| 03:27.22 | Trepalium | I'm using on of those clone cards for testing stuff. I'm not sure I'd want to use it in a production system, but it answers calls from the PSTN, and that's all I need right now. |
| 03:27.38 | topping | yah, that's what i was thinking... thanks |
| 03:28.09 | Trepalium | And besides, what's $8 + S/H. |
| 03:28.16 | topping | exactly |
| 03:28.48 | topping | anyone ever tried to make asterisk SIP client functionality look exactly like a ATA-150 to the peer? |
| 03:29.21 | topping | (trying to connect to v0nag3) |
| 03:29.46 | PatrickDK | topping, heh, not possible |
| 03:29.55 | topping | howso? |
| 03:30.00 | *** join/#asterisk terrapen (~cjs@fw-01.satx.bikeworld.net) |
| 03:30.06 | PatrickDK | they encrypt it |
| 03:30.13 | PatrickDK | they also make random changes |
| 03:30.26 | myconid | So I can accept broadvoice calls.. |
| 03:30.27 | PatrickDK | you will have the same problems the people who hack directtv service have |
| 03:30.30 | myconid | they hear me.. |
| 03:30.32 | topping | but if they weren't, let's say |
| 03:30.35 | myconid | but I cant hear them |
| 03:30.43 | PatrickDK | if they wheren't, then it would work fine |
| 03:30.53 | PatrickDK | but they don't do it like that, so it won't work |
| 03:30.59 | topping | well, there's some SIP headers that need changing |
| 03:31.09 | topping | like telling them that I am an asterisk box is not a great idea |
| 03:31.11 | *** join/#asterisk MikeJ[Jayden] (~ircatjerr@pcp02795302pcs.roylok01.mi.comcast.net) |
| 03:31.13 | PatrickDK | there is ALOT of stuff that needs changing |
| 03:31.22 | topping | not that much |
| 03:31.25 | PatrickDK | hmm, you don't tell them your an asterisk box |
| 03:31.36 | PatrickDK | unless you didn't setup your sip config right |
| 03:31.40 | PatrickDK | or you just don't care |
| 03:32.03 | mikegrb | topping: for starters you don't even know your account auth name or password |
| 03:32.03 | topping | well, i need to tell them that i am an ata-150 |
| 03:32.12 | topping | i don't? |
| 03:32.16 | mikegrb | no |
| 03:32.18 | terrapen | good evening |
| 03:32.27 | topping | why don't i? |
| 03:32.48 | mikegrb | topping: because they don't give it out |
| 03:32.48 | topping | anyway |
| 03:32.49 | *** join/#asterisk montoya (montoya@200.195.84.238) |
| 03:32.50 | drsperm | I thought vonage was based on mac |
| 03:32.59 | mikegrb | haw haw |
| 03:33.01 | mikegrb | no |
| 03:33.07 | mikegrb | the config file is |
| 03:33.13 | mikegrb | which is encrypted |
| 03:33.18 | drsperm | ah...I got part of it right... |
| 03:33.23 | myconid | why doesnt vonage do sip. |
| 03:33.29 | topping | they do |
| 03:33.30 | myconid | anyone have BV working? |
| 03:33.36 | drsperm | vonage = bastards |
| 03:33.51 | topping | is there a way to change the SIP header fields or do i have to recompile it? |
| 03:33.55 | *** join/#asterisk brc007 (~brian@brc.base.supporter.pdpc) |
| 03:33.57 | drsperm | probably to control growth... |
| 03:33.59 | mikegrb | topping: it won't work |
| 03:34.05 | mikegrb | topping: you don't have the auth information |
| 03:34.15 | topping | mikegrb: you said that already |
| 03:34.26 | mikegrb | and you still don't have it |
| 03:34.39 | topping | why did you erase it from my hard drive? |
| 03:34.57 | myconid | I dont think broadvoice supports * anymore |
| 03:35.05 | topping | so as i was saying... |
| 03:35.23 | brc007 | myconid, yes they do |
| 03:35.24 | mikegrb | you can't do it |
| 03:35.28 | myconid | they most certainly dont. |
| 03:35.37 | brc007 | eh |
| 03:35.38 | myconid | cvs head doesnt work with BV anymore. |
| 03:35.38 | mikegrb | myconid: they never did officially |
| 03:35.47 | myconid | mikegrb: funny.. their webpage says they do. |
| 03:35.58 | myconid | and support in the past has helped me with issues. |
| 03:36.03 | mikegrb | yes |
| 03:36.09 | myconid | but now * doesnt work |
| 03:36.13 | myconid | lol. |
| 03:36.35 | mikegrb | myconid: there website has said that it will connect to asterisk but asterisk is not offically supported |
| 03:36.37 | drsperm | ok..so if broadvoice is not supported...who to use then? |
| 03:36.44 | brc007 | drsperm, you need local dids? |
| 03:36.46 | brc007 | if not, |
| 03:36.47 | brc007 | ~nufone |
| 03:36.48 | jbot | i heard nufone is Visit http://www.nufone.net for an excellent, native IAX termination service. |
| 03:36.49 | Trepalium | It's the same nonsense that cellular providers do with their cell phones. You have a perfectly good phone from your former cell provider, now buy a new one. |
| 03:37.05 | myconid | does anyone here run broadvoice? |
| 03:37.09 | drsperm | k..I will look.. |
| 03:37.24 | drsperm | myconid: someone was on this am...said they loved it. |
| 03:37.31 | myconid | I love it to |
| 03:37.34 | myconid | it works great. |
| 03:37.40 | myconid | I wish it worked with * though. |
| 03:37.54 | drsperm | I thought they were using it...but maybe not. |
| 03:38.05 | myconid | I can get calls.. but the caller cant hear me |
| 03:38.09 | myconid | and outbound calls fail. |
| 03:38.16 | myconid | e config file is |
| 03:38.16 | myconid | <mikegrb> which is encrypted |
| 03:38.16 | myconid | <drsperm> ah...I |
| 03:38.18 | myconid | oops sorry. |
| 03:38.26 | mikegrb | myconid: probably a nat issue |
| 03:38.30 | myconid | Yea.. |
| 03:38.32 | myconid | Except I dont use it. |
| 03:38.43 | drsperm | hmm..pots sounds pretty good... |
| 03:38.43 | mikegrb | ahh |
| 03:38.50 | myconid | Replace accountid with your account, password with your password and extension with one of your accessible extensions in the dial plan. <- what does this mean? |
| 03:38.55 | myconid | register => <accountid>@sip.broadvoice.com:<password>:<account id>@sip.broadvoice.com/<extension> |
| 03:39.03 | ManxPower | The phones are not in production yet. |
| 03:39.04 | myconid | in [default] ? |
| 03:39.18 | ManxPower | But they are on desks 8-) |
| 03:39.40 | Trepalium | If you're using Xten X-Lite, make sure transmit silence is on. |
| 03:39.41 | drsperm | Hey...nufone.net is powered by Asterisk.... |
| 03:39.42 | drsperm | :) |
| 03:39.56 | *** join/#asterisk Nukemizer (~Nuke@66.237.85.58.ptr.us.xo.net) |
| 03:39.58 | Trepalium | It can prevent you from transmitting sound if it's not on. |
| 03:40.01 | myconid | Treplalium: tried x-crap and an iad. neither work |
| 03:40.10 | ManxPower | drsperm, un yes and owned by the guy that wrote the chan_skinny and chan_h323 that are included with the Asterisk source. |
| 03:40.50 | mikegrb | drsperm: how else would they allow iax delivery of calls? |
| 03:41.04 | drsperm | cool... |
| 03:41.20 | drsperm | mikegrb: huh? |
| 03:41.25 | *** join/#asterisk tessier_ (~treed@wsip-68-224-172-77.sd.sd.cox.net) |
| 03:41.55 | mikegrb | drsperm: iax is asterisk only, currently |
| 03:42.09 | drsperm | cool...I didn't know... |
| 03:42.17 | myconid | It most certainly is not. |
| 03:42.21 | drsperm | I am researching everything right not...not deploying... |
| 03:42.22 | Trepalium | Given the middle letter stands for asterisk... |
| 03:42.23 | myconid | There are many iax clients. |
| 03:42.34 | myconid | who says something closed cant support it? |
| 03:42.34 | drsperm | that is in a few days...after I get my voip phones in... |
| 03:42.46 | drsperm | I just deployed my server at the colo today...now to config |
| 03:43.14 | drsperm | So I guess my next question is iax or sip ? answer iax... |
| 03:43.25 | ManxPower | drsperm, just install it. |
| 03:43.31 | drsperm | installed. |
| 03:43.35 | drsperm | time to config... |
| 03:43.39 | drsperm | running gentoo... |
| 03:45.43 | *** join/#asterisk t3t (~t3t@galley.pangalacticgargleblaster.com) |
| 03:46.14 | *** part/#asterisk terrapen (~cjs@fw-01.satx.bikeworld.net) |
| 03:46.18 | Qwell | t3t: love the domain :p |
| 03:47.24 | mikegrb | topping: read the fsking docs |
| 03:47.32 | topping | mikegrb: put a cork in it |
| 03:47.35 | mikegrb | topping: hell there is a fsking example in the damn config file |
| 03:47.51 | topping | you just don't get it |
| 03:47.54 | mikegrb | you obviously have put 0 researtch into this |
| 03:48.10 | topping | you obviously don't know what the hell you are talking about |
| 03:48.24 | CoaxD | SHAL KEK NEM RON! |
| 03:48.26 | drsperm | nufone seems pricey... |
| 03:48.30 | mikegrb | ok, you come back when you admin several asterisk boxes |
| 03:48.38 | topping | ooooh |
| 03:48.40 | topping | so manly |
| 03:48.44 | mikegrb | drsperm: it's quality compared to other providers though |
| 03:48.53 | drsperm | k...good to know... |
| 03:48.54 | drsperm | thanks. |
| 03:49.18 | mikegrb | drsperm: I've used cheaper places but still pay more and use them, even for home service |
| 03:49.31 | CoaxD | how much cheaper is cheaper |
| 03:49.39 | mikegrb | like .6 cents/min |
| 03:50.00 | CoaxD | because marginal service is not what I'd call cheaper |
| 03:50.03 | CoaxD | I call that more expensive |
| 03:50.11 | CoaxD | mikegrb: .6 cents per minute. hah. |
| 03:50.13 | mikegrb | heh |
| 03:50.17 | Qwell | mikegrb: y0 |
| 03:50.18 | mikegrb | that much cheaper |
| 03:50.19 | *** join/#asterisk DrRighteous (~DrRighteo@ool-182c867b.dyn.optonline.net) |
| 03:50.23 | mikegrb | 1.4 cents I think |
| 03:50.25 | CoaxD | mikegrb: Are incoming calls free with the company that does .6c/min? |
| 03:50.27 | mikegrb | Qwell: hey, sup |
| 03:50.35 | DrRighteous | hey all |
| 03:50.42 | mikegrb | CoaxD: 1.4/c outgoing only |
| 03:50.42 | CoaxD | mikegrb: Yeah. but if it doesn't WORK sometimes, it is more EXPENSIVE. Get my point? |
| 03:50.58 | CoaxD | mike: So whats .6c/min? |
| 03:51.13 | Trepalium | He meant 0.6c/min cheaper than nufone. |
| 03:51.16 | CoaxD | OH |
| 03:51.17 | *** join/#asterisk Grooby (~Grooby@ip24-250-126-171.dc.dc.cox.net) |
| 03:51.19 | CoaxD | I get it |
| 03:51.33 | mikegrb | CoaxD: I get you point, that's why I pay nufone 2 cents per minute |
| 03:51.39 | CoaxD | mike: OKay. understood ;) |
| 03:51.58 | Tr0j4N | pardon, what's the command to copy a whole directory? |
| 03:52.12 | drsperm | cp -a /dir /dest |
| 03:52.14 | Trepalium | cp -a |
| 03:52.26 | Tr0j4N | ah, thanks |
| 03:52.41 | Trepalium | Drat, too slow. :) |
| 03:52.59 | drsperm | hey...I may not know * but linux...no prob... |
| 03:53.03 | mikegrb | haw haw |
| 03:53.04 | drsperm | :0 |
| 03:53.17 | Nukemizer | Jaffa KREE |
| 03:53.29 | mikegrb | topping has such a low self esteem he feels it is necessary to msg me to defend himself by talking about how worthless I am |
| 03:54.12 | Trepalium | Why not just /ignore him and be done with it? |
| 03:54.39 | Nukemizer | SHOVA |
| 03:54.39 | mikegrb | it's quite entertaining |
| 03:54.50 | CoaxD | Nukemizer: YOU DO NOT SPEAK PROPER GOA'ULD SIR! |
| 03:54.56 | CoaxD | Nukemizer: SHOL'VA!#@$ |
| 03:55.01 | Nukemizer | lol |
| 03:55.02 | CoaxD | Nukemizer; *chuckle* |
| 03:55.07 | mikegrb | ha ha |
| 03:55.34 | CoaxD | apparently, next episode of SGA, Teyla turns into a WRAITH? Hmmmm.. |
| 03:55.44 | Qwell | CoaxD: spoiler alert! |
| 03:55.51 | Qwell | not that I watch it, or anything |
| 03:56.02 | CoaxD | Qwell: No spoiler. It was on the SCIFI trailer for friday |
| 03:56.06 | Qwell | :p |
| 03:56.22 | CoaxD | speaking of which, i'm gonna have to poke at the tivo to update the lineups |
| 03:56.24 | Nukemizer | Telya is HOT |
| 03:56.37 | CoaxD | Nukemizer; You're telling me. There's gotta be some Teyla porn around SOMEWHERE.. |
| 03:56.44 | Nukemizer | you need to get Tivo 2 go |
| 03:56.55 | CoaxD | Nukemizer: Why? |
| 03:57.01 | CoaxD | Nukemizer; I can watch it on my TV just fine ;) |
| 03:57.09 | Nukemizer | can pull your hsows to laptop or PC |
| 03:57.10 | topping | i was more concerned about mikegrb's self-righteousness and inability to let other people answer when he knows not what he speaks of |
| 03:57.13 | Nukemizer | shows |
| 03:57.26 | opus___ | who are the main developers for asterisk? |
| 03:57.27 | CoaxD | Nukemizer; No point for me.. I like watching TV on TV |
| 03:57.42 | CoaxD | opus: Mark Spencer is the lead developer and initial author of the project |
| 03:57.53 | *** join/#asterisk Lethargicclown (~chatzilla@ool-18bee80e.dyn.optonline.net) |
| 03:57.53 | opus___ | does he irc here? |
| 03:57.59 | CoaxD | opus: ('kram' here on irc) |
| 03:58.02 | Qwell | opus___: kram |
| 03:58.05 | Qwell | erm, what he said |
| 03:58.21 | Nukemizer | CoaxD, I have Kids that take over TV .. Tivo is the way to go however |
| 03:58.26 | *** join/#asterisk thodal (nwhere@pool-151-203-187-197.wma.east.verizon.net) |
| 03:58.30 | mikegrb | Qwell and Beirdo are the second and third most contributers |
| 03:58.30 | hardwire | hi |
| 03:58.34 | CoaxD | Nukemizer: You need 2 tivos and 2 tv's :) |
| 03:58.38 | Qwell | I don't think so, heh |
| 03:58.39 | hardwire | my last name is spencer |
| 03:58.41 | hardwire | but I am not mark |
| 03:58.44 | hardwire | I only wish.. right? |
| 03:58.46 | Nukemizer | Coaxd, got em :) |
| 03:58.51 | opus___ | thanks |
| 03:58.52 | Qwell | hardwire: I sure wouldn't want to be |
| 03:58.52 | CoaxD | Nukemizer: Sweet! |
| 03:59.01 | hardwire | Qwell: heh ok! |
| 03:59.04 | CoaxD | Nukemizer: Tho with MythTV, you wouldn't need to pay $26/mo in order to watch TV |
| 03:59.11 | Qwell | people are downright mean to him |
| 03:59.12 | CoaxD | Nukemizer: You could have as many frontends as you wanted |
| 03:59.12 | Nukemizer | coaxD very sweet indeed |
| 03:59.14 | hardwire | I wouldn't mind being somebody else right now |
| 03:59.15 | hardwire | my tooth just split in two |
| 03:59.18 | hardwire | oww |
| 03:59.21 | thodal | ouchies |
| 03:59.21 | Grooby | <--- f around w/ his mythtv right now |
| 03:59.23 | CoaxD | hardwire: Oh god. *groan* |
| 03:59.24 | Grooby | trying to get HD to work |
| 03:59.32 | mikegrb | CoaxD: I've got a nice myth setup |
| 03:59.45 | mikegrb | well, did anyway |
| 03:59.47 | CoaxD | mikegrb: I did at one time, too. Had too many problems with PVR-350 decoder |
| 03:59.55 | CoaxD | mikegrb: Tivo had what i wanted at the time. I bit |
| 03:59.55 | t3t | Qwell: thanks :) |
| 03:59.55 | mikegrb | I've got a 250 |
| 04:00.01 | CoaxD | mike: I have a 350 and a 250 |
| 04:00.06 | CoaxD | mike: THey jiust sit here. *g* |
| 04:00.10 | topping | the 150 and 500 are working as of the last few days |
| 04:00.14 | Nukemizer | went with Beyond TV currently.. could get the cylces to diddle with myth TV. But Myth Tv does look good |
| 04:00.25 | mikegrb | we now pay the cable company pvr as it does hd and was like $3/month |
| 04:00.48 | mikegrb | and a dlink media player that will do high def output |
| 04:00.50 | Nukemizer | * is more fun for the moment ... Do updates for VON get posted here anywhere ? |
| 04:00.59 | hardwire | heh |
| 04:01.03 | hardwire | I wanna be at VON |
| 04:01.09 | hardwire | instead of this god forsaken island. |
| 04:01.17 | hardwire | atleast I am headed for dry land soon |
| 04:01.25 | topping | what island? |
| 04:01.26 | sivana | ? |
| 04:01.28 | Nukemizer | Need a Worm whole .. Jack has one right now |
| 04:01.32 | hardwire | I uh.. tapped a few people going to VON.. I will monitor them remotely |
| 04:01.33 | hardwire | mwa ha ha haaaa |
| 04:01.42 | hardwire | topping: st paul island, ak |
| 04:01.53 | topping | wow, and cold to boot this time of year |
| 04:02.14 | hardwire | not bad |
| 04:02.17 | hardwire | not as bad as anchorage |
| 04:02.21 | hardwire | humidity helps with that |
| 04:02.24 | hardwire | islands are always kinda steamy. |
| 04:02.36 | topping | ic |
| 04:03.00 | mikegrb | hardwire: indeed, I lived on an island off south carolina for a while |
| 04:03.07 | mikegrb | now by the beach in florida |
| 04:03.11 | hardwire | that would be nice! |
| 04:03.17 | opus___ | damn, server is fscked |
| 04:03.20 | hardwire | mikegrb: yeh.. can I have your job.. whatever it is? |
| 04:03.22 | thodal | can someone anwser a stupid noobie question? |
| 04:03.25 | mikegrb | not as many flying insects in florida |
| 04:03.33 | *** part/#asterisk Grooby (~Grooby@ip24-250-126-171.dc.dc.cox.net) |
| 04:03.35 | hardwire | mikegrb: do you have phobias? |
| 04:03.37 | mikegrb | hardwire: heh, I babysit about 45 servers and 2k ysers |
| 04:03.42 | mikegrb | and work at home |
| 04:03.46 | thodal | is there anywherev were i can get alist of voip termination providers>? |
| 04:03.48 | hardwire | mikegrb: ah |
| 04:03.51 | mikegrb | take the ibook and sit by the water |
| 04:03.53 | hardwire | I babysit real people |
| 04:03.54 | hardwire | it seems |
| 04:03.57 | hardwire | oh well |
| 04:04.02 | mikegrb | I think I agree there |
| 04:04.05 | mikegrb | sometimes it seems |
| 04:04.19 | hardwire | I do internal support for our company |
| 04:04.21 | hardwire | sat based comms and voip |
| 04:04.24 | hardwire | stuff like that |
| 04:04.27 | mikegrb | had someone open a support ticket because yum wouldn't work with a corrupt rpm |
| 04:04.34 | hardwire | teaching them to use a voip system is like pulling teeth |
| 04:04.40 | hardwire | see.. |
| 04:04.44 | hardwire | you dial only 3 numbers |
| 04:04.48 | hardwire | OOOHHHH! |
| 04:04.59 | *** join/#asterisk terrapen (~cjs@fw-01.satx.bikeworld.net) |
| 04:05.00 | hardwire | they don't understand push to dial |
| 04:05.06 | terrapen | woohoo check it out |
| 04:05.07 | hardwire | and they don't type fast enough for auto dial |
| 04:05.08 | terrapen | http://slashdot.org/comments.pl?sid=141718&cid=11872925 |
| 04:05.23 | *** join/#asterisk slePP (~slepp@S01060040f48412ad.ed.shawcable.net) |
| 04:05.27 | Nukemizer | Any AMP masters online ? |
| 04:05.32 | opus___ | damn it my wm is frozen while nfs is down |
| 04:05.44 | mikegrb | hardwire: heh, even my wife didn't take long to catch on |
| 04:06.06 | hardwire | mikegrb: well she probably had better tech growing up than the people out here |
| 04:06.15 | hardwire | they hadn't seen TV until 20 hours it was mainstream in the lower 48 |
| 04:06.17 | mikegrb | hardwire: now she prefers sip hard phones to the cordless phones as she doesn't have to dial with the ata |
| 04:07.02 | hardwire | 20 years |
| 04:07.02 | hardwire | heh |
| 04:07.02 | hardwire | it seems |
| 04:07.02 | terrapen | my phone auction ends in 26 min |
| 04:07.02 | terrapen | hopefully i win |
| 04:07.02 | Trepalium | I've managed to successfully install AMP, but that's all. |
| 04:07.03 | hardwire | I hope you win! |
| 04:07.03 | hardwire | heh |
| 04:07.03 | terrapen | and i get my new apartment centerpeice |
| 04:07.03 | mikegrb | she never would touch any of the sip hardphones before |
| 04:07.03 | terrapen | :) |
| 04:07.03 | Nukemizer | I want to use the AMP interface to ring in a Zap channel and ring back out to another Zap Channel |
| 04:07.03 | mikegrb | showed her and she was like wow, that is so much better |
| 04:07.03 | hardwire | mikegrb: so you made it a pain in the ass to use anything else? |
| 04:07.04 | mikegrb | now it is just about all she uses |
| 04:07.05 | topping | terrapen: best to bid five seconds before the end of auction :-) |
| 04:07.07 | hardwire | really nice |
| 04:07.27 | terrapen | i had my dad put a snipe on it |
| 04:07.34 | topping | lol |
| 04:07.39 | *** join/#asterisk FryGuy- (fryguy@c-24-23-19-33.client.comcast.net) |
| 04:07.40 | terrapen | sniperight rules |
| 04:07.45 | mikegrb | hardwire: well, the ATAs recognize 7,8,9 or any other 4 digit number and dial imidiatly |
| 04:08.07 | mikegrb | 7, 8, and 9 do disa with no auth for fwd, enum, and outgoing |
| 04:08.20 | hardwire | auck |
| 04:08.22 | hardwire | flight leaving |
| 04:08.23 | mikegrb | on the sip hardphone she can just dial the number she wants |
| 04:08.25 | hardwire | must go |
| 04:08.25 | hardwire | heh |
| 04:08.26 | hardwire | bbl |
| 04:08.31 | mikegrb | have fun |
| 04:08.50 | *** part/#asterisk HeppyCat (~unknown@cpe-24-164-217-41.jam.res.rr.com) |
| 04:08.58 | terrapen | i wish i could get more people interested in my anti-phisher underground campaign |
| 04:09.14 | opus___ | terrapen how so |
| 04:09.26 | topping | terrapen: i thought it was rad... hook it up to procmail! |
| 04:09.59 | *** join/#asterisk Mike (~mike@201.135.48.217) |
| 04:10.13 | terrapen | opus, see this: |
| 04:10.16 | terrapen | http://slashdot.org/comments.pl?sid=141718&cid=11872925 |
| 04:10.22 | terrapen | i need to refine it |
| 04:10.31 | Mike | guys does asterisk have some way to mask the voice, like have some effect to make it like disguise? |
| 04:10.34 | terrapen | and establish some sort of mailing list where phishing emails are traded |
| 04:10.52 | terrapen | and "attacks" coordinated |
| 04:10.52 | Mike | imagine like the guys from the movie scream that had something that disrupt the voice? |
| 04:10.59 | Trepalium | Slight DoS potential on that script... |
| 04:11.17 | terrapen | trep, yep, "Denial of Scam" |
| 04:11.17 | opus___ | "attacks" |
| 04:11.36 | terrapen | first off, my attack (if you call it that) is not bandwidth-consuming |
| 04:11.41 | terrapen | only about 10KB/s |
| 04:11.53 | terrapen | all it does is pollute their collected data with garbage |
| 04:11.59 | opus___ | legally you couldn't hack'em unless the local government declared them illegal one by one i would think... then the bounty law would take in effect and you could do as you wish |
| 04:12.02 | terrapen | to thwart their collection routines |
| 04:12.07 | opus___ | oh bandwidth:) |
| 04:12.07 | terrapen | i dont want to hack them |
| 04:12.20 | topping | mike: it's possible, you would have to get into the codec stream |
| 04:12.22 | terrapen | i just want to make it difficult for them to steal people's identities |
| 04:12.29 | *** join/#asterisk shepherd (matt@pcp01541028pcs.huntsv01.al.comcast.net) |
| 04:12.36 | topping | be some coding tho |
| 04:12.39 | Mike | topping, why wouldnt asterisk put it as a feature? |
| 04:12.48 | opus___ | terrapen you could also flood them with worthless info which would increase the signal to noise ratio |
| 04:12.56 | opus___ | decrease |
| 04:13.02 | Mike | topping, this way people can use masks so they wont reconize there voices |
| 04:13.07 | topping | Mike: asterisk is configurable enough that it wouldn't have to be made an explicit feature |
| 04:13.12 | terrapen | thats what im doing, opus |
| 04:13.15 | terrapen | oooo... |
| 04:13.16 | topping | you could build it yourself and link it in |
| 04:13.20 | Trepalium | One suggestion would be to use a database of common names for firstname, lastname, and username, and restrict the card_number, etc to numerals only. |
| 04:13.21 | opus___ | oh |
| 04:13.22 | terrapen | i like the idea of a voice masking application |
| 04:13.26 | terrapen | that is an awesome idea |
| 04:13.39 | terrapen | combined with caller*ID spoofing, i could have fun with that |
| 04:13.44 | topping | haha |
| 04:13.55 | opus___ | speaking about caller id spoofing |
| 04:13.58 | topping | i've been reading asterisk source tonight |
| 04:14.02 | opus___ | does anybody have a setup to do that |
| 04:14.03 | terrapen | trep, yep, I'm doing some of that already |
| 04:14.14 | terrapen | opus: it's called SetCallerIDNum() |
| 04:14.18 | Corydon76-home | terrapen: easily done. It's called LPC10 |
| 04:14.18 | opus___ | really |
| 04:14.26 | terrapen | trep, though I need to add support for a common names dictionary |
| 04:14.27 | opus___ | do you need a digital PRI T1 |
| 04:14.28 | Trepalium | Mixed-case alphanumeric garbage can be a little too easy to filter. |
| 04:14.31 | shepherd | but.. it talkes a t1 |
| 04:14.32 | shepherd | yeah |
| 04:14.35 | terrapen | LPC-10? |
| 04:14.42 | opus___ | hmmm.. |
| 04:14.47 | shepherd | takes also :) |
| 04:14.50 | terrapen | trep: i agree |
| 04:14.52 | Mike | topping, i though this could be a usefull feature for a lot of people |
| 04:14.54 | Corydon76-home | Yeah, makes you sound like Mr Roboto... |
| 04:14.55 | terrapen | trep: want to help me code this? |
| 04:15.17 | shepherd | opus: or you can get a channel from someone who has a t1 |
| 04:15.18 | opus___ | you could have a 900 number 'please select your voice' |
| 04:15.20 | Trepalium | Nope. I suck at writing perl. Can hardly make sense of it most of the time. |
| 04:15.22 | topping | hahah |
| 04:15.26 | terrapen | imagine what a private investigator would pay for the ability to do this: |
| 04:15.38 | myconid | is IAX encrypted? |
| 04:15.39 | topping | yes indeed |
| 04:15.46 | techie | metalic as fuck |
| 04:15.51 | shepherd | myconid: no |
| 04:15.51 | opus___ | shepherd you know anyone who would sell me one? |
| 04:16.00 | shepherd | opus: nuphone |
| 04:16.00 | terrapen | call a client's (cheating) spouse with the spoofed caller*id of the suspected person he/she is cheating with |
| 04:16.01 | DrRighteous | hey anyone here ever play with PortaOne softswitch software? |
| 04:16.04 | Mike | terrapen, i got a phone that makes this mask and other 3 masks for 35 bucks |
| 04:16.08 | shepherd | it just has to be in iax |
| 04:16.11 | myconid | its 2005 and ppl cant write encrypted protocols by default? |
| 04:16.15 | terrapen | and call that person with the spoofed caller*id of the cheating spouse |
| 04:16.15 | shepherd | and you can change it to whatever you want |
| 04:16.18 | opus___ | shepherd really,what protocol do they use |
| 04:16.19 | terrapen | and bridge the two calls together |
| 04:16.22 | terrapen | and record the results |
| 04:16.24 | opus___ | do they use SIP? |
| 04:16.36 | shepherd | i don't think you can do it with sip |
| 04:16.43 | opus___ | IAX? |
| 04:16.49 | shepherd | but nuphone uses iax |
| 04:16.54 | opus___ | oh |
| 04:17.03 | terrapen | is that evil or what? |
| 04:17.08 | terrapen | dammit im an evil bastard tonight |
| 04:17.17 | topping | terrapen: what do you usually code in? |
| 04:17.22 | terrapen | Perl |
| 04:17.33 | terrapen | that script i posted on slashdot is not a good example of my code |
| 04:17.40 | terrapen | its just something i banged out in a few minutes |
| 04:18.03 | opus___ | is nuphone more stable then broadvoice? |
| 04:18.04 | shepherd | mycon: voip uses a ton of bandwidth.. i'm sure you could figure out a way to make it encrypted if you want |
| 04:18.21 | shepherd | nuphone is just a long distance provider |
| 04:18.25 | terrapen | i like nufone |
| 04:18.30 | terrapen | super stable for me |
| 04:18.33 | shepherd | and 1800 number |
| 04:18.37 | shepherd | provider |
| 04:18.49 | Trepalium | Why not just tunnel it via IPsec or similar if you're worried about encryption? |
| 04:18.53 | terrapen | and a michigan DID provider |
| 04:19.09 | myconid | Trepalium:..wow.. "why not just tunnel telnet with ipsec.. you dont need ssh" |
| 04:19.26 | topping | terrapen: but only $25 more to get a 866 number |
| 04:19.27 | opus___ | yeah, is there an easy way to encrypt sip? |
| 04:19.29 | myconid | if you want security in your car, get a club.. we dont offer door locks. |
| 04:19.33 | shepherd | hmm.. |
| 04:19.43 | roamer323 | the sp3k supports encrypted calls right out of the 'ATA' box |
| 04:19.48 | shepherd | i wonder how you send encrypted udp packets without screwing everything up |
| 04:20.04 | *** join/#asterisk W1thdraw (~Withdraw@ip68-5-125-44.oc.oc.cox.net) |
| 04:20.06 | opus___ | nc -u :) |
| 04:20.17 | terrapen | its not 25$ for an 866 |
| 04:20.17 | topping | opus___: what about ipsec with opportunistic tunnel setup? |
| 04:20.18 | myconid | My cisco phone does it to the concentrator. |
| 04:20.19 | terrapen | its free |
| 04:20.22 | terrapen | and $0.02/min |
| 04:20.28 | opus___ | topping - ipsec isn't really on demand |
| 04:20.46 | mikegrb | topping: uh, toll free is free |
| 04:20.46 | terrapen | its $25 to get a custom toll-free number |
| 04:20.50 | terrapen | ie., you pick the number |
| 04:20.51 | shepherd | ipsec would be forcing voip to do error checking |
| 04:20.55 | terrapen | if you let them pick for you, its free |
| 04:21.05 | topping | terrapen: u sure about that? |
| 04:21.08 | mikegrb | topping: why don't you figure out what you are talking about before you go spotting bull shit |
| 04:21.20 | topping | mikegrb: fuck off |
| 04:21.23 | terrapen | topping: real sure |
| 04:21.27 | mikegrb | I have several toll free dids from them |
| 04:21.29 | mikegrb | all free |
| 04:21.32 | terrapen | i have several from them |
| 04:21.33 | terrapen | yep |
| 04:21.34 | mikegrb | just like it says on the web page |
| 04:21.40 | terrapen | but it is $0.02/min |
| 04:21.43 | terrapen | for incoming |
| 04:21.50 | terrapen | but thats a swell deal, imho |
| 04:21.55 | mikegrb | indeed it is |
| 04:22.03 | *** join/#asterisk BurnedOutGeek (nobody@res-66-191-172-161.spa.sc.charter.com) |
| 04:22.11 | opus___ | broadvoice has it for like .20 cents a minute... thats like 5 minutes for a penny |
| 04:22.11 | opus___ | :) |
| 04:22.22 | BurnedOutGeek | any asterisk devs around in here tonight? |
| 04:22.23 | opus___ | atleast thats what the website claims |
| 04:22.33 | opus___ | i haven't seen a better deal since burger king had .99 cent whoppers |
| 04:22.37 | drsperm | ok....so does nufone offer inbound and outbound? |
| 04:22.46 | mikegrb | opus___: but you must pay a monthly fee |
| 04:22.50 | mikegrb | drsperm: yes |
| 04:22.54 | shepherd | drsperm: yes |
| 04:22.56 | terrapen | sperm, of course |
| 04:23.00 | topping | yes, it looked at the time like 866 was a vanity charge but indeed it is for a custom number |
| 04:23.01 | roamer323 | asterisk devs are boozin@VON |
| 04:23.02 | terrapen | check rates.csv |
| 04:23.03 | *** join/#asterisk soundguy (~soundguy@zeus.soundguy.id.au) |
| 04:23.07 | topping | mikegrb wins! |
| 04:23.08 | drsperm | thanks ....I missed the boat on that one... |
| 04:23.11 | topping | the fans go wild! |
| 04:23.18 | drsperm | opus___: too much mayo |
| 04:23.35 | Tr0j4N | how do you add to your PATH variable? |
| 04:23.41 | topping | opus___: anyway opportunistic encryption grabs pubkeys from DNS |
| 04:23.56 | topping | and if each side finds them, it sets up the tunnel |
| 04:24.00 | opus___ | topping what arch/distro does it best? |
| 04:24.03 | terrapen | nufone's toll free service allowed me to do this: 866-859-7359 |
| 04:24.07 | terrapen | call that number :) |
| 04:24.07 | roamer323 | BV does not charge per min on incoming calls |
| 04:24.16 | topping | i'm using redWall on my firewall right now |
| 04:24.33 | topping | it's a CD based redhat 9 distro with everything configured |
| 04:24.58 | opus___ | terrapen thas cooL! |
| 04:25.03 | terrapen | :-) |
| 04:25.05 | terrapen | thx |
| 04:25.14 | terrapen | it updates every 5s |
| 04:25.15 | opus___ | what tts was that |
| 04:25.18 | terrapen | ? |
| 04:25.21 | terrapen | tts? |
| 04:25.24 | topping | terrapen: nice work |
| 04:25.45 | terrapen | i need to pay for some real voice prompts for that |
| 04:25.53 | terrapen | my texan accent fucks it all up, really |
| 04:26.02 | niZon | is it really worth the effort to try and get a cisco 79xx with an SIP image? |
| 04:26.12 | niZon | or will everything work fine using skinny? |
| 04:26.14 | drsperm | terracon: cool...but get a cowgirl to be the voice... |
| 04:26.22 | drsperm | surly I can find a few here in San Antonio.... |
| 04:26.24 | terrapen | that would be my sister :) |
| 04:26.32 | terrapen | you are in SA? |
| 04:26.36 | terrapen | i am in SA |
| 04:26.47 | drsperm | Cool...I am in Boerne...20 miles north. |
| 04:26.51 | terrapen | no way! |
| 04:26.54 | drsperm | yes way. |
| 04:26.55 | terrapen | i used to live in Bandera |
| 04:26.58 | terrapen | (bantucky!) |
| 04:27.10 | drsperm | I am sorry to hear that...we always kicked your ass in Football... |
| 04:27.13 | drsperm | f:) |
| 04:27.13 | terrapen | hahahah |
| 04:27.14 | opus___ | fuck, this server is going to fsck for the 4th time |
| 04:27.17 | terrapen | i went to Alamo Heights |
| 04:27.22 | terrapen | you still probably kicked our ass |
| 04:27.30 | terrapen | though not last year :P |
| 04:27.35 | drsperm | Well....I have dated a few 09'er girls... |
| 04:27.51 | terrapen | "Look, honey, we're in Burn, TX" |
| 04:27.56 | drsperm | I used to work over on Harry Wurzbach for 9 years... |
| 04:28.01 | terrapen | dumb tourist pronounciation of boerne |
| 04:28.04 | drsperm | 09'er was my shortcut.. |
| 04:28.05 | terrapen | nice |
| 04:28.11 | terrapen | my family owns bike world |
| 04:28.15 | terrapen | dunno if youve ever been theere |
| 04:28.16 | opus___ | is there any way to skip fsck check? |
| 04:28.25 | drsperm | cool...I help funded your college tuition... |
| 04:28.33 | topping | terrapen: are you using IAX or SIP with nufone (presume IAX) |
| 04:28.43 | shepherd | i think nuphone only does iax |
| 04:28.52 | terrapen | you know whats strage? i've been on irc for like 12 years now and never met anyone from around here..till i come to #asterisk...and i've met three this month |
| 04:28.57 | topping | they do sip now according to the signup sheet |
| 04:28.59 | terrapen | topping, IAX |
| 04:29.03 | topping | cool thx |
| 04:29.05 | terrapen | dsperm, why thank you |
| 04:29.08 | terrapen | :P |
| 04:29.10 | shepherd | oh wait |
| 04:29.13 | shepherd | they do support sip too |
| 04:29.17 | drsperm | yeah...I talked with a SA guy this am... |
| 04:29.51 | topping | eventually i'll get enough things right that mikegrb will keep his pie hole shut |
| 04:30.04 | drsperm | terrapen: we may have to meet up some time...hack some wifi at starbucks... |
| 04:30.06 | PTG123 | hey when i hit * it asks me for a mailbox #, how can i make it so it knows that? |
| 04:30.14 | terrapen | you bet |
| 04:30.25 | terrapen | its impossible to bust into starbucks tho :) |
| 04:30.30 | terrapen | but the quarry is now open 24/7 |
| 04:30.48 | drsperm | cool... |
| 04:30.51 | shepherd | PTG: agi |
| 04:30.53 | shepherd | hehe |
| 04:30.58 | drsperm | some of the EZ's around town have free wifi... |
| 04:31.00 | terrapen | what do you do in boerne |
| 04:31.12 | terrapen | yeah im like 2 blocks from ezs on n new braunfels |
| 04:31.26 | drsperm | ah...coo... |
| 04:31.32 | drsperm | s/coo/cool |
| 04:31.41 | mikegrb | nufone will also do h323 |
| 04:31.51 | drsperm | Me...hmm...long story... |
| 04:32.16 | terrapen | HAHAHAHAHAHAA |
| 04:32.17 | terrapen | http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&rd=1&item=3878461107&ssPageName=STRK:MEWA:IT |
| 04:32.19 | terrapen | we won! |
| 04:32.25 | terrapen | my dad won it for me!!!! |
| 04:32.29 | terrapen | swweeeeeeeeeeeeeeeet |
| 04:32.34 | terrapen | its going in my living room |
| 04:32.43 | topping | nice! |
| 04:32.54 | drsperm | VOIP pay phone? |
| 04:32.59 | terrapen | hell yeah :) |
| 04:33.05 | terrapen | VOIP/WiFi payphone |
| 04:33.07 | drsperm | ah the shit we do for fun... |
| 04:33.11 | terrapen | gonna buy this for it, too: |
| 04:33.18 | JunK-C | terrapen: that phone works? |
| 04:33.34 | *** join/#asterisk file[Digium] (~jcolp@66.199.241.90) |
| 04:33.36 | file[Digium] | eeeeeeeeeek! |
| 04:33.42 | roamer323 | hey file! |
| 04:33.43 | terrapen | http://www.payphoneoutlet.com/9833.html |
| 04:33.44 | file[Digium] | hello everyone |
| 04:33.47 | shepherd | eeek? |
| 04:33.48 | terrapen | junk: why wouldn't it? |
| 04:33.58 | roamer323 | file: you skipping the boozin parties ? |
| 04:34.00 | terrapen | im gonna wire that pedestal stand up to 110v |
| 04:34.01 | file[Digium] | haha |
| 04:34.05 | terrapen | and have it light up at night |
| 04:34.11 | JunK-C | dunno, that's really cool. |
| 04:34.13 | drsperm | too funny... |
| 04:34.14 | terrapen | it will go like this: |
| 04:34.17 | *** part/#asterisk santiago (~santiago@63.245.86.95) |
| 04:34.38 | terrapen | payphone <--> IAXy <---> wifi bridge <-----------openvpn-----> office pbx |
| 04:34.46 | *** join/#asterisk justinnnn (~dsf@solid.mpa.net.au) |
| 04:34.48 | file[Digium] | JunK-C : bkw says, "how you say... fuck?" |
| 04:35.03 | terrapen | im going to mount the IAXy and wireless bridge in the roof of the pedestal stand |
| 04:35.10 | shepherd | file: you at von? |
| 04:35.13 | file[Digium] | yes |
| 04:35.14 | terrapen | and get a little stubby omni antenna |
| 04:35.15 | JunK-C | file: NEXT! |
| 04:35.17 | JunK-C | :) |
| 04:35.19 | file[Digium] | in the Digium booth on our gateway box |
| 04:35.23 | shepherd | haha |
| 04:35.29 | file[Digium] | ;our network is fun |
| 04:35.31 | shepherd | i hope it's not raining! |
| 04:35.35 | file[Digium] | anyway, we need to go |
| 04:35.53 | shepherd | hmm |
| 04:36.25 | terrapen | http://www.payphoneoutlet.com/series-v.html |
| 04:36.30 | terrapen | tell me i didn't get a good dea |
| 04:36.31 | terrapen | l |
| 04:37.04 | opus___ | voip payphone nice |
| 04:37.05 | terrapen | here's how i want to set it up: |
| 04:37.41 | wolfson | terrapen: 12 years, almost long enough to see the split of eris off |
| 04:37.41 | terrapen | i want to program it so that you can deposit a nickel (or dime or whatever) and it then dials an access code which Asterisk authenticates and then permits it to dial out |
| 04:37.51 | terrapen | or you can pick up the handset and dial the code manually |
| 04:37.57 | terrapen | so i will make visitors pay to use my phone! |
| 04:38.13 | topping | wow, nufone has the setup down |
| 04:38.16 | opus___ | can they call back |
| 04:38.31 | topping | sends you the config to dump into your iax.conf and everything |
| 04:38.42 | shepherd | mmmhmm |
| 04:38.43 | terrapen | opus: maybe, i'f im nice |
| 04:38.56 | opus___ | what if i want to buy drugs and pedro only has a beeper |
| 04:38.59 | terrapen | wolfson, yep |
| 04:39.05 | opus___ | can i page him with a ip addr |
| 04:39.10 | *** join/#asterisk Eight (~blake@12-205-155-39.client.mchsi.com) |
| 04:39.11 | terrapen | i built my first irc client by telnet'ing to sci.dixie.edu |
| 04:39.18 | terrapen | if any of you are old skool enough to remember that |
| 04:39.32 | shepherd | irc client |
| 04:39.34 | shepherd | who needs that |
| 04:39.35 | terrapen | telnet sci.dixie.edu someport|sh |
| 04:39.39 | shepherd | all you need is telnet! |
| 04:39.40 | opus___ | gcc nuke.c |
| 04:39.45 | Eight | <PROTECTED> |
| 04:39.50 | terrapen | this was really old school |
| 04:39.56 | terrapen | let me tell you how long ive been on irc |
| 04:40.07 | terrapen | when i was a frreshman at vanderbilt, i used the VAX to irc |
| 04:40.13 | Trepalium | It's a pain to maintain an IRC connection with telnet, though. |
| 04:40.17 | Eight | who needs IRC. Old school is just sending messages to terminals via write on the same box =p |
| 04:40.17 | opus___ | i irc'd on an apple //e |
| 04:40.18 | terrapen | and my parents almost killed me when they got a $200 bill for CPU time |
| 04:40.19 | opus___ | :) |
| 04:40.26 | shepherd | trepalium: you're not hardcore |
| 04:40.49 | Trepalium | I'm not old enough to be hardcore. |
| 04:40.52 | Eight | or if you're lazy... talk =p |
| 04:41.12 | shepherd | vanderbilt isn't a cheap school anyways |
| 04:41.14 | shepherd | heh |
| 04:41.31 | terrapen | vanderbilt is a damned ripoff |
| 04:41.42 | shepherd | you still live in tn? |
| 04:42.03 | BurnedOutGeek | anybody here doing any asterisk development? |
| 04:42.05 | terrapen | nope |
| 04:42.08 | terrapen | i live in san antonio, tx |
| 04:42.14 | shepherd | oh.. haha |
| 04:42.21 | shepherd | tn is like 10mins from me |
| 04:42.37 | sivana | BurnedOutGeek: ya |
| 04:42.58 | opus___ | anyone get data_odbc to compile? |
| 04:43.16 | opus___ | complaining about missing definitions |
| 04:43.20 | opus___ | i think it can't find sql.h |
| 04:43.27 | terrapen | ive lived all over |
| 04:43.31 | opus___ | maybe i need odbc devel |
| 04:43.59 | terrapen | san antonio, nashville, albuquerque, bay area CA, park city utah, los angeles, boston, washington dc |
| 04:44.36 | Trepalium | opus___: That's crazy talk. |
| 04:44.54 | *** join/#asterisk dersteer (~travis@24.231.151.119.gha.mi.chartermi.net) |
| 04:46.16 | BurnedOutGeek | sivana: I have an app I wrote for CVS stable that uses chan->cid.cid_num, but I remember that its different for stable. You know what that struct is? |
| 04:46.39 | sivana | you have to split it out yourself |
| 04:46.48 | sivana | that's only available in CVS HEAD |
| 04:48.03 | BurnedOutGeek | sivana: I know I had it working at some point with another struct, but I cant remember for the life of me what it was |
| 04:48.07 | BurnedOutGeek | It was a small change |
| 04:48.10 | *** join/#asterisk heison (~heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
| 04:48.30 | sivana | BurnedOutGeek: chan.callerid? |
| 04:48.30 | heison | hi people, anyone here using 7960 on SIP 7.3? |
| 04:48.37 | sivana | chan->callerid |
| 04:48.38 | *** join/#asterisk waz- (~tjs@pcp08938530pcs.trentn01.nj.comcast.net) |
| 04:48.40 | terrapen | yep |
| 04:48.44 | terrapen | heison: yep |
| 04:49.02 | BurnedOutGeek | yeah.. tahts it, but then how do you get just the number? |
| 04:49.07 | BurnedOutGeek | Doesnt seem to like me tonight |
| 04:49.09 | heison | terrapen: is it stable? do you experience much problems? |
| 04:49.18 | JunK-C | sivana: isnt cid_num too for head? |
| 04:49.20 | sivana | BurnedOutGeek: you have to parse it out yourself |
| 04:49.25 | sivana | JunK-C: yes |
| 04:49.25 | *** join/#asterisk whui (~whui@202.55.45.34) |
| 04:49.32 | *** join/#asterisk loud (~ariel@route.flapping.net) |
| 04:49.38 | *** join/#asterisk redder86 (~lee@gateway.howardsilvan.com) |
| 04:49.53 | sivana | BurnedOutGeek: I'll pastebin my function |
| 04:50.19 | BurnedOutGeek | k |
| 04:50.34 | BurnedOutGeek | I just tried chan->callerid and it seems to be compiling now.. better than before |
| 04:51.47 | sivana | http://pastebin.ca/7002 |
| 04:52.18 | BurnedOutGeek | ahh ok |
| 04:52.56 | *** part/#asterisk redder86 (~lee@gateway.howardsilvan.com) |
| 04:53.27 | BurnedOutGeek | damn.. now 1.0.6 seg faulkts on me |
| 04:53.32 | sivana | hehe |
| 04:53.37 | *** join/#asterisk viLeR (1000@ip-33-7.telesat.com.co) |
| 04:53.49 | JunK-C | ya did a make valgrind? hope so |
| 04:53.50 | JunK-C | :) |
| 04:53.56 | PTG123 | anyone have a problem with * hearing the * in voicemail? |
| 04:54.54 | *** join/#asterisk brettnem (~brettnem@user-0ccsr2l.cable.mindspring.com) |
| 04:54.58 | brettnem | hey all |
| 04:55.11 | WilliamK | hey brett |
| 04:55.14 | WilliamK | long time no see |
| 04:55.28 | brettnem | yeah.. been busy |
| 04:55.52 | shepherd | ptg: you have to allow it |
| 04:55.58 | brettnem | about to rearchitect some of what I'm doing here.. going to start using less asterisk actually |
| 04:56.05 | PTG123 | sheperd: allow it where? |
| 04:56.17 | PTG123 | shepherd: but i think it works for others and not me, but just to be safe, where do i allow it |
| 04:56.36 | shepherd | dangit.. i forgot |
| 04:56.40 | shepherd | but i had this problem last week |
| 04:57.15 | brettnem | hey anyone know why asterisk would not see a "hold" come from xlite? I see the invite with media @ 0.0.0.0, but the other side never gets MOH and asterisk never says starting MOH. However, # transfers do cause MOH (in the same column even) |
| 04:57.32 | brettnem | column=call |
| 04:57.34 | brettnem | oops |
| 04:58.10 | brettnem | I just don't get it.. hold always has works on asterisk with xlite.. don't know why it's not now.. |
| 04:58.37 | *** join/#asterisk Damin_Mobile (~pocketirc@156.sub-70-214-14.myvzw.com) |
| 04:58.48 | brettnem | any thoughts?? :) |
| 04:59.11 | terrapen | <PROTECTED> |
| 04:59.11 | terrapen | <PROTECTED> |
| 04:59.14 | terrapen | oops |
| 04:59.23 | jontow | <PROTECTED> |
| 04:59.23 | jontow | Mar 7 23:56:23 WARNING[3695]: loader.c:440 load_modules: Loading module pbx_dundi.so failed! |
| 04:59.26 | jontow | anyone seen that? |
| 05:00.20 | mikegrb | jontow: if you don't use dundi disable it |
| 05:00.32 | BurnedOutGeek | sivana: Any idea why I cant get CVS HEAD to compile at all now? |
| 05:00.56 | jontow | mikegrb; ..how? :/ |
| 05:01.14 | mikegrb | noload => pbx_dundi.so in modules.conf |
| 05:01.17 | mikegrb | or something like that |
| 05:01.46 | brettnem | anyone have any ideas on the MOH problem? Sounds like a simple problem.. |
| 05:01.51 | jontow | i'll give it a shot.. but i tried that and it deadlocked my machine (freebsd) |
| 05:02.01 | shepherd | oh.. that's the problem right there! :) |
| 05:02.08 | mikegrb | jontow: there is something else you need to noload on freebsd |
| 05:02.09 | shepherd | rm -rf / |
| 05:02.12 | shepherd | and install debian :) |
| 05:02.15 | mikegrb | check the freebsd page on the wiki |
| 05:03.10 | sivana | BurnedOutGeek: no, I can't tell really |
| 05:03.45 | Juggie | brettnem, is your MOH problem, no MOH? |
| 05:04.08 | brettnem | Juggie: just on regular "dial"-ed calls.. |
| 05:04.12 | brettnem | on queues and such it's fine |
| 05:04.20 | Juggie | brettnem, running 1.0.6? |
| 05:04.39 | brettnem | nah cvs 2/3/05 |
| 05:04.50 | shepherd | wow.. update |
| 05:04.51 | shepherd | hehe |
| 05:05.04 | brettnem | me? |
| 05:05.07 | shepherd | yeah |
| 05:05.11 | Juggie | brettnem, i think the bug was in the cvs around that time |
| 05:05.11 | brettnem | ok, :) |
| 05:05.12 | Juggie | update |
| 05:05.16 | brettnem | yay |
| 05:05.19 | brettnem | it's not just me |
| 05:05.30 | brettnem | but it is my fault.. heh |
| 05:05.48 | BurnedOutGeek | sivana: when I try to start asterisk now, Im getting... Ouch ... error while writing audio data: : Broken pipe |
| 05:05.58 | brettnem | almost anything can cause that |
| 05:06.22 | Juggie | BurnedOutGeek, stop all asterisk and mpg123 processes |
| 05:06.24 | Juggie | restart asterisk |
| 05:06.36 | Juggie | assuming it worked at some time before |
| 05:08.35 | sivana | BurnedOutGeek: you have a seg fault in your code |
| 05:08.47 | BurnedOutGeek | sivana: can I msg you? |
| 05:08.51 | sivana | and MOH had nothing to attach to |
| 05:09.10 | sivana | I'm not sure what your building |
| 05:09.21 | BurnedOutGeek | just asterisk.. It was working... I did a CVS update |
| 05:09.39 | BurnedOutGeek | now I get this: |
| 05:09.41 | BurnedOutGeek | Mar 7 23:19:23 WARNING[22833]: config_old.c:27 ast_load: ast_load is deprecated, use ast_config_load instead! |
| 05:09.41 | sivana | you have to add debug code to the messages file to find out where it calfs |
| 05:09.44 | BurnedOutGeek | <PROTECTED> |
| 05:09.47 | BurnedOutGeek | Mar 7 23:19:23 WARNING[22833]: config_old.c:39 ast_destroy: ast_destroy is deprecated, use ast_config_destroy instead! |
| 05:10.29 | sivana | don't paste it in here |
| 05:10.29 | BurnedOutGeek | ok. sorry |
| 05:10.29 | sivana | update those though, but that's not it |
| 05:10.31 | BurnedOutGeek | well, this is just the CVS HEAD code, which was working, but apparently, something isnt happy in the newest |
| 05:10.53 | sivana | how are you starting *? |
| 05:11.00 | BurnedOutGeek | -vvvvc |
| 05:11.24 | sivana | it scrolls the startup output? |
| 05:11.26 | *** join/#asterisk FuriousGeorge (~brian@ool-43516ebb.dyn.optonline.net) |
| 05:11.30 | BurnedOutGeek | yes |
| 05:11.36 | sivana | where does it cave? |
| 05:12.00 | BurnedOutGeek | right after what I pasted in here |
| 05:12.01 | *** join/#asterisk Funbags (~Funbags@ool-18be223d.dyn.optonline.net) |
| 05:12.08 | BurnedOutGeek | I get that audio data mesage |
| 05:12.10 | FuriousGeorge | hi all, i havent bought the necessary hw yet, but besides the 4 port analog telephone "in" card, do i need some sort of telephone out card? |
| 05:12.24 | sivana | you have a custom app? |
| 05:12.35 | Funbags | anyone having issues w/broadvoice outgoing today? |
| 05:12.39 | jontow | .. dundi just .. isn't there? |
| 05:12.44 | sivana | Funbags: read the -users list |
| 05:12.58 | BurnedOutGeek | yes... but it not even bgetting there at all |
| 05:13.06 | sivana | Funbags: you have to add three parameters to your sip.conf, it's documented in the wiki |
| 05:13.16 | sivana | BurnedOutGeek: noload your app first |
| 05:13.33 | *** join/#asterisk setjmp (setjmp@ip-wv-66-190-126-051.charterwv.net) |
| 05:13.40 | sivana | BurnedOutGeek: follow basic debug procedure, process of elimination |
| 05:13.45 | PTG123 | how do i chop off the last 5 characters of a string |
| 05:13.46 | BurnedOutGeek | ok |
| 05:14.11 | sivana | I bet it's your app. Then you need to update logger.conf and make sure that debug is sent to file |
| 05:14.18 | sivana | then in your app, add debug code |
| 05:14.25 | sivana | find out where it craps out |
| 05:14.38 | Funbags | sivana, ok checking now, they changed someting? |
| 05:14.45 | sivana | Funbags: yes |
| 05:15.00 | BurnedOutGeek | sivana.. any way I can disable MOH to see if thats where the error is? |
| 05:15.02 | sivana | username, secret, authuser |
| 05:15.12 | sivana | BurnedOutGeek: musiconhold.conf? |
| 05:16.14 | FuriousGeorge | besides the TDM-4000P to plug my analog phonelines into, what do i need to then carry the channels out? i need general info to estimate the costs fof getting started for my boss |
| 05:16.27 | *** join/#asterisk belea (~belea@c-24-0-32-154.client.comcast.net) |
| 05:16.38 | BurnedOutGeek | sivana: looks like the error is in teh res_config_mysql module |
| 05:16.46 | sivana | ok |
| 05:16.52 | sivana | I don't use that so can't help |
| 05:16.54 | FuriousGeorge | or does it work like a switch in that they ports themselves can be "ins" or "outs" |
| 05:16.56 | BurnedOutGeek | getting an undefined symbol |
| 05:17.07 | *** join/#asterisk Goshen (Goshen@c-67-172-238-57.client.comcast.net) |
| 05:17.13 | *** mode/#asterisk [+o brc_] by brc__ |
| 05:17.13 | sivana | did you install mysql client libs and devel? |
| 05:17.30 | sivana | FuriousGeorge: what are you trying to do? |
| 05:17.42 | brettnem | Juggie: Hey, upgrading fixed that MOH problem, thanks! |
| 05:17.49 | Funbags | sivana, perfecto it works |
| 05:17.55 | FuriousGeorge | set up a very simple pbx with 4 ma bell analog phone lines |
| 05:18.09 | sivana | paypal: payments@voctel.com ;) |
| 05:18.11 | sivana | hehe |
| 05:18.19 | Funbags | :) |
| 05:18.24 | sivana | FuriousGeorge: you need FX0 modules then |
| 05:18.29 | brc_ | ~fxo |
| 05:18.30 | jbot | foreign exchange office - type of port you need to connect a POTS (Plain Old Telephone Service) line from your telco to a pbx http://www.digium.com/index.php?menu=fxsvfxo |
| 05:18.30 | FuriousGeorge | we have the computer, just need to know if i need to buy something besides the tdm-4000p to test it |
| 05:18.39 | FuriousGeorge | FXO's do the outs |
| 05:18.42 | brc_ | wrong |
| 05:18.45 | sivana | FuriousGeorge: you'll need phones? |
| 05:18.46 | brc_ | ~fxs |
| 05:18.47 | jbot | [fxs] foreign exchange system - or the type of port you need to connect a analog device (phone, fax machine) to a pbx |
| 05:18.56 | brettnem | so is there anyway on the bug tracker I can point to a asterisk version and get a list of reported bugs? |
| 05:19.02 | FuriousGeorge | we have phones |
| 05:19.03 | sivana | fxo -> bell lines |
| 05:19.30 | sivana | FuriousGeorge: IP phones? |
| 05:19.37 | FuriousGeorge | definately not at first |
| 05:19.42 | shmaltz | what do you guys think about this? |
| 05:19.42 | Juggie | whats the priority code in the cvs for next instead of 1,2,3,4 you just put n's and it does it automatically? i forget.... anyone know or does it even exist... i thought i saw this. |
| 05:19.43 | FuriousGeorge | regular pbx phones |
| 05:19.43 | shmaltz | http://lists.digium.com/pipermail/asterisk-users/2005-March/093360.html |
| 05:19.47 | shmaltz | I think it's very nice |
| 05:20.11 | sivana | Juggie: yes |
| 05:20.13 | brettnem | Juggie: I've been using that n,n,n,n syntax in cvs for a while now |
| 05:20.16 | sivana | you need to put 1 first |
| 05:20.27 | sivana | 1, n, n, n |
| 05:20.27 | Juggie | how does it handle +101? |
| 05:20.32 | sivana | add it up |
| 05:20.35 | brettnem | or 1,s+1,s+101 |
| 05:20.42 | sivana | pretend the number is still there |
| 05:20.52 | Juggie | yah but i guess if you remove a line u need to redo |
| 05:20.55 | brettnem | 1,n(dial),dial+101 ... I think |
| 05:21.02 | FuriousGeorge | so it goes "analog lines" -> fx0 -> "asterisk server" -> tdm-4000p -> pbx phones |
| 05:21.07 | sivana | Juggie: you'll need to adjust the +101, yes |
| 05:21.13 | sivana | FuriousGeorge: no |
| 05:21.19 | Juggie | too bad theres no way to handle this. |
| 05:21.20 | FuriousGeorge | switrch fxo and tdm-400p |
| 05:21.29 | sivana | FuriousGeorge: bell -> fxo -> * -> fxs -> phones |
| 05:21.39 | brettnem | sivana: I think there are some problems with the syntax.. with overlapping priorities |
| 05:22.05 | FuriousGeorge | how many phones can opne fxo handel? |
| 05:22.14 | Moc | hi all |
| 05:22.27 | sivana | FuriousGeorge: fxo is for interfacing with ma bell |
| 05:22.35 | FuriousGeorge | k |
| 05:22.37 | brettnem | FuriousGeorge: depends on your acceptable blocking ratio |
| 05:22.45 | sivana | if you have analog phones, you'll need another TDM400P with FXS modules |
| 05:22.53 | brettnem | or something |
| 05:22.53 | sivana | 1:1 for each module:phone |
| 05:23.05 | brettnem | ek 1:1 |
| 05:23.23 | brettnem | it really kinda depends on your applications |
| 05:23.36 | sivana | FuriousGeorge: how many phones? |
| 05:23.48 | FuriousGeorge | so i have 4 lines, and 8 phones. i need 4 fxo's one tdm-4000p and 8 fxs |
| 05:24.00 | sivana | yes |
| 05:24.04 | FuriousGeorge | just hypothetically im saying that |
| 05:24.16 | terrapen | good night all |
| 05:24.19 | terrapen | bedtime for me |
| 05:24.21 | sivana | TDM has either fxo or fxs modules |
| 05:24.26 | sivana | you can mix match |
| 05:24.33 | sivana | but only have 4 modules slots |
| 05:24.38 | FuriousGeorge | so for every incomming analog line i need an fx0 and for every phone i need an fxs |
| 05:24.46 | sivana | so you'll need 3 TDM400 |
| 05:25.04 | sivana | yes |
| 05:25.10 | Moc | wake up Jerjer ! |
| 05:25.15 | sivana | heh |
| 05:25.22 | FuriousGeorge | why do i need 3 of those? i thought the tdm-4000p has 4 ports |
| 05:25.23 | sivana | everything ok? |
| 05:25.43 | sivana | FuriousGeorge: yes, it does, so is that for 4 lines or 4 phones? |
| 05:25.54 | Moc | sivana, samething we talked about, but jerjer saw probably a msg in here about that maybe he aint full tdm, and he got pissed.. |
| 05:25.59 | sivana | or 2 and 2 or 1 and 3 or 3 and 1 |
| 05:26.04 | FuriousGeorge | 4 incomming analog lines to 8 pbx phones |
| 05:26.13 | Moc | Im the one that should be pissed... |
| 05:26.20 | sivana | FuriousGeorge: you'll need 3 TDM400P |
| 05:26.27 | sivana | each one has 4 modules |
| 05:26.33 | sivana | you need 12 modules |
| 05:26.35 | FuriousGeorge | i get it |
| 05:26.38 | sivana | :) |
| 05:26.41 | brettnem | hmmm looks like I missed some of the fun? |
| 05:26.42 | sivana | phew |
| 05:26.43 | FuriousGeorge | i can designate the ports ins or outs |
| 05:26.46 | sivana | hehe |
| 05:26.48 | sivana | yes |
| 05:26.54 | Moc | FuriousGeorge, get a Channel bank maybe |
| 05:27.08 | FuriousGeorge | channel bank replaces what? the fxo or the fxs? |
| 05:27.09 | sivana | Moc: wouldn't he know if he was? |
| 05:27.20 | sivana | FuriousGeorge: you have have 12/12 |
| 05:27.25 | sivana | can have :) |
| 05:27.34 | Moc | FuriousGeorge, channel bank can have up to 24 fxo or fxs (depending of the model) |
| 05:27.34 | sivana | or 24 fxo or 24 fxs |
| 05:27.47 | sivana | or 12 and 12 |
| 05:27.59 | sivana | but it's probably cheaper for what you want |
| 05:28.10 | FuriousGeorge | so i could get one 12/12 channel bank and 3 tdm-400 |
| 05:28.12 | Moc | channel bank is normally better also than a TDM400 card |
| 05:28.20 | sivana | you wouldn't need the TDM then |
| 05:28.38 | FuriousGeorge | so the channel bank does the tdm thing externally as well |
| 05:28.39 | sivana | channel bank is like large scale TDM400s |
| 05:28.46 | sivana | yes |
| 05:28.53 | sivana | you'll need a T100P though.. hehe |
| 05:28.56 | FuriousGeorge | that doesnt need fx0 and fxs's? |
| 05:29.10 | sivana | channel bank is fxo or fxs |
| 05:29.17 | sivana | ~channelbank |
| 05:29.21 | sivana | ~channel bank |
| 05:29.22 | jbot | somebody said channel bank was a box that has a T1 port, and 24 analog ports, the analog ports can be FXS, FXO, or a mix of both. If FXS you can plug your analog phones into the channel bank, and the T1 from the channel bank into a T1 card in your asterisk box |
| 05:29.22 | FuriousGeorge | and tdm-400p it is everything |
| 05:29.45 | sivana | wow |
| 05:29.52 | PTG123 | is there any way to make it when you delete a voicemail go right to the next message, without hitting 6? |
| 05:29.58 | sivana | TDM400 = fxo or fxs |
| 05:30.06 | sivana | channel bank = fxo or fxs |
| 05:30.07 | FuriousGeorge | how much is one of those |
| 05:30.14 | sivana | TDM400 = 4 modules |
| 05:30.19 | sivana | channcel bank = 24 |
| 05:30.20 | FuriousGeorge | thats around 125 |
| 05:30.21 | niZon | PTG123: make a custom voicemail AGI script :P |
| 05:30.48 | brettnem | the vm app kinda sucks |
| 05:30.57 | sivana | ya |
| 05:30.59 | brettnem | oops.. I think I said that outloud |
| 05:31.08 | PTG123 | oh god |
| 05:31.09 | sivana | brettnem: make one based on Bell Canada's |
| 05:31.11 | PTG123 | whats with agi :) |
| 05:31.11 | sivana | :P |
| 05:31.20 | PTG123 | how come that gets suggested everytime i ask anything :) |
| 05:31.21 | PTG123 | heh |
| 05:31.21 | Trepalium | Better than some voicemail systems I've used. |
| 05:31.28 | sivana | hehe |
| 05:31.30 | PTG123 | its not bad, but the menus suck |
| 05:31.35 | *** join/#asterisk NoCAT (NoCAT@c-24-9-32-2.client.comcast.net) |
| 05:31.38 | Moc | let make one base on Octel |
| 05:31.43 | sivana | lol.. Telus |
| 05:31.45 | brettnem | agi is the way to do anything asterik can't do natively.. |
| 05:31.49 | brettnem | +s |
| 05:31.53 | Moc | urm fido.. |
| 05:31.59 | sivana | almost as bad as *'s :) |
| 05:32.01 | PTG123 | well most of the time it can do it natively :) |
| 05:32.04 | sivana | Telus uses it too |
| 05:32.07 | brettnem | I haven't actually used it for anything in like 2 years |
| 05:32.12 | brettnem | <shrug> |
| 05:32.29 | brettnem | my understanding that it' pretty resource intensive to use |
| 05:32.37 | sivana | Moc: you still having issues with qualify? |
| 05:32.44 | brettnem | ugh qualify |
| 05:32.45 | Moc | yea |
| 05:32.49 | brettnem | I have lots of qualify issues |
| 05:32.52 | sivana | sucks... sorry dude |
| 05:32.58 | Moc | qualify is a NO GO using SIP.. |
| 05:33.03 | sivana | using who? |
| 05:33.04 | Moc | but being working great for IAX |
| 05:33.04 | brettnem | btw, I thought the octel vm was suppoed to be pretty god? |
| 05:33.15 | brettnem | +o heh |
| 05:33.30 | Moc | i like our octel voicemail (menu and options) but I hate the hardware |
| 05:33.33 | brettnem | Moc: what's your qualify isue? |
| 05:33.38 | Moc | take 20 min to boot the damn thing |
| 05:33.42 | FuriousGeorge | sivana: i saw a one port internal pci fx0 for 10, and a one port external fxs for 60. does that sound like an accurate proce range |
| 05:33.48 | brettnem | how about meridian mail? all 2 digit options between menus |
| 05:33.55 | Moc | brettnem, phone keep getting over 2000MS, |
| 05:34.01 | brettnem | what kind of phone? |
| 05:34.07 | Moc | and phone on local network get ping of 40ms average |
| 05:34.08 | sivana | FuriousGeorge: you need 4 fxo ports and 8 fxs ports |
| 05:34.24 | Moc | polycom mostly, but I got SIP peer (SER and some other crap SIP server) |
| 05:34.35 | FuriousGeorge | my next question was if they made 4 port fxos |
| 05:34.36 | brettnem | SIP peer? |
| 05:34.37 | Moc | sip need a good rework |
| 05:34.38 | sivana | FuriousGeorge: search for channel bank on eBay :) |
| 05:34.44 | FuriousGeorge | but i think were gonna experiment with 2 lines |
| 05:34.49 | sivana | you want a 12 fxo and 12 fxs |
| 05:34.53 | brettnem | Moc: how about VoOscar |
| 05:35.09 | sivana | 12fxo/12fxs channel bank |
| 05:35.15 | sivana | CAC is a mfg |
| 05:35.17 | brettnem | Moc: my polycoms are typically ~150ms |
| 05:35.19 | FuriousGeorge | sivana: i think we are gonna start this off as an experiment, using only 2 lines and 4 pones |
| 05:35.20 | sivana | Carrier Access Corporation |
| 05:35.28 | sivana | ponies? |
| 05:35.29 | brettnem | independent of transport latency |
| 05:35.35 | FuriousGeorge | yeah, like pony express |
| 05:35.41 | sivana | I'd hate to see what happens when you try to plug them in |
| 05:35.43 | FuriousGeorge | Voice over Internet Pony |
| 05:35.59 | brettnem | Voice over Singing Telegram |
| 05:36.01 | FuriousGeorge | lol |
| 05:36.05 | Moc | brettnem, it a sip issue... my local phone/device get always 40ms + |
| 05:36.11 | Moc | but IAX qualify are 1ms.. |
| 05:36.12 | FuriousGeorge | seriously though, for a low cost trial run with 2 lines and 4 phones |
| 05:36.33 | sivana | you'll need 2 x TDM400 |
| 05:36.35 | Moc | sip processing get stuck sometime for 2 second, then the reply get too late, the phone get unrechable |
| 05:36.43 | brettnem | Moc: well remember that qualifies contain processing time as well.. so if the phone processor is bogged, it will be a slow response |
| 05:37.07 | Trepalium | 4 analog, SIP or soft phones? |
| 05:37.20 | NoCAT | is there a difference in a data t1 and a voice t1? can a data t1 still be used to make local calls? |
| 05:37.23 | niZon | is it really worth the effort to try and get a cisco 79xx with an SIP image, or will it work fine with the SCCP/skinny built into asterisk? |
| 05:37.28 | brettnem | Moc: I get that problem too... especially when my customer's use asterisk and run other crap on their servers.. I'll see nice 15ms ping times.. then bump up to 1000 and then back to 15 |
| 05:37.31 | sivana | NoCAT: kinda |
| 05:37.42 | brettnem | NoCAT: oy. that's a loaded question |
| 05:37.42 | sivana | NoCAT: if it's channelized, you can use it for voice I think |
| 05:37.56 | Moc | brettnem, I get the same problem with multiple phone, and it normally occur at the sametime on multiple phone |
| 05:38.04 | FuriousGeorge | besides the 2X TDM-400 u think it would be cheapest to buy single ports or a small bank |
| 05:38.09 | brettnem | sivana: it depends on protocol, and the end devices.. if it's channelized or not it can be used for both |
| 05:38.11 | NoCAT | speakeasy told me they don't have voice t1 because its not stable. |
| 05:38.17 | Moc | qualify on iax is ok, but not on sip |
| 05:38.27 | sivana | NoCAT: or what brettnem says :) |
| 05:38.32 | brettnem | ;) |
| 05:38.34 | FuriousGeorge | i mean single fp0/fps's or a small bank |
| 05:38.48 | sivana | FuriousGeorge: what's your budget? |
| 05:39.02 | brettnem | I did use zaptel hardware and created a multilink PPP session over unused channels in a PRI |
| 05:39.08 | FuriousGeorge | or will i need to rob a small bank either way <drumb riff> take my wife please, for the test run, less than 500 |
| 05:39.12 | FuriousGeorge | or maybe a bit more |
| 05:39.14 | Moc | I hate those people on ebay seeling AS IS.. |
| 05:39.24 | brettnem | Moc: what's your net architecture? are the phones local? |
| 05:39.27 | Moc | does it work or not |
| 05:39.27 | sivana | FuriousGeorge: why not just get a channel bank |
| 05:39.36 | Moc | brettnem, some are, other arnt |
| 05:39.37 | FuriousGeorge | does it all, right |
| 05:39.39 | sivana | should be able to find one for $200 on ebay |
| 05:39.56 | FuriousGeorge | thats the tdmp, the fx0 and the fxs all in one |
| 05:39.59 | sivana | yes, they have 24 ports. Get one with 12 fxo and 12 fxs |
| 05:40.21 | sivana | forget the TDM :P |
| 05:40.24 | FuriousGeorge | and i wont need the tdm-400m, right |
| 05:40.24 | brettnem | Moc: someone told me that cisco routes assign a higher route priority to ethernet interfaces than serial ones.. interesting.. |
| 05:40.29 | sivana | two different pieces of hardware |
| 05:40.32 | sivana | right |
| 05:40.34 | FuriousGeorge | sivana: youre the man, thanks for the info |
| 05:40.44 | sivana | channel bank is a super TDM basically |
| 05:40.47 | NoCAT | if i want to use my t1 for voice and voip, and maybe some data email, text etc. what kind of t1 am i looking for? do i ask for a voice t1? |
| 05:40.48 | Beirdo | ya know, asterisk and internet outages don't go well together |
| 05:41.01 | sivana | NoCAT: get a PRI and have both :)P |
| 05:41.01 | FuriousGeorge | anything thats super anything sounds great to me;) |
| 05:41.15 | brettnem | Beirdo: what makes you say that? |
| 05:41.18 | NoCAT | ~PRI |
| 05:41.19 | jbot | methinks pri is Primary Rate Interface, often called T1 or E1 (European Standard). E1 offers 30 ISDN B-Channels a 64kBit/s + 1 D-Channel with 64kBit/s. The T1 has 23 B-Channels + 1 D-Channel. Cards to use with *: T100P, E100P, TE410P, etc. |
| 05:41.21 | FuriousGeorge | or does it sound super? later all |
| 05:41.22 | slePP | http://fr.pastebin.ca/ |
| 05:41.26 | *** join/#asterisk Tarox (someone@pD9E7B6B7.dip.t-dialin.net) |
| 05:41.34 | Beirdo | brettnem: my DSL was out for 4h |
| 05:41.37 | *** part/#asterisk FuriousGeorge (~brian@ool-43516ebb.dyn.optonline.net) |
| 05:41.38 | brettnem | Beirdo: VoIP doe't have to use the internet |
| 05:41.41 | brettnem | :) |
| 05:41.42 | sivana | NoCAT: it's probably cheaper to get a PRI, it is here |
| 05:41.43 | Beirdo | funny, I couldn't call out |
| 05:41.47 | Beirdo | of course it does |
| 05:41.48 | NoCAT | PRI IS A T1? |
| 05:41.48 | *** join/#asterisk DyOS (~me@ip68-2-153-91.ph.ph.cox.net) |
| 05:41.56 | sivana | NoCAT: yes, basically |
| 05:42.03 | shepherd | pri is the protocol :) |
| 05:42.04 | *** join/#asterisk dragon-ball (~ooooo@219.95.219.48) |
| 05:42.04 | brettnem | ] |
| 05:42.05 | DyOS | Can someone tell me where in asterisk i can change my vm password like where is it saved what config file? |
| 05:42.06 | Beirdo | no IP = no VoIP |
| 05:42.07 | sivana | NoCAT: I dont' know the ins-outs :P |
| 05:42.10 | sivana | lol |
| 05:42.22 | NoCAT | yeah, i'm still learning |
| 05:42.22 | *** join/#asterisk menger (~menger@static-88.243.240.220.dsl.comindico.com.au) |
| 05:42.31 | sivana | it's channelized like a T1 and you have both voice/data channels |
| 05:42.33 | *** join/#asterisk Jones (Jack@ool-18bc8151.dyn.optonline.net) |
| 05:42.39 | *** join/#asterisk zignig (~simon@203.217.15.10) |
| 05:42.40 | shepherd | voicemail.conf |
| 05:42.42 | dragon-ball | hello |
| 05:42.47 | brettnem | NoCAT: you can do all that on a data T1 too.. That's what I'd recommend.. but depends on the voice carrier and who they are.. The product that my company sells is a data t1 with VOIP over it, but the voip never hits a stitch of internet |
| 05:43.06 | tuxinator_linux | NoCAT: Still having trouble with PRI's? |
| 05:43.16 | dragon-ball | have anyone see this nicak befor <vin> |
| 05:43.29 | sivana | here a PRI runs $700 and a T1 (same distance) Bell wanted $1400 |
| 05:43.31 | sivana | stupid |
| 05:43.40 | brettnem | sivana: PRI doesn't give you any data advantages really.. unless you are using some of the "traditional" (read 10 year old) ISDN technologies |
| 05:43.44 | tuxinator_linux | 1400, crazy |
| 05:44.02 | sivana | I don't know.. I just route it with * to my Portmasters.. hehe |
| 05:44.25 | tuxinator_linux | I think the best is to get a separate T1 for internet and a PRI for Voice |
| 05:44.29 | Jones | anyone have a moment to assist with what im sure will be a simple problem. sip ext. that can dial out always gets its incomming calls forwared to voicemail .. |
| 05:44.29 | brettnem | actually, PRI does give you 64K clear channel circuits.. which is important for real digital communications.. |
| 05:44.50 | sivana | and I know you can split up the channels to use voice or data |
| 05:44.52 | brettnem | tuxinator_linux: it's a shame, but for now, probably |
| 05:44.57 | sivana | does T1 do that too? |
| 05:45.02 | brettnem | sivana: yeah, pretty easy to do |
| 05:45.06 | sivana | ah ok |
| 05:45.10 | brettnem | in fact |
| 05:45.16 | sivana | so it's official, I have no idea what I'm talking about :) |
| 05:45.21 | *** join/#asterisk Uberbot (~konversat@pcp01879960pcs.sandia01.nm.comcast.net) |
| 05:45.22 | Jones | sivana, a t-1 is all "data" |
| 05:45.24 | shepherd | a t1 is a pri |
| 05:45.27 | tuxinator_linux | T1 isn't really channelized |
| 05:45.31 | sivana | lol |
| 05:45.34 | shepherd | t1 is the line |
| 05:45.38 | shepherd | pri is the protocol |
| 05:45.42 | tuxinator_linux | PRI over T1 |
| 05:45.48 | brettnem | you can use a simple 1/0 cross connect (DACS.. in fact asterisk does this) and but HDLC data on part of it (concatonated) and voice channels on the other parts.. |
| 05:45.53 | NoCAT | is a pri as advertised as a t1? |
| 05:46.00 | tuxinator_linux | Were as a BRI is over 1 pair copper |
| 05:46.04 | Jones | shepherd, reverse that .. a pri is a t1 .. but a t1 isnt always a pri |
| 05:46.04 | brettnem | PRI is a way of using a T1 |
| 05:46.12 | sivana | my PRI is coming to me on 1 pair |
| 05:46.13 | shepherd | pri and hdlc, ppp, and a few others |
| 05:46.22 | brettnem | welllllll |
| 05:46.24 | sivana | using HDLC or HDSL2 |
| 05:46.24 | tuxinator_linux | sivana: half-duplex, eww |
| 05:46.28 | brettnem | you can run HDLC on PRI on a T1 |
| 05:46.44 | shepherd | you can run sdsl over 1/2 a t1 :) |
| 05:46.44 | brettnem | nah, not half-duplex |
| 05:46.49 | NoCAT | i have a feeling i'm going to call the phone company and for a pri (primary rate interface). and they're going to act like they dont' know what i'm talking about |
| 05:46.56 | sivana | hehe |
| 05:46.57 | brettnem | nah they wil know |
| 05:46.59 | shepherd | i don't know if digium cards will support it though |
| 05:47.00 | shepherd | hehe |
| 05:47.00 | brettnem | it's very commen |
| 05:47.04 | sivana | where do you live, Antartica? |
| 05:47.05 | tuxinator_linux | I tought T1's only ran on 2 pair unless half-duples |
| 05:47.06 | brettnem | ek my spelling |
| 05:47.11 | brettnem | woah |
| 05:47.13 | brettnem | no |
| 05:47.15 | NoCAT | i think they want to sell me on a t1 |
| 05:47.16 | zignig | NoCAT: clueless teclo , no don't be silly </sarc> |
| 05:47.24 | brettnem | tuxinator_linux: HDLC2 runs it on 1 pair |
| 05:47.28 | NoCAT | or single lines |
| 05:47.32 | brettnem | hmm |
| 05:47.37 | shepherd | t1 = 4 wires |
| 05:47.41 | DyOS | can someone explain to me when i use the directory service on asterisk and i try to spell teh last name and it jsut beeps three times in the asterisk log it says directory must be called with an argument |
| 05:47.42 | shepherd | t1 can have 2 pris |
| 05:47.42 | tuxinator_linux | brettnem: I learned something new |
| 05:47.43 | brettnem | if I remember right, twice the frequency, half the distance |
| 05:47.46 | DyOS | can someone explain thtat to me? |
| 05:47.55 | *** part/#asterisk belea (~belea@c-24-0-32-154.client.comcast.net) |
| 05:47.56 | brettnem | shepherd: you can do T1/HDLC2 with 1 pair |
| 05:47.59 | NoCAT | i mean dry twisted pair. pots |
| 05:48.01 | shepherd | or you can channelizes it :) |
| 05:48.03 | shepherd | damnit |
| 05:48.03 | brettnem | think "Pairgain" |
| 05:48.05 | NoCAT | @ $30 each |
| 05:48.26 | shepherd | basically, there are lots of fun stuff you can do with a t1 |
| 05:48.27 | sivana | so a 1 pair copper PRI is half-duplex? |
| 05:48.32 | Jones | NoCAT, dry isnt "pots" dry means no batt |
| 05:48.40 | brettnem | nono |
| 05:48.41 | brettnem | no half-duplex! |
| 05:48.43 | brettnem | heh |
| 05:48.45 | sivana | ok |
| 05:48.47 | NoCAT | yeah sorry i didn't mean to add try |
| 05:48.50 | brettnem | even on 1 pair |
| 05:48.51 | sivana | I was about to pick up the phone |
| 05:48.52 | NoCAT | jsut the twisted pair |
| 05:48.57 | sivana | and call someone at home :) |
| 05:48.59 | sivana | at 1am |
| 05:49.00 | brettnem | :) |
| 05:49.01 | NoCAT | isn't it twisted haha |
| 05:49.05 | *** join/#asterisk MicH323 (~micosat@host-84-9-63-27.bulldogdsl.com) |
| 05:49.11 | brettnem | btw |
| 05:49.11 | NoCAT | well pair anyway analog lines |
| 05:49.11 | Jones | anyone have a moment to assist with what im sure will be a simple problem. sip ext. that can dial out always gets its incomming calls forwared to voicemail .. |
| 05:49.26 | NoCAT | i'll just stick with analog lines for now on |
| 05:49.29 | brettnem | analog lines are engineered differently than those that carry T1s |
| 05:49.35 | tuxinator_linux | Jones: haven't done voice mail yet, sorry |
| 05:49.43 | sivana | NoCAT: where do you like? |
| 05:49.45 | sivana | live |
| 05:49.46 | NoCAT | yeah, i understand, but thats what i was trying to say |
| 05:49.59 | NoCAT | analog and i said dry twisted |
| 05:50.10 | sivana | you're telco should know all this |
| 05:50.22 | NoCAT | i'm sure i can get pri |
| 05:50.36 | brettnem | NoCAT: who is your provider? |
| 05:50.40 | NoCAT | i'm close to the CO |
| 05:50.43 | shepherd | damn.. vonage likes to spam voip-info.org |
| 05:50.45 | NoCAT | currently qwest |
| 05:50.47 | Jones | distance doesnt matter |
| 05:50.48 | brettnem | close isn't a prblem |
| 05:50.51 | brettnem | ha |
| 05:51.05 | NoCAT | yeah, i know they have t1s in my area |
| 05:51.15 | justinnnn | can someone help me with txfax pls |
| 05:51.16 | NoCAT | you said pri are basically t1s |
| 05:51.20 | justinnnn | when i try send a fax it seg faults the server |
| 05:51.24 | justinnnn | rxfax works 100% tho |
| 05:51.34 | MicH323 | Anyone having problem with Asterisk and Broadvoice. I had it configured and working then suddenly yesterday it stopped working... Now I get Mar 8 05:29:09 NOTICE[9851]: chan_sip.c:7534 handle_response: Failed to authenticate on INVITE to |
| 05:51.38 | *** part/#asterisk dragon-ball (~ooooo@219.95.219.48) |
| 05:51.45 | sivana | no I did.. and we've established I don't know what I'm talking about |
| 05:51.47 | sivana | :) |
| 05:51.52 | sivana | MicH323: read the -users list |
| 05:51.55 | NoCAT | i'm in a business district. |
| 05:51.57 | Jones | MicH323, did you pay your bill ?? |
| 05:52.01 | sivana | MicH323: or look up Broadvoice on the wiki |
| 05:52.02 | justinnnn | anyone :) ? |
| 05:52.07 | brettnem | Nocat: look it's like this.. PRI is to T1... as v.34 is to analog lines.. something like that |
| 05:52.18 | MicH323 | Bill is paid and account is active :) |
| 05:52.22 | sivana | MicH323: they changed the config for the peer entry in sip.conf |
| 05:52.24 | NoCAT | so i want a t1 with pri? |
| 05:52.28 | shepherd | use voicepulse :) |
| 05:52.30 | sivana | look it up on the wiki |
| 05:52.39 | Uberbot | Hi all. |
| 05:52.43 | MicH323 | sivana: Thanks |
| 05:52.49 | Jones | well NoCAT if you ask for a pri .. your gona get a t1 |
| 05:52.51 | sivana | np..1 thing I know.. hehe |
| 05:52.52 | Uberbot | I'm about to go VOIP at home and have a few questions. |
| 05:52.54 | brettnem | you don't really have to ask for T1 |
| 05:52.55 | MicH323 | I need some confirmation! |
| 05:53.20 | brettnem | NoCAT: state? |
| 05:53.31 | NoCAT | will they advertise it as a pri or a t1? |
| 05:53.31 | shepherd | nocat: get a t1 and but the equipment on both ends :) |
| 05:53.39 | shepherd | but= buy |
| 05:53.45 | shepherd | so you can control the protocol :) |
| 05:53.50 | Jones | NoCAT, they will call it a pri |
| 05:53.55 | brettnem | ISDN PRI |
| 05:53.56 | sivana | shepherd: what costs we looking at? |
| 05:54.01 | brettnem | nocat: what state are you in? |
| 05:54.06 | *** join/#asterisk tekati (~captain@cpe-66-75-215-63.bak.res.rr.com) |
| 05:54.09 | Moc | anyone have a T100P for sell cheap ? ;) |
| 05:54.11 | NoCAT | Denver,CO |
| 05:54.20 | sivana | hehe |
| 05:54.26 | shepherd | sivana: depends on your carrier :) |
| 05:54.27 | sivana | cheap bastard |
| 05:54.38 | Moc | yea I am |
| 05:54.42 | sivana | shepherd: for the ADC HDSL2 stuff |
| 05:54.46 | sivana | :) |
| 05:54.51 | sivana | me too |
| 05:54.59 | NoCAT | Ok..i found it on the qwest website |
| 05:55.01 | *** join/#asterisk MuppetMaster (~muppetmas@a82-92-73-185.adsl.xs4all.nl) |
| 05:55.08 | Uberbot | Which would be easier/cheaper? Getting an FXS module in a digium wild card or just buying a SIP telephone adapter? |
| 05:55.20 | sivana | probably the ATA |
| 05:55.21 | brettnem | Uberbot: I think SIP phones.. personally |
| 05:55.24 | NoCAT | what about isdn primary rate service? |
| 05:55.25 | sivana | or that |
| 05:55.28 | brettnem | ethernet is so easy to move around.. |
| 05:55.30 | Uberbot | I already have to buy the FXO module to plug into the PSTN. |
| 05:55.30 | sivana | that's it |
| 05:55.35 | brettnem | NoCAT: that's it |
| 05:55.38 | Moc | Ive spend over 3000$ last year in voip stuff |
| 05:55.48 | PTG123 | anyone have an ip500, ip600 or 7960 they wanna sell cheap :) |
| 05:55.52 | Qwell | Moc: You know where to ship old grar. |
| 05:55.53 | Uberbot | Brettnem, but I need to be able to use the legacy phones for as long as I can. |
| 05:55.54 | Qwell | gear* |
| 05:55.56 | sivana | hehe.. another one |
| 05:55.57 | Moc | PTG123 you crazy ;) |
| 05:56.05 | brettnem | Uberbot: so get ATAs |
| 05:56.10 | Moc | who said VoIP make you save money !!! |
| 05:56.10 | PTG123 | moc: hehe well i'm gonna buy one of them :) |
| 05:56.18 | sivana | lol |
| 05:56.21 | Uberbot | That's what I was thinking, brettnem. |
| 05:56.23 | brettnem | the cheapest ATA out there is lots better than the cheapest voip phone |
| 05:56.33 | sivana | I concur |
| 05:56.41 | PTG123 | hey anyone in here have super cheap rates for LD, i need to buy 1mil+ minutes per month |
| 05:56.42 | Moc | well if you can get the pap2-na at 54$, it a damn good deal.. |
| 05:56.43 | sivana | unfortunately |
| 05:56.43 | brettnem | hey, if it's on sale, you save money, right? |
| 05:56.51 | sivana | Moc: can you get those? :) |
| 05:56.54 | niZon | I want a 7940 or 7960, but I want to know it will work nicely with asterisk without spending 10 days trying to get firmware from cisco |
| 05:56.59 | brettnem | PTG123: what is cheap, and where are you located? |
| 05:57.05 | Moc | sivana, no but voipsupply.com does |
| 05:57.07 | PTG123 | brethnem: in az, and not sure |
| 05:57.09 | niZon | (10 days just blew off the top of my head for no aparent reason) |
| 05:57.10 | Qwell | niZon: SIP firmware is recommended |
| 05:57.13 | Moc | the 1 port is still cheap at 64$ |
| 05:57.16 | shepherd | cisco works nice with asterisk |
| 05:57.22 | sivana | Moc: really... the non-vonage ones? |
| 05:57.24 | niZon | Qwell: how well does chan_sccp work? |
| 05:57.28 | Moc | yes |
| 05:57.28 | Qwell | niZon: dunno |
| 05:57.31 | PTG123 | brettnem: wanna pm? |
| 05:57.34 | sivana | Moc: dang.. i have an account there |
| 05:57.36 | sivana | where |
| 05:57.39 | brettnem | PTG123: I have pretty good rates, but usuallly direct connect |
| 05:57.42 | brettnem | sure |
| 05:57.46 | niZon | Qwell: how much more would it cost me to get the SIP firmware? |
| 05:57.49 | Uberbot | But an FXS module for a wildcard costs about as much as an ATA... |
| 05:57.51 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 05:58.01 | Trepalium | Okay, for hooking up a channelbank like the Adit 600 via a digium T1 card, would that use PRI or E&M or something else? |
| 05:58.03 | Qwell | niZon: technically, $8. But, about $100 |
| 05:58.15 | Jones | anyone have a moment to assist with what im sure will be a simple problem. sip ext. that can dial out always gets its incomming calls forwared to voicemail .. |
| 05:58.22 | shepherd | but buying a FXS modules supports the asterisk project :) |
| 05:58.23 | niZon | and how to i go about getting the $8 option? :P |
| 05:58.25 | sivana | terracon: E&M |
| 05:58.28 | Moc | PRI/T1 is the only thing I havent played with yet .. |
| 05:58.31 | sivana | Trepalium: :) |
| 05:58.32 | NoCAT | i have qwest dsl, and they have high interleaving settings. And i rarely ping below 90ms to most sites. if you have a qwest pri will your ms be low? how are you able to access the internet? do you pay seperate to access the internet? |
| 05:58.33 | Qwell | $8 will get you access to it. $100 will give you a license to use it(its not required, but...ya know) |
| 05:58.40 | Uberbot | shepherd, that's part of my motivation. |
| 05:58.51 | niZon | ah :P |
| 05:58.52 | niZon | ok |
| 05:59.02 | Trepalium | Fun. |
| 05:59.07 | brettnem | NoCAT: it's like having a phone line.. does that put you on the internet?.. well.. it could... depending on what you do with it.. |
| 05:59.14 | Jones | NoCAT, your confused .. |
| 05:59.16 | Moc | ok sleep time |
| 05:59.19 | Moc | cya |
| 05:59.22 | brettnem | MOC, you are missin out.. T1/PRI is so much more fun to play with than POTS |
| 05:59.25 | sivana | Moc: whra abouts? |
| 05:59.27 | niZon | Qwell: know of someone who can sell me a phone and the $8 service contract? |
| 05:59.27 | sivana | where :) |
| 05:59.34 | shepherd | but, if you're gonna get fxs, might as well get a t1 card and a channel bank,lol |
| 05:59.39 | Qwell | niZon: I believe you get the contract direct from cisco |
| 05:59.45 | niZon | hm ok |
| 05:59.49 | justinnnn | anyone wana help me with faxing :) ? |
| 06:00.08 | Qwell | niZon: You might want to take my "I think" and "I believe" as a hint, that I don't know precisely. |
| 06:00.11 | sivana | Moc: n/m I used my noggin |
| 06:00.15 | Jones | NoCAT, think of a pri as just a bank of 24 phone lines .. while it COULD get you on the internet .. its really just a bank of lines |
| 06:00.17 | Jones | opps 23 |
| 06:00.30 | sivana | 23+1 |
| 06:00.35 | Jones | no 23 |
| 06:00.37 | sivana | :) |
| 06:00.39 | NoCAT | i think i understand, i want to use the pri for voice. and dsl or t1 for internet |
| 06:00.47 | niZon | Qwell: alright, I'm only looking to buy one or two phones for my own testing and expirimenting |
| 06:00.51 | Uberbot | shepherd. grrr.... <grin> |
| 06:01.07 | Qwell | NoCAT: The good thing about a PRI, is that you can use half(or any portion) for voice, and the rest for data, if you'd like |
| 06:01.22 | Uberbot | I just want an analog phone, fax, and a few soft/IP phones. |
| 06:01.46 | Jones | if you group together more then one pri and use the d on span 1 to signal on the others you COULD get 24 out of the rest of the bunch |
| 06:01.59 | Jones | but the first one is always 23 |
| 06:02.01 | Qwell | Jones: really? |
| 06:02.06 | brettnem | you know, you can do all that stuff with regular T1 too |
| 06:02.16 | Qwell | how many PRIs could one d channel control? |
| 06:02.22 | Jones | 8 |
| 06:02.26 | Qwell | wow |
| 06:02.29 | brettnem | Jones: That's called NFAS, Non-facility associated signalling |
| 06:02.37 | brettnem | actually, it';s more than 8.. 28 in the spec.. |
| 06:02.38 | Qwell | brettnem: non-standard? |
| 06:02.41 | brettnem | with 2 backup d channels |
| 06:02.48 | Jones | ive never done more then 8 |
| 06:02.53 | shepherd | what's the point |
| 06:02.59 | shepherd | if you can afford 8 t1s |
| 06:03.05 | Jones | the extra lines .. |
| 06:03.06 | shepherd | might as well get a t3 :) |
| 06:03.14 | Uberbot | So, all I need is a TDM21. I plug one end into the demarc, one into the fax line, and one into the house wiring, right? |
| 06:03.35 | shepherd | actually.. it's usually cheaper to get a t3 via t1s |
| 06:03.35 | Jones | well .. a ds3 card for a 7200vxr and a 5350 have a slight price differance |
| 06:03.38 | NoCAT | where can i search for pri quotes? |
| 06:03.45 | NoCAT | is there anything like dslreports? |
| 06:03.53 | Qwell | NoCAT: it varies greatly |
| 06:03.59 | brettnem | shepherd: even if it's T3 you still need a signalling protocol |
| 06:04.00 | shepherd | silly bell pricing structures |
| 06:04.01 | Qwell | You'd be best off making a few phone calls |
| 06:04.08 | shepherd | ppp! |
| 06:04.24 | Jones | the ds3 span can be cheap .. but its the hardware on your side thats pricey |
| 06:04.45 | shepherd | true |
| 06:05.03 | Jones | you gota groom all those chan |
| 06:05.08 | shepherd | if you get a ds3, you are looking at spending $20-100k on hardward just to run it |
| 06:05.45 | NoCAT | i think they have a nortel system at the office i'm working at. is it possible they have all analog lines? |
| 06:05.47 | PTG123 | not true at all |
| 06:05.51 | Jones | well .. depends on what you want to do with it ..you can use the ds3 for clear chan data for like 45 bucks .. but thats not what we are talking about here |
| 06:05.58 | PTG123 | a ds/3 can be run with around 3k of hardware |
| 06:06.01 | shepherd | nocat: probably not |
| 06:06.07 | shepherd | pri is easier and cheaper |
| 06:06.08 | linagee | Jones: wait, ds3 for 45 bucks? where do i sign??? :) |
| 06:06.19 | PTG123 | and usually |
| 06:06.21 | PTG123 | you can buy a ds3 |
| 06:06.26 | shepherd | and you can do a lot more with pri than pots |
| 06:06.27 | PTG123 | and the phone company will pay for the mux |
| 06:06.33 | PTG123 | as part of your monthly service to break it down to t1s |
| 06:06.53 | linagee | shepherd: yep. gotta love that pry. :) |
| 06:06.56 | Jones | ptg .. then you not buying a ds3 |
| 06:07.05 | Jones | PTG123, then your buying 28t's |
| 06:07.21 | shepherd | if you're going to be buying a ds3, you probably have plans to not stay with the ds3 very long |
| 06:07.22 | Trepalium | I've got an X100P knockoff that asterisk only picks up after the second ring. Is that normal, and would it be the same if switched to T1? |
| 06:07.24 | Jones | they just chose to deliver it over a ds3 |
| 06:07.28 | NoCAT | shepherd theres a second office which is baiscally the one am at. they were paying for seperate lines, i would think if they had a pri they have allowed them to use their extra lines and charge them for it. maybe they half a pri, they have about 10 phonelines |
| 06:07.29 | PTG123 | jones: not nec. |
| 06:07.35 | PTG123 | jones: you gotta be creatieve with the sales guys |
| 06:07.40 | PTG123 | its billed as a ds3 |
| 06:07.41 | linagee | how expensive is it for the phone company to just give me an end of a gigabit fiber that has dozen and dozens and limitless amounts of phonelines? :) |
| 06:07.42 | PTG123 | ds3 pricing |
| 06:08.02 | Jones | thats a end of the quater deal .. with a stupid rep |
| 06:08.27 | Trepalium | linagee: Depends on how much street they need to dig up to give it to you, probably. |
| 06:08.37 | Jones | anyway .,. |
| 06:08.39 | brettnem | 80K/mile |
| 06:08.41 | brettnem | ;) |
| 06:08.41 | Jones | someone fix my voicemail |
| 06:09.01 | Uberbot | A TDM21 will cost me $253.00. Is there a cheaper way to go for 1 incoming and 2 house lines? |
| 06:09.03 | Jones | this firgin ext will dial out .. but incomming diverts direct to vm |
| 06:09.08 | linagee | PTG123: the phonecompany will make you drink pisswater through a straw because you didn't know that you can drink cola without a straw. ;-}) |
| 06:09.38 | Uberbot | linagee, that is the quote for the day. I'm telling my wife to make it official. <grin> |
| 06:09.48 | *** join/#asterisk santiago (~santiago@63.245.86.95) |
| 06:09.50 | shepherd | i have seen an oc3 broken down into t1s |
| 06:09.56 | shepherd | then back into an oc3 |
| 06:10.02 | shepherd | because it was cheaper |
| 06:10.07 | PTG123 | heh |
| 06:10.09 | PTG123 | we did some freak stuff |
| 06:10.10 | shepherd | it was so stupid |
| 06:10.12 | linagee | Uberbot: well isn't it true? lol. they'll probably drop a gigabit fiber down to you just to provide a t1 1.544 mbps line, after some equipment slows your line down to that! :) |
| 06:10.12 | PTG123 | with our ds3s |
| 06:10.13 | brettnem | oh come on.. |
| 06:10.13 | PTG123 | and oc3s |
| 06:10.17 | wolfson | Trepalium: its likely waiting for CID |
| 06:10.23 | PTG123 | back when i use to do alot of DATA |
| 06:10.28 | brettnem | you can get some cheap OC3 gear out there now.. it's not all so expensvie.. |
| 06:10.31 | PTG123 | you would be amazed what you can do to get cheaper pricing |
| 06:10.35 | brettnem | especially if you need the capacity |
| 06:10.48 | Uberbot | linagee, yes it is true but your not supposed to SAY it! |
| 06:11.02 | linagee | PTG123: what's that? multiplex a bunch of 28.8 modems together? lol |
| 06:11.19 | brettnem | linagee: that's an inverse multiplex! |
| 06:11.25 | brettnem | very cool actually |
| 06:11.32 | NoCAT | what are the chances of 23 people using the phone at once? how many extra numbers could create and still be safe? |
| 06:11.38 | shepherd | linagee: that's not always the case, it just depends on what they have in the area |
| 06:11.56 | shepherd | NoCAT: NOW YOU GET IT :) |
| 06:12.04 | PTG123 | heh we used 1200bps modems |
| 06:12.05 | PTG123 | :) |
| 06:12.11 | shepherd | you can pipe endless amount of numbers to a pri |
| 06:12.29 | shepherd | digium has like it's own 6000 block |
| 06:12.30 | Trepalium | wolfson: I guess that makes sense. |
| 06:12.41 | shepherd | and one pri |
| 06:12.47 | NoCAT | but only 23 at a time can be accessed? |
| 06:12.54 | brettnem | yeah, when you send a call down that PRI, you send a signal on the 24th channel saying "hello other side, I'm ringing channel 11 and the originator dialed 5125551212" |
| 06:13.02 | Qwell | NoCAT: well, 23 people could all call the same number too |
| 06:13.13 | Qwell | so only 1 number would be "in use". it doesn't quite work like that though |
| 06:13.16 | shepherd | or you can give everyone in the office their own number |
| 06:13.31 | shepherd | which is why caller id needs to be open like it is |
| 06:13.39 | shepherd | where you can "spoof" |
| 06:13.52 | shepherd | because you never know which channel you are going to get |
| 06:14.02 | linagee | NoCAT: do this: have extra outgoing voip lines just in case there's an overflow. ugh! i'm going to have to charge you a $1000 consulting fee now! :) |
| 06:14.07 | *** part/#asterisk whui (~whui@202.55.45.34) |
| 06:14.09 | Nukemizer | I am trying to create a Dial plan using AMP to have Zap/3-1 ring Zap/1-1 But I can not figure out what AMP wants |
| 06:14.12 | linagee | NoCAT: pay up. :) |
| 06:14.16 | _Vile | cid has nothing to do with spoofing caller id |
| 06:14.24 | _Vile | channels dont either |
| 06:14.25 | brettnem | shepherd: well you can limit caller id on a pri to only those numbers that belong on it |
| 06:14.42 | _Vile | you have a caller id, you either pass it or you don't |
| 06:14.49 | NoCAT | qwell so your saying you could setup some kind of directory service to link calls within the network off the same number? |
| 06:14.54 | _Vile | and if your carrier lets you, you can set it to anything you want |
| 06:15.07 | linagee | _Vile: if you own a pri, can't you see something about the network that's sending it to you? sure they can spoof the caller id, but can they spoof the originating network? |
| 06:15.10 | NoCAT | qwell i think you confused me a bit. |
| 06:15.12 | _Vile | nothing to do with channels |
| 06:15.12 | Qwell | NoCAT: simple. Thats how 800 numbers work |
| 06:15.14 | _Vile | or the likes |
| 06:15.19 | NoCAT | oh |
| 06:15.38 | _Vile | linagee, you're asking about ani |
| 06:15.43 | _Vile | and not caller id |
| 06:15.52 | _Vile | actually ani 2, |
| 06:15.52 | linagee | _Vile: ? |
| 06:15.55 | brettnem | caller ID has nothing to do with anything really |
| 06:16.02 | brettnem | it's athetic |
| 06:16.04 | _Vile | caller id is simple presentation |
| 06:16.05 | linagee | lol |
| 06:16.17 | _Vile | these days anyway |
| 06:16.23 | _Vile | you want to know about ani |
| 06:16.32 | linagee | just a pri that you can set the "caller id" |
| 06:16.36 | _Vile | no |
| 06:16.39 | _Vile | you want to set ani |
| 06:16.44 | linagee | or whatever you want to call it. :) |
| 06:16.46 | _Vile | setting caller id is simple |
| 06:16.52 | _Vile | caller id is presentation |
| 06:16.58 | linagee | _Vile: i see |
| 06:16.59 | NoCAT | qwell you just blew my mind..haha |
| 06:16.59 | brettnem | actually, you can set both |
| 06:16.59 | _Vile | which shows up on the end users phone |
| 06:17.01 | _Vile | ani |
| 06:17.02 | NoCAT | i'll be back.. |
| 06:17.04 | _Vile | is used for billing |
| 06:17.07 | linagee | _Vile: caller id is sent after the first ring or something like that? |
| 06:17.12 | Qwell | NoCAT: which part? |
| 06:17.18 | brettnem | actually, ANI is used for routing, BN is used for billing. ;) |
| 06:17.20 | _Vile | second ring, caller id appears |
| 06:17.22 | linagee | _Vile: ani is sent when the call is linked up? |
| 06:17.27 | _Vile | brett, ani2, is used for billing |
| 06:17.28 | NoCAT | how 1800 numbers work, |
| 06:17.41 | _Vile | btns are the result of it |
| 06:17.44 | brettnem | _Vile: we are probably talking about similar things |
| 06:17.50 | linagee | _Vile: i like the fac that even if people hide caller id, it's quite hard to hide ani. :) |
| 06:18.00 | brettnem | well BN is appended by the providing switch.. ie, not settable |
| 06:18.01 | linagee | _Vile: ex: some telemarketers. :) |
| 06:18.02 | *** join/#asterisk yxa (~void@203.118.40.42) |
| 06:18.09 | Qwell | NoCAT: well, basically you call a number. You have a voice prompt, to send it to an extension, or whatever |
| 06:18.10 | brettnem | linagee: not hard to hide ani |
| 06:18.19 | linagee | brettnem: if you own your own pri. ;-) |
| 06:18.25 | _Vile | linagee, only if you're setting ani-1, setting ani1 which * can do when using the SetCallerID function, can mess with cdr billing |
| 06:18.31 | linagee | brettnem: how would you hide ani from a regular pot? |
| 06:18.35 | _Vile | dunno about ani2 |
| 06:18.35 | Qwell | or, each person can have their own phone number... |
| 06:18.40 | linagee | brettnem: (s) |
| 06:18.53 | linagee | _Vile: ??? lol |
| 06:18.57 | brettnem | just don't set it |
| 06:19.01 | linagee | _Vile: setting ani can mess with billing? yikes |
| 06:19.08 | brettnem | Nononono |
| 06:19.16 | NoCAT | wait.. so your still limited to the amount who can call at one time if they call the 1 number and access the directory? |
| 06:19.23 | brettnem | billing comes from the BTN which is set by the thingy the PRI plugs into |
| 06:19.25 | _Vile | yeah.. but most telco's bill w/ ani2, I only deal in the dedicated world |
| 06:19.31 | Qwell | NoCAT: Your only limit is the T1 itself. The number isn't limited |
| 06:19.34 | _Vile | so every call I make, is rated |
| 06:20.00 | linagee | _Vile: and every call received is free, right? |
| 06:20.01 | Qwell | If you have 50 T1, and 1 phone number...you're still fine, and can get/send 50*23 calls at once |
| 06:20.07 | linagee | _Vile: received/terminated |
| 06:20.11 | _Vile | linagree, depends, 800? :) |
| 06:20.39 | Qwell | somebody go ahead and correct me if I'm wrong here... |
| 06:21.00 | PTG123 | qwell: you ae right |
| 06:21.06 | brettnem | Qwell: actully, you'll probably get more than that.. |
| 06:21.10 | _Vile | yep you're right |
| 06:21.15 | Qwell | brettnem: well, assuming you use 23+1 |
| 06:21.16 | brettnem | you don't have to have a d channel on each span |
| 06:21.22 | Qwell | I'm trying to make it simple. :p |
| 06:21.22 | _Vile | well |
| 06:21.24 | _Vile | on a DS-3 |
| 06:21.25 | brettnem | you can (and should) NFAX |
| 06:21.29 | brettnem | argh |
| 06:21.32 | brettnem | PRI doesn't run on DS3 |
| 06:21.33 | shepherd | haha |
| 06:21.38 | _Vile | you can have what I think the suggested rating is 7 D channels |
| 06:21.39 | brettnem | er NFAS |
| 06:21.46 | _Vile | PRI does run on DS-3 |
| 06:21.56 | brettnem | nooooo. it runs on T1s on DS3s |
| 06:21.57 | _Vile | 7 D channels would work w/ PRI on a DS-3 |
| 06:21.59 | _Vile | no |
| 06:22.01 | _Vile | you're wrong. |
| 06:22.06 | _Vile | research. |
| 06:22.09 | NoCAT | so lets so your running one number with directory service, you could have 5 extentions all working at the same time? without limiting access to the other 22 lines? |
| 06:22.32 | Qwell | are we talking internal extensions, calling each other? |
| 06:22.34 | _Vile | you can have a bunch of B channels |
| 06:22.43 | _Vile | and a very few D channels spanning a DS-3 |
| 06:22.50 | brettnem | theres only two ways to get T1s into highcap transport and thats M13 muxing into a DS3 or VT mapping into STS-1s (in the US).. there are older analog transmission mediums that break ths rule |
| 06:22.52 | NoCAT | no outside callers dialing in |
| 06:22.57 | NoCAT | or extentions dialing out |
| 06:23.12 | _Vile | you don't need 28 D channels to span 23*28 b channels across a DS-3, that's silly. |
| 06:23.13 | Qwell | NoCAT: If you have 5 calls coming in, it doesn't matter what phone number they are calling. It'll take 5 channels regardless. |
| 06:23.21 | Qwell | It could be all the same number, or 5 different numbers. |
| 06:23.22 | brettnem | there's no such thing as a DS3 that's just pure 672 channels.. it's not standard at least |
| 06:23.28 | MicH323 | Does anyone now of any Test G.729 or G.723.1 codes that I can use that DOSENT require the Intel IPP libraries |
| 06:23.29 | brettnem | of course you don't |
| 06:23.42 | linagee | why do 800 numbers cost more? do they have to be routed special or something? is there really a greater cost to the telco? |
| 06:23.44 | NoCAT | qwell, ok i'm sorry thats the part i was getting confused with. |
| 06:23.52 | brettnem | ie: (N)on (F)acility (A)ssociated (S)ignalling |
| 06:24.02 | NoCAT | thank you. |
| 06:24.04 | Qwell | NoCAT: phone numbers mean squat to a T1, and the capacity |
| 06:24.05 | brettnem | linagee: it's a weird business |
| 06:24.08 | NoCAT | sorry, i took so much of your time. |
| 06:24.16 | _Vile | brettnem, 672 DSS channels provies 56k voice channels on a DS-3, all dialable.......? |
| 06:24.17 | Qwell | np, I'm sure I've asked dumber questions |
| 06:24.17 | brettnem | phone numbers are also easy to get |
| 06:24.25 | linagee | brettnem: soon it will fade away with wifi and voip. :) muhahaha! :) |
| 06:24.35 | brettnem | DSS? |
| 06:24.37 | brettnem | ek! |
| 06:24.37 | linagee | brettnem: add mesh networks into the pot |
| 06:24.57 | linagee | lol |
| 06:25.05 | _Vile | it's what your local phone company uses, if you use your local phone company for voice. |
| 06:25.05 | brettnem | you mean DDS? |
| 06:25.23 | brettnem | no it isn't! heh.. not in the US |
| 06:25.28 | linagee | Qwell: get a big pot. add some wifi, some wifi routing, some voip protocols, some eager users. bake at 350F for 25 minutes. serves millions. :) |
| 06:25.31 | _Vile | i'm in the US |
| 06:25.38 | _Vile | you are limited to a 56kbit channel |
| 06:25.42 | _Vile | unless you are on ISDN |
| 06:25.42 | brettnem | nooooo |
| 06:25.46 | _Vile | tell me I'm wrong :) |
| 06:25.47 | brettnem | or SS7 |
| 06:25.52 | brettnem | you are wrong :) |
| 06:25.57 | _Vile | tell me why |
| 06:26.00 | brettnem | we use SS7 ISUP signalling |
| 06:26.05 | Qwell | isn't it that modems are limited to 56k? |
| 06:26.08 | linagee | _Vile: because you've been a bad boy? |
| 06:26.09 | *** join/#asterisk erwinism (~yoly_beth@210.213.143.73) |
| 06:26.19 | _Vile | then you're switching |
| 06:26.25 | Qwell | because a standard call is 64k, right? |
| 06:26.25 | _Vile | and you're not a standard user |
| 06:26.32 | _Vile | standard call is 56k |
| 06:26.36 | erwinism | hello, can i use the intel modem clone FXO to be my fxo card? |
| 06:26.39 | _Vile | unless it is BRI or PRI |
| 06:26.40 | brettnem | Qwell: it has to do with the frequency capacity of the copper cabling and RBS signalling in older T-carrier systems |
| 06:26.44 | _Vile | at which point it is digital |
| 06:26.45 | brettnem | no |
| 06:26.52 | linagee | _Vile: where did you pull 56k from |
| 06:26.53 | _Vile | using 64k |
| 06:27.02 | brettnem | _Vile: 56k came from RBS which took bits from the T1 to do signalling |
| 06:27.06 | Qwell | brettnem: I'm not arguing it, just asking, from what I've heard |
| 06:27.11 | _Vile | robbed bit signalling |
| 06:27.13 | linagee | _Vile: standard call is a line capable of holding 4kHz or whatnot |
| 06:27.14 | _Vile | rbs |
| 06:27.14 | brettnem | right |
| 06:27.18 | brettnem | SS7 does NOT rob bits |
| 06:27.25 | _Vile | ss7 is switching |
| 06:27.32 | _Vile | not standard user phone lines. |
| 06:27.39 | Jones | ss7 is nfas |
| 06:27.40 | shepherd | 64k |
| 06:27.40 | brettnem | ISUP trunks carry 64K DS0s |
| 06:27.42 | brettnem | yes |
| 06:27.46 | _Vile | it's not 64k |
| 06:27.52 | Jones | the signal doesnt have to be anywhere near the call |
| 06:27.53 | _Vile | only on digital |
| 06:27.56 | _Vile | is it 64k |
| 06:27.57 | brettnem | Jones: correction A-links are NFAS. :) F-Links are not. :) |
| 06:28.07 | _Vile | which is why you pay more for a BRI |
| 06:28.10 | _Vile | than you do a DS0 |
| 06:28.11 | brettnem | ISUP DS0s are 64K |
| 06:28.11 | Qwell | a-links, f-links... |
| 06:28.17 | brettnem | haha |
| 06:28.19 | linagee | if i have the number from the (2 inch by 2 inch?) box on my wall, is there any way to see how fast the connection is? |
| 06:28.19 | _Vile | yes |
| 06:28.22 | Qwell | TOO MANY ACRONYMS |
| 06:28.25 | _Vile | ISUP DS0' |
| 06:28.29 | _Vile | s are 64k! |
| 06:28.32 | _Vile | but not analog bish! |
| 06:28.38 | brettnem | a-link=access links |
| 06:28.42 | Jones | a ds0 is 64 |
| 06:28.45 | brettnem | they carry analog data.. if requested |
| 06:28.48 | Jones | no matter how you slice it |
| 06:28.50 | _Vile | a ds0 is not 645 |
| 06:28.52 | _Vile | 64 |
| 06:28.55 | Jones | yes it is |
| 06:28.59 | _Vile | no it is not. |
| 06:28.59 | Qwell | brettnem: and f-link? |
| 06:29.04 | Jones | yes it is |
| 06:29.04 | linagee | in other words, on a t1 line with the numbering on the box at the point of entry, can i do anything with that T1 identifier number? |
| 06:29.07 | Jones | go look it up |
| 06:29.07 | _Vile | no it isn't. |
| 06:29.08 | zignig | yes its not! |
| 06:29.12 | _Vile | you go look it up |
| 06:29.15 | Jones | you only get 56k usable .. |
| 06:29.16 | _Vile | I know my tools. |
| 06:29.19 | Jones | but teh ds0 is 64 |
| 06:29.20 | zignig | get a room ladies. |
| 06:29.21 | zignig | :P |
| 06:29.22 | brettnem | a- access, b-bridge,c-cross,d-diagnoal,e-extended,f-facility |
| 06:29.25 | _Vile | you are looking at inband signalling |
| 06:29.30 | Qwell | brettnem: excellent, thanks |
| 06:29.36 | _Vile | not possible to achieve 64kbit signalling using inband |
| 06:29.39 | shepherd | although each DS-0 has a 64 kbps data rate (8 bits X 8 kHz), only 56 kbps is available for data (7 bits X 8 kHz) |
| 06:29.41 | Qwell | I won't remember those tomorrow though, unfortunately |
| 06:29.43 | brettnem | a DS0 is DEFINATELY 64K |
| 06:29.44 | Jones | A DSZERO IS 64 .. |
| 06:29.47 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
| 06:29.52 | _Vile | 64kbit usability, rather with inband |
| 06:29.52 | *** join/#asterisk tessier (~treed@203.210.209.73) |
| 06:30.04 | _Vile | if you're using inband |
| 06:30.07 | brettnem | it is not possible to achieve 64 on RBS! |
| 06:30.14 | brettnem | but no one uses it anymore!! |
| 06:30.15 | Jones | _Vile, if your gona use the names .. then use them right .. a ds0 is 64 |
| 06:30.19 | _Vile | then you have to submit to part of that data rate for signalling |
| 06:30.26 | Jones | the same way a ds1 is 1.554 mb |
| 06:30.42 | brettnem | _Vile: not if we take the signalling OUT OF BAND |
| 06:30.44 | brettnem | like SS7 |
| 06:30.48 | brettnem | or ISDN |
| 06:30.48 | Qwell | because 64k*24=1.554mb? |
| 06:30.54 | _Vile | brett, then you' |
| 06:31.05 | brettnem | I'm... |
| 06:31.08 | _Vile | re looking at PRI, 23 channels with a D channel for signalling |
| 06:31.12 | _Vile | or SS7 for signalling |
| 06:31.14 | brettnem | OR SS7 |
| 06:31.15 | brettnem | yes |
| 06:31.21 | brettnem | and....?? |
| 06:31.23 | _Vile | maybe ds0's are 64 k |
| 06:31.28 | Jones | even pri .. as we established before .. can use nafs signaling |
| 06:31.28 | brettnem | :) |
| 06:31.34 | brettnem | right |
| 06:31.36 | _Vile | but they are used in the complex of inband |
| 06:31.39 | _Vile | normally. |
| 06:31.42 | Jones | ds0 IS 64k |
| 06:31.48 | brettnem | _Vile: they? |
| 06:31.53 | _Vile | ds0's. |
| 06:32.06 | brettnem | no |
| 06:32.08 | brettnem | I disagree |
| 06:32.14 | brettnem | the DS0 is used for inband signalling for PRI |
| 06:32.21 | brettnem | er.. |
| 06:32.23 | brettnem | ISDN |
| 06:32.24 | Jones | out |
| 06:32.28 | _Vile | you're a cock munch |
| 06:32.36 | _Vile | you have to be kidding |
| 06:32.41 | brettnem | Jones.. in the T, OUT the DS0. ;) |
| 06:32.48 | _Vile | no |
| 06:32.56 | _Vile | a PRI is 24 DS0' |
| 06:32.57 | _Vile | s |
| 06:33.03 | _Vile | 1 used for signal |
| 06:33.04 | brettnem | in pri a regular 64K DS0 is used for signalling |
| 06:33.07 | _Vile | 23 used voice data |
| 06:33.11 | Jones | not always _Vile |
| 06:33.13 | _Vile | or for data |
| 06:33.15 | brettnem | right |
| 06:33.22 | shepherd | in MOST situations :) |
| 06:33.28 | brettnem | _Vile: did you know that ISDN protocol specifies "H" Channels.. no shit! |
| 06:33.32 | _Vile | 23B+1D combinations are |
| 06:33.33 | brettnem | not just B and D |
| 06:33.37 | _Vile | never heard of H |
| 06:33.40 | Jones | dont forget the qbit chans |
| 06:33.57 | Qwell | qbit...h channels... |
| 06:33.58 | *** join/#asterisk SexyKen (~sexyken@c-67-161-5-149.client.comcast.net) |
| 06:34.01 | brettnem | go look it up! Telcordia NWT-TR-1057 |
| 06:34.08 | brettnem | or somewhere really close to that |
| 06:34.21 | Jones | and if you toggle the z and r bit you get the caller pregnant |
| 06:34.27 | SexyKen | Does Asterisk log Hold Time for calls/queues anywhere? |
| 06:34.47 | _Vile | Sexy, no. |
| 06:34.51 | _Vile | see ya |
| 06:34.53 | shepherd | sexy: agi ;) |
| 06:34.56 | NoCAT | would it be possible to setup numbers up to have priority over certain lines. and if a call is coming in on that line and no lines are available it will drop the other line so that the call will come through? |
| 06:34.57 | brettnem | and H-channel was basically 3 B-channels concatonated together to form a big channel.. mainly used for video conference calling |
| 06:35.03 | SexyKen | •shepherd• What do you mean agi? |
| 06:35.14 | shepherd | learn it, use it |
| 06:35.14 | brettnem | pretty cool, it could be requested and routed through the PSTN on demand.. |
| 06:35.17 | Jones | anyway .. _Vile do you finaly admint that a ds0 is 64k no matter what you do to it |
| 06:35.17 | _Vile | but that's lame and why would anyone use that for standard voice |
| 06:35.35 | _Vile | jones, yes a ds0 is 64kbit unless it's limited by inband |
| 06:35.35 | brettnem | what is lame? |
| 06:35.48 | brettnem | by RBS |
| 06:35.51 | _Vile | but 64kbit none the less |
| 06:35.53 | erwinism | hello can i use Generic X100P |
| 06:36.00 | Jones | it will ALWAYS carry 64k of information .. even if you use inband signaling |
| 06:36.05 | _Vile | yep |
| 06:36.05 | MicH323 | Does anyone now of any Test G.729 or G.723.1 codes that I can use that DOSENT require the Intel IPP libraries |
| 06:36.05 | brettnem | right |
| 06:36.09 | erwinism | hello can i use Generic X100P Intel IA92 WinModem to be my FXO card? |
| 06:36.18 | brettnem | I'm about to get nyquist on this |
| 06:36.27 | *** join/#asterisk jmhunter (~jmhunter@64.77.199.223) |
| 06:36.27 | *** mode/#asterisk [+o jmhunter] by ChanServ |
| 06:36.40 | jmhunter | ~seen kram |
| 06:36.45 | jbot | kram <~mark@kram.digium.sponsor.pdpc> was last seen on IRC in channel #asterisk, 3d 3h 9s ago, saying: 'oh :)'. |
| 06:36.48 | shepherd | jmhunter: he's at von |
| 06:36.53 | jmhunter | no shit |
| 06:36.56 | _Vile | long time, I think he's at von |
| 06:36.59 | jmhunter | i need a ticket |
| 06:37.03 | _Vile | err |
| 06:37.03 | _Vile | ok |
| 06:37.09 | brettnem | well, this has been a very engaging chat.. :) |
| 06:37.14 | jmhunter | lol |
| 06:37.17 | shepherd | you in san jose? |
| 06:37.22 | jmhunter | yep |
| 06:37.24 | Jones | SOMEONE FIX MY VMAIL |
| 06:37.30 | Damin_Mobile | jmhunter: kramis just on the other side of the table from me |
| 06:37.31 | Jones | so go |
| 06:37.31 | Sedorox | .... |
| 06:37.32 | brettnem | btw, I'm a class 4/5 switch engineer in texas. :) |
| 06:37.41 | jmhunter | damin |
| 06:37.43 | Jones | im a geek in NY |
| 06:37.49 | jmhunter | ask him if he can front a pass for me |
| 06:37.52 | _Vile | brett, hey, I am curious every heard of the old summa fours? |
| 06:37.53 | brettnem | hello geek in ny. :) |
| 06:37.59 | _Vile | vco 80 junk piles? |
| 06:38.01 | brettnem | unfortunately, yes |
| 06:38.02 | justinnnn | txfax... anyone wana help me |
| 06:38.06 | justinnnn | or fix it for $$ :) ? |
| 06:38.08 | justinnnn | anyone atall :) |
| 06:38.11 | Damin_Mobile | jmhunter: we are at pf changs... |
| 06:38.13 | Jones | I had me a point code .. way back when .. |
| 06:38.15 | jmhunter | kramis i like that |
| 06:38.16 | brettnem | I knew it as a vco 4k |
| 06:38.21 | brettnem | I have a few PCs |
| 06:38.23 | brettnem | :) |
| 06:38.25 | _Vile | vco4k is the compact version |
| 06:38.26 | jmhunter | whatis pfchangs |
| 06:38.31 | _Vile | vco 80 is the big fucker :) |
| 06:38.34 | brettnem | mmm pfchang |
| 06:38.36 | shepherd | chineese resturant |
| 06:38.42 | _Vile | one rack tall |
| 06:38.45 | brettnem | _Vile: we gave cisco a year to make one work |
| 06:38.47 | _Vile | 80 T1s |
| 06:38.50 | jmhunter | well im 40 mins away |
| 06:38.51 | _Vile | yeah |
| 06:38.51 | brettnem | what? |
| 06:38.54 | brettnem | 80 T is big? |
| 06:38.54 | *** join/#asterisk mark64 (mark64@bb220-255-34-207.singnet.com.sg) |
| 06:38.57 | brettnem | 1 rack is big? |
| 06:38.59 | mark64 | Hi All |
| 06:38.59 | _Vile | cisco dropepd it immediately |
| 06:39.00 | brettnem | hm |
| 06:39.01 | _Vile | 1 rack is big |
| 06:39.07 | soundguy | does someone have the Cisco IP SoftPhone here? |
| 06:39.08 | brettnem | really??!? |
| 06:39.10 | *** join/#asterisk j0 (~dan@S010600105a04ed8d.va.shawcable.net) |
| 06:39.13 | _Vile | for me anyway |
| 06:39.15 | brettnem | hm |
| 06:39.17 | _Vile | I do have a room to fill |
| 06:39.18 | Qwell | cisco has a softphone? |
| 06:39.22 | justinnnn | cof cof txfax |
| 06:39.24 | soundguy | Yes |
| 06:39.27 | _Vile | but not for something meta has |
| 06:39.29 | brettnem | heh, I installed a switch that was 27 23" bays |
| 06:39.30 | mark64 | can someone tell me how to check if a certain channel is being use by certain user already? |
| 06:39.37 | Damin_Mobile | jjmhunter; no can do. |
| 06:39.39 | _Vile | nice |
| 06:39.40 | j0 | is it possible to have 2 peers for making outbound calls and load balance between the 2? |
| 06:39.42 | _Vile | 5e? |
| 06:39.45 | brettnem | not a phone switch.. |
| 06:39.56 | Qwell | brettnem: Would you mind a message or two? |
| 06:39.57 | shepherd | mark zap, sip, iax? |
| 06:39.59 | mark64 | i want to limit to 1 call for my prepaid system |
| 06:40.03 | mark64 | zap |
| 06:40.11 | _Vile | im working on a class 5 solution |
| 06:40.22 | brettnem | solution?? :) |
| 06:40.23 | _Vile | trying to use *, but it isn't mature enough yet |
| 06:40.28 | brettnem | qwell.. shoot. :) |
| 06:40.29 | _Vile | well |
| 06:40.30 | _Vile | meta |
| 06:40.34 | _Vile | is my goal |
| 06:40.35 | _Vile | $150k |
| 06:40.35 | brettnem | haha I have a solution.. |
| 06:40.41 | brettnem | get rd of it.. ;) |
| 06:40.43 | shepherd | heh.. no clue, hah |
| 06:40.48 | mark64 | hehe |
| 06:40.55 | mark64 | no one done that before? |
| 06:40.56 | PTG123 | anyone know if its possible to do extension mobility or roaming in asterisk? |
| 06:40.57 | _Vile | I can't use *, not reliable enough |
| 06:41.05 | shepherd | well you can see it in realtime |
| 06:41.06 | _Vile | hardware or software |
| 06:41.08 | brettnem | PTG123: sure, why not |
| 06:41.11 | mark64 | how? |
| 06:41.17 | brettnem | _Vile: have you looked at SER? |
| 06:41.19 | shepherd | asterisk -vvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvcg |
| 06:41.20 | PTG123 | brettnem: you know how to do it? |
| 06:41.21 | shepherd | :D |
| 06:41.34 | _Vile | that'd be a good solution, in fact, a perfect solution for sip clients |
| 06:41.36 | brettnem | PTG123: depends on the actually impmentation, but it shouldn't be hard |
| 06:41.38 | mark64 | err.. is there some way to list it thru AGI? |
| 06:41.39 | _Vile | but I'm dealing with pots |
| 06:41.53 | _Vile | dlcs |
| 06:41.56 | _Vile | etc |
| 06:42.07 | shepherd | aaah! |
| 06:42.09 | PTG123 | brettnem: well .. :) Is their already an existing way to do it?: |
| 06:42.12 | PTG123 | i can't find anything on the wiki |
| 06:42.12 | shepherd | mark: show channels |
| 06:42.17 | shepherd | that's for everything |
| 06:42.19 | brettnem | well asterisk dialplan stuff.. |
| 06:42.32 | mark64 | sheperd: show channels doesn't show the callerid eh |
| 06:42.50 | _Vile | so i dunno |
| 06:42.58 | _Vile | ser won't work for me |
| 06:43.01 | brettnem | PTG123: tell me how you want it to work |
| 06:43.04 | brettnem | _Vile: why? |
| 06:43.27 | PTG123 | well |
| 06:43.36 | PTG123 | basically i want each phone to always be logged into a server |
| 06:43.47 | PTG123 | but i can log into a website, and type in my login and pwrd, and it changes to use my sip account |
| 06:43.50 | PTG123 | to that phone instead |
| 06:44.00 | PTG123 | so that way the cdr is all correct |
| 06:44.03 | PTG123 | matches my sip account, etc |
| 06:44.05 | PTG123 | as well |
| 06:44.10 | PTG123 | althought i guess thats based on account code |
| 06:44.11 | PTG123 | hmm |
| 06:44.16 | mark64 | it shows the dialed number.. but i need the calling party so i can check if he's online/using a line already |
| 06:44.43 | brettnem | PTG123: look at DISA |
| 06:44.51 | PTG123 | DISA? |
| 06:45.05 | _Vile | well I'm trying not to do a one to one using asterisk, I gotta have modem capability, none of this technology supports any of that until foip is out.. and even then it's questionable. I'm looking to a build a switch out of asterisk, and it's basically too immature. |
| 06:45.25 | _Vile | so I gotta involve oversubscription, (supported) |
| 06:45.26 | PTG123 | DISA: looks more like sopmething i would want to originate a call from outside the system |
| 06:45.31 | *** join/#asterisk _tekati_ (~captain@cpe-66-75-215-63.bak.res.rr.com) |
| 06:45.48 | shepherd | fax over ip? |
| 06:45.58 | brettnem | oh |
| 06:46.01 | brettnem | ok |
| 06:46.12 | mark64 | nope.. i have a sip client. then i want to terminate pstn calls thru asterisk |
| 06:46.14 | _Vile | sec let me get the rfc |
| 06:46.16 | brettnem | so you want to "do something" and then have my SIP phone number right the phone I'm next to |
| 06:46.21 | Mavvie | fax over IP is such a non issue. |
| 06:46.24 | brettnem | FoIP.. crappy |
| 06:46.28 | Mavvie | tty emulation with zaptel, that's the issue! |
| 06:46.33 | mark64 | any help? |
| 06:46.33 | _Vile | no |
| 06:46.39 | shepherd | why does anyone use fax anymore |
| 06:46.42 | shepherd | yeah.. for real |
| 06:46.48 | _Vile | the issue is the amount of interrupts these cards using |
| 06:46.50 | PTG123 | if we had T38 it would not be crappy |
| 06:46.51 | Mavvie | shepherd: email still doesn't have a legal status |
| 06:46.55 | _Vile | and packing them into 1Us isn't easy |
| 06:46.56 | _Vile | or cheap |
| 06:46.58 | brettnem | I made an extension where I could call into the system, type a pw and then calls to my office rang in my office AND at the caller id where I made the call from |
| 06:47.04 | brettnem | T38 kinda sucks |
| 06:47.15 | shepherd | mavv: pdf? |
| 06:47.21 | PTG123 | kind of sucks ? |
| 06:47.24 | PTG123 | it just sends raw binary data |
| 06:47.27 | PTG123 | what more could you want |
| 06:47.31 | shepherd | you can digitally sign pdfs |
| 06:47.33 | brettnem | well it's not very well implemented |
| 06:47.40 | PTG123 | heh |
| 06:47.42 | shepherd | and it's legally binding |
| 06:47.43 | brettnem | AND it's not real time.. I don't think.. |
| 06:47.45 | PTG123 | its not implemented at all in asterisk :) |
| 06:47.46 | Mavvie | shepherd: PDF doesn't have a legal status, nor does it show the transmission path. |
| 06:47.48 | brettnem | I get 37 and 38 mixed up |
| 06:47.52 | PTG123 | yah its realtime |
| 06:47.55 | brettnem | ah |
| 06:47.56 | PTG123 | its just another codec |
| 06:48.05 | PTG123 | that just doesn't compress the data like audio |
| 06:48.14 | PTG123 | so you are guaranteed to have the binary data exact |
| 06:48.21 | brettnem | right |
| 06:48.21 | PTG123 | so if you have a box on one end connected directly to a t1 |
| 06:48.23 | _Vile | 37 is fucked |
| 06:48.28 | _Vile | 37 will never work |
| 06:48.31 | PTG123 | 2000 miles away, you get an exact perfect connection |
| 06:48.31 | _Vile | tested |
| 06:48.38 | brettnem | but I hear that implementations vary widely |
| 06:48.39 | _Vile | 38 has a chance |
| 06:49.22 | brettnem | I think 38 is the way to go |
| 06:49.27 | brettnem | 37 is f-d |
| 06:49.37 | PTG123 | heh |
| 06:49.42 | PTG123 | i didn't even know there is a 37 |
| 06:49.44 | PTG123 | is there a 36? :) |
| 06:49.51 | _Vile | i cant find my document, but research "modem usage" and "faxing" and VoIP. VoIP is meant to bring down the quality of voice level services to lower the bandwidth used, which is why these RFCs are made, to try to find a way around that |
| 06:49.52 | brettnem | heh, probably |
| 06:49.58 | _Vile | there's one unpublished I think |
| 06:50.01 | _Vile | that could help |
| 06:50.10 | _Vile | 38 is the way, if anything is |
| 06:50.21 | _Vile | its protocol works, if anything else does |
| 06:50.29 | shepherd | voip can use more bandwidth too :) |
| 06:50.32 | brettnem | "VoIP is meant to bring down the quality of voice level services to lower the bandwidth used" - I don't agree.. maybe this defines a codec? |
| 06:50.47 | shepherd | bascially, use a better codec :) |
| 06:50.48 | brettnem | _Vile: Look for FoIP |
| 06:50.58 | _Vile | is it foip i'm looking for? |
| 06:51.00 | brettnem | you could use a 130K MP3 codec.. right? |
| 06:51.04 | shepherd | you aren't limited to 64k with voip |
| 06:51.08 | brettnem | I think so.. there i an ITU doc for it too I think |
| 06:51.09 | shepherd | like you are with bell |
| 06:51.20 | _Vile | but no, the codec only uses more bandwidth than the standard analog channel |
| 06:51.27 | PTG123 | mp3 works by knowing what the entire file is before it encodes |
| 06:51.30 | _Vile | sec lemme look for foip |
| 06:51.36 | PTG123 | you need a sliding window codec for live video/voice |
| 06:51.43 | shepherd | _vile: but for fax, it might be necessary |
| 06:51.57 | brettnem | _Vile: depends on the codec if it uses more.. IP on the other hand adds quite a bit to the overhead.. |
| 06:51.59 | _Vile | fax uses 14.4k on the average |
| 06:52.01 | _Vile | bits per second |
| 06:52.12 | shepherd | _vile: has nothing to do with that |
| 06:52.17 | _Vile | tell me you're joking. |
| 06:52.24 | shepherd | 14.4k on a 64k channel |
| 06:52.26 | brettnem | well.. actually it kinda does |
| 06:52.37 | brettnem | there is a specral issue here.. |
| 06:52.51 | brettnem | there are only two codecs that can pass fax data transparently |
| 06:52.59 | brettnem | oh |
| 06:53.02 | brettnem | you mean compared to t38 |
| 06:53.07 | _Vile | no |
| 06:53.16 | brettnem | shepherd: right? |
| 06:53.17 | _Vile | sec lemme find my doc |
| 06:53.30 | brettnem | caue then it looks more like 300K of data.. not 14.4kbps |
| 06:53.43 | _Vile | ps? |
| 06:53.58 | brettnem | it's just a chunk of data |
| 06:54.04 | shepherd | _vile: why not just convert to tiff, send the tiff as data, and deliver on the other side |
| 06:54.06 | _Vile | ahh |
| 06:54.06 | brettnem | and it's going over... whatever your link is |
| 06:54.07 | _Vile | ok |
| 06:54.12 | *** part/#asterisk jmhunter (~jmhunter@64.77.199.223) |
| 06:54.27 | brettnem | sooo.. perhaps you should take into account the spead of your imaging device if you wanted realtime transmission? |
| 06:54.37 | _Vile | there is an RFC on FoIP, that doesn't describe t38 or t37 |
| 06:54.51 | _Vile | I think it's more meant for modem connections |
| 06:54.58 | _Vile | but I can't for shit find it here |
| 06:55.34 | _Vile | actually |
| 06:55.40 | _Vile | I don't think it's meant for faxing |
| 06:55.45 | brettnem | there is a seperate ITU spec for MoIP |
| 06:55.48 | _Vile | but only for modem connections, but still relavant |
| 06:55.50 | _Vile | MoIP |
| 06:55.51 | brettnem | tak about f-d |
| 06:55.55 | brettnem | +l |
| 06:56.28 | _Vile | relevant |
| 06:56.38 | _Vile | I'm going to smoke |
| 06:57.20 | brettnem | ss7 |
| 06:57.24 | _Vile | shep, why not connect Fax Machines to LAN connections, and use them only as scanner/e-mail devices? |
| 06:57.45 | linagee | what is today? female day or something? (google logo) |
| 06:57.47 | _Vile | because the industry isn't made that way, and you won't change it. |
| 06:58.01 | brettnem | fax machines are really low quality. |
| 06:58.14 | j0 | linagee: hm. interesting |
| 06:58.27 | j0 | check out the alt text for the logo |
| 06:58.38 | shepherd | _vile: so, you save the fax as tiff |
| 06:58.39 | brettnem | I have a kick ass copy machine (yes, you heard that right) in my office.. I can put a stack of 100 double sided pages in it, give it my email address and it emails me a pdf. |
| 06:58.45 | shepherd | deliver tiff as data |
| 06:58.53 | shepherd | call the number on the other side and deliver :) |
| 06:59.26 | shepherd | simple solution for fax oip :) |
| 06:59.39 | brettnem | how about.. go to the computer store.. buy a POS $30 USB scanner and save yourself a headache.. ;) |
| 06:59.58 | brettnem | shepherd: there is a software interface that is missing |
| 07:00.43 | shepherd | brett: get-r-done.. kthx |
| 07:01.10 | brettnem | a fax machine looking device that was a SIP endpoint, spoke T.38 and where you could enter in a phone number would kick ass.. but only if people actually supported t38 |
| 07:01.22 | brettnem | ;) |
| 07:01.30 | shepherd | all i am saying.. |
| 07:01.31 | _Vile | shep, that doesn't work for standard fax machines, does it? -- otherwise I'd be implementing a solution like that... that'd work from company to company transmissions, though, from scanners. |
| 07:01.44 | shepherd | you can probably write an agi script for asterisk to support faxes |
| 07:01.55 | shepherd | but.. it couldn't be done via voip |
| 07:02.03 | shepherd | and there really is no way to confirm if it got sent |
| 07:02.07 | linagee | j0: how sexist. when is international men's day?? |
| 07:02.13 | brettnem | shepherd: think SER |
| 07:02.22 | _Vile | brett, look at as5300 + t.38 + hylafax and openh232 |
| 07:02.25 | _Vile | works |
| 07:02.26 | shepherd | but you would have to have 2 asterisks |
| 07:02.30 | shepherd | one on each side of the internet |
| 07:02.44 | brettnem | _Vile: yeah, I need to send that fax data out a SS7 connection.. what do you recommend for that? :) |
| 07:03.12 | _Vile | the same using sip connections to isdn boxes connected to your ss7 network |
| 07:03.15 | zignig | linagee: we are chavanist pigs , we don't deserve one.... :P |
| 07:03.27 | _Vile | rather, s/sip/pri |
| 07:03.36 | brettnem | oh |
| 07:03.40 | linagee | zignig: who's a chavanist pig |
| 07:03.48 | brettnem | ok.. so I go T.38/SIP into the AS5300 and PRI out? |
| 07:03.53 | _Vile | yep |
| 07:03.55 | _Vile | all the way |
| 07:03.56 | zignig | linagee: all men. |
| 07:03.58 | brettnem | it's a bit sloppy, but I guess it'd work |
| 07:04.04 | linagee | zignig: not me. :) |
| 07:04.06 | zignig | linagee: <joke> |
| 07:04.22 | linagee | zignig: /me doesn't swing that way |
| 07:04.23 | _Vile | expensive, too |
| 07:04.30 | brettnem | what is expensive? |
| 07:04.39 | *** join/#asterisk soulz- (~soulz@host-137-132-45-118.imcb.nus.edu.sg) |
| 07:04.41 | linagee | brettnem: money. :) |
| 07:04.46 | soulz- | hello all |
| 07:04.48 | _Vile | as5300, I shouldn't be saying 5300, you should be spending $ on 5350 |
| 07:04.52 | _Vile | for the good DSP's |
| 07:04.53 | linagee | brettnem: just look at the prime rate! ;-) |
| 07:04.58 | soulz- | i am having trouble with bv, and no support from bv:( |
| 07:05.27 | soulz- | anyone on bv at the moment? |
| 07:05.33 | _Vile | and despite popular belief, stick a max of 4 T1s into all of those boxes, not 8 |
| 07:05.34 | brettnem | heh prime |
| 07:05.47 | *** join/#asterisk shaZwaz (~adnans@203.81.196.167) |
| 07:05.49 | SexyKen | Asterisk doesn't have real-time for queues? |
| 07:05.54 | shaZwaz | hi ppl |
| 07:06.27 | shmaltz | hi shaZwaz |
| 07:06.28 | brettnem | I've been looking at using something like the AS5300 as a media gateway for a while.. |
| 07:06.40 | shaZwaz | I think initially its for SIP /IAX and the dialplan |
| 07:06.46 | _Vile | it's good for echo can |
| 07:06.59 | _Vile | in fact |
| 07:07.07 | brettnem | I got cans in my switch.. thank god |
| 07:07.24 | _Vile | I've yet to hear an echo canceller better than cisco built.. I'm talking from the switch companies etc... they hired some smart people |
| 07:07.39 | _Vile | well, it's on par at least |
| 07:07.42 | _Vile | but damn good |
| 07:08.01 | _Vile | what switch are you running? |
| 07:08.16 | brettnem | Sonus and Tekelec |
| 07:08.28 | _Vile | tekelec is cool, how's the sonus doing for you? |
| 07:08.41 | _Vile | simpler than tek? |
| 07:08.45 | brettnem | heh. honestly, I think the tekelec is a piece of junk.. the sonus is nice |
| 07:09.03 | _Vile | I had it backwards, thought the sonus was junk and vice versa :) |
| 07:09.18 | _Vile | hows the interface? |
| 07:09.33 | brettnem | well I don't use it myself that much.. |
| 07:09.41 | SexyKen | voip-info.org down? |
| 07:09.43 | brettnem | tekelec's management interface runs on windows |
| 07:09.45 | _Vile | what about the admin side? |
| 07:09.47 | brettnem | I can't believe it |
| 07:09.52 | _Vile | that's sad |
| 07:09.55 | _Vile | should be all nix |
| 07:10.01 | brettnem | crashes all the time |
| 07:10.08 | _Vile | sonus? |
| 07:10.12 | brettnem | it doesn't actually depend on it real time |
| 07:10.15 | *** join/#asterisk file (~file@251.134.218.209.transedge.com) |
| 07:10.17 | brettnem | the sonus is a champ.. good stuff |
| 07:10.25 | _Vile | how's that interface? |
| 07:10.30 | _Vile | I'm about to spend $200k so asking |
| 07:10.32 | SexyKen | What is tekelec and sonus? |
| 07:10.45 | brettnem | 200k?? that's a good deal, what are you getting?? |
| 07:10.51 | _Vile | looking at meta |
| 07:10.54 | _Vile | currently |
| 07:11.02 | brettnem | the sonus one looks pretty good.. haven't used it myself.. |
| 07:11.14 | brettnem | but intuitive.. more than anything, I like their architecture.. |
| 07:12.05 | brettnem | SexyKen: Commerical carrier grade "Softswitchws" |
| 07:12.07 | brettnem | -w +e |
| 07:12.19 | brettnem | _Vile: have you looked at Verisign's SIP-7? |
| 07:12.25 | _Vile | the metaswitch is pretty cheap, not the cheapest.. if I had to go with a cheap solution, I'd buy a bunch of vco 80's :) |
| 07:12.33 | _Vile | and application everything |
| 07:12.34 | _Vile | yeap |
| 07:12.36 | SexyKen | •brettnem• So they're like asterisk but better? |
| 07:12.37 | _Vile | w/ asterisk |
| 07:12.39 | _Vile | and SER |
| 07:12.42 | _Vile | bur the problem is |
| 07:12.53 | _Vile | I can't rely on customers being 1-1 on a given * box |
| 07:13.01 | _Vile | s/bur/but |
| 07:13.11 | _Vile | I need the *POTS* |
| 07:13.16 | _Vile | DLC customers |
| 07:13.27 | _Vile | to be on a 1-*whatever* system |
| 07:13.32 | _Vile | or a reliable 5 9's switch |
| 07:13.36 | brettnem | SexyKen: well.. let's put it this way.. ONE Sonus terminal takes up about 1/2 of a 7 foot rack and can terminate 420 T1.. and do SS7.. |
| 07:13.54 | brettnem | _Vile: you know asterisk can do GR-303 |
| 07:13.59 | brettnem | you can do 10-1 with that |
| 07:14.00 | _Vile | yes I do |
| 07:14.03 | _Vile | but no |
| 07:14.07 | _Vile | 10 to 1 box! |
| 07:14.12 | _Vile | that's the problem |
| 07:14.19 | _Vile | 10-1 subscription ratio |
| 07:14.25 | _Vile | but 10 on one box |
| 07:14.30 | brettnem | wha? |
| 07:14.32 | _Vile | which means, one box goes down |
| 07:14.41 | _Vile | 10 customers per port goes down |
| 07:14.45 | brettnem | look , you just gotta architect it right |
| 07:14.55 | _Vile | I know, and I've tried |
| 07:15.03 | _Vile | I priced an * solution at $100k |
| 07:15.13 | brettnem | actually, with T1s it's tricky |
| 07:15.13 | _Vile | but it doesn't act reliably |
| 07:15.16 | _Vile | yep |
| 07:15.18 | _Vile | and DLCs |
| 07:15.26 | _Vile | and per port management etc |
| 07:15.28 | brettnem | ah.. the DLC is the easy part I think |
| 07:15.36 | _Vile | feature wise, it's an awesome system |
| 07:15.38 | brettnem | OSS? |
| 07:15.42 | brettnem | what DLC you using? |
| 07:15.50 | _Vile | but when you get into relaibility solutions |
| 07:15.55 | _Vile | it gets difficult |
| 07:15.57 | _Vile | when using POTS |
| 07:16.00 | _Vile | but not SER |
| 07:16.00 | brettnem | _Vile: why don;'t you like SER? |
| 07:16.04 | _Vile | pairgain all of the way |
| 07:16.05 | brettnem | heh |
| 07:16.18 | brettnem | like "Pairgain" (tm) pairgain? |
| 07:16.23 | MuppetMaster | Who is using SER with Asterisk? |
| 07:16.23 | _Vile | PairGain |
| 07:16.24 | SexyKen | I want to start offering cell phones via my VOIP PBX |
| 07:16.24 | _Vile | DLC |
| 07:16.28 | _Vile | RTs |
| 07:16.29 | _Vile | etc |
| 07:16.39 | _Vile | Sexy |
| 07:16.51 | _Vile | look into 802.11b transmitters |
| 07:16.52 | MuppetMaster | I am having a problem getting Video (H.263) to work behind a firewall/NAT with Asterisk, so I am looking to something that would allow me to setup my own NAT proxy. Any recommendations? |
| 07:16.54 | brettnem | MuppetMaster: me |
| 07:16.59 | _Vile | and 802.11b phones |
| 07:17.11 | SexyKen | •viLeR• That's wireless, I mean cellular :-) |
| 07:17.21 | PTG123 | heh cellular requires major licenses |
| 07:17.24 | _Vile | same diff |
| 07:17.25 | PTG123 | to have attennas etc |
| 07:17.30 | brettnem | MuppetMaster: what client? |
| 07:17.34 | PTG123 | its a pretty involved process |
| 07:17.41 | PTG123 | which is why all these companies are merging |
| 07:17.42 | *** join/#asterisk tomas- (~tomas@pcp0010083068pcs.frnklv01.nj.comcast.net) |
| 07:17.43 | brettnem | _Vile: what is wrong with SER? |
| 07:17.45 | PTG123 | they double their prescence |
| 07:17.49 | PTG123 | without all the shitwork |
| 07:17.54 | _Vile | brett, nothing, when you're using SIP |
| 07:17.56 | MuppetMaster | brettnem: Eyebeam. I can use Eyebeam with http://www.ineen.com with no problem, as well as with FWD and their NAT. |
| 07:18.01 | SexyKen | •PTG123• I can lease portions of other providers antennas |
| 07:18.10 | PTG123 | sexyken: you can use someone elses network yes |
| 07:18.13 | _Vile | but when I'm connecting DS0's from Land Lines to a Switching system, I can't just connect them to SER |
| 07:18.15 | MuppetMaster | brettnem: But with Asterisk alone (even though I have nat=yes, verify=yes) no luck. |
| 07:18.21 | brettnem | MuppetMaster: there is an asterisk patch for Eyebeam H.263 or whatever they use |
| 07:18.32 | brettnem | _Vile: yes you can |
| 07:18.32 | justinnnn | anyone want to help me with faxing please ??? |
| 07:18.37 | justinnnn | anyone at all :) ??? |
| 07:18.40 | justinnnn | money to give out ppls |
| 07:18.46 | brettnem | money? |
| 07:18.46 | justinnnn | just loking for someone to give it to..... |
| 07:18.47 | brettnem | where |
| 07:18.48 | PTG123 | you know speaking of eyebeam |
| 07:18.50 | MuppetMaster | justinnnn: I have fax working with spandsp. |
| 07:18.50 | PTG123 | my eyebeam broke |
| 07:18.53 | PTG123 | it won't register at all |
| 07:18.53 | _Vile | there will always be a failure point, and until * gets onto a 5 9's platform, should I, as a business manager, risk that? |
| 07:18.56 | justinnnn | fix mine then pls |
| 07:18.59 | PTG123 | i don't even get any messages in the cli |
| 07:19.09 | MuppetMaster | justinnnn: What problem are you having? |
| 07:19.30 | brettnem | _Vile: I'm architectuing my wholesale voip network now.. it' ser+asterisk based and fully redundant |
| 07:19.35 | erwinism | hello can i use Generic X100P Intel IA92 WinModem to be my FXO card? |
| 07:19.44 | tomas- | hi, I'm completely ignorant of how PBX's work, but dont PBX's run over T1 lines? is it possible to have a PBX system and data Internet line function over the same T1 line? |
| 07:19.49 | MuppetMaster | PTG1243: I found to get Eyebeam working behind a NAT (not the same with X-Pro) I had to have nat=yes and verify=yes in the [general] section of my sip.conf. |
| 07:19.52 | brettnem | PTG123: have you used tethereal to see if you are getting the registrations at all? |
| 07:20.06 | _Vile | and I've talked to bkw about this, and the rest of the development crew, a lot... but * at this point, is a switching PBX platform, and not a customer switching platform when it comes to POTS.. it doesn't run on gigabit switching backplanes, etc. |
| 07:20.07 | shepherd | asterisk is 5 9's (but they don't advertise that) |
| 07:20.07 | MuppetMaster | tomas: PBX may be T1/E1, Analog, VoIP, etc. |
| 07:20.10 | brettnem | tomas-: short answer: yes |
| 07:20.16 | _Vile | but it does work. |
| 07:20.22 | _Vile | And I trust it w/ Business. |
| 07:20.27 | brettnem | _Vile: so don;'t use it as a customer switching platform! |
| 07:20.34 | _Vile | but it ain't ready for the switching market. |
| 07:20.38 | brettnem | use SER there |
| 07:20.43 | _Vile | heh, I don't :) |
| 07:20.44 | PTG123 | MuppetMaster: easier said then done, since i don't use text confs.. but let me figure out how to do that in the peer record :) and no i haven't brett.. is that easy to use? |
| 07:20.44 | MuppetMaster | brettnem: So any recommendtions on Eyebeam/H.263 and a NAT with Asterisk? Could SER foot the bill? |
| 07:21.13 | MuppetMaster | PTG123: Yes, that can be done on a section by section level as well. |
| 07:21.22 | _Vile | I don't know how to cross POTS to SER and keep full modem compatibility. |
| 07:21.25 | PTG123 | MuppetMaster: how do you mean? |
| 07:21.34 | brettnem | PTG123: yes, easy.. apt-get install ethereal then tethereal -R "udp.port==5060" |
| 07:21.38 | *** join/#asterisk twisted (~twisted@twisted-professional-pdpc.developer.and.friend.of.asterisk) |
| 07:21.38 | *** mode/#asterisk [+o twisted] by ChanServ |
| 07:21.39 | _Vile | s/SER (SER -> SER to SIP) |
| 07:21.42 | MuppetMaster | PTG123: So the endpoint that is Eyebeam behind a NAT could have the configuration locally. So for example, if you have [3002] as your Eyebeam extension, you can drop theose options there. |
| 07:21.46 | _Vile | twised |
| 07:21.46 | brettnem | MuppetMaster: get the asterisk patch for Eyebeam.. it's new |
| 07:21.49 | PTG123 | MuppetMaster: if i put it in general will it do it for all? |
| 07:21.55 | shepherd | _vile: how about not using sip :) |
| 07:22.03 | MuppetMaster | brettnem: I did, but that is not the problem. The problem is the Firewall/NAT, Eyebeam seems quite sensitive. |
| 07:22.07 | brettnem | _Vile: SER never see the media path.. why would it effect modems? |
| 07:22.10 | PTG123 | MuppetMaster: you don't understand, i build the peers with my own custom code :) directly in chan_sip.c |
| 07:22.12 | shepherd | sip isn't the only thing out there |
| 07:22.19 | MuppetMaster | PTG123: Ah. Good fun. |
| 07:22.24 | brettnem | ew |
| 07:22.29 | _Vile | SIP would, would I go ulaw all the way using ser? |
| 07:22.32 | brettnem | open some ports. :) |
| 07:22.36 | MuppetMaster | PTG123: May I ask why? |
| 07:22.39 | brettnem | anything you want |
| 07:22.45 | _Vile | sip doesn't require a codec, right. |
| 07:22.51 | MuppetMaster | brettnem: I have opened all the usual suspects, including 2979 for H.263, still no joy. |
| 07:22.55 | _Vile | but I' |
| 07:23.10 | brettnem | can you sniff the network where eyebeam is and see what it is using? |
| 07:23.16 | MuppetMaster | brettnem: The audio codec is fine, so are my Sipuras and X-Pro, it is just the H.263 video that can not seem to get through. |
| 07:23.17 | tomas- | okay, so as a guy with decent computer networking background (not really any telephony), and a little help from the local phone company technician (as far as getting the hardware set up), is it easy enough to get an open source pbx system up and running so i can run a t1 line into my home for voice+data, having several phone lines (each with different phone numbers), for business AND personal use (ie home business) |
| 07:23.26 | MuppetMaster | brettnem: Good idea, I will give that a shot. |
| 07:23.31 | linagee | brettnem: be careful with sniffing. they are watching your every move |
| 07:23.36 | _Vile | m still not sure whether or not using ulaw (*maximum* 14.4k fax, when standard telco can achieve higher), would be the *same* as a switch environment. |
| 07:23.37 | brettnem | yeah, well that's a different port and protocol |
| 07:23.40 | MuppetMaster | tomas: Short answer, yes. |
| 07:23.45 | PTG123 | MuppetMaster: um because i have my own realtime implementation basically to work with alot of other stuff i am doing |
| 07:23.49 | brettnem | yeah, don't sniff someone elses network.. I didn't tell you to do it. :) |
| 07:24.00 | MuppetMaster | PTG123: Interesting. |
| 07:24.14 | brettnem | _Vile: ulaw codec is identical to the bandwidth of the PSTN |
| 07:24.26 | brettnem | in fact, in actual practice it's quite a bit better |
| 07:24.32 | MuppetMaster | _Vile: In fact it is what the PSTN uses. |
| 07:24.35 | justinnnn | someone please fix my txfax :) |
| 07:24.39 | tomas- | MuppetMaster: how would i go about understanding (i.e. education via the Internet) the LONG answer? |
| 07:24.41 | PTG123 | ok |
| 07:24.42 | brettnem | except the PSTN calls it PCM |
| 07:24.44 | MuppetMaster | justinnnn: What is the problem you are having? |
| 07:24.46 | PTG123 | eyebeam is sending my internal ip |
| 07:24.46 | _Vile | I know pstn uses ulaw |
| 07:24.48 | PTG123 | instead of external |
| 07:24.50 | PTG123 | how do if ix that |
| 07:24.52 | brettnem | that'll do it |
| 07:24.57 | MuppetMaster | tomas: A few places to look: http://www.voip-info.com |
| 07:24.59 | brettnem | nat= yes might fix it |
| 07:25.03 | shepherd | _vile: why don't you try it |
| 07:25.07 | MuppetMaster | tomas: http://www.asteriskdocs.org |
| 07:25.12 | _Vile | shep, I have |
| 07:25.12 | shepherd | and let us know how it goes :) |
| 07:25.13 | MuppetMaster | tomas: http://asterisk.xvoip.com |
| 07:25.19 | brettnem | _Vile: just saying that the codec ulaw is == to PSTN ulaw |
| 07:25.22 | shepherd | try iax and ulaw |
| 07:25.34 | _Vile | I've tried zap and ulaw |
| 07:25.40 | MuppetMaster | IAX works great through firewalls, much better than SIP. |
| 07:25.44 | brettnem | _Vile: and? |
| 07:26.01 | _Vile | and it works great for faxing up to 14.4k, perfect except for echo issues on some calls |
| 07:26.04 | _Vile | dropped calls |
| 07:26.08 | PTG123 | autodetect ip address isn't working |
| 07:26.09 | MuppetMaster | tomas: Sorry, that is http://www.voip-info.org, not .com. |
| 07:26.09 | _Vile | PRI problems |
| 07:26.11 | PTG123 | stupid |
| 07:26.12 | _Vile | and I can go on :) |
| 07:26.24 | shepherd | vile: echo can? |
| 07:26.25 | brettnem | pri problems?! |
| 07:26.31 | brettnem | like what? |
| 07:26.34 | _Vile | I think the dropped calls are a result of "PRI 1-23 resetting" |
| 07:26.44 | MuppetMaster | justinnnn: I recommend posting your fax problem to http://asterisk.xvoip.com |
| 07:26.47 | _Vile | shep, echotraining is set at 800 |
| 07:26.52 | brettnem | they do reset regularly.. that's normal and doesn't effect operation. |
| 07:26.53 | _Vile | upstream has echo tools turned on |
| 07:27.09 | _Vile | brett, good to know, TY |
| 07:27.11 | brettnem | where do you hear the echo |
| 07:27.31 | _Vile | first 10 seconds of the call, and if any additional "noise" is added to the call |
| 07:27.33 | PTG123 | its that nat stuff |
| 07:27.38 | PTG123 | well let me see how their code does it |
| 07:27.40 | tomas- | so i take it that running my phone lines over my internet data t1 is what's called "VoIP"? |
| 07:28.01 | _Vile | tomas, yes, voice over IP would be considered running voice over IP |
| 07:28.29 | brettnem | _Vile: <doubletake> |
| 07:28.42 | _Vile | plus I have a cool problem of random numbers giving a fast busy |
| 07:28.44 | MuppetMaster | tomas: TDM is what is considered 'traditional' telephony, when you run it over IP, then it is indeed VoIP. |
| 07:28.51 | _Vile | I need to update to the latest CVS |
| 07:28.53 | brettnem | nI mean, what side of the call do you hear echo? |
| 07:28.56 | _Vile | or stable (whichever) |
| 07:29.00 | MuppetMaster | tomas: Have a look at http://www.wikipedia.org for some definitions, most of them are there actually. |
| 07:29.11 | shepherd | cvs |
| 07:29.16 | _Vile | brett, I've never heard echo on my side, but I've had user reports of hearing echo |
| 07:29.23 | _Vile | and I've had caller reports of it |
| 07:29.30 | brettnem | ah ha |
| 07:29.36 | _Vile | so, my attempts at tracking it down are at most limited |
| 07:29.37 | brettnem | if THEY hear echo.. then it's YOUR problem |
| 07:29.54 | brettnem | ie: you are (for some reason) sending their audio back to them |
| 07:30.05 | brettnem | it's non-intuitive |
| 07:30.13 | _Vile | yeah and I know that, but if I'm sending audio back to them at random events, and they're sending audio back to me on occassion |
| 07:30.18 | brettnem | did you check your echo settings? there are a few |
| 07:30.18 | _Vile | what would be your verdict? |
| 07:30.26 | _Vile | yeah I set them all up right |
| 07:30.33 | _Vile | according to my original PRI testing |
| 07:30.34 | *** join/#asterisk vipor (vipor@ip68-111-216-186.sd.sd.cox.net) |
| 07:30.37 | vipor | Hello |
| 07:30.48 | tomas- | office-grade phones (with all the extra buttons) can run analog, right? Is there really a difference between analog and digital? which is better to go with? |
| 07:30.50 | vipor | I have a quick *?*... |
| 07:30.52 | _Vile | unless that fucker CLEC decided to play with me |
| 07:31.10 | brettnem | nah |
| 07:31.15 | brettnem | hey |
| 07:31.17 | brettnem | be nice to clec |
| 07:31.18 | _Vile | I manage 8 7940s in the office |
| 07:31.19 | brettnem | s |
| 07:31.20 | brettnem | :) |
| 07:31.22 | _Vile | I'm a clec |
| 07:31.24 | MuppetMaster | tomas: Depends on your needs. There are enhanced sets that run SIP/VoIP with an ethernet interface. You could also take your existing Analog and connect them to an ATA and convert to VoIP that way. |
| 07:31.28 | brettnem | oh. me too. :) |
| 07:31.31 | _Vile | we're all being fucked with |
| 07:31.31 | linagee | tomas-: digital is always better |
| 07:31.35 | _Vile | fucking UNE going away |
| 07:31.41 | brettnem | Wahoo! |
| 07:31.45 | brettnem | more business for me |
| 07:31.47 | brettnem | ;) |
| 07:31.49 | _Vile | I'm going to sue those fucks as soon as I get enough money |
| 07:31.59 | brettnem | we sue em all the time |
| 07:32.03 | linagee | _Vile: what is a 7940? |
| 07:32.05 | _Vile | I have 3 pending |
| 07:32.07 | brettnem | you said you are in denver? |
| 07:32.10 | MuppetMaster | tomas: So if it is a question of preserving existing investment, then maybe analog with a channel bank for VoIP conversion would be fine. If you are going to acquire new, then VoIP phones that support SIP are the only way to go. Examples of ATAs and SIP phones may be found at http://www.sipura.com |
| 07:32.12 | _Vile | linagree, ebay for it |
| 07:32.15 | _Vile | brett, no oregon |
| 07:32.16 | brettnem | you much be non-fac? |
| 07:32.18 | brettnem | ah |
| 07:32.26 | _Vile | I have 3 against qwest |
| 07:32.27 | linagee | _Vile: clecs are always suing clecs from what i hear |
| 07:32.32 | _Vile | nonono |
| 07:32.37 | _Vile | clecs against ilecs |
| 07:32.42 | brettnem | right |
| 07:32.43 | vipor | <PROTECTED> |
| 07:32.43 | vipor | <PROTECTED> |
| 07:32.43 | vipor | <PROTECTED> |
| 07:32.43 | vipor | <PROTECTED> |
| 07:32.44 | _Vile | ilecs are bending us over |
| 07:32.48 | _Vile | right now |
| 07:32.52 | MuppetMaster | vipor: Yes, it is the Monitor cmd. |
| 07:32.53 | _Vile | because of that nigger bush admin |
| 07:32.59 | vipor | <PROTECTED> |
| 07:32.59 | brettnem | _Vile: are you non-fac or fac? |
| 07:33.08 | _Vile | brett, fac? |
| 07:33.09 | linagee | _Vile: ilec? |
| 07:33.20 | brettnem | _Vile: facilities based or non facilities.. |
| 07:33.25 | _Vile | linagree, Qwest, BellSouth, Verizon, etc etc |
| 07:33.27 | MuppetMaster | vipor: On the standard modem side, not sure how Asterisk would handle that. Needs to be a specific chipset to access the device. |
| 07:33.28 | linagee | _Vile: someone slashdotted ebay. :( |
| 07:33.32 | _Vile | I colo, etc |
| 07:33.40 | _Vile | I'm facilities based |
| 07:33.48 | _Vile | and they're raising prices on me |
| 07:33.48 | vipor | ic |
| 07:33.49 | MuppetMaster | vipor: But Asterisk and hardware are not my thing, I use Asterisk purely as a Softswitch with external ATAs and Softphones. |
| 07:33.50 | linagee | _Vile: ah. 7940 is just a cisco phone? |
| 07:33.53 | _Vile | without me signing contracts |
| 07:33.57 | _Vile | lin, yes |
| 07:34.02 | *** join/#asterisk mark64 (mark64@bb220-255-34-207.singnet.com.sg) |
| 07:34.09 | brettnem | yeah UNE-P going away is going to look interesting.. |
| 07:34.16 | linagee | _Vile: i thought it was some sort of CO equipment. :) |
| 07:34.18 | brettnem | we are developing some alternative UNE-P stuff.. |
| 07:34.19 | _Vile | une-p, I'm not worried about une-p |
| 07:34.22 | _Vile | une-p is gone |
| 07:34.27 | *** part/#asterisk Uberbot (~konversat@pcp01879960pcs.sandia01.nm.comcast.net) |
| 07:34.28 | _Vile | I'm worried about une in general |
| 07:34.30 | *** join/#asterisk erwinism (~pogz@210.213.143.73) |
| 07:34.36 | _Vile | and the ilecs transferring all of us to resale |
| 07:34.38 | MuppetMaster | _Vile: What is une? |
| 07:34.51 | linagee | uunet? |
| 07:34.52 | _Vile | un bundeled network elements |
| 07:34.53 | brettnem | now that's bullshit |
| 07:34.58 | vipor | If I could get it to see the modem , do you think it would support what I need? |
| 07:34.58 | brettnem | heh, loaded question |
| 07:35.00 | erwinism | hello, what are those ports to enable in the firewall for my SIP clients to login? |
| 07:35.02 | _Vile | I don't think so, brett |
| 07:35.09 | _Vile | I think that's going to happen in 5 years |
| 07:35.12 | brettnem | nah |
| 07:35.14 | _Vile | what's happening now? |
| 07:35.16 | _Vile | 1996 is gone |
| 07:35.17 | _Vile | . |
| 07:35.33 | linagee | _Vile: 1984 was a typo! :) |
| 07:35.38 | mark64 | anyone writing agis here? |
| 07:35.42 | brettnem | hey look.. I have MY interconnection agreement.. it is NOT T2A.. |
| 07:35.42 | _Vile | 1984 was fun. |
| 07:36.08 | _Vile | well, when they change it on you and tell you to bend over further, tell me that I'm wrong :) |
| 07:36.30 | brettnem | I will |
| 07:36.32 | _Vile | until then, you' |
| 07:36.35 | _Vile | re right |
| 07:36.39 | brettnem | ;) |
| 07:36.42 | tomas- | MuppetMaster: im actually moving to an area where they seem to only offer T1 (in the mountains), and it's going to cost $400 for the T1 to be cost-justified on their end, so i need to come up with as many ways as possible to maximize the thing. i intend on running a business from home, so having a PBX would enhance my professional image...im on a serious budget, so the cheaper, the better. I'd only need 3 voice phones + 1 fax, but |
| 07:36.45 | _Vile | give it a year, :) |
| 07:37.03 | brettnem | oh.. in a year, I'll be interconneting with ISDN instead of SS7 |
| 07:37.06 | erwinism | any ideas on my problem? |
| 07:37.11 | _Vile | I do cost effective data T1s at $99 |
| 07:37.14 | brettnem | I'm the one bending them over.. really.. watch for us |
| 07:37.26 | _Vile | tell Qwest to compete with me |
| 07:37.28 | brettnem | does that include transport and mileage? |
| 07:37.35 | mark64 | hello? |
| 07:37.36 | MuppetMaster | tomas: Then, I would recommend a few things. First, ensure you get a router for the T1 that supports QOS to prioritize the VoIP. |
| 07:37.36 | _Vile | includes that and BW |
| 07:37.38 | brettnem | port costs, switching |
| 07:37.43 | _Vile | everything |
| 07:37.50 | _Vile | we run our own HDLC equipment |
| 07:37.53 | brettnem | who does your transport? |
| 07:37.59 | brettnem | ah.. you collo |
| 07:38.03 | MuppetMaster | tomas: Then, you may want to look at the Sipura 2100s, as they are going to support T.38 Fax over Internet standard in the firmware soon. |
| 07:38.04 | _Vile | qwest |
| 07:38.05 | _Vile | yep |
| 07:38.05 | brettnem | so limited availability |
| 07:38.11 | _Vile | limited |
| 07:38.15 | _Vile | 11kfeet |
| 07:38.19 | brettnem | ek |
| 07:38.23 | _Vile | $169 beyond our range |
| 07:38.23 | MuppetMaster | tomas: Also, have a look at asteriskathome.sourceforge.net for a self-installing web interface version of Asterisk |
| 07:38.29 | *** part/#asterisk X-Gen (~x-gen@rrba-146-120-223.telkomadsl.co.za) |
| 07:38.29 | linagee | _Vile: T1 for $99?? |
| 07:38.40 | _Vile | lina, $169 where we can't reach you yes |
| 07:38.50 | PTG123 | i';ll take 2 t1s vile |
| 07:38.50 | PTG123 | :) |
| 07:38.52 | brettnem | _Vile: I don't think that $99 can make money.. I've done this transport thing quite a bit |
| 07:38.54 | MuppetMaster | tomas: What you are doing is absolutely viable, not expensive (besides the T1, as the ATAs are ~US$89.95 each and support 2 lines each). |
| 07:38.55 | linagee | _Vile: i thought a T1 was like $500/mo |
| 07:38.59 | linagee | _Vile: at LEAST |
| 07:39.01 | _Vile | brett, I'm making money |
| 07:39.04 | _Vile | lina, times change. |
| 07:39.14 | linagee | _Vile: what changed with that |
| 07:39.19 | PTG123 | if you have a t1, get your phone service via a sip or iax handoff, do't use normal telephones |
| 07:39.34 | MuppetMaster | tomas: With Asterisk you may then have voicemail (vmail to email), call prompting, recording capabilities, call transfer, conference facilitilies, etc, etc. |
| 07:39.35 | _Vile | lina, *shrug*, economy I guess. |
| 07:39.36 | brettnem | what's your UNE charge? |
| 07:39.37 | erwinism | i am already allowing port 5060 UDP but the softphone cant still connect. what else should i enable? |
| 07:39.41 | brettnem | oh wait.. HDLC loops are cheap |
| 07:39.43 | linagee | PTG123: what if i want a backup? :) |
| 07:39.49 | _Vile | brett, $15 for ds0, $87 for t1 |
| 07:39.51 | brettnem | ok.. I can buy that |
| 07:39.53 | MuppetMaster | tomas: I use Asterisk as my SoHo PBX, but I live in Amsterdam and my 4MB ADSL is only 49 Euros/month. |
| 07:40.02 | brettnem | it's HDLC, so how many collos do you have? |
| 07:40.07 | PTG123 | linagee: have your voip provider route it to a cell phone or voicemail if it detects you rline is down |
| 07:40.11 | _Vile | brett, ?? |
| 07:40.21 | _Vile | we have a 14x14 cage |
| 07:40.22 | justinnnn | can someone help me with faxing pls |
| 07:40.30 | _Vile | in the CO |
| 07:40.31 | brettnem | ok, so like in the CBD |
| 07:40.32 | justinnnn | rxfax is working... txfax isnt still.. :( |
| 07:40.34 | MuppetMaster | tomas: My setup is running an Asterisk instance at http://www.easyspeedy.com on SuSE 9.2 with Eyebeam Softphones and a Sipura 2000 and Sipura 3000 (since I don't have naked DSL, still want to run that POTS line into the Asterisk). |
| 07:40.35 | _Vile | cross connecting on the frame |
| 07:40.45 | _Vile | mdf |
| 07:40.50 | brettnem | _Vile: do you know of EPGN? Alpheus communications? |
| 07:40.57 | _Vile | nope |
| 07:41.00 | linagee | PTG123: hrm. i know a friend that would do that, but how many commercial VoIP providers would do that? |
| 07:41.03 | brettnem | bah.. texas firm |
| 07:41.06 | MuppetMaster | tomas: But no reason you can not run it on some old laptop, as this is what I use to do. Just wanted something a little more stable/secure that could handle calls in case my ADSL/Router went down. |
| 07:41.27 | _Vile | all I know is, my une prices are going to go up in a few years :) |
| 07:41.36 | linagee | PTG123: and besides, an emergency analog line out is not for missed calls, it's for 911 if your net connection goes out |
| 07:41.41 | brettnem | I build their fiber optic network. over 100 collocations IN SBC end offices across texas. about 90% were homerunned OC-48 |
| 07:41.42 | PTG123 | linagee: i would think everyone.. when we launch we will :) |
| 07:41.43 | MuppetMaster | justinnnn: You still have not posted what your problem actually is. To say txfax is not working is not quite enough to go off of to help you. |
| 07:41.44 | justinnnn | ill fix ur adsl router... u fix my txfax... |
| 07:41.51 | _Vile | nice |
| 07:41.56 | brettnem | and the loop cost just keeps going down |
| 07:42.01 | _Vile | I can't buy UDF anymore |
| 07:42.02 | *** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
| 07:42.03 | justinnnn | muppermaster... when we send the fax.. it seg faults the server.. |
| 07:42.05 | PTG123 | linagee: use your cell phone in that case |
| 07:42.08 | _Vile | unless it's retail |
| 07:42.15 | PTG123 | voip saves you money, but if you are buying an extra line, doesn't save you much |
| 07:42.16 | justinnnn | im using libtif 3.7.1, spandsp 10pre2 |
| 07:42.16 | Qwell | Night all, PTG123 |
| 07:42.18 | linagee | PTG123: and if cell networks jam? |
| 07:42.22 | PTG123 | qwell: night :) |
| 07:42.24 | justinnnn | cvs from middle of january... |
| 07:42.31 | MuppetMaster | justinnnn: Sounds like a bug. I would recommend ensuring you are on the latest CVS HEAD and then try again. |
| 07:42.35 | _Vile | I think that's illegal, but I have an FCC document telling me it isn't |
| 07:42.46 | brettnem | UDF? |
| 07:42.52 | linagee | PTG123: if i am calling someone within the same phone switch, the call should pretty much go through unless the entire switch is jammed or the batteries and gas are dead |
| 07:42.52 | brettnem | distribution feeder? |
| 07:42.59 | MuppetMaster | justinnnn: If that does not work, then I would recommend running in debug mode, and then doing a bt/bt full on the core file and reporting to Mantis. |
| 07:43.05 | MuppetMaster | justinnnn: Segfaults are never good. |
| 07:43.14 | _Vile | however, those fucks telling me that I have to switch to QPP immediately or face a $3 charge per line that I have with using their UNE switched service, *IS* illegal |
| 07:43.16 | linagee | do COs have gas generator backups? |
| 07:43.25 | _Vile | and I'm going to sue the fuck out of them for it |
| 07:43.27 | PTG123 | linagee: what are you saying? :) |
| 07:43.27 | _Vile | UDF |
| 07:43.34 | _Vile | Unbundled Dark Fiber |
| 07:43.41 | brettnem | wtf is UNE switched service? sounds like UNE-P or retail |
| 07:43.43 | brettnem | resale |
| 07:43.45 | PTG123 | linagee: your t1 goes over the same coper as your phoneline |
| 07:43.47 | brettnem | ah |
| 07:43.47 | linagee | _Vile: dark fiber. lol |
| 07:43.49 | PTG123 | assuming you have backup power at your place |
| 07:43.51 | _Vile | UNE-P which is now QPP |
| 07:43.53 | PTG123 | and your voip provider has backups |
| 07:43.56 | linagee | _Vile: gotta eat your fiber so it stays dark. :) |
| 07:43.56 | PTG123 | it should be as stable |
| 07:43.58 | _Vile | for Qwest anyway |
| 07:43.59 | mark64 | i'm writing a prepaid system in perl.. i'm using sip clients, using asterisk to terminate pstn calls. anyone got any idea to block simulataneous call from same user? |
| 07:44.03 | _Vile | UNE-P is gone |
| 07:44.05 | MuppetMaster | justinnnn: Daemons should never segfault.... |
| 07:44.18 | dfunnell | Hi guys, who wants a real challenge? |
| 07:44.22 | brettnem | _Vile: yeah. that whole EPGN net I built was UDF |
| 07:44.24 | MuppetMaster | dfunnell: Fire. |
| 07:44.28 | justinnnn | yips, thats why i thought it might be an issue :) |
| 07:44.29 | mark64 | i need to find a way to see who is using the channels |
| 07:44.33 | linagee | PTG123: i don't have a t1. i have DSL. that goes down if the CO loses connection to my ISP |
| 07:44.35 | brettnem | about 8,000 miles of it! just in texas |
| 07:44.35 | _Vile | brett, udf is gone. |
| 07:44.41 | _Vile | sad shit man |
| 07:44.42 | brettnem | not for me! |
| 07:44.51 | _Vile | you have 18 months |
| 07:44.54 | PTG123 | linagee: well if you wanna spend the extra money go for it :) |
| 07:44.56 | _Vile | so enjoy it :) |
| 07:45.00 | PTG123 | linagee: i like my voip only setup |
| 07:45.01 | PTG123 | heh |
| 07:45.02 | brettnem | nah.. |
| 07:45.06 | brettnem | it'll be longer than that |
| 07:45.10 | dfunnell | MuppetMaster: Aha! Have BRI ISDN running with CAPI. When first channel is in use * will not recognise that second channel is ringing. |
| 07:45.15 | linagee | PTG123: yeah. why not. spend extra money. have a direct copper pair right to the 911 call center. LOL. |
| 07:45.15 | brettnem | I work for a badass mo-fo |
| 07:45.17 | _Vile | what's your e-mail addy, I |
| 07:45.21 | mark64 | please help :) |
| 07:45.23 | _Vile | 'll send you the FCC direction |
| 07:45.25 | linagee | PTG123: then you can buy a special red colored phone and everything. :) |
| 07:45.33 | _Vile | sayin exactly how long you have :) |
| 07:45.39 | MuppetMaster | dfunnell: Sorry, Asterisk and internal hardware are not my thing, never done it. I use Asterisk purely as a softswitch with ATAs and softphones... |
| 07:45.40 | tomas- | MuppetMaster: is that sipura thing separate from the pbx card? are there pbx cards that work directly with the analog phones? including fax? |
| 07:45.42 | linagee | PTG123: and buy a special signal light with specialized logo. :) |
| 07:45.44 | PTG123 | linagee: i'll use smoke signals |
| 07:45.53 | MuppetMaster | dfunnell: Keeps it easy and lets me run on many more types of hardware (ie - old laptop). |
| 07:45.59 | _Vile | unless you bought resale |
| 07:46.01 | PTG123 | linagee: how do you do help i have fallen and can't get up in smoke signals |
| 07:46.04 | _Vile | then I wouldn't worry about it |
| 07:46.07 | _Vile | see |
| 07:46.10 | brettnem | pfft resale.. |
| 07:46.12 | dfunnell | MuppetMaster: No worries, doesn't seem to be many using ISDN. |
| 07:46.16 | dfunnell | Any other takers? |
| 07:46.17 | _Vile | the whole object of the ilec currently is to kill UNE |
| 07:46.17 | brettnem | I think I used to pay $0.35/mile |
| 07:46.19 | _Vile | that's it. |
| 07:46.21 | brettnem | sure |
| 07:46.26 | linagee | PTG123: i pay about $2.50/mo for an emergency landline |
| 07:46.27 | PTG123 | i just wanna let everyonbe know if a nuclear bomb hits here, my servers will still be online they tell me :) |
| 07:46.29 | brettnem | I don't think it will last.. really |
| 07:46.29 | _Vile | pay off politicians |
| 07:46.31 | _Vile | kill clecs |
| 07:46.33 | _Vile | kill une |
| 07:46.36 | linagee | PTG123: the highest price is the FXO/FXS/etc |
| 07:46.36 | _Vile | call it a day. |
| 07:46.37 | justinnnn | if anyone wants to get monies helping me fix app_txfax please email me at justinm@mpa.net.au :))) |
| 07:46.39 | brettnem | pfft |
| 07:46.46 | brettnem | damn bush |
| 07:46.50 | MuppetMaster | tomas: Fax over IP is still not perfect, as you really need the T.38 capability. And yes, the ATAs from Sipura are external. You may run Asterisk without any internal hardware at all using it only as a softswitch. If you want internal hardware, then you may want to look at http://www.digium.com cards. |
| 07:46.51 | PTG123 | linagee: well if its that cheap, thats different. |
| 07:46.57 | _Vile | it ain't bush here |
| 07:46.57 | tomas- | are the fancy "digital" phones ones that work directly with a pbx without a analog-to-digital converter? |
| 07:47.02 | _Vile | though, I don't like the faggot |
| 07:47.05 | brettnem | haha |
| 07:47.07 | linagee | PTG123: sure it's metered if i have outgoing calls |
| 07:47.11 | _Vile | no harm bkw |
| 07:47.12 | linagee | _Vile: don't insult faggots |
| 07:47.19 | _Vile | no harm lin |
| 07:47.21 | MuppetMaster | justinnnn: If it is segfaulting on the latest CVS HEAD, you will simply be paying someone to post a bug on Mantis. |
| 07:47.22 | linagee | PTG123: and sure i can only call within the area code |
| 07:47.24 | _Vile | though I don't like the fucker |
| 07:47.35 | brettnem | _Vile: be nice to fuckers |
| 07:47.35 | _Vile | he ain't at fault here |
| 07:47.38 | MuppetMaster | justinnnn: Unless of course they may fix the bug and post it. |
| 07:47.40 | brettnem | ;) |
| 07:47.42 | _Vile | god damnit |
| 07:47.45 | brettnem | sorry |
| 07:47.46 | brettnem | hahaha |
| 07:47.48 | _Vile | he ain't at fault here |
| 07:47.57 | _Vile | it's those fucking lobbyists |
| 07:48.26 | _Vile | that those billion dollar companies spending millions onf lobbying. |
| 07:48.29 | brettnem | well.. I feel that our organization is uniquely positioned in the market.. <sigh> |
| 07:48.31 | MuppetMaster | tomas: Are you looking to use the T1 for voice and data? Do you want the T1 to be used for TDM networking and then IP as well? I would say it would be better just to use the T1 as IP and mix your voice and data with a QOS capable router. |
| 07:48.33 | _Vile | and buying each other |
| 07:48.38 | MuppetMaster | tomas: Which 'mountains' are you going to be in? |
| 07:48.53 | brettnem | _Vile: so what do you think.. think SBC should switch to AT&T? |
| 07:48.55 | _Vile | the *only* reason Verizon can *buy* MCI now is because of recent laws |
| 07:48.58 | brettnem | but I hope so |
| 07:49.01 | brettnem | er boy |
| 07:49.05 | _Vile | it was *ILLEGAL* a year a go. |
| 07:49.14 | iceyp | hey guys, is there any issue with doing a Dial(IAX2/user@localhost/extension) ? |
| 07:49.26 | brettnem | oh who cares.. there is nothing left of that market |
| 07:49.26 | *** join/#asterisk Rival (~rival@66.177.249.219) |
| 07:49.35 | tomas- | MuppetMaster: green mountains, in vermont |
| 07:49.37 | brettnem | although |
| 07:49.39 | _Vile | I think no long distance companies should ever be involved in local service |
| 07:49.42 | MuppetMaster | iceyp: Doesn't look at it, but of course depends on your config. |
| 07:49.44 | _Vile | it kills competition |
| 07:49.44 | brettnem | they do have quite an infrastructure |
| 07:49.57 | dfunnell | Anyone know CAPI/ISDN? |
| 07:49.59 | brettnem | yeah.. that whole 271 relief was a mistake |
| 07:50.03 | brettnem | dfunnell: maybe |
| 07:50.13 | _Vile | dfunnell, probably not, |
| 07:50.16 | MuppetMaster | tomas: So, in the US. You may want to look at the likes of http://www.teliax.com to provide you with PSTN interconnection, then you may use the T1 as pure data and not have to worry about internal PBX cards. Will make your life much simpler. |
| 07:50.22 | dfunnell | brettnem: Will pay in beer for help! |
| 07:50.29 | brettnem | hmm ebeer |
| 07:50.35 | iceyp | MuppetMaster reason being is, i'm using one DB and will be housed on 2 boxs... it will dial the local box first then the second box second, as i'd be unsure what server they connected to |
| 07:50.36 | MuppetMaster | tomas: I use TelIAX for termination in and out of Denver, never had a problem, great quality and inexpensive. |
| 07:50.38 | _Vile | dfunnell, you need to do better than that for this crowd |
| 07:51.03 | MuppetMaster | iceyp: Ah. Have you thought of using DUNDi to create a federated PBX? May be applicable. |
| 07:51.09 | _Vile | brett, apparently needs crack. Iceyp is a glue sniffer, and Muppet needs some flexoril. |
| 07:51.09 | dfunnell | brettnem: Can't get * to recognise when second BRI channel is ringing - second channel works and it is dialling in, just can't get * to figure it out. |
| 07:51.18 | iceyp | MuppetMaster nah i hadnt |
| 07:51.21 | brettnem | federated PBX.. that just sounds cool |
| 07:51.27 | MuppetMaster | Don't it though. |
| 07:51.32 | dfunnell | _Vile: It's pretty good beer! |
| 07:51.42 | brettnem | what do you mwan.. you say it is dialing in (you mean out?) |
| 07:51.43 | _Vile | dude it just won't cut it |
| 07:51.49 | _Vile | but ship me a case, btw. |
| 07:52.05 | MuppetMaster | I have used DUNDi between two internal PBXs with a private numbering plan, works great, creates transperancy and ease of management. Having said this, I don't actually need it for myself, so more of a POC. |
| 07:52.06 | iceyp | actually i may be able to get the perl script to re write the Dial depending on which box its being run from |
| 07:52.20 | MuppetMaster | iceyp: Definately. |
| 07:52.22 | _Vile | blah |
| 07:52.22 | _Vile | fuck the ilec |
| 07:52.25 | _Vile | love the clec |
| 07:52.35 | _Vile | and the ixc should never exist |
| 07:52.37 | _Vile | im going to bed. |
| 07:52.38 | brettnem | heh.. </soapbox> ;) |
| 07:52.40 | iceyp | MuppetMaster does this mean the user will be assigned a singal extension number and the call will be sent to which ever server they connected to? |
| 07:52.44 | tomas- | MuppetMaster: well, im not familiar with my options, truthfully.... all i want is t1 for my internet (which i'll have a few web servers running off of), as well as handling my voice for 3 voice lines and 1 fax line... it would be extremely nice to have a PBX that can handle the 3 lines for voicemail, etc... how it happens, i dont care... |
| 07:52.51 | MuppetMaster | iceyp: Yes. |
| 07:52.53 | _Vile | later |
| 07:52.54 | brettnem | _Vile: you're reminding me of dennis leary |
| 07:53.03 | dfunnell | brettnem: Can dial in via ISDN and can dial out. If first channel is in use, though, then can't manage to dial in on second. |
| 07:53.03 | _Vile | you know |
| 07:53.07 | _Vile | that guy had some sense |
| 07:53.15 | brettnem | I liked him.. heh |
| 07:53.23 | dfunnell | brettnem: Can dial out (from * to world) always, inc. using second channel. |
| 07:53.38 | iceyp | MuppetMaster got a description of how its done? |
| 07:53.43 | brettnem | have you done a span debug? |
| 07:53.48 | MuppetMaster | tomas: Then, definatly you should go with a T1 only as data, and let Asterisk provide your PBX functions interconnected to the likes of TelIAX. Then hang ATAs (ie - Sipuras) and Softphones (ie - Eyebeam from http://www.xten.com, may also do h.263 video) and you have a perfect SoHo system. |
| 07:53.51 | brettnem | what does your provider say is happening? |
| 07:54.06 | dfunnell | brettnem: On phone now, for example. If someone rings me then they get ring tone and ISDN NTU shows its ringing, just * doesn't register an incoming call. |
| 07:54.11 | brettnem | MuppetMaster: eyebeam work with conferencing?? |
| 07:54.15 | dfunnell | brettnem: Provider tells me it's working ok. |
| 07:54.21 | MuppetMaster | tomas: You could use asteriskathome.sourceforge.net on an old piece of kit and be off and running in no time. Just remember to get a router that supports QOS to prioritize the VoIP across that T1. |
| 07:54.31 | MuppetMaster | brettnem: Have not actually tried at this stage. |
| 07:54.39 | brettnem | woah S-interface? |
| 07:54.39 | dfunnell | brettnem: Tend to believe them as can see NTU flashing a ring. |
| 07:55.04 | *** join/#asterisk scanna (~scannachi@81-174-16-211.f5.ngi.it) |
| 07:55.06 | brettnem | dfunnell: what country? |
| 07:55.11 | iceyp | MuppetMaster what if i just add a dialplan? |
| 07:55.15 | dfunnell | brettnem: New Zealand. |
| 07:55.18 | iceyp | how could i do it |
| 07:55.24 | dfunnell | brettnem: Use euro ISDN. |
| 07:55.27 | brettnem | didn't smell like US... heh |
| 07:55.32 | brettnem | "us" har |
| 07:55.35 | brettnem | oy it's late |
| 07:55.39 | brettnem | hmm |
| 07:55.50 | dfunnell | brettnem: Is that complement or an insult ;-) |
| 07:56.00 | brettnem | I think it's an insult.. |
| 07:56.06 | brettnem | but I think it was insulting me |
| 07:56.06 | dfunnell | ;) |
| 07:56.09 | brettnem | heh |
| 07:56.13 | MuppetMaster | iceyp: Still not entirely clear on what exactly you are trying to do to provide a precise dialplan that would support it. Let me understand. You have two systems with extensions hanging off them and you want to be able to dial both systems from one of those extensions in order to have one 'virtual' PBX. Correct? |
| 07:56.14 | dfunnell | Heh heh. |
| 07:56.21 | tomas- | MuppetMaster: well, i truthfully HAVE TO get my phone service from the T1 provider, because if i don't spend $100 on top of the $300 T1 line per month, then they wont give it to me |
| 07:56.23 | brettnem | did you try a span debug? |
| 07:56.41 | MuppetMaster | iceyp: If this is the case, a very simple way would be to have two Dial strings in the dialplan, one to try local first and if that fails try the second switch. |
| 07:56.44 | brettnem | tomas-: what region are you in? |
| 07:56.49 | dfunnell | brettnem: Nope, what is this span debug magic you speak of? |
| 07:56.56 | brettnem | hmm |
| 07:56.57 | MuppetMaster | brettnem: He will be in the Green Mountains of Vermont. |
| 07:57.10 | brettnem | will be? am I stuck in the past or what? |
| 07:57.11 | *** join/#asterisk erwinism (~pogz@210.213.143.73) |
| 07:57.26 | brettnem | dfunnell: well, for pri I'd do "pri debug span 1" or something of the like |
| 07:57.27 | MuppetMaster | iceyp: So, you would execute Dial(IAX/user@localhost) if fail (n+1) Dial(IAX/user@remotehost) |
| 07:57.35 | dfunnell | brettnem: How many typing hands have you got, anyway? |
| 07:57.41 | erwinism | my asterisk server is behind firewall, what ports should i enable in the router? |
| 07:57.43 | MuppetMaster | icepy: I use something similar as I try to do an ENUM, and if that fails then it goes to an alternate dial string. |
| 07:57.50 | brettnem | 2.5 |
| 07:57.51 | dfunnell | brettnem: From the * console? |
| 07:57.54 | brettnem | yep |
| 07:57.56 | MuppetMaster | icepy: But if you use DUNDi, this may all happen automagicaly. |
| 07:58.14 | brettnem | oh dundi.. |
| 07:58.14 | MuppetMaster | erwinism: Depends on what is in your sip.conf, iax.conf and rtp.conf. |
| 07:58.20 | brettnem | I wish I were impressed with you |
| 07:58.32 | MuppetMaster | erwinism: Defalut value for SIP is 5060, rtp will be a range, just look in your rtp.conf file. |
| 07:58.34 | tomas- | MuppetMaster: that ASSUMES they can even DO VoIP |
| 07:58.46 | MuppetMaster | brettnem: You don't sound like a fan of DUNDi? |
| 07:58.50 | brettnem | I'd like to see failover built into dundi |
| 07:58.54 | brettnem | no I'm not particularly |
| 07:58.56 | MuppetMaster | tomas: Who are the they? |
| 07:59.02 | *** join/#asterisk _m_ (~m@fbta199.fbta.uni-karlsruhe.de) |
| 07:59.05 | brettnem | dundilookup never came up to speed, did it? |
| 07:59.10 | erwinism | MuppetMaster i use the defaults |
| 07:59.14 | MuppetMaster | tomas: Your telco providing the T1 does not have to do VoIP, as they are just providing you with a dumb data tunnel. |
| 07:59.18 | iceyp | MuppetMaster yeah, problem being I use 1 database and currently have... Dial(IAX2/ape@apevoip.fast.co.nz/1000000,30,tr) then below that Dial(IAX2/wix@wixvoip.fast.co.nz/1000000,30,tr) |
| 07:59.26 | mark64 | is there a way to look at the realtime cdr? i noticed the cdr is only written after call completion |
| 07:59.33 | brettnem | if SER supported DUNDi.. perhaps I'd reconsider.. |
| 07:59.42 | iceyp | but i think if i dia; apevoip.fast.co.nz/1000000 its going to loop to the same extension and keep looping |
| 07:59.45 | mark64 | asterisk must be storing this temporarily somewhere right? |
| 07:59.53 | MuppetMaster | iceyp: Even with a database you still have priorities that are going to be followed. So if your first dial command is at 2, then your second dial command should be at 103 (n+1). |
| 07:59.58 | MuppetMaster | brettnem: Ah, I see. |
| 08:00.07 | MuppetMaster | brettnem: Write the interface, it is an open standard! ;) |
| 08:00.18 | tomas- | MuppetMaster: I know they dont NEED to provide VoIP, but I need to have an extra $100 in services from them on TOP of my T1 for them to even get me set up (since I'm somewhat near the edge of their service area) |
| 08:00.18 | brettnem | mark64: you can show channels.. show channel SIP/myphone-da2d |
| 08:00.24 | iceyp | n+1? |
| 08:00.34 | brettnem | MuppetMaster: yeah.. I'm a user.. not a coder. ;) |
| 08:00.40 | tomas- | MuppetMaster: just some rinky dink Internet/phone provider in Vermont: www.sover.net . Really nothing too professional looking |
| 08:00.44 | mark64 | hi brettnem.. show channels doesn't show my the callerid of the dialling party |
| 08:00.50 | MuppetMaster | brettnem: Me too... Not much beyond some PHP/MySQL these days... |
| 08:00.59 | brettnem | mark64: show channel <channel name> |
| 08:01.05 | MuppetMaster | tomas: Ah, I see, but that is unrelated to whether 'they' do VoIP. |
| 08:01.06 | mark64 | i have sip client dialing a pstn number |
| 08:01.08 | tomas- | MuppetMaster: There also IS Verizon, i think they own the lines |
| 08:01.17 | brettnem | MuppetMaster: yeah me too.. blah |
| 08:01.19 | iceyp | MuppetMaster : |
| 08:01.19 | iceyp | exten => 1000000,1,Dial(IAX2/ape@apevoip.fast.co.nz/1000000,30,tr) ; |
| 08:01.20 | iceyp | exten => 1000000,2,Dial(IAX2/wix@wixvoip.fast.co.nz/1000000,30,tr) |
| 08:01.20 | iceyp | exten => 1000000,3,Macro(vmessage,1000000) |
| 08:01.20 | iceyp | exten => 1000000,4,Hangup |
| 08:01.37 | MuppetMaster | iceyp: Effectively, what ever priority you first dial command is at (let's say 4) add 101 to your second dial commmand which is picked up if the first one fails. So 105. |
| 08:01.46 | brettnem | why so many 0's.. yow.. I'd never keep track |
| 08:01.53 | iceyp | ahh ok |
| 08:02.02 | iceyp | brettnem i expect a lot of signups |
| 08:02.08 | MuppetMaster | iceyp: So what you just displayed, the first dial would be 1, the second should be 102 and your macro 103, keep in mind that you would need to duplicate your Macro after each dial if necessary. |
| 08:02.09 | brettnem | I like using that ${DIALSTATUS} variable |
| 08:02.10 | iceyp | its a NZ FWD type service |
| 08:02.19 | *** join/#asterisk jmhunter (~jmhunter@64.77.199.223) |
| 08:02.19 | *** mode/#asterisk [+o jmhunter] by ChanServ |
| 08:02.31 | brettnem | iceyp: hope you got some SER in there |
| 08:02.34 | MuppetMaster | iceyp: Have a look at the Wiki at the Dial command, there are more details in there on how to use multiple dials in the same extension of a dialplan. |
| 08:02.39 | iceyp | brettnem nah |
| 08:02.44 | brettnem | ek! |
| 08:02.51 | MuppetMaster | iceyp: Yes, you definately would want SER to scale. |
| 08:03.06 | MuppetMaster | iceyp: But I guess there are more sheep there than people, so maybe no big deal... |
| 08:03.15 | brettnem | HA |
| 08:03.19 | mark64 | Brettnem: thanks show channel worked!.. any idea to grab it from agi? |
| 08:03.30 | iceyp | MuppetMaster my issue is how does it dial the local server if its IAX2/ape@apevoip.fast.co.nz/1000000 |
| 08:03.35 | MuppetMaster | mark64: Which AGI library ar you using? |
| 08:03.40 | mark64 | perl |
| 08:03.42 | iceyp | when the iax server receives that its going to look at it's self |
| 08:03.46 | tomas- | MuppetMaster: well i dont know how to spend $100/mo unless i get several phone lines from them, etc... |
| 08:03.47 | iceyp | then dial IAX2/ape@apevoip.fast.co.nz/1000000 again |
| 08:04.01 | MuppetMaster | tomas: What exactly is the $100/month for? |
| 08:04.06 | brettnem | try channel status? |
| 08:04.14 | MuppetMaster | iceyp: You lost me. |
| 08:04.57 | tomas- | MuppetMaster: can asterisk work with multi-line analog input? the $100 is so I pay them $400/month ($300 for the T1 alone) so it's cost justified for THEM for the whole support thing.... |
| 08:04.59 | mark64 | nada :) |
| 08:05.13 | iceyp | ok lets consider this from a singal server setup, what would happen if i dialed IAX2/ape@apevoip.fast.co.nz/1000000 |
| 08:05.21 | brettnem | ok guys.. need sleep |
| 08:05.23 | MuppetMaster | Mark64: Have a look at phpagi.sourceforge.net, it gives a class that allows access to channel_status within an AGI. You may then convert that into any language you like |
| 08:05.31 | dfunnell | brettnem: No BRI debug, but found CAPI DEBUG - proceeds to freak the fuck out when call is made over BRI. |
| 08:05.38 | dfunnell | terminates in - Ouch ... error while writing audio data: : Broken pipe |
| 08:05.39 | iceyp | that would connect to the same server and lookup 1000000 in the same context, then try dial the extension or IAX2/ape@apevoip.fast.co.nz/1000000 again? |
| 08:05.44 | dfunnell | Segmentation fault |
| 08:05.47 | MuppetMaster | tomas: So the $100 is just a premium for being in the Green Mountains with no additional benefits? |
| 08:05.49 | brettnem | oops |
| 08:05.57 | brettnem | maybe a driver problem |
| 08:05.58 | dfunnell | Then CAPI seems to roll over and die ;-( |
| 08:06.08 | MuppetMaster | iceyp: I thought you were trying to dial a local server and if it failed dial the second server? |
| 08:06.12 | *** join/#asterisk datareactor (~datareact@203.81.192.53) |
| 08:06.18 | brettnem | MuppetMaster: luxury tax |
| 08:06.19 | MuppetMaster | iceyp: Why try the same local extension twice? |
| 08:06.30 | dfunnell | Could be. Although debug messages look ok, just lots of them. |
| 08:06.38 | tomas- | MuppetMaster: no, i mean i need to spend $100 in OTHER services (i figure 3 voice lines, 1 fax, long distance, etc) they provide |
| 08:06.44 | MuppetMaster | brettnem: T1s are a luxury in Vermont? Wow. Yes, Asterisk may handle Analog input, you may either use a Sipura 3000, or an internal card. |
| 08:06.52 | MuppetMaster | tomas: Ah, I see. |
| 08:06.53 | brettnem | goodnight all.. dfunnell good luck.. look real close at those logs |
| 08:07.01 | dfunnell | Ok, cheers mate. |
| 08:07.12 | MuppetMaster | tomas: And you are looking to cost justify an Asterisk implementation based on not having to pay the $100/month. What is your ROI horizon? |
| 08:07.12 | brettnem | MuppetMaster: read "sarcasm" ;) sorry.. tired.. fading away |
| 08:07.19 | iceyp | well i was oing to try local, then the next host, but both servers use the same db, so the next host will be IAX2/nexthost , however when being run from that second host, it will check locally, then IAX2/nexthost being it's self |
| 08:07.31 | MuppetMaster | brettnem: Have sweet dreams. |
| 08:07.45 | brettnem | oh I will... I will... I will.... |
| 08:07.49 | iceyp | unless i modify the secondhost DNS to think the secondhost is the primary |
| 08:08.00 | *** join/#asterisk mark64 (mark64@bb220-255-34-207.singnet.com.sg) |
| 08:08.17 | dfunnell | Anyone else have a clue about ISDN/CAPI? |
| 08:08.20 | MuppetMaster | iceyp: But the hosts have difference host names, no? I would definately recommend setting up a dialplan that gives each Asterisk instance a unique set of numbers, otherwise you are asking for confusion. |
| 08:09.19 | *** part/#asterisk jmhunter (~jmhunter@64.77.199.223) |
| 08:10.16 | tomas- | MuppetMaster: oh who knows what sort of business i'll be bringing in....scary, but by june i need to at LEAST make that much money per month in my business(es) so i at least break even (I work full-time for an employer, so this is a side business until it grows full-time).... if i dont go with the T1 data + internal server hosting + 3 phone lines + 1 fax + LD + PBX, then i'm more or less stuck with dialup, not a good thing when im |
| 08:11.03 | MuppetMaster | tomas: No, definatly want broadband. And what you are looking to do is feasible with Asterisk, the key is just to take some old hardware/buy used, so you don't have that to factor into your business case. |
| 08:11.12 | MuppetMaster | tomas: Are you in a valley or on a mountain top? |
| 08:11.39 | MuppetMaster | tomas: Meaning, maybe there is someone providing some type of wireless into the Vermont mountains (beautiful area by the way, was there last spring) stayed on Lake Willoughby. |
| 08:11.59 | MuppetMaster | tomas: Wishful thinking, i Know... |
| 08:12.09 | MuppetMaster | tomas: There is always satellite, but would not recommend voice over that. |
| 08:12.21 | tomas- | MuppetMaster: wireless is out of the question. expensive as is, i dont know if i'll get a decent LOS, and no potential to grow like the T1 + voice + pbx |
| 08:13.39 | tomas- | MuppetMaster: maybe if i make the decision to spend $400 on internet + voice per month starting June, I will actually pressure myself into working hard enough that i can bring up a sustainable business instead of sitting pretty in my comfortable corporate-style zombie employee lifestyle |
| 08:14.10 | MuppetMaster | tomas: Nothing like creating a fire under your ass to help motivate yourself. |
| 08:15.05 | MuppetMaster | tomas: Do you plan to have other folks in your home office, or are you going to be using 3 phones yourself? |
| 08:15.32 | *** join/#asterisk tafazzi (~Dario@eporroj0-1.customer.nettuno.it) |
| 08:15.36 | MuppetMaster | tomas: Keep in mind, the likes of TelIAX can provide you with a business service as well, so you just have ATAs/Softphones and let them worry about the Asterisk stuff. |
| 08:15.42 | MuppetMaster | tomas: Of course costs a bit more. |
| 08:15.43 | mark64 | anyone know how to call "sip show channels" in perl AGI? |
| 08:16.16 | MuppetMaster | mark64: Not in PERL, but in PHP. Did you take a look at phpagi.sourceforge.net? Should be easy enough to convert there 'channel_status' class into a Perl script. |
| 08:16.39 | mark64 | ok i'll check it out.. thanks! |
| 08:16.45 | MuppetMaster | mark64: Good luck! |
| 08:17.58 | iceyp | MuppetMaster well both servers have the same ip address , anycast ip ranges, so if ur in wellington u get connected to that box, if ur in auckland, u get connected to that box |
| 08:18.07 | iceyp | setting the dialplan is done via the DB |
| 08:18.27 | MuppetMaster | iceyp: Ah, ok. |
| 08:18.35 | MuppetMaster | iceyp: Are you using Realtime for this? |
| 08:18.39 | iceyp | and because they both use the same DB, it'll do DIAL localhost, then DIAL secondaryhost |
| 08:19.00 | MuppetMaster | iceyp: This is how you are doing load balancing? |
| 08:19.14 | iceyp | now when a user connects to secondaryhost and dials an extension, its doing DIAL localhost (all fine), then DIAL secondaryhost (which is its' self) |
| 08:19.17 | MuppetMaster | iceyp: Now I understand. |
| 08:19.46 | tomas- | MuppetMaster: well, it'd be cool to find someone to help me out, but if anyone, just 1 or 2 people, and i'm not sure if i want to get into the partnership thing anyway. anyway, what would be going on is my personal phone, business phone for my employer, business phone for my home business and fax for whatever (as well as long distance, etc) |
| 08:19.48 | Rival | this asterisk is pretty sweet |
| 08:19.51 | iceyp | so from the secondaryhost perhaps i should modify the /etc/hosts file to think secondaryhost is actualy the primary server |
| 08:20.00 | *** join/#asterisk AgiNamu (~AgiNamu@216.230.151.230) |
| 08:20.12 | MuppetMaster | iceyp: The problem you will have, is that the Asterisk instance will not be able to tell the difference between the two dials as they both dial the same hostname. Not sure how you sort that out. |
| 08:20.15 | AgiNamu | How do you handle billing call forwarding? |
| 08:20.51 | MuppetMaster | tomas: With Asterisk, you may have multiple incoming phones/lines but only need one physical phone to handle all of those. |
| 08:20.58 | iceyp | MuppetMaster : |
| 08:20.59 | iceyp | exten => 1000000,1,Dial(IAX2/1000000,30,tr) ; |
| 08:20.59 | iceyp | exten => 1000000,2,Dial(IAX2/wix@wixvoip.fast.co.nz/1000000,30,tr) |
| 08:20.59 | iceyp | exten => 1000000,3,Macro(vmessage,1000000) |
| 08:20.59 | iceyp | exten => 1000000,4,Hangup |
| 08:21.21 | iceyp | so on the wixvoip.fast.co.nz/1000000 i'll set the dns to actualy not point to irs self but to the primary box |
| 08:21.26 | iceyp | this will work, yes? |
| 08:21.34 | MuppetMaster | tomas: So, for example, I have a number in the UK I use for business, a number I use in Spain for personal, a number in the US for personal, a number in Holland for business. Each answers with a different greeting and then routes based on CLI or what is selected. |
| 08:21.46 | MuppetMaster | So it looks like I have multiple 'lines' but I only really need one phone for me and one for my wife. |
| 08:22.10 | MuppetMaster | tomas: So you don't really need 3 physical phones unless you have the need for 3 people in your home office. This could bring down the investment. |
| 08:23.04 | MuppetMaster | iceyp: In theory yes, but you still need to keep the (n+1) in mind to keep it clean. So it should be 1, 102, 103, 104, etc. |
| 08:23.08 | MuppetMaster | In terms of the priorities. |
| 08:23.14 | iceyp | yeah |
| 08:23.16 | iceyp | will do |
| 08:23.17 | Eight | Phones = Number of people. Lines = max( number of public numbers, number of concurrent calls) |
| 08:23.18 | MuppetMaster | iceyp: Athough it should work. |
| 08:23.19 | iceyp | i already modified the db |
| 08:23.26 | iceyp | sweet |
| 08:23.28 | iceyp | lets home |
| 08:23.31 | iceyp | hope* |
| 08:23.34 | MuppetMaster | iceyp: Let us know how it works. |
| 08:23.34 | tomas- | MuppetMaster: my budget is like $0, because not only am i buying a house that'll cost me up to $2000/mo, but im not sure as to my workflow with my employer (could get as low as 20-30 hr/week), and my contract is up for renewal in july/august (which im fairly sure they'll keep me, but STILL, its quite a bit of money pressure to ignore), i'm already going to be racking up credit cards, i'm also going to owe another loan that will hel |
| 08:23.37 | iceyp | is SER BSD friendly? |
| 08:24.02 | shepherd | ask #ser :) |
| 08:24.06 | tuxinator_linux | check the #SER channel |
| 08:24.11 | AgiNamu | ForkCDR just makes a copy of the CDRs? |
| 08:24.22 | Rival | tomas well if work != 40hr find sidework |
| 08:24.27 | MuppetMaster | tomas: So, your best bet is to variabalize your cost through OpEx as opposed to CapEx and use a service like TelIAX which provides the equivalent to VoIP centrex services. |
| 08:24.46 | MuppetMaster | iceyp: Ys. |
| 08:24.50 | MuppetMaster | iceyp: Yes that is. |
| 08:24.52 | Rival | i dunno about u but i put in 60-80 hrs a week easy but then again no family |
| 08:25.24 | Rival | is there a list of recommended asterisk supported voip providers |
| 08:25.25 | MuppetMaster | tomas: Then use the free X-Lite from http://www.xten.com with a USB headset on your existing computer. |
| 08:25.35 | Rival | xlite is nice |
| 08:25.47 | MuppetMaster | tomas: Then you only have the cost of the T1 line and the TelIAX service. |
| 08:25.52 | *** join/#asterisk Verliba (~Miranda@250.1-14-84.ripe.coltfrance.com) |
| 08:25.58 | tomas- | Rival: yeah, well it depends on how 'hard' i work -- i.e. how much of the hour im really doing work....lately ive been pushing 50 hours/week, but it will DEFINITELY go down to maybe even 15 hrs/week once in a while |
| 08:26.40 | tomas- | MuppetMaster: well, you've forgot the $100 in services the internet/phone company requires me to use, or no t1 at all |
| 08:27.02 | MuppetMaster | Rival: Yes, there are several. I personally use and recommend http://www.teliax.com. I have used http://www.nufone.net, would not personally recommend, but some swear by them. There are many more that may be found on the VoIP/Asterisk Wiki. |
| 08:27.08 | Rival | tomas ya i dunno what kinda job ya got but if im stuck in one office where i have to babysit something i take out my laptop plug into their network and work on other companies stuff via vpn ect |
| 08:27.16 | MuppetMaster | tomas: Ah, so the 100 is not optional. |
| 08:27.41 | Rival | thanks muppetmaster |
| 08:27.53 | tomas- | my choices are T1 + home-based servers + home/business phone lines + pbx OR dialup + home/business phone lines |
| 08:28.18 | MuppetMaster | tomas: Understood. Remoteness comes with a cost, unfortunately. |
| 08:29.13 | iceyp | is ser do everything that asterisk does? |
| 08:29.20 | iceyp | does ser* |
| 08:31.26 | MuppetMaster | iceyp: No. |
| 08:31.36 | Rival | tomas is there any way you can come up with to make up the cost of the t1 |
| 08:31.49 | Rival | perhaps you could come up with a semi profitable way to use your bandwith |
| 08:31.55 | MuppetMaster | iceyp: SER is a SIP Proxy primarily, which is good for scale and connecting two endpoints. When things like voicemail or other PBX functions are needed, that is when you put Asterisk in the stream. |
| 08:32.19 | iceyp | MuppetMaster but ser will assign users extensions etc? |
| 08:32.22 | iceyp | dialplans and more? |
| 08:32.32 | MuppetMaster | iceyp: FWD uses SER as the primary 'router' and then Asterisk to provide additional functions. This allows scale. |
| 08:32.32 | tomas- | there is then the prepwork of MOVING, which for me is going to take 3 weeks to move and settle in (i have to go to california to work on my parent's house and pick up most of my furniture, drive to new jersey to pick up the rest, and then to vermont, via rental truck, ITSELF costing $4,000), which means i'm potentially out of work for 3 weeks! which i TOTALLY cannot afford. especially going into so much debt....i can GET dialup, bu |
| 08:32.37 | MuppetMaster | iceyp: Yes. |
| 08:32.40 | tomas- | ie physical driving across the country |
| 08:32.59 | iceyp | MuppetMaster so serweb is a full functionable signup system for users |
| 08:33.10 | MuppetMaster | iceyp: More or less. |
| 08:33.20 | iceyp | and allows them to login and check their status etc |
| 08:33.22 | iceyp | u used it? |
| 08:33.30 | MuppetMaster | iceyp: But you would still need to provide some backend to merge your SER/Asterisk configs to provide the 'enhanced' services. |
| 08:33.48 | MuppetMaster | iceyp: No. So I am speaking theory out of my ass here, but have seen cases where it worked. |
| 08:34.13 | iceyp | well all i want to start with is the extnsion numbers and routing calls to other peers such as sipphone, fwd etc |
| 08:34.19 | MuppetMaster | iceyp: BTW - NZ is very nice, not too long ago I was in Christchurch for a wedding. Where are you located? |
| 08:34.28 | iceyp | auckland : |
| 08:34.30 | iceyp | :) |
| 08:34.40 | iceyp | I originally from south africa, i love it here |
| 08:34.57 | MuppetMaster | iceyp: SER can do that for you, and would probably scale better and be easier to manage. You may then add Asterisk to the mix as you want to provide voicemail, conferencing, etc. |
| 08:35.04 | dfunnell | Bloody South Africans - country is infested with them ;-) |
| 08:35.21 | MuppetMaster | iceyp: I am an Australia fan myself, as NZ really is the edge of the earth. I have spent some time in Auckland, great place, just so far away from everywhere. |
| 08:35.32 | tomas- | Rival: oh, i INTEND on running a set of web servers/services from my t1 line at home....essentially web-based portal services (i.e. "turnkey websites" for the layman), with anywhere up to $100/mo and $300 or so setup fee...so if i get only a FEW people new per month, it's paid for....i know there's stuff like InterLand and HomeStead, but i plan on working locally, definitely over the phone and maybe face-to-face, local advertising, |
| 08:35.44 | MuppetMaster | iceyp: Originally from the US myself, my wife is Spanish, and she would never move 'downunder' as it is just too far away from the family. I would in a heartbeat. |
| 08:35.58 | dfunnell | iceyp: Hear that? An AU fan? I kinda like Canda more than the US, how about you? |
| 08:36.25 | MuppetMaster | dfunnell: Yeah, yeah, yeah. If you like it cold and don't mind getting trampled by moose. |
| 08:36.32 | tomas- | i seriously need a business credit card, and maybe a business loan to pull this all off |
| 08:36.35 | dfunnell | Well put |
| 08:36.40 | Rival | tomas good luck but its a growing industry =p |
| 08:36.50 | Rival | tomas avoid credit cards |
| 08:36.52 | MuppetMaster | tomas: Vermont may have some small business loans you may want to look into. |
| 08:36.55 | Rival | take out a loan |
| 08:37.07 | MuppetMaster | tomas: Credit cards=pure evil. The bane of America at the moment. |
| 08:37.13 | iceyp | same, hehe |
| 08:37.18 | MuppetMaster | tomas: I shudder at the lack of spending on the amount of debt... |
| 08:37.32 | MuppetMaster | lack of savings I mean. |
| 08:37.50 | MuppetMaster | credit providers will give cards to a cat right now |
| 08:37.54 | MuppetMaster | bastards |
| 08:37.59 | MuppetMaster | don't get me started |
| 08:38.12 | tomas- | well, interest rates ARE higher for CC's |
| 08:39.52 | dfunnell | Ok guys, I'm going to go and commit suicide now because my customer is waiting on a PABX that I can't get working. If anyone has any flashes of brillance re. CAPI/ISDN then drop me a line, will ya? |
| 08:39.53 | tomas- | and paying back a loan is more predictable, with a known monthly payment (putting adjustable rates aside) |
| 08:40.08 | Zeeek | be patriotic! get in debt, now! |
| 08:40.09 | MuppetMaster | dfunnell: Good luck, apologies could not help on that one. |
| 08:40.30 | MuppetMaster | All good Americans are in debt. |
| 08:40.33 | MuppetMaster | It is your duty. |
| 08:40.46 | dfunnell | MuppetMaster: Thanks anyway MM! |
| 08:41.25 | Zeeek | you don't think banks make money the more you give them in your savings? No, it's LOANS and CC that makes them rich |
| 08:41.41 | Zeeek | so DON'T SAVE - you're ruinging the economy |
| 08:41.42 | tomas- | just when i get OUT of debt, i buy a house, get a loan for the down payment, spend $400/mo on internet/phone, and rack up my credit cards again to maximize my cash |
| 08:41.48 | MuppetMaster | Zeeek: But it is that savings money that gives them the liquidity to make those loans... |
| 08:41.49 | Zeeek | BUY MORE DIGIUM STUFF NOW |
| 08:42.12 | Zeeek | yes, but they are happier to see you in debt than with a big savings acct |
| 08:42.21 | MuppetMaster | tomas: Don't consider a mortgage simply debt, it is really a financial instrument that allows you to have capital working in multiple places at once. |
| 08:42.49 | Zeeek | debt is like cholestorol - there's "good" and "bad" |
| 08:42.50 | tomas- | well it WILL provide facilities to operate 3 separate businesses |
| 08:43.03 | MuppetMaster | tomas: At least that is how I use it, especially with interest rates where they are. In Holland you may buy a house with 0 down and even get a loan for the closing costs. Then you simply make payments. So a house with zero equity. Sounds crazy, but then you get to write everything off taxes. |
| 08:43.12 | MuppetMaster | Zeeek: Exactly. |
| 08:43.21 | Zeeek | good is money for investment, buying a home, college for the kiddies. Bad is DVD player, vacation you can't afford, and voIP gadnets |
| 08:43.44 | Zeeek | and the advertising world plays to idiots who don't know the diff |
| 08:43.52 | Zeeek | "buy now pay later" |
| 08:44.04 | MuppetMaster | Zeeek: Well, my voip gadnets have reduced my phone bill from ~1000 Euros/month to less than 100, and that is mostly for the mobile when I am out of the house. |
| 08:44.11 | Zeeek | sounds a lot like the virgins in heaven argument, doesn't it? |
| 08:44.19 | *** join/#asterisk erwinism (~pogz@210.213.143.73) |
| 08:44.19 | tomas- | oh im totally out of the loop of entertainment/materialism, except for a few things here and there like dvd box sets/movies once in a while |
| 08:44.29 | erwinism | hello, what does this error means: Unable to create channel of type 'SIP' |
| 08:44.33 | MuppetMaster | Zeeek: Yes, America is about creating a global consumer society. Folks get that confused with cultural exports. Not the same thing. |
| 08:44.36 | Zeeek | the key: "can't afford" |
| 08:44.45 | MuppetMaster | erwinism: Means that the SIP channel you are trying to access is not present. |
| 08:45.03 | *** join/#asterisk RoyK (~roy@83.80-203-29.nextgentel.com) |
| 08:45.04 | MuppetMaster | erwinism: So, for example if you are dialing an extension on Asterisk, it means that extension is currently not registered. For example. |
| 08:45.19 | Zeeek | look at it this way: you have 512megs RAM? You can do stuff with the swap file. But there is point where you have no more resources |
| 08:45.43 | *** join/#asterisk NoCAT (NoCAT@c-24-9-32-2.client.comcast.net) |
| 08:45.45 | MuppetMaster | Zeeek: Well, then you just create more money and let the Asians buy your debt. |
| 08:45.50 | MuppetMaster | Zeeek: See, easy as pie. |
| 08:45.50 | Zeeek | heh |
| 08:46.02 | Zeeek | speaking of memory... wait what did I wanna say... |
| 08:46.27 | Zeeek | R=unning only asterisk 1.0.6 I now see RAM use creep up to all in a few days |
| 08:46.29 | tomas- | MuppetMaster, Rival: thanks for the help/advice.... i gotta get going, its 3:45am |
| 08:46.39 | Zeeek | anyone have any memory leaks issues? |
| 08:47.32 | MuppetMaster | Zeeek: I have seen that with Asterisk, there does appear to be a memory leak somewhere, but it is not consistent. |
| 08:47.39 | erwinism | MuppetMaster how can i register SIP channel? |
| 08:47.56 | Zeeek | I couold reboot every night at 3AM or something |
| 08:47.59 | NoCAT | have you guys ever heard of reselling directtv access? is it possible? i read something about how direct will allow to resell access in apartments |
| 08:48.07 | MuppetMaster | Zeeek: I use the latest head. And sometimes I may go days without seeing a creep, and then other times it happens in hours. Never really pegged it down. |
| 08:48.14 | MuppetMaster | Tomas: Good luck! |
| 08:48.46 | MuppetMaster | NoCAT: There is some guy who has done the concept of location shifting (as opposed to just time shifting) for folks, but don't recall the name of the service right now. |
| 08:49.05 | Zeeek | Muppet: |
| 08:49.06 | Zeeek | Mem: 515460k av, 65864k used, 449596k free, 0k shrd, 2940k buff |
| 08:49.06 | Zeeek | Mem: 515460k av, 68500k used, 446960k free, 0k shrd, 3260k buff |
| 08:49.07 | MuppetMaster | Erwinism: You need a SIP endpoint to register with that SIP profile, so a softphone an ATA, another Asterisk server, etc. |
| 08:49.23 | MuppetMaster | Zeeek: You look okay right now. |
| 08:49.32 | Zeeek | that's about 45 minutes |
| 08:49.35 | NoCAT | muppetmaster: Hi, nice to met you. thanks |
| 08:49.43 | Zeeek | 30 megs in 45 min |
| 08:49.45 | tomas- | MuppetMaster: how can a pbx function over a t1 line if not VoIP? I thought you implied such (referring to a router w/QoS) |
| 08:49.53 | MuppetMaster | NoCAT: Likewise, I think there was something in Wired recently on the subject, you might search there. |
| 08:50.04 | Zeeek | no big deal but this morning there was 65 megs free |
| 08:50.34 | Zeeek | IOW it was revered, 65M free 480M used! |
| 08:50.40 | Zeeek | reversed |
| 08:50.42 | MuppetMaster | Tomas: It is possible to have a T1 split between certain channels (via a MUX/DEMUX) that takes some of the channels (there are 23 available at about 56K/64K each) and uses them for traditional TDM and then others for data. |
| 08:51.09 | MuppetMaster | tomas: So the same circuit could be providing 'traditional' telephony and the others data for internet access. |
| 08:51.29 | MuppetMaster | Zeeek: Happens to me about once a week, even on the latest CVS HEAD. Hard to pinpoint, so can't really post a bug. |
| 08:52.03 | MuppetMaster | tomas: Or just have the whole circuit allocated for data and use VoIP (much better if you have a QOS capable router). |
| 08:52.08 | Zeeek | Well I am not running anything beides asterisk and mp3 |
| 08:52.14 | tomas- | MuppetMaster: so each phone line would require a separate channel? |
| 08:52.19 | Eight | MuppetMaster: I've had this channel open for a while in the background, and every time I glance at it you're asking yet another random question. So my question is: How come you mind isn't totally numb by now? =p |
| 08:52.57 | Eight | tomas-: Yup. |
| 08:53.22 | tomas- | MuppetMaster: you mean using VoIP will allow me to shift the full capacity over to the data (ie my server network) when the phones arent in use? |
| 08:53.29 | tomas- | and not waste channels |
| 08:54.14 | Eight | tomas-: If you just use the T1 as pure data, and then use VoIP for your phone service... yup. |
| 08:54.33 | MuppetMaster | tomas: Correct, in TDM. In VoIP, no, just certain bandwidth on shared channels. |
| 08:54.57 | MuppetMaster | tomas: Exactly. |
| 08:55.00 | NoCAT | muppetmaster i'm trying to google for location shifting tv, i can't seem to much information |
| 08:55.21 | tomas- | ok thanks guys |
| 08:55.22 | MuppetMaster | tomas: But you want QOS (Quality of Service) so that when a call does come, your other data gets downshifted to ensure the quality of the voice. |
| 08:55.27 | MuppetMaster | Eight: Asking or answering? |
| 08:55.49 | Eight | MuppetMaster: errr, answering. |
| 08:55.55 | Eight | MuppetMaster: I guess it's MY mind that's numb =p |
| 08:55.56 | tomas- | and all I need to worry about is QoS on MY router? what about the other end of the data line? |
| 08:56.45 | MuppetMaster | Eight: Ah, this keeps the mind nimble. Don't do it often on the IRC, I actually prefer forums where it stays 'on record' for others to be able to access later. |
| 08:56.56 | MuppetMaster | Eight: A place like http://asterisk.xvoip.com. |
| 08:57.40 | MuppetMaster | tomas: There in lies the rub, there is no way currently to guarantee that on the wilds of the internet. Theoretically routers will support QOS prioritization through the stream, but no guarantee. |
| 08:58.06 | MuppetMaster | tomas: Things will get better with ipv6 if we ever get there. In the meantime, you would need to use a private network (ie - a Level3) to get guarantees of service level. |
| 08:58.24 | MuppetMaster | tomas: Just not possible on the internet. Having said this, I use the internet for all of my VoIP and never have a problem. |
| 08:58.30 | MuppetMaster | tomas: But there is a risk. |
| 08:59.05 | Eight | VoIP: Because getting your website DDOSed wasn't enough. =) |
| 09:00.21 | MuppetMaster | Eight: Good point. |
| 09:00.21 | tomas- | MuppetMaster: but if my t1 provider also does the VoIP, then does that decrease the risk? |
| 09:00.21 | MuppetMaster | Also the concept of SPIM coming about, which is SPAM for VoIP, as calls are free for telemarketers if they can find your open SIP server. |
| 09:00.21 | tomas- | so then a good firewall would help |
| 09:00.23 | *** join/#asterisk Fabe (~spamhere@pD95B0BF7.dip0.t-ipconnect.de) |
| 09:00.29 | MuppetMaster | tomas: Depending on how they provide it, it may or may not. |
| 09:00.37 | MuppetMaster | tomas: Yes, firewalls are important. |
| 09:00.50 | MuppetMaster | tomas: But I would worry about a DDOS attack on your provider before yourself. |
| 09:01.17 | MuppetMaster | tomas: One of the reasons I have multiple outbound providers, have a PHP script that does LCR and failover if a provider does not respond. |
| 09:01.40 | MuppetMaster | Well, I have to feed my monkey. |
| 09:01.45 | MuppetMaster | Bye |
| 09:01.52 | tomas- | thanks for the help |
| 09:01.58 | *** part/#asterisk topping (~topping@dhcp024-210-082-196.columbus.rr.com) |
| 09:02.13 | Eight | So Broadvoice are "The gusy to use" for Asterisk VoIP <-> PSTN termination? |
| 09:02.20 | Eight | erm, guys. |
| 09:02.40 | Zeeek | Eight a lot of people are having trouvbe with BV |
| 09:02.47 | Eight | Zeeek: oh? |
| 09:02.48 | Zeeek | check the mailing list carefully |
| 09:02.49 | MuppetMaster | Eight: I would recommend http://www.teliax.com |
| 09:03.06 | Zeeek | Eight what exactly do you need in a provider? |
| 09:03.12 | godsmoke | I would recommend Voicepulse Connect over Teliac |
| 09:03.13 | *** join/#asterisk shepherd (matt@pcp01541028pcs.huntsv01.al.comcast.net) |
| 09:03.15 | godsmoke | Teliax* |
| 09:03.25 | Zeeek | I have tried several providers |
| 09:03.58 | Eight | At this point just VoIP to PSTN termination that plays well with Asterisk for me tinker with. |
| 09:04.05 | Eight | But I'm poking around with using it for something more. |
| 09:04.14 | godsmoke | need a local did? |
| 09:04.22 | Zeeek | that's the bug question |
| 09:04.27 | Zeeek | big question |
| 09:04.29 | MuppetMaster | For OB termination only you can not beat http://www.voipjet.com |
| 09:04.38 | Zeeek | lowest price in town |
| 09:04.44 | Eight | For the tinkering, no. For something more I'd prefer to be able to port existing numbers consistently. |
| 09:04.50 | Zeeek | good performance in my experience |
| 09:04.55 | godsmoke | then I recommend voicepulse connect |
| 09:05.06 | shepherd | voicepulse connect! |
| 09:05.08 | Zeeek | VP connect is good and they do IAX |
| 09:05.12 | godsmoke | yes |
| 09:05.18 | godsmoke | and local dids for reasonable price |
| 09:05.18 | Zeeek | and have two servers |
| 09:05.30 | Zeeek | only in about 24 states tho |
| 09:05.39 | godsmoke | yeah, obviously check your area |
| 09:05.46 | godsmoke | note |
| 09:05.54 | godsmoke | voicepulse connect is DIFFERENT from voicepulse |
| 09:05.54 | Eight | thanks for the references guys. |
| 09:06.05 | Zeeek | yes you'll have to dig for VP connect |
| 09:06.07 | godsmoke | "connect" is their IAX/experienced user system -- pay per minute |
| 09:06.13 | ta[i]nted | u can't beat BV prices for unlimited call to certain countries |
| 09:06.30 | ta[i]nted | u just can't |
| 09:06.34 | ta[i]nted | it's not possible |
| 09:06.37 | Eight | ya, a friend of mine is pretty enthused about calling the netherlands. |
| 09:06.43 | MuppetMaster | ta[i]nted: I do, with http://www.voipuser.org |
| 09:06.55 | ta[i]nted | MuppetMaster how |
| 09:07.02 | ta[i]nted | how does voipuer.org beat it |
| 09:07.14 | ta[i]nted | u have to route minutes to use minutes don't u |
| 09:08.18 | *** join/#asterisk mbranca (~matteo@81.208.92.210) |
| 09:09.00 | MuppetMaster | Yes, but I have no problem with that. |
| 09:09.09 | Eight | ok, little confused... VoicePlus Connect is offered by who exactly? |
| 09:09.15 | ta[i]nted | Update : Outbound calls are currently capped to a maximum duration of 10 mins. This is just to help reduce the February defecit. We'll be upping it again as soon as we are able to. |
| 09:09.21 | MuppetMaster | I use an 0844 number for inbound calls from the UK to Holland. The caller saves money and I add to the pool. |
| 09:09.28 | ta[i]nted | MuppetMaster you are comparing cars to bicycles |
| 09:10.25 | MuppetMaster | No |
| 09:10.25 | MuppetMaster | That is only temporary |
| 09:10.29 | RoyK | anyone using the context= setting in queue.conf? |
| 09:10.41 | ta[i]nted | MuppetMaster where are there inbound rates? |
| 09:10.59 | MuppetMaster | 3pc/min |
| 09:11.01 | ta[i]nted | their |
| 09:11.11 | ta[i]nted | is that just for UK DID? |
| 09:11.12 | MuppetMaster | for 0844 with SIP only termination |
| 09:11.21 | modulus_ | hi tainted |
| 09:11.28 | modulus_ | do you like perl? |
| 09:11.28 | ta[i]nted | modulus_ hi |
| 09:11.35 | ta[i]nted | i like perl very much |
| 09:11.42 | modulus_ | shall we perl then? |
| 09:12.13 | ta[i]nted | lol |
| 09:12.27 | ta[i]nted | modulus_ is there a way to see how good a colo is? |
| 09:12.37 | modulus_ | yeah |
| 09:12.48 | ta[i]nted | can u check to see if these guys oversubscribe their lines? |
| 09:12.48 | modulus_ | ask them if they have admins who perl |
| 09:12.48 | Zeeek | modulus_ Clean that UP! |
| 09:13.47 | modulus_ | tainted, looks like a web hosting co. |
| 09:13.54 | ta[i]nted | modulus_ he claims single-homed 10Mbps unmetered to cogent for 450 + rackspace |
| 09:14.09 | ta[i]nted | modulus_ i visited their facilities near koreatown |
| 09:14.27 | modulus_ | tainted, address? |
| 09:14.29 | ta[i]nted | modulus_ just wondering if there was a way to see if he oversubscribes his perl |
| 09:14.40 | ta[i]nted | 3250 Wilshire Blvd Suite#301 |
| 09:14.40 | ta[i]nted | Los Angeles, CA 90010 |
| 09:15.28 | modulus_ | was there a noc, or command center? |
| 09:15.44 | ta[i]nted | i have choice of single-homed to cogent or multi-homed through mzima.. $450 vs $1000 |
| 09:15.54 | ta[i]nted | yea but it was unmanned at the time i was there |
| 09:16.26 | modulus_ | who gave you a tour of the place? |
| 09:16.40 | *** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk) |
| 09:16.49 | ta[i]nted | owner |
| 09:16.53 | modulus_ | name? |
| 09:16.54 | *** join/#asterisk jayCampbell (~jay@jet.got.net) |
| 09:17.10 | slePP | anyone think i should make an 'image pastebin'? |
| 09:17.29 | slePP | or would it just get abused to no end with stupid junk? |
| 09:17.40 | ta[i]nted | it looked like a lot of his clientele was pr0n |
| 09:17.44 | modulus_ | tainted, to be honest i didn't know there were colos in ktown |
| 09:17.45 | ta[i]nted | a LOT |
| 09:17.51 | modulus_ | tainted, that's all colo business |
| 09:17.56 | modulus_ | and mailers too |
| 09:17.58 | ta[i]nted | corner of vermont & wilshire |
| 09:18.04 | jayCampbell | who do i pester for a discount to friday's Meet Asterisk |
| 09:19.07 | ta[i]nted | modulus_ do u colo * at your buddy's place? |
| 09:19.29 | modulus_ | yup |
| 09:19.45 | modulus_ | they'll match the price probably |
| 09:19.54 | modulus_ | and better bandwidth probably too |
| 09:20.16 | ta[i]nted | i'll see if he drops the ball |
| 09:20.33 | skrusty | morning |
| 09:20.53 | modulus_ | ta[i]nted, how much b/w you thinking of pushing? |
| 09:21.24 | ta[i]nted | start with 10Mbps unmetered.. go from there |
| 09:21.45 | modulus_ | go to blacksun.net tell them modulus sent you |
| 09:21.47 | ta[i]nted | gotta see what my boxes can handle |
| 09:22.04 | modulus_ | mebbe i'll get a cut |
| 09:22.22 | modulus_ | we can iax peer our boxen |
| 09:22.34 | modulus_ | and perl happily off into the sunset |
| 09:22.41 | ta[i]nted | does your box go anywhere? |
| 09:22.46 | ta[i]nted | or just to your desk phone |
| 09:22.47 | ta[i]nted | lol |
| 09:22.53 | shepherd | heh |
| 09:22.55 | shepherd | just fwd |
| 09:23.14 | modulus_ | my box runs their noc/office customer's offices as well |
| 09:23.20 | Rival | is there any way to stress test your pbx? |
| 09:23.34 | modulus_ | pbx can get stressed? |
| 09:23.38 | ta[i]nted | Rival what do u want to test |
| 09:23.38 | shepherd | rival: you can buy a call generator |
| 09:23.42 | shepherd | but those get expensive |
| 09:23.58 | Rival | load / lag / errors ect |
| 09:24.16 | ta[i]nted | * is very sensitive to cpu load i've noticed |
| 09:24.29 | shepherd | yeah |
| 09:24.31 | modulus_ | indeed |
| 09:24.51 | shepherd | it's best to run * on a dedicated server |
| 09:24.58 | Rival | mine is |
| 09:25.14 | ta[i]nted | u can literally hear some retard on the lan's p0rn packets |
| 09:25.16 | Rival | but its only a 1ghz / 396mb ram |
| 09:25.27 | modulus_ | anyone here run * on a dual athlon? |
| 09:25.49 | shepherd | i have before |
| 09:25.56 | Rival | i was just wondering if you guys have redundant pbx's |
| 09:25.58 | shepherd | wait |
| 09:26.14 | Rival | would suck to have downtime for a business |
| 09:26.40 | *** join/#asterisk afe ([1e27ZnmmE@c-e616e055.123-1-64736c12.cust.bredbandsbolaget.se) |
| 09:28.06 | channan | Rival, my friend has that service, can design up to 99.999% uptime. Would you like the contact? |
| 09:28.17 | Rival | nah |
| 09:28.21 | Rival | im still r/d |
| 09:28.49 | *** join/#asterisk Jas_Williams (~Jason@host81-155-66-178.range81-155.btcentralplus.com) |
| 09:29.08 | erwinism | can anyone help me make a sip.conf and extensions.conf for 3 users only. i just need a sample |
| 09:29.11 | channan | ok... |
| 09:29.34 | slePP | what's everyone's favourite movie on DVD? |
| 09:29.46 | modulus_ | pr0n? |
| 09:29.48 | MuppetMaster | Fear and loathing in Las Vegas. |
| 09:29.53 | slePP | perhaps not pr0n :> |
| 09:29.56 | slePP | MuppetMaster: not a bad one |
| 09:30.10 | modulus_ | shaving private ryan |
| 09:30.20 | modulus_ | or was it "saving ryan's privates" |
| 09:30.22 | ta[i]nted | shaving ryan's privates |
| 09:30.24 | erwinism | Jusrassic Pork |
| 09:30.27 | modulus_ | heh |
| 09:30.33 | slePP | http://pastebin.ca/contest.php |
| 09:30.35 | slePP | that's why :P |
| 09:30.38 | modulus_ | forrest hump |
| 09:30.54 | channan | forrest gump for sure |
| 09:31.10 | slePP | nor jurassic pork :> though that'd definitely be interesting to see the logistics of |
| 09:31.32 | ta[i]nted | slePP u made pastbin? |
| 09:32.03 | slePP | and people use it. i'm very pleased with it. |
| 09:32.10 | NoCAT | does anyone know of any irc channels, to discuess wifi technology? |
| 09:32.12 | slePP | and so, i'm doing a dvd giveaway, but i need suggestions of good movies. |
| 09:32.15 | ta[i]nted | how many channels use it? |
| 09:32.22 | slePP | by last guess, maybe 10-20 |
| 09:32.25 | slePP | a few irc networks use it |
| 09:32.27 | slePP | and a few random people |
| 09:32.42 | erwinism | i use wifi |
| 09:32.45 | slePP | and then there are 4 subdomains to pastebin for private users |
| 09:33.20 | shepherd | we need an astbin :( |
| 09:33.27 | *** join/#asterisk tzafrir (~tzafrir@62.90.10.53) |
| 09:33.42 | slePP | so fear and loathing in las vegas and forrest gump.. are those the only two movies anyone likes? :P |
| 09:33.45 | NoCAT | i found one.. #wireless |
| 09:33.50 | NoCAT | on this server |
| 09:33.51 | slePP | but not everyone likes dubbed films |
| 09:33.56 | ta[i]nted | NoCAT try #wireless |
| 09:33.56 | slePP | maybe i could give a selection |
| 09:34.01 | *** join/#asterisk Delvar (~irc@83.146.53.34) |
| 09:34.05 | MuppetMaster | I hate dubbed films. |
| 09:34.06 | ta[i]nted | NoCAT do a /list |
| 09:34.29 | modulus_ | slepp are you new to irc? |
| 09:34.29 | Delvar | moo |
| 09:34.35 | shepherd | lord of the rings |
| 09:34.36 | shepherd | heh |
| 09:34.42 | shepherd | startrek! |
| 09:34.49 | slePP | modulus_: god no |
| 09:34.55 | ta[i]nted | slePP give away the incredibles |
| 09:34.59 | slePP | modulus_: i've been running an irc network for 5 years or more |
| 09:35.18 | slePP | ooh. the incredibles. that's a good one. |
| 09:35.44 | modulus_ | i only own 2 dvds, donnie brasco and princess bride |
| 09:35.45 | ta[i]nted | Seven Samurai |
| 09:35.46 | Delvar | Lords of the Ring (gay porno) |
| 09:35.57 | slePP | hehe |
| 09:36.04 | slePP | ta[i]nted: i don't even know of that one |
| 09:36.11 | ta[i]nted | it's good |
| 09:36.15 | shepherd | Ass Masters :( |
| 09:36.23 | shepherd | like astmasters only ass |
| 09:36.25 | shepherd | :( |
| 09:36.31 | ta[i]nted | hey now |
| 09:36.50 | ta[i]nted | where did that frown come from |
| 09:36.59 | shepherd | bad joke |
| 09:37.13 | ta[i]nted | find the closest mirror |
| 09:37.29 | shepherd | haha |
| 09:37.30 | ta[i]nted | and say into it, i'm good enough, i'm smart enough, and darn it people like me |
| 09:37.38 | shepherd | i once saw this sing in the bathroom |
| 09:37.41 | shepherd | aksdjf |
| 09:37.44 | shepherd | i once saw this sign in the bathroom |
| 09:37.54 | modulus_ | doesn't he say "gosh darn it"? |
| 09:37.55 | shepherd | it said, "the only joke is the one you are holding" |
| 09:38.07 | shepherd | :( |
| 09:38.31 | shepherd | Paris Hilton Porno |
| 09:38.33 | shepherd | on DVD! |
| 09:38.34 | slePP | heh |
| 09:38.36 | slePP | people like that? |
| 09:38.38 | slePP | she's scrawny |
| 09:38.41 | slePP | she'd just hurt in bed |
| 09:38.47 | shepherd | we've already seen it anwyays |
| 09:39.07 | ta[i]nted | lol |
| 09:39.15 | ta[i]nted | did i join #asterisk |
| 09:39.34 | modulus_ | i'm hungry |
| 09:39.35 | slePP | no, yer in #Asstricks |
| 09:39.45 | ta[i]nted | slePP i think we should buy u a dvd for your pastebin contribution |
| 09:39.51 | *** join/#asterisk BuzzBud (~adoroar@90.62-97-254.bkkb.no) |
| 09:40.05 | slePP | i'm all up for presents :> |
| 09:40.12 | slePP | but i still have to decide what the users get :P |
| 09:40.23 | ta[i]nted | they might be from the walmart value bin however... :( |
| 09:40.31 | slePP | hehehe |
| 09:40.34 | slePP | some of those movies rule |
| 09:40.45 | ta[i]nted | it's not true |
| 09:40.51 | ta[i]nted | name one good movie that is under $5 |
| 09:40.55 | slePP | uhm |
| 09:41.01 | slePP | Mars Attacks |
| 09:41.03 | slePP | is it in the bin yet? |
| 09:41.08 | shepherd | drop dead gorgeous |
| 09:41.19 | shepherd | best wal-mart movie bin ever! |
| 09:41.47 | slePP | Trains, Planes and Something... |
| 09:41.52 | ta[i]nted | autos |
| 09:41.56 | slePP | ah, yeh. that one :> |
| 09:41.58 | ta[i]nted | automobiles |
| 09:42.02 | slePP | 'May I go to the bathroom?' |
| 09:42.05 | ta[i]nted | is that value binned? |
| 09:42.06 | slePP | 'Yes, Ruprick' |
| 09:42.12 | slePP | i have a copy that has a $3 sticker on it |
| 09:42.20 | slePP | but that may be from a movie store. it's pretty beat up |
| 09:42.28 | slePP | VHS, too |
| 09:42.33 | ta[i]nted | value.pastebin.ca |
| 09:43.05 | slePP | i could just take random selections of the IMDb top 250 |
| 09:45.44 | slePP | well, i got The Incredibles, Fear and Loathing in Las Vegas, Forrest Gump, Donnie Darko and Mars Attacks! |
| 09:46.00 | *** join/#asterisk darkskiez (~darkskiez@usergc137.dsl.pipex.com) |
| 09:47.08 | Eight | Does anyone here have any experience with Qwest's VoIP services? |
| 09:47.47 | Eight | This question is open to anyone NOT here as well I suppose. But they're likely to have difficulty responding ;-) |
| 09:48.30 | Eight | http://www.qwest.com/smallbusiness/products/voip/index.html |
| 09:48.40 | Eight | very light on the details. |
| 09:49.04 | Eight | This is also interesting, but anemic http://www.qwest.com/smallbusiness/products/voip/integratedaccess.html |
| 09:49.24 | modulus_ | snoRRRE |
| 09:52.01 | shepherd | voip is just cheaper |
| 09:52.12 | shepherd | when you do data |
| 09:52.20 | shepherd | :D |
| 09:52.33 | Eight | "do data"? |
| 09:53.03 | shepherd | well.. i guess it's just cheaper all the way around.. cheaper to manager, etc |
| 09:55.24 | modulus_ | i wish i could watch some star trek |
| 09:57.56 | modulus_ | <PROTECTED> |
| 09:57.56 | modulus_ | <PROTECTED> |
| 09:58.04 | modulus_ | my dsl sucks donkey balls |
| 09:58.41 | *** join/#asterisk Inferna (~sasha@194.158.51.171) |
| 10:01.08 | *** join/#asterisk hajekd (~hajekd@mail.idoox.com) |
| 10:01.30 | *** join/#asterisk wasim (~wasim@203.81.217.160) |
| 10:02.40 | datareactor | is there list all hardware voip phones |
| 10:02.53 | datareactor | i want to do some comparsion |
| 10:03.15 | wasim | datareactor: voip-info.org has some of them |
| 10:03.19 | *** join/#asterisk TheJudge (JTR@209-203-52-3.network.ods.co.za) |
| 10:03.33 | TheJudge | hello all |
| 10:03.51 | wasim | modulus_: irani pistachios are some of the best |
| 10:04.00 | TheJudge | I am having a problem with my asterisk server and sip phone |
| 10:04.08 | TheJudge | I get a crackel on the line ? |
| 10:04.15 | modulus_ | no wonder why they're expensive |
| 10:04.16 | *** join/#asterisk soulz- (~soulz@host-137-132-45-118.imcb.nus.edu.sg) |
| 10:04.19 | soulz- | hello all |
| 10:04.21 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) |
| 10:04.36 | soulz- | still stuck with the bv problem, is anyone using bv? |
| 10:04.38 | datareactor | thanks wasims |
| 10:04.39 | wasim | modulus_: yeah, here they are about $10 / kg |
| 10:04.47 | ta[i]nted | soulz- yep |
| 10:04.55 | TheJudge | yes used to use bv |
| 10:05.02 | soulz- | tainted: is ur sip show peers working? |
| 10:05.04 | ta[i]nted | wasim u are in iran? |
| 10:05.07 | ta[i]nted | soulz- yep |
| 10:05.15 | soulz- | thejudge: what are u using now? |
| 10:05.20 | wasim | ta[i]nted: nope, just east |
| 10:05.29 | TheJudge | moved to voiptalk.org |
| 10:05.41 | TheJudge | calls seem better support more codecs |
| 10:05.42 | soulz- | tainted: phark man, my sip i registering, but my sip.broadvoice.com is unreachable |
| 10:05.46 | soulz- | i dunno why |
| 10:05.53 | TheJudge | however calls seem to have a crackle on them ? |
| 10:06.06 | soulz- | in sip show peers |
| 10:06.12 | erwinism | how can i make in the extensions.conf when someone dials 2000 it will ring the username 'erwin' ? |
| 10:06.18 | ta[i]nted | soulz- did u set proxy in your /etc/hosts |
| 10:06.30 | soulz- | tainted: i tried all the proxys |
| 10:06.36 | *** join/#asterisk Jer13261 (~Jer@rdu57-251-152.nc.rr.com) |
| 10:06.42 | soulz- | mia, dca, lax chi |
| 10:06.44 | ta[i]nted | both incoming and outgoing down? |
| 10:06.52 | soulz- | yup |
| 10:07.02 | TheJudge | yip sometimes you just dont register switch on and off |
| 10:07.09 | ta[i]nted | did u add authuser = bvphonenum |
| 10:07.16 | soulz- | broadvoice/212202xxxx 147.135.0.128 255.255.255.255 5060 UNREACHABLE |
| 10:07.29 | ta[i]nted | Jer13261 ebay |
| 10:07.44 | soulz- | i have authuser authname, username |
| 10:08.05 | Jer13261 | and search for? :) |
| 10:08.07 | ta[i]nted | did u call their techs |
| 10:08.23 | wasim | Jer13261: get a TDM40B |
| 10:08.34 | soulz- | yup, no answer, i will call again at 9am us time |
| 10:09.15 | ta[i]nted | Jer13261 'fxo card' |
| 10:09.29 | TheJudge | can I ask, there is packet lengh ? set at 20 ms for my codec ? |
| 10:09.45 | TheJudge | what happens when I increase and decrease it ? |
| 10:09.51 | Jer13261 | no........wait....i thought FXS was to connect to an analog phone....FXO was to connect ao a outside line....right? |
| 10:10.03 | Jer13261 | so i'd want FXS then |
| 10:10.21 | wasim | Jer13261: yep |
| 10:10.30 | TheJudge | FXO to phone FXS tp swictboard |
| 10:10.41 | *** join/#asterisk FaithX (~Faithful@202-6-145-116.ip.adam.com.au) |
| 10:10.50 | ta[i]nted | no he needs FXO |
| 10:11.13 | ta[i]nted | n/m |
| 10:11.20 | ta[i]nted | i assume everyone uses voip |
| 10:11.50 | modulus_ | voip? what's that? |
| 10:11.59 | modulus_ | sounds like a new sandwich |
| 10:12.01 | Jer13261 | do i need FXS or FXO...i want to go from my voip provider to my desk phone... |
| 10:12.14 | TheJudge | FXO |
| 10:12.16 | Jer13261 | as my current POS setup at work @^)(#&@# |
| 10:12.19 | ta[i]nted | FXO |
| 10:12.31 | Jer13261 | its an IXJ heh |
| 10:13.03 | FaithX | which is the first in the Lord of the Rings Movie series??? |
| 10:13.11 | modulus_ | The fellowship of the ring |
| 10:13.22 | modulus_ | The Two towers |
| 10:13.25 | modulus_ | the return of the king |
| 10:14.13 | FaithX | thanks |
| 10:14.32 | Jer13261 | how about an X100P? |
| 10:16.37 | wasim | Jer13261: thats to connect a phone line to |
| 10:16.47 | wasim | Jer13261: you need an FXS or an IAXY |
| 10:16.51 | Eight | Wait... the hardware device needs the "FXS" label to talk to a desk phone. But the configuration of the port in asterisk is FXS_ks, right? |
| 10:17.01 | Eight | err, FXO in the asterisk config |
| 10:17.08 | Eight | I hate it when I confuse the only point I'm trying to make =p |
| 10:18.11 | datareactor | what are the top 5 Voip phones in market |
| 10:18.22 | wasim | farfon :P |
| 10:19.01 | datareactor | Cisco grandstream Sipura etc |
| 10:19.19 | Zeeek | hi wasim, wassup? |
| 10:19.21 | wasim | cisco, polycom, iptel, grandstream |
| 10:19.28 | wasim | heya Zeeek , did you try the new firmware? |
| 10:19.32 | Zeeek | yes sir |
| 10:19.34 | Jer13261 | damn as FXO is only like $10 lol |
| 10:19.38 | Zeeek | I'm waiting for my salary though |
| 10:19.40 | datareactor | thanks wasim |
| 10:20.13 | Jer13261 | if IXJ supported CID i'd be set as i have 3 of them already :P |
| 10:20.19 | wasim | whee ... francs galore! |
| 10:20.56 | shaZwaz | hey wasim |
| 10:21.17 | shaZwaz | 235/6 howz that !! |
| 10:21.28 | wasim | shaZwaz: i don't think inzi was out |
| 10:21.42 | shaZwaz | not again |
| 10:27.09 | *** join/#asterisk xyharley (~daecon@xyharley.dsl.xmission.com) |
| 10:30.25 | *** join/#asterisk soundguy (~soundguy@zeus.soundguy.id.au) |
| 10:30.43 | *** join/#asterisk xyharley (~daecon@xyharley.dsl.xmission.com) |
| 10:32.50 | *** join/#asterisk marak (~twist@ndn-165-143-245.telkomadsl.co.za) |
| 10:33.32 | wasim | as long as we can bat the day out, i think we'll be ok |
| 10:33.49 | wasim | 25ish overs left? |
| 10:34.55 | marak | hi all. thanks for the help yesterday. i have run into an interesting thing. in our office we have 7 people running off firefly softphones. 4 are having no problems quality is good dispite 3 seconds of echo training in beginning of call. 2 however have incredibly bad call quality. we are all on the same hub.when they call voicemail quality is perfect but grabbing and outside line it is totally inaudible |
| 10:36.00 | shaZwaz | there goes another one |
| 10:36.48 | wasim | marak: does that happen on the 6th caller, or is it the same 2 people, everytime irregardless of the number of users? |
| 10:37.09 | shaZwaz | wasim may be an hour or more play still left for the day |
| 10:37.26 | marak | always the same 2 users with the audio problems. |
| 10:37.41 | wasim | shaZwaz: they same they can give 'em an extra half an hour |
| 10:38.11 | wasim | marak: what about when they call someone else on the lan? |
| 10:38.48 | shaZwaz | I doubt Naveed's bating capabilities |
| 10:39.00 | tuxinator_linux | marak: all the same comps? |
| 10:39.28 | tuxinator_linux | marak: are they dells? |
| 10:39.33 | wasim | marak: what happens if you use one of the 4 healthy ones config on the non healthy one? |
| 10:39.37 | marak | testing call on lan now. not same machines |
| 10:39.47 | marak | but decent machines the one is running a sb live |
| 10:39.56 | shaZwaz | wasim with ptcl announcing sms services over its network, sh'd the sms app in * work now ? tried ? |
| 10:40.42 | marak | a lan call is perfect sound quality. |
| 10:41.07 | wasim | shaZwaz: it should work on zapata |
| 10:41.16 | tuxinator_linux | Pulling this one out of my butt.... could the other two machines have a sub-par soundcard or NIC? |
| 10:41.51 | shaZwaz | and with the phones too I guess , the ones ptcl has brought in the market |
| 10:42.02 | shaZwaz | are they the ADSI ones ? |
| 10:42.40 | shaZwaz | they are offering sms, email, fax on these sets |
| 10:42.50 | tuxinator_linux | marak: is the * config the same for all the phones? |
| 10:43.00 | wasim | no, i don't think they are ADSI |
| 10:43.13 | shaZwaz | huh ? |
| 10:43.28 | wasim | i haven't been paying attention, too buy with the GSM stuff and CDMA, but i will shortly |
| 10:43.30 | marak | yes firefly softphone. i have tried the denoiseoptions in the phones but nothing changes. |
| 10:43.49 | *** join/#asterisk wizard2000 (~dang@fubar.arcbox.com) |
| 10:43.58 | marak | would you try set them up using sip softphone to see if it changes |
| 10:44.12 | wizard2000 | i have a problem with a TDM card... it doesn't seem to hangup for a while after the call has ended... |
| 10:44.22 | shaZwaz | wasim its not GSM / CDMA , I am talking about these features on Anlaog lines, what ptcl says |
| 10:44.39 | tuxinator_linux | marak: I'm not sure if you can set the quality per account in * |
| 10:44.56 | tuxinator_linux | I am still pretty new to * |
| 10:45.08 | marak | i havent seen anything for it. |
| 10:45.16 | Jer13261 | cab abtie \\be tekk me if DIALOGIC D/40B would be supported? |
| 10:45.22 | Jer13261 | can anyone tell me if... |
| 10:45.29 | shaZwaz | wasim are there any demo farfon units available ? |
| 10:45.29 | tuxinator_linux | Have you watch the * console? |
| 10:45.38 | wasim | shaZwaz: yeah, i know, i've been busy on gsm/cdma work, so couldn't devote time to the fixed line stuff |
| 10:45.49 | wasim | shaZwaz: yep |
| 10:46.25 | shaZwaz | can I collect one ? |
| 10:46.50 | wasim | shaZwaz: geopos? |
| 10:46.51 | tuxinator_linux | marak: I would watch the an * console to see if there is any differenced between calls |
| 10:46.57 | wizard2000 | anyone? |
| 10:47.00 | shaZwaz | lhr man |
| 10:47.06 | wasim | shaZwaz: sure |
| 10:47.08 | tuxinator_linux | marak: make sure the verbosity is pretty high |
| 10:48.36 | *** join/#asterisk ckruetze (ckruetze@cpc1-cmbg7-5-0-cust31.cmbg.cable.ntl.com) |
| 10:49.27 | datareactor | top 5 ip phone Cisco SIPURA Polycom snom iptel Please comment |
| 10:50.24 | wizard2000 | when i call the line with the TDM on it, it picks up fine... when i hang up from the phone i'm calling from, asterisk does not hangup the call... it still thinks that the line is connected... |
| 10:50.48 | marak | i have watched asterisk no difference. what is verbosity ? |
| 10:51.14 | wasim | wizard2000: you need far end disconnect supervision |
| 10:51.31 | wizard2000 | wasim: how do i do this please? |
| 10:51.45 | *** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
| 10:51.56 | wasim | wizard2000: either ks if your telco supports it, call supervision, or busydetect, basically play with it |
| 10:51.59 | tuxinator_linux | marak: asterisk -vvvvvvvvvvvvvvvvvvvvr Increase verbosity (multiple v's = more verbose) |
| 10:52.13 | *** join/#asterisk Newbie___ (some@218.111.220.69) |
| 10:52.15 | tuxinator_linux | marak: 'asterisk --help' |
| 10:52.42 | Newbie___ | hi guys, IAX vs SIP which is the preferred ? |
| 10:52.45 | wizard2000 | wasim: thanks :) |
| 10:52.54 | wasim | Newbie___: IAX |
| 10:52.55 | Newbie___ | i meant IAX2 |
| 10:52.56 | Jer13261 | IAX |
| 10:53.04 | tuxinator_linux | IAX |
| 10:53.06 | Newbie___ | because of firewall ? |
| 10:53.15 | wasim | Newbie___: because of braindead SIP |
| 10:53.18 | tuxinator_linux | better protocol |
| 10:53.46 | *** join/#asterisk zeedo (~zeedo@www.bsrf.org.uk) |
| 10:53.50 | Newbie___ | i was told that IAX2 handles firewall better than SIP |
| 10:53.54 | *** join/#asterisk soulz- (~soulz@host-137-132-45-118.imcb.nus.edu.sg) |
| 10:53.55 | tuxinator_linux | yes |
| 10:53.59 | soulz- | hello all |
| 10:54.06 | Newbie___ | roger, thanks guys |
| 10:54.14 | Rival | you know i was looking at these CISCO IP PHONE 7970G |
| 10:54.15 | tuxinator_linux | Hello Mr Soulz- |
| 10:54.26 | Rival | and i was thinking god damn theres no way im paying 500$ for a phone |
| 10:54.41 | Rival | then i looked at my cell phone and thought wtf i already did |
| 10:54.45 | Makenshi | a firewall with a proper stateful sip proxy should have no trouble with sip |
| 10:54.46 | tuxinator_linux | no SIP on 7970's |
| 10:54.59 | soulz- | can anyone can help me? i am using bv, and sip show peers, shows the bv server, sip.broadvoice.com is unreachable |
| 10:54.59 | tuxinator_linux | Rival: ^ |
| 10:55.10 | Makenshi | that reminds me, i still need to try porting LInksys's ip_conntrack_sip module into my firewall |
| 10:55.15 | soulz- | but i can ping, trace, and i can register |
| 10:55.26 | shepherd | what's the point of a firewall if you have a port open |
| 10:55.37 | Rival | 7940 has sip right? |
| 10:56.04 | tuxinator_linux | yes |
| 10:56.16 | tuxinator_linux | 7940 and 7960 |
| 10:56.28 | tuxinator_linux | 103 or something for the SIP license |
| 10:56.35 | tuxinator_linux | USD |
| 10:56.56 | tuxinator_linux | I am looking to get a few 7960G's to play with |
| 10:57.16 | Rival | 7960 have video? |
| 10:57.26 | Rival | or is that just the 70 |
| 10:57.51 | tuxinator_linux | Rival: video, no I don't think so |
| 10:58.24 | tuxinator_linux | Rival: http://cisco.com/en/US/products/hw/phones/ps379/index.html |
| 10:58.25 | Rival | oh well vid confercing is over rated imho |
| 10:58.35 | tuxinator_linux | Rival: yep |
| 10:58.41 | tuxinator_linux | I don't care for it |
| 10:59.03 | tuxinator_linux | what is the fun if you can make stupid faces while talking to people |
| 10:59.08 | Rival | hehe |
| 10:59.08 | tuxinator_linux | can't |
| 10:59.16 | slePP | http://pastebin.ca/contest.php |
| 10:59.24 | tuxinator_linux | 7960's start around $300 USD |
| 11:00.36 | tuxinator_linux | then you need to SIP firmware add anonther $100 USD |
| 11:00.42 | Rival | hehe |
| 11:00.48 | Rival | figures |
| 11:00.57 | Rival | no one knows how to milk money like cisco |
| 11:01.04 | Rival | think they hold the patent on it |
| 11:01.10 | tuxinator_linux | the 7970's look nice, but they are not releasing SIP firmware yet |
| 11:01.14 | Zeeek | slePP are you the pastebin guy? |
| 11:01.37 | slePP | (how does no one seem to know that? :) |
| 11:01.45 | Zeeek | I got a phone call a while back from someone who found my number in pastebin |
| 11:01.50 | Zeeek | thru google |
| 11:01.54 | wasim | oh no |
| 11:01.54 | slePP | nice. need the item deleted? |
| 11:01.56 | tuxinator_linux | I didn't know |
| 11:02.00 | Zeeek | it would be good to make it noindex, nofollow |
| 11:02.07 | slePP | ones marked for expire are noindex, nofollow |
| 11:02.10 | Zeeek | yeah, from Google :) |
| 11:02.17 | slePP | it's mentioned in about 3 places that that is the case :P |
| 11:02.36 | Zeeek | how did they get it in google then? |
| 11:02.41 | datareactor | top 5 ip phone Cisco SIPURA Polycom snom iptel plz commment |
| 11:02.43 | slePP | did you mark it to expire? |
| 11:03.13 | Zeeek | it was just a piece of SMS dialplan |
| 11:03.16 | Zeeek | no |
| 11:03.20 | slePP | <PROTECTED> |
| 11:03.28 | slePP | yeh. people tend not to read that bit, i've found. |
| 11:03.31 | Zeeek | shouldn't it all expire in like an hour or something? |
| 11:03.34 | tuxinator_linux | marak: the cisco phones look to be more durable than the others, it may be worth it. |
| 11:03.41 | slePP | no, it's all permanently stored, for reference reasons |
| 11:03.47 | Zeeek | no one reads anything on wab pages, especially usefull ones like pastebin |
| 11:03.47 | slePP | there's some very handy bits in there |
| 11:03.54 | Rival | hehe do you deal with people that get pissed off tux? |
| 11:04.19 | Zeeek | anyway, it's a funny thing - I had the SMS center callerID in there and the guy calls and said "why are you calling me?" |
| 11:04.19 | Rival | if i catch a employee banging company equipment i ream them out |
| 11:04.29 | tuxinator_linux | Rival: laugh at the for taking things to seriously |
| 11:04.33 | slePP | Zeeek: he searched his own number out? |
| 11:05.09 | ta[i]nted | Rival why would ur employees bang company equipment |
| 11:05.11 | Rival | tux never can be too certian ive thrown my nextel through a wall |
| 11:05.18 | tuxinator_linux | I never a problem like that with my employees |
| 11:05.29 | Zeeek | he saw the 0101666666 or something called him at 3AM. He googled for that numnber and found it in my pastebin post |
| 11:05.39 | Inferna | did anybody use grandstream phone, it's cheap, is it good? |
| 11:05.42 | Zeeek | it's the French SMS center that calls with messages |
| 11:05.42 | tuxinator_linux | Rival: I buy expensived stuff, and it works well |
| 11:05.47 | slePP | Zeeek: i should make a cookie, and if you don't have it set, it BLINKS the damn message at you :P |
| 11:05.57 | Zeeek | Inferna GS is cheap and good but there are now more phones in that range |
| 11:06.07 | Rival | i tell ya those nextel i860's arent as flimsy as i thought they looked |
| 11:06.20 | Zeeek | slePP no, if you don't have the cookie, it goes to a condition of use page |
| 11:06.28 | Zeeek | which you have to check |
| 11:06.30 | Rival | i imagine ip phones are gonna come down in price a lot over the next 5 years |
| 11:06.38 | tuxinator_linux | yep |
| 11:06.43 | slePP | Zeeek: that's expecting a lot of people who don't read in the first place, isn't it? |
| 11:07.01 | Zeeek | ya but they won't get any further ntil they read your conditions page |
| 11:07.05 | Zeeek | I'd go for that |
| 11:07.13 | shaZwaz | hey Zeeek |
| 11:07.15 | slePP | not a bad idea |
| 11:07.16 | Zeeek | and it will protect you from info theft suits |
| 11:07.23 | slePP | now go enter the draw |
| 11:07.24 | Zeeek | make the text SHORT |
| 11:07.35 | Zeeek | shipping to Europe is included? |
| 11:07.46 | slePP | sure, we don't have any trade embargos and it can't be more'n $20 |
| 11:08.01 | slePP | besides, isn't there an amazon.co.uk? :> |
| 11:08.11 | Zeeek | "you gree to use this service without guarantee blah blah not vbbe held respoinsible blah balh" |
| 11:08.14 | Rival | whats your guys take on ip communications being insecure? |
| 11:08.25 | Zeeek | IT IS! |
| 11:08.36 | Zeeek | deal with it |
| 11:08.37 | slePP | mmmmmm. phone sex |
| 11:09.23 | tuxinator_linux | slePP: ewww |
| 11:09.26 | slePP | heh |
| 11:09.52 | tuxinator_linux | Rival: I hear that more security is going to placed into the protocols, but I imagine they are pretty secure now |
| 11:10.09 | Rival | ya i suppose as secure as your weakest link |
| 11:10.21 | wasim | security is on the network part, there is talk of encrypted payload and encrypted protocol stuff, buts its all in the lab right now |
| 11:10.40 | tuxinator_linux | Rival: all of our VoIP will be on a private network, so I haven't been to concerned with it |
| 11:10.49 | Rival | ya |
| 11:10.56 | slePP | k, well, 4:10am |
| 11:10.58 | slePP | i'm going to bed |
| 11:11.05 | tuxinator_linux | 4:10 here also |
| 11:11.15 | slePP | funny how those time zones work ;> |
| 11:11.16 | Rival | 6:11am |
| 11:11.19 | Rival | just wakin up |
| 11:11.20 | wasim | :) |
| 11:11.31 | tuxinator_linux | I will be going down soon |
| 11:12.01 | shepherd | you can always use vpn to somewhat protect voip traffic |
| 11:12.02 | slePP | but, that's just extra detail |
| 11:12.11 | tuxinator_linux | slePP: terry who? |
| 11:12.12 | Rival | well thats already in place |
| 11:12.28 | slePP | tuxinator_linux: terry pratchett |
| 11:12.31 | Rival | but thats all well and good for interoffice connections |
| 11:12.36 | slePP | british fellow, writes the Discworld series |
| 11:12.44 | slePP | some of the best books i've ever read |
| 11:12.47 | Rival | oh |
| 11:12.48 | slePP | alongside douglas adams' stuff |
| 11:12.58 | Rival | i thought you were talking about that girl from desperate housewives |
| 11:13.25 | tuxinator_linux | What about William Gibson? |
| 11:13.32 | Zeeek | Shades |
| 11:13.35 | Rival | me either |
| 11:13.43 | Rival | but i know the girls name is close to that |
| 11:13.54 | Zeeek | Necromancer |
| 11:13.57 | tuxinator_linux | slePP: He coined the term cyberspace |
| 11:14.04 | slePP | but i've never read any |
| 11:14.06 | Zeeek | no that was John... |
| 11:14.21 | slePP | and so i hear all about the technogeek novels |
| 11:14.31 | shepherd | hax0r your gibson? |
| 11:14.36 | slePP | that's where it comes from |
| 11:14.43 | Rival | is that like startrekie school |
| 11:14.54 | slePP | you'd think so |
| 11:14.57 | Rival | hehe |
| 11:15.01 | slePP | but it's how humanities come together with technology |
| 11:15.08 | slePP | and so they do things like text analysis using perl, etc. |
| 11:15.18 | slePP | how cyberpunk novels have influenced culture, and vice versa |
| 11:15.24 | Rival | heh |
| 11:15.35 | slePP | they discuss people that implant things into themselves (one key person of which i forget the name) |
| 11:15.35 | Rival | i can see use for that |
| 11:16.01 | tuxinator_linux | You remember how everyone put "pi" symbols on their web pages when that one move came out? |
| 11:16.07 | slePP | The Net |
| 11:16.10 | slePP | with Sandra Bullock |
| 11:16.19 | tuxinator_linux | She's kinda cute |
| 11:16.21 | slePP | and i don't remember that anyone ever did that, but i can see it happening :> |
| 11:16.23 | slePP | she is, sometimes. |
| 11:16.26 | slePP | sometimes she's nasty |
| 11:16.30 | tuxinator_linux | yep |
| 11:16.50 | tuxinator_linux | slePP: The "pi" was everywhere, it was funny |
| 11:16.59 | slePP | wow. 2 entries to the draw. this is going to be an easy win for someone :> |
| 11:17.02 | godsmoke | you don't need to put it in quotes :) |
| 11:17.05 | godsmoke | it's a real word |
| 11:17.09 | slePP | hehe |
| 11:17.16 | Rival | anyone else having viewing problems @ slashdot with firefox? |
| 11:17.16 | slePP | perhaps it's in quotes to denote that the symbol should be there |
| 11:17.25 | slePP | Rival: slashdot is evil. you should purge it from your life. |
| 11:17.27 | tuxinator_linux | hey god, your right |
| 11:17.29 | godsmoke | riksta: loads just fine |
| 11:17.30 | shepherd | rival: www.slashdot.org works |
| 11:17.34 | shaZwaz | wasim hey this gr8 chekout http://64.78.24.200/cricket/sportztv/espntv.htm |
| 11:17.41 | Rival | weird |
| 11:17.48 | Rival | mines all screwed on all the office computers |
| 11:17.57 | godsmoke | up-to-date? |
| 11:18.04 | shepherd | rival: it's like it crashed |
| 11:18.11 | shepherd | it was doing that to me earlier |
| 11:18.22 | *** join/#asterisk xorol (~x@213.219.182.88.fixedpower.by.edpnet.be) |
| 11:18.23 | xorol | ello |
| 11:18.39 | xorol | does anybody now a voip provider for mobile cell traffic in belgium ? |
| 11:18.43 | slePP | Spiderman 2 is on the top 250 on imdb |
| 11:18.43 | slePP | ack |
| 11:18.46 | slePP | what is the world coming to |
| 11:18.54 | tuxinator_linux | the chick is cute |
| 11:19.05 | godsmoke | slePP: people still read slashdot -- what is the world coming to? |
| 11:19.11 | slePP | godsmoke: yeh... |
| 11:19.12 | tuxinator_linux | owww |
| 11:19.25 | godsmoke | heh |
| 11:19.31 | tuxinator_linux | what is wrong with slashdot? |
| 11:19.40 | slePP | it's just full of silliness that is pointless |
| 11:19.42 | marak | changing the tx and rx levels in zapata file has cleared up the calls on these 2 phones considerably but of course the bad sound fix has been suplemented by lower sound volume any ideas why only these 2 phone are so badly affected. the calls are now at least makeable from these 2 extensions. |
| 11:19.44 | godsmoke | yea |
| 11:19.49 | Rival | hehe |
| 11:19.53 | godsmoke | also |
| 11:19.57 | godsmoke | it's a reckless tool |
| 11:20.03 | Rival | still a viable source for news |
| 11:20.05 | godsmoke | the mods don't give any consideration to small server |
| 11:20.07 | godsmoke | servers* |
| 11:20.15 | slePP | it runs like a 'small little community site', when it is anything but |
| 11:20.15 | Rival | slepp where do you go for your news source then |
| 11:20.19 | marak | i have scanned the astrisk console with iax debuging calls are in gsm format on all cals |
| 11:20.22 | slePP | and so it runs rampant and accomplishes nothing |
| 11:20.28 | godsmoke | Rival: google news, baby |
| 11:20.28 | slePP | Rival: CBC |
| 11:20.43 | godsmoke | the slashdot effect can be a crushing tool |
| 11:20.51 | godsmoke | and the mods are not responsible AT ALL |
| 11:20.58 | godsmoke | so, I don't support it with my visit |
| 11:21.04 | slePP | yeh |
| 11:21.07 | slePP | and all the 'FIRST POST' crap |
| 11:21.17 | slePP | 'wow, look at you, chief...' |
| 11:21.36 | shaZwaz | damn sami's gone |
| 11:21.40 | godsmoke | yeh -- I remember that homebrew barcode scanning system for opening doors |
| 11:21.48 | godsmoke | and it completely brought down his network for days |
| 11:21.48 | slePP | plus i find the layout painful, but that's just me... i mean, lookit pastebin... that's painful too :> |
| 11:22.16 | slePP | it took 4 days for it to die down, after about 100gb of transfer |
| 11:22.24 | slePP | (this was on my home connection) |
| 11:22.32 | slePP | and it was merely in the 80th comment of some post |
| 11:22.45 | slePP | slashdot is the equivalent of mob mentality |
| 11:22.58 | *** join/#asterisk mady (~mady@61.11.24.250) |
| 11:23.02 | godsmoke | anyway |
| 11:23.03 | marak | ah 1 more question is there a way to speed up the echo cancellation in the beginning of a call. i seem to remeber reading something about getting asterisk to do some sort of sound ping before a call to check echo |
| 11:23.06 | godsmoke | enough ranting :) |
| 11:23.25 | slePP | now, you want a _real_ news site |
| 11:23.28 | slePP | http://www.gravitysucks.org/ |
| 11:23.33 | Rival | i hardly ever read the user posts |
| 11:23.37 | slePP | (k, it was cool in the day) |
| 11:23.43 | mady | hello denon:can i chat with u? |
| 11:23.47 | godsmoke | so, let me get this straight -- when one buys a cisco phone, all new and in the box from a decent company -- this doesn't come with this special service contract, which allows me to grab sip firmware? |
| 11:23.59 | slePP | godsmoke: don't think so |
| 11:24.03 | godsmoke | that's just nuts |
| 11:24.11 | slePP | you'd have to request the software license, i think |
| 11:24.15 | godsmoke | I'm so glad I don't work with cisco (the company) on a daily basis |
| 11:24.20 | Rival | and pay money |
| 11:24.27 | godsmoke | yeah, I heard it's like $75/year |
| 11:24.31 | godsmoke | for firmware |
| 11:24.37 | Rival | its a annual contract? |
| 11:24.40 | Rival | shit that blows |
| 11:24.47 | godsmoke | slePP: only for "international customers" |
| 11:24.51 | godsmoke | and it takes weeks to get your account |
| 11:24.52 | tuxinator_linux | $8 to get in the door |
| 11:25.05 | slePP | ah |
| 11:25.07 | godsmoke | if it was seriously $8, and I could get it tomorrow -- I wouldn't bitch |
| 11:25.24 | tuxinator_linux | I second that god |
| 11:25.35 | godsmoke | :) |
| 11:25.37 | slePP | or pap2's :> i have two boxes of those in my living room |
| 11:25.38 | mady | hello all: i am looking for technical help in asterisk configuration |
| 11:25.46 | slePP | godsmoke: it exists, you just have to find it ;> |
| 11:25.50 | godsmoke | slePP: I'm sure |
| 11:25.51 | shepherd | firmwarez |
| 11:25.51 | shepherd | hehe |
| 11:25.52 | godsmoke | haven't looked |
| 11:25.59 | slePP | shepherd: sounds like porn of young ladies |
| 11:25.59 | godsmoke | don't get my ciscos until wed |
| 11:26.01 | *** join/#asterisk Mother_ (~m@53.Red-217-126-93.pooles.rima-tde.net) |
| 11:26.12 | Rival | what ones ya go with godsmoke |
| 11:26.13 | wasim | mady: we're looking for financial help |
| 11:26.13 | godsmoke | was hoping they come with a cd or something |
| 11:26.15 | slePP | which sip firmware are you needing? you didn't get sip on them when you ordered? |
| 11:26.17 | Mother_ | hi all |
| 11:26.18 | godsmoke | Rival: the 7960s |
| 11:26.24 | Rival | nice |
| 11:26.27 | godsmoke | well ... see |
| 11:26.30 | godsmoke | no idea |
| 11:26.33 | Rival | hehe |
| 11:26.34 | godsmoke | there was no option for such things |
| 11:26.35 | tuxinator_linux | I see slePP on gravity |
| 11:26.36 | Rival | didnt ask? |
| 11:26.38 | Mother_ | godsmoke: you'll have to get one of those $8 service contracts |
| 11:26.43 | shepherd | does cisco not use h323? |
| 11:26.51 | slePP | tuxinator_linux: aye. i wrote it |
| 11:26.53 | slePP | ArBase2 |
| 11:26.54 | Mother_ | so you can download the SIP images off the Cisco support site |
| 11:26.57 | godsmoke | Mother_: I've heard it's quite a pain to even get one |
| 11:27.01 | godsmoke | Mother_: we know you can |
| 11:27.11 | slePP | shepherd: cisco uses SCCP |
| 11:27.16 | Mother_ | godsmoke: yes it is, but it's the only way :( |
| 11:27.21 | godsmoke | not really |
| 11:27.31 | godsmoke | I'm sure it's on the p2p networks somewhere -- or whatever |
| 11:27.34 | Rival | wheres the home brew ip phones =p |
| 11:27.38 | Mother_ | well sorry, it's the only *legal* way |
| 11:27.39 | Mother_ | :D |
| 11:27.50 | Rival | i wanna make my own phone |
| 11:28.01 | godsmoke | Mother_: why deal with such small details? :) |
| 11:28.08 | wasim | Rival: farfon.com |
| 11:28.10 | Mother_ | I wanna make my own full size steam locomotive, but don't have the time yet |
| 11:28.23 | RoyK | Rival: wasim's phones are nice |
| 11:28.29 | Mother_ | godsmoke: ph33r the Cisco Police and their black helicopters |
| 11:28.38 | FaithX | Heh XTen Lite works under wine |
| 11:28.42 | godsmoke | Mother_: I have the fbi waiting outside in a verizon truck -- like I care |
| 11:28.45 | shaZwaz | anyone download the fastsms app ? |
| 11:28.59 | FaithX | Better than any othe the Linux Phones |
| 11:29.02 | shaZwaz | downloaded |
| 11:29.09 | Zeeek | fyLinksys routers are locked and now only work with Cosco phones. Plus they check if the SIP image is legit and freeze the phone if not. |
| 11:29.34 | Mother_ | Zeeek: whaaaa? how can they check it's legit? |
| 11:29.46 | Zeeek | they do a dig legit.com |
| 11:29.46 | tuxinator_linux | checksum? |
| 11:30.01 | Zeeek | like rDNS only better |
| 11:30.12 | stustu | It seems that one can schedule an outgoing call by setting mtime on files that are put in the /var/asterisk/outgoing directory. Is this a documented feature that one can depend on? |
| 11:30.31 | Zeeek | you can copy any .call file to there |
| 11:30.34 | shepherd | hehe.. wow asterisk supports it! |
| 11:30.38 | Mother_ | well I know the images are signed etc. etc. but you are not given an individual image file for your phone |
| 11:30.39 | shepherd | http://www.voip-info.org/wiki-Asterisk+SCCP+channels |
| 11:31.07 | Mother_ | so unless they have your phone's MAC address and check if you have a service contract for that particular phone, I don't know |
| 11:31.13 | Zeeek | Mother your phone has a sooper secret password that only Crisco knows |
| 11:31.30 | tuxinator_linux | "letmein" |
| 11:31.33 | Zeeek | and when they detect your stolen SIP image, they lock up the phone |
| 11:31.38 | Mother_ | hmmmmm conspiracy policies heh |
| 11:31.51 | Zeeek | you guys are ready to believe this crock? :) |
| 11:31.54 | wasim | 300/8! |
| 11:32.00 | Mother_ | not really |
| 11:32.08 | Mother_ | it's a nice morning laugh tho :) |
| 11:32.24 | sambal | Zeeek: i added legit.com to my own dns server, so linksys can't resolve ;) |
| 11:32.31 | FaithX | Hey anyone got zaphfc running? |
| 11:32.32 | Mother_ | next they'll be putting small pyro charges inside the phones and detonating them remotely |
| 11:32.56 | Zeeek | hosts file is a wonderful thing too |
| 11:33.00 | Rival | isnt it illegal for them to phone home? |
| 11:33.15 | tuxinator_linux | Why would it be Rival |
| 11:33.19 | Mother_ | Rival: almost any piece of software you install on your PC nowadays phones home |
| 11:33.19 | Rival | or is it in the eula? |
| 11:33.26 | stustu | Zeeek: Yes, I know I can copy call files there, but is the fact that asterisk waits processing it until its mtime a feature, or does it just happen to do so? |
| 11:33.36 | sambal | the eula is big enough, so it can be there ;) |
| 11:33.37 | Mother_ | even windoze calculator phones home...well...almost |
| 11:33.39 | Rival | id consider it datamining |
| 11:33.58 | stustu | I have a cron job that puts files there, but to me it seems I could get rid of the cron job very easily... |
| 11:34.22 | Mother_ | I regularly see software now that phones home even while it's being installed, it doesn't wait for you to run it |
| 11:34.59 | Rival | yea |
| 11:35.01 | sambal | ET phone home |
| 11:35.02 | *** part/#asterisk marak (~twist@ndn-165-143-245.telkomadsl.co.za) |
| 11:35.03 | wizard2000 | anyone here that has disconnect supervision or busy detect working properly in the UK? |
| 11:35.20 | Rival | ya |
| 11:35.31 | shaZwaz | anyione used fastsms ? |
| 11:36.33 | Rival | jesus christ if they show those damn dolphins on tv ever again ill cancel my tv subscription |
| 11:36.53 | slePP | dolphins? |
| 11:36.59 | slePP | shaZwaz: i will be in about 3 days |
| 11:37.05 | tuxinator_linux | It's slePP: http://www.gravitysucks.org/images/collections/vanc-1-jul-27-2001/high/sleppDrive.jpg |
| 11:37.11 | shaZwaz | 3 days ? |
| 11:38.07 | wasim | i think we're gonna bowl soon |
| 11:38.16 | shaZwaz | :) |
| 11:38.42 | *** join/#asterisk antifuchs (~asf@walrus.boinkor.net) |
| 11:38.43 | antifuchs | hi all |
| 11:38.54 | slePP | tuxinator_linux: heh. http://geeksanon.ca/slepp-vix.jpg |
| 11:38.58 | slePP | me'n'my fiance |
| 11:39.23 | shaZwaz | I think it will a short test |
| 11:39.26 | shaZwaz | :) |
| 11:40.10 | *** join/#asterisk ennuyeux72 (~ennuyeux7@83.146.53.34) |
| 11:40.24 | tuxinator_linux | slePP: cute girl |
| 11:40.52 | slePP | thank ya :> |
| 11:40.55 | *** join/#asterisk h4mm3r` (~h4mm3r@81-208-60-202.fastres.net) |
| 11:40.58 | slePP | now if i showed you the _rest_ of those vacation pictures ;> |
| 11:41.20 | slePP | http://geeksanon.ca/netgal/ |
| 11:41.28 | antifuchs | I've got a pretty strange problem with asterisk's voicemail application and sound output. I installed the german voicemail sound files that the City of Pfortzheim offers, and now the voicemail app hangs up when it's supposed to play some sound files, like INBOX.gsm. at the time of the hangup, i get "WARNING[603601920]: Wait failed (Resource temporarily unavailable)" in the full log. our sip server is running on FreeBSD. |
| 11:41.56 | Mother_ | did you just type all that or did you paste from somewhere? ;) |
| 11:41.58 | antifuchs | english sound files that come with asterisk work perfectly. did anybody else have similar problems? |
| 11:42.03 | antifuchs | Mother_: typed it in (: |
| 11:42.16 | ta[i]nted | slePP those mountains look very familiar |
| 11:42.23 | slePP | ta[i]nted: where're you at? |
| 11:42.29 | slePP | cuz if you're in my time zone, they probably are |
| 11:42.33 | *** join/#asterisk kks (~kks@203.115.210.253) |
| 11:42.51 | ta[i]nted | well i'm in california now |
| 11:42.54 | Mother_ | antifuchs: make sure you have installed the right version of mpg123 |
| 11:42.58 | ta[i]nted | but they look strangely familiar |
| 11:43.11 | antifuchs | Mother_: these are .gsm sound files... /-: |
| 11:43.23 | Mother_ | ah sorry, was getting confused with MOH |
| 11:43.25 | Mother_ | apologies |
| 11:43.26 | slePP | ta[i]nted: think Salt Lake City |
| 11:43.30 | antifuchs | no problem (: |
| 11:43.30 | slePP | which is about 15 minutes away from there |
| 11:43.34 | tuxinator_linux | slePP: I see Arizona, I live there |
| 11:43.35 | *** join/#asterisk kks (~kks@203.115.210.253) |
| 11:43.45 | slePP | yes, and a bit of arizona. about 15-20 minutes of it |
| 11:43.55 | Mother_ | I tried to make some gsm files with Audition and converting on linux and it didn't work |
| 11:44.04 | Mother_ | * didn't hang, but they sounded horrible |
| 11:44.09 | *** join/#asterisk jbAU (~johnhewit@203-173-20-3.dyn.iinet.net.au) |
| 11:44.32 | slePP | it was a very long drive |
| 11:44.46 | antifuchs | for me, they work pretty well |
| 11:44.53 | tuxinator_linux | I like road trips |
| 11:44.58 | Mother_ | does this happen with any file, or with the first file it tries to play? |
| 11:45.01 | antifuchs | asterisk is pretty flexible when it comes to different sound formats |
| 11:45.04 | Mother_ | yeah |
| 11:45.06 | tuxinator_linux | But I am flying to VON, for the sake of time |
| 11:45.11 | shaZwaz | wasim u got it ! |
| 11:45.15 | antifuchs | you can use .wav and .alaw and whatever you want as a format. that's nice |
| 11:45.26 | slePP | no one wants me to go |
| 11:45.32 | slePP | (or rather, no one wanted to pay for me :) |
| 11:45.37 | *** join/#asterisk jedirl (~fdsafasdf@213.162.200.226) |
| 11:45.38 | tuxinator_linux | I am mostly going for Meet * |
| 11:45.38 | jedirl | Hello |
| 11:45.52 | jedirl | I'm trying to convert some .wav files to asterisk's .alaw format |
| 11:46.08 | tuxinator_linux | slePP: I don't work for anyone, so it had to come out of my pockey |
| 11:46.09 | jedirl | any hint on how can I do it? |
| 11:46.22 | jedirl | sox is quite extensive and I don't know how to use it... |
| 11:46.26 | antifuchs | jedirl: I just did that. "sox foo.wav -A foo_alaw.wav", then renaming foo_alaw.wav to foo.alaw did the trick for me |
| 11:46.57 | slePP | tuxinator_linux: i don't even know where VON is, let alone how much it is |
| 11:47.01 | slePP | san fran, perhaps |
| 11:47.11 | jedirl | antifuchs: thanks |
| 11:47.18 | tuxinator_linux | slePP: San Jose, close |
| 11:48.05 | tuxinator_linux | Its a hop skip and a jump for me |
| 11:48.26 | slePP | not bad then |
| 11:48.30 | slePP | 20+ hour drive for me |
| 11:48.42 | tuxinator_linux | it would be 12 hour driver for me |
| 11:49.13 | tuxinator_linux | I wish I could fly myself, just for the fun of it, but I am not complete with my private pilot certificate yet |
| 11:49.39 | Mother_ | oh I did the PPL a few years ago |
| 11:49.47 | Mother_ | don't have time to fly though :( |
| 11:50.03 | Mother_ | good luck with it |
| 11:50.54 | slePP | okie dokie, bedtime |
| 11:55.08 | wizard2000 | grrrrrrr |
| 11:55.10 | wizard2000 | telcos suck |
| 11:55.19 | wizard2000 | BT doesn't know what disconnect supervision is |
| 11:55.44 | jbAU | what is it? |
| 11:56.14 | Mother_ | wizard2000: telcos don't know they are telcos |
| 11:56.32 | Mother_ | they think they are a "service company" |
| 11:56.46 | *** join/#asterisk shadebob (~shadebob@rnis-162-206-192-81.marocconnect.com) |
| 11:56.54 | wizard2000 | heh |
| 11:56.57 | wizard2000 | doesn't help |
| 11:56.57 | Mother_ | mine swiftly removed callerid from the inbound lines when they were configured in a hunt group |
| 11:57.03 | wizard2000 | busydetect ain't working either :( |
| 11:57.16 | Mother_ | you'll probably have to play with the tones in the source and recompile |
| 11:58.00 | Mother_ | I was trying to get it to work in Spain, and had the same problems, basically VM will record empty messages until it times out, but I never got around to modifying the source |
| 11:58.32 | wizard2000 | yeah, this is the problem i have... in the UK obviously |
| 11:58.44 | wizard2000 | but this is just a test... when we go live we'll have an E1 trunk |
| 11:59.30 | Mother_ | for example |
| 11:59.31 | Mother_ | http://lists.digium.com/pipermail/asterisk-dev/2005-January/008702.html |
| 12:02.27 | *** join/#asterisk zotz (~zotz@24.231.32.191) |
| 12:02.49 | Mother_ | to which I have a question, when it is said to loadmod the TDM400 with OPERMODE=SPAIN, where does this instruction go? Do I have to change wctdm.c and rebuild? |
| 12:03.37 | ManxPower | ~docs |
| 12:03.38 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 12:03.50 | Mother_ | ergh |
| 12:04.11 | ManxPower | modprobe wcfxs opermode=SPAIN |
| 12:04.14 | shaZwaz | is smsq available in 1.0.3 ? |
| 12:04.20 | Mother_ | ManxPower: thanks a bunch |
| 12:04.57 | *** join/#asterisk JerJer[mobile] (~jj@65.173.197.109) |
| 12:05.58 | ManxPower | You can thank me by finding me a job in Europe. |
| 12:06.07 | Mother_ | lol I can try |
| 12:06.12 | ManxPower | 8-) |
| 12:06.13 | Mother_ | what kind of job? |
| 12:06.25 | ManxPower | see my /away message |
| 12:06.30 | shaZwaz | I guess not |
| 12:07.38 | Mother_ | lol I guess you should change can to will at the end :D |
| 12:08.43 | ManxPower | fied |
| 12:08.47 | ManxPower | fixed |
| 12:11.27 | Mother_ | OK, I know a company here that does Asterisk work, I'll drop them the line |
| 12:11.41 | Mother_ | tech work in Spain is really bad lately though, specially pay is crap |
| 12:14.26 | shadebob | someone had an example of zaptel.conf and zapata.conf for a server with 2 te110p? |
| 12:14.34 | Rival | whats the cheapest ata supported by asterisk |
| 12:15.14 | shepherd | depends on how many you buy :) |
| 12:15.25 | Rival | 1-2 |
| 12:15.28 | *** join/#asterisk Luhiwu (~marsosa@200.63.89.248) |
| 12:15.29 | shepherd | oh.. |
| 12:15.32 | shepherd | i don't know then |
| 12:15.33 | Rival | just for testing |
| 12:15.36 | elric | does the TE405P work on FreeBSD? I am using a TDM400P right now at work but we might deploy asterisk and replace our panasonic pbx |
| 12:16.12 | shepherd | te405p is sketchy on freebsd |
| 12:16.21 | shepherd | like the rest of asterisk |
| 12:16.29 | shepherd | on freebsd |
| 12:17.17 | CleanerX | anyone got a sip uri to test my setup? |
| 12:17.37 | CleanerX | a voicemail uri or something like that would be nice |
| 12:17.40 | CleanerX | meetme etc |
| 12:17.48 | shepherd | cleanerx: you could sign up with fwd :) |
| 12:17.52 | Mother_ | later all |
| 12:18.00 | Rival | 100$ for a digium ata iax seems a bit expensive |
| 12:18.16 | CleanerX | no i want a technical test |
| 12:18.48 | CleanerX | and a fwd wont help me in this case |
| 12:20.10 | *** join/#asterisk Vlasis (user@titan.physics.auth.gr) |
| 12:20.26 | Vlasis | hello |
| 12:20.31 | Luhiwu | Anyone here sells (or know where can i buy) origination in the US? |
| 12:20.34 | *** join/#asterisk mog_home (~mog_home@146.229.176.173) |
| 12:20.37 | stustu | I can't dial extensions that include a pound sign from X-Lite. The same extensions work from Zap interfaces... Anyone? |
| 12:20.42 | mog_home | Good morning |
| 12:21.00 | CleanerX | stustu, take a look at the kind of dtmf signaling |
| 12:21.10 | shepherd | luhiwu: nufone, voicepulse, broadvoice |
| 12:21.10 | Vlasis | I wanted to ask you if anyone knows where we can BUY licenses and binaries for a proper G723.1 implementation from 20 to 30 channels? |
| 12:21.36 | kks | in channel oh323 (not gk) how can i restrict only incomming from certail IP? |
| 12:21.42 | stustu | CleanerX: You mean the DTMF settings in X-Lite, right? |
| 12:21.47 | Darwin35 | Top the morning to you all. Rise and shine ladies |
| 12:22.50 | CleanerX | stustu, sip.conf |
| 12:23.03 | Luhiwu | shepherd: thanks, do you know if any of those will sell me bulk? i'm looking for offering US numbers with my service here in Argentina |
| 12:23.05 | CleanerX | there is inband and rfc and another one |
| 12:23.13 | *** join/#asterisk clive- (~pirch@rndf-146-55-139.telkomadsl.co.za) |
| 12:23.36 | shepherd | luhiwu: i'm sure they will |
| 12:23.44 | mog_home | mwatt why are you up? |
| 12:23.46 | Luhiwu | shepherd: thanks again :) |
| 12:23.47 | CleanerX | QUESTION for the whole round: what countries do you know to use ISDN predominantly? |
| 12:23.48 | mog_home | matt* |
| 12:23.53 | shepherd | mog: i have no idea |
| 12:23.53 | shepherd | hehe |
| 12:25.12 | shepherd | vlasis: http://www.voip-info.org/wiki-Asterisk+codecs |
| 12:27.38 | ManxPower | Vlasis, licenses for G723.1 is about $30,000 |
| 12:28.09 | Vlasis | shepherd: I already tried those implementeations... they are not compatible with all G723.1 codec implementations like for example ATAs etc |
| 12:28.18 | visik7 | can I use an High quality codec like ogg or mp3 ? I have a 100mbit so why don't use it ? |
| 12:28.28 | Vlasis | ManxPower: I know, the same are for G729 |
| 12:28.30 | visik7 | maybe it's useless but just to try |
| 12:28.46 | Vlasis | ManxPower: I'm looking for a party that can sell licenses in smaller bundles, like Digium does for G729 |
| 12:29.01 | antifuchs | visik7: it's very likely to be very useless, unless you want to make high-quality music calls (: |
| 12:29.07 | ManxPower | http://www.dspg.com/technology/LicensePricing.html |
| 12:29.07 | shadebob | how can I install 2 te110p on a same server? |
| 12:29.11 | CleanerX | visik7, try speex |
| 12:29.12 | Darwin35 | ok if bri is for isdn why put it in the base and not make it a addon ? |
| 12:29.13 | visik7 | antifuchs infact :) |
| 12:29.13 | antifuchs | ogg and mp3 are optimized for music, not speech |
| 12:29.30 | ManxPower | Vlasis, The patent holders of G723.1 are not interested in licensing their tech to smaller companies. |
| 12:29.31 | CleanerX | visik7, these codecs don't encode in realtime |
| 12:29.38 | Darwin35 | I know isdn is still used even though its outdated |
| 12:29.38 | antifuchs | um. unpreicse. vorbis and mp3 are (: |
| 12:29.38 | ManxPower | shadebob, yes |
| 12:29.59 | CleanerX | Darwin35, that completely incorrect |
| 12:30.02 | visik7 | CleanerX why not ? I encode an mp3 in less the time of the song |
| 12:30.19 | shadebob | ManxPower : in my zaptel.conf i don't known how to setup 2 cards? |
| 12:30.45 | Vlasis | shepherd, ManxPower: so the pointers on the wiki about the G723.1 are the onlyones available? |
| 12:30.46 | CleanerX | visik7 on your p4 3,6 GHz |
| 12:30.47 | ManxPower | shadebob, The channels on the second card will begin at 25 for T-1 |
| 12:30.58 | visik7 | CleanerX yes |
| 12:31.05 | visik7 | CleanerX what's the problem ? |
| 12:31.08 | ManxPower | Vlasis, Any F723.1 codec for Asterisk is UNLICENSED. |
| 12:31.18 | ManxPower | ..e.r..G723,1 |
| 12:31.21 | CleanerX | visik7 and then you have an interrupt in your os and audio gets dropped |
| 12:31.36 | visik7 | CleanerX it's just a test I don't want to use it in a production environment |
| 12:32.02 | visik7 | mmm |
| 12:32.05 | CleanerX | because of their structure they are simply not used |
| 12:32.37 | Vlasis | ManxPower: I know what you mean, this is why I asked for regular, licensed codec for G723.1 to pay for. I just don;t want more than 30 coders/decoders...:( |
| 12:32.52 | ManxPower | Vlasis, What you want is not available. |
| 12:32.57 | shepherd | visik: licensing restrictions prevent us from using the mp3 codec |
| 12:33.02 | Vlasis | ManxPower: I thought so...:( |
| 12:33.55 | antifuchs | oh. another question pops up |
| 12:34.06 | visik7 | shepherd ok nevermind mp3 but what about vorbis ? |
| 12:34.17 | shepherd | hmmm |
| 12:34.23 | ManxPower | visik7, Why do you want MP# or Ogg? |
| 12:34.25 | Darwin35 | speex is the key man |
| 12:34.28 | shadebob | ManxPower : hmmm ok... It's E1 line so span=1,1,0, cas, ami, crc4 bchan=1-15,17-31,32-46,48-62 dchan=16,47? |
| 12:34.31 | shepherd | manx: better quality :) |
| 12:34.41 | ManxPower | visik7, What does it give you that other codecs don't? |
| 12:34.46 | antifuchs | recently, I'm flooded (on the asterisk console even at debug/verbose = 0) with these messages: |
| 12:34.46 | antifuchs | Mar 8 11:18:37 NOTICE[135413760]: Disconnecting call 'SIP/sip_proxy_out_1-b0b1' for lack of RTP activity in 430740 seconds |
| 12:34.46 | antifuchs | Mar 8 11:18:37 NOTICE[135413760]: Disconnecting call 'SIP/a_user_name-9e57' for lack of RTP activity in 430740 seconds |
| 12:34.54 | shepherd | cisco phones support better quality, just don't know what codec they use |
| 12:34.54 | antifuchs | the weird thing is, it's counting UP |
| 12:34.55 | stustu | CleanerX: I tried all three DTMF variations in SIP.conf, but it doesn't work. Is DTMF even used when setting up the original call? |
| 12:34.57 | visik7 | ManxPower for high-quality music calls |
| 12:35.10 | antifuchs | and not down. |
| 12:35.11 | ManxPower | visik7, What is a music call? |
| 12:35.20 | visik7 | using * as a streaming server |
| 12:35.24 | antifuchs | I have not found any way to terminate that call on the cosole |
| 12:35.31 | ManxPower | The highest quality codec is ulaw and alaw. That is what the telcos use. |
| 12:35.43 | antifuchs | anybody know how this happens? and how to terminate the call? |
| 12:35.45 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
| 12:36.17 | ManxPower | antifuchs, The message is pretty obvious. See "soft hangup" in the CLI |
| 12:36.52 | antifuchs | ManxPower: I know what it says. but I have no idea why it's counting up. |
| 12:37.11 | antifuchs | ManxPower: but soft hangup is a good tip. thanks (: |
| 12:37.17 | ManxPower | antifuchs, Asterisk is no longer getting audio from the device, but got no disconnect request from the device |
| 12:37.37 | antifuchs | ok, that makes sense |
| 12:37.50 | antifuchs | the co-worker says he disconnected his phone while he talked |
| 12:38.09 | ManxPower | Tell him to stop hurting himself. |
| 12:39.17 | antifuchs | yeah (: |
| 12:39.21 | antifuchs | he's hurting me too, though (: |
| 12:39.41 | antifuchs | these messages have been showing up on the admin console for more than a day now |
| 12:39.52 | antifuchs | and it's still counting up |
| 12:40.01 | antifuchs | it's like the zebra in disney's alice in wonderland |
| 12:40.07 | *** part/#asterisk Vlasis (user@titan.physics.auth.gr) |
| 12:40.27 | antifuchs | "one lap to go, two laps to go, three laps to go, four laps to go, ..." as it runs around the tree |
| 12:41.00 | kks | in channel oh323 (not gk) any way to restrict only incomming from certain IP address? |
| 12:41.34 | ManxPower | antifuchs, Fix the network problems |
| 12:41.37 | shepherd | iptables? |
| 12:41.40 | shepherd | heh |
| 12:41.42 | antifuchs | ManxPower: ha ha |
| 12:42.02 | antifuchs | btw, the soft hangup didn't help either |
| 12:42.26 | antifuchs | it says 'requesting hangup', and the messages still keep buzzing by |
| 12:42.34 | DrRighteous | HAHAHAHAHA HAHAHAH AHHAHAAHAHAHAHA AHAHAHA <--(evil laugh over his new Cuba route) |
| 12:42.41 | ManxPower | antifuchs, As I said, it's really a network problem.; |
| 12:44.36 | antifuchs | ManxPower: and from the things I'm seeing, it's a problem in asterisk. please see http://paste.lisp.org/display/6403 |
| 12:45.11 | ManxPower | antifuchs, I see nothing in that pastebin to indicate an asterisk problem. |
| 12:45.32 | ManxPower | I see a packet filtering router/firewall between your server and the SIP clients |
| 12:45.52 | antifuchs | very well |
| 12:48.23 | antifuchs | so it's just my inferior mind seeing a countdown that counts up (: |
| 12:48.39 | Zeeek | all minds are inferior to ManxPower's |
| 12:48.49 | antifuchs | ...for a call that is supposed to have happened 119 hours ago (: |
| 12:49.15 | antifuchs | and 173 hours ago, respectively |
| 12:54.19 | *** join/#asterisk RoyKa (~roy@213.160.242.69) |
| 12:55.05 | riksta | is anyone interested in helping me with the ongoing development of ADM http://adm.hamnett.org |
| 12:56.12 | *** join/#asterisk Conductor (~thomas@62.8.240.132) |
| 12:58.09 | jedirl | When I convert my .wav files to a-law format with sox, I get a "click" at the beginning of the play of each file |
| 12:58.51 | ManxPower | antifuchs, I have given you my advice. I don't care if you follow it or not |
| 12:59.11 | antifuchs | ManxPower: thanks. |
| 13:00.20 | *** join/#asterisk Inferna (~sasha@194.158.51.171) |
| 13:00.27 | hajekd | someone here has sipura 841? |
| 13:08.46 | JunK-C | jedirl: the click could be remove, ive that when converting ulaw to gsm. |
| 13:12.50 | shadebob | How I can transfert an incoming call (E1 TE110P) to a channel bank (T1 TE110P) without answer call. A simple exten=>s,1,Dial(my extension, 20) work? I can't test now because I haven't hardware |
| 13:12.53 | hajekd | looks like sipura's speaker/mike is quite poor |
| 13:13.51 | Conductor | -- Got SIP response 482 "Loop Detected" back from 10.0.8.93 |
| 13:14.00 | JerJer[mobile] | don't loop |
| 13:14.04 | Zeeek | ya |
| 13:14.11 | Conductor | JerJer[mobile], it is no loop... |
| 13:14.13 | *** part/#asterisk antifuchs (~asf@walrus.boinkor.net) |
| 13:14.22 | Conductor | JerJer[mobile], at least not really |
| 13:14.32 | Zeeek | wizard, please calm down and tell us |
| 13:14.41 | wizard2000 | busydetect ain't working |
| 13:14.50 | JerJer[mobile] | don't use it |
| 13:14.53 | wizard2000 | so it doesn't hangup the line when the remote party hangs up |
| 13:15.05 | wizard2000 | which causes a problem |
| 13:15.08 | Zeeek | what country? |
| 13:15.09 | Conductor | JerJer[mobile], the thing is, that i registered asterisk as a sip-extension |
| 13:15.11 | wizard2000 | UK |
| 13:15.16 | wizard2000 | British Telecom |
| 13:15.20 | JerJer[mobile] | have your telco turn on disconnect supervsion |
| 13:15.23 | Zeeek | have you seen the many messages about this on the ML? |
| 13:15.34 | wizard2000 | JerJer[mobile]: they don't know what that is |
| 13:15.39 | Zeeek | have you seen the many messages about this on the ML? |
| 13:15.47 | wizard2000 | Zeeek: yeah, but no answers that i can see |
| 13:15.54 | Zeeek | there are sone things to try |
| 13:15.56 | JerJer[mobile] | look harder |
| 13:16.01 | Zeeek | also email Digium support |
| 13:16.12 | Zeeek | that's what they're there for |
| 13:16.21 | JerJer[mobile] | don't bug digium until you've done your own research |
| 13:16.30 | wizard2000 | JerJer[mobile]: does 2 days count? |
| 13:16.30 | Zeeek | [they must be sick of hearing about BT though] |
| 13:16.44 | *** join/#asterisk eipi (~eipi@100-172-114-200.fibertel.com.ar) |
| 13:16.46 | wizard2000 | this is research for a magazine article |
| 13:17.07 | Zeeek | JerJer I have to differ with you this time, Digium needs to either publish the BT solutions or respond to the queries |
| 13:17.09 | wizard2000 | so its pretty important it gets sorted... or no 60,000 copies of an * article will be published |
| 13:17.49 | JerJer[mobile] | loadzone=uk |
| 13:17.59 | JerJer[mobile] | i've setup countless boxes on that side of the pond |
| 13:18.06 | Zeeek | theu ssh ? |
| 13:18.09 | JerJer[mobile] | not one has issues hanging up |
| 13:18.13 | JerJer[mobile] | um yes |
| 13:18.55 | Zeeek | this is interesting http://lists.digium.com/pipermail/asterisk-users/2004-November/070860.html |
| 13:19.07 | *** join/#asterisk marak (~twist@196.46.64.143) |
| 13:19.14 | JerJer[mobile] | and not one uses busydetect |
| 13:19.17 | *** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net) |
| 13:19.23 | ariel_ | mornint |
| 13:19.28 | marak | hi all. |
| 13:19.39 | ariel_ | ./mornint morning all |
| 13:19.58 | Zeeek | wizard2000 |
| 13:20.39 | marak | quicky, have managed to sort out most of the echo problems on the 1 softphone. the last problem now comes in as follows. when no speaking they can hear the other side with no problems. when they speak the sound comes back into their own headphone all distorted and scrtachy |
| 13:20.44 | wizard2000 | let me try it quickly |
| 13:21.04 | Zeeek | look up disconnect supervision on the wiki |
| 13:21.12 | Zeeek | there are several comments that may help |
| 13:21.40 | wizard2000 | i have read all that |
| 13:21.44 | JerJer[mobile] | or simply plug in a lighted telephone and have someone else hang up on you |
| 13:21.51 | Zeeek | http://lists.digium.com/pipermail/asterisk-users/2002-May/002460.html |
| 13:22.16 | JerJer[mobile] | if the lights DO NOT go out you DO NOT have disconnection supervision on the line and no PBX out there will be able to reliably hang up that lnie |
| 13:22.33 | wizard2000 | there is no disconnect supervision, as i say |
| 13:22.37 | wizard2000 | BT don't even know what it is |
| 13:22.40 | Zeeek | well this is one of those topics like Broadvoice+Asterisk where a lot of people have solved it and no one has written the definitive how-to |
| 13:22.46 | wizard2000 | and i spoke to a so-called engineer |
| 13:22.48 | JerJer[mobile] | bullshit - they know what it is |
| 13:23.12 | Zeeek | the engineer was prolly a jalitor trying to impress his girl |
| 13:23.17 | wizard2000 | hehe |
| 13:23.27 | Zeeek | the truth is out there somewhere |
| 13:23.46 | Zeeek | once you find it, you'll forget about it too and the next person won't know |
| 13:24.11 | stevekstevek | wizard2000: try "disconnect supervision, mate" |
| 13:24.32 | stevekstevek | or "disconnect supervision, cheerio" :) |
| 13:24.33 | wizard2000 | stevekstevek: i am from london, thats the exact phrase i used! lol |
| 13:24.38 | stevekstevek | heh |
| 13:24.40 | JerJer[mobile] | the truth is: add disconnect supervision. Then it will magically work |
| 13:24.48 | wizard2000 | and if i work it out, it shall be written in a magazine |
| 13:24.51 | tzanger | dammit |
| 13:24.55 | wizard2000 | so i'd call that pretty well documented |
| 13:24.57 | tzanger | can a bug marshall reset my password on mantis? |
| 13:24.58 | JerJer[mobile] | this is not an asterisk problem |
| 13:25.02 | Zeeek | no don't do that, charge people for the answer |
| 13:25.14 | JerJer[mobile] | this is a telco problem |
| 13:25.26 | Zeeek | remember, the hooker sells what she has, then keeps it to sell again |
| 13:25.34 | Conductor | JerJer[mobile], when using a sip-phone, i dont get a loop-error. only when i use the *-manager. How come? |
| 13:25.45 | wizard2000 | so how do you suggest i get BT to enable disconnect supervision? |
| 13:25.47 | JerJer[mobile] | something is looping |
| 13:26.06 | JerJer[mobile] | wizard2000: tell them you have a pbx connected to the line and it requires disconnect supervsiion |
| 13:26.15 | JerJer[mobile] | very trival |
| 13:26.19 | Zeeek | that sounds pretty grown up! |
| 13:26.40 | wizard2000 | JerJer[mobile]: i did that |
| 13:26.52 | JerJer[mobile] | then it should be on your line |
| 13:26.57 | wizard2000 | nope |
| 13:27.07 | wizard2000 | they claimed to have never heard about that |
| 13:27.13 | tzanger | nevermind I got it |
| 13:27.17 | JerJer[mobile] | then throw away your computer and telephony hardware and bake cookies for a living |
| 13:27.26 | Zeeek | ouch |
| 13:27.34 | wizard2000 | JerJer[mobile]: yes, i think i'll do that |
| 13:27.48 | wizard2000 | it would mean not dealing with such helpful people as yourself |
| 13:28.16 | JerJer[mobile] | i'm telling you the precise solution to your problem |
| 13:28.31 | wizard2000 | yeah, i'll write that for the magazine, sure they'd love it |
| 13:28.32 | JerJer[mobile] | if you don't like it - leave |
| 13:28.50 | tzanger | wizard2000: what, precisely, is the issue? |
| 13:28.59 | JerJer[mobile] | fxo isn't hanging up |
| 13:29.02 | wizard2000 | tzanger: hangup detection |
| 13:29.05 | JerJer[mobile] | cuz there is no disconnect supervision on the line |
| 13:29.11 | JerJer[mobile] | very trivial |
| 13:29.18 | wizard2000 | yup, and BT claim not to know what it is |
| 13:29.21 | tzanger | wizard2000: do you have a multimeter or something to read voltage? |
| 13:29.31 | JerJer[mobile] | or very simply a lighted phone |
| 13:29.33 | Zeeek | <PROTECTED> |
| 13:29.38 | wizard2000 | tzanger: fraid not, neither |
| 13:29.50 | *** join/#asterisk arkaoss (trilluser@69.37.244.103.adsl.snet.net) |
| 13:29.56 | wizard2000 | some people say that it has worked with busy tone detection |
| 13:29.59 | JerJer[mobile] | then call it "battery reversal" or "battery removal" |
| 13:30.02 | *** part/#asterisk jks (~jks@0x503e4c12.arcnxx4.adsl-dhcp.tele.dk) |
| 13:30.05 | arkaoss | woah.. |
| 13:30.08 | JerJer[mobile] | that is a hack |
| 13:30.14 | Zeeek | wizard200 did you read the above before? |
| 13:30.15 | wizard2000 | yes, its a hack |
| 13:30.15 | JerJer[mobile] | get disconnect supervision on your line |
| 13:30.19 | tzanger | wizard2000: disconnect supervision comes in a couple of flavours -- both involve some kind of modification to the voltage on the line when the far-end hangs up |
| 13:30.19 | wizard2000 | but if it works... |
| 13:30.28 | JerJer[mobile] | it doesn't work |
| 13:30.31 | JerJer[mobile] | as you have experienced |
| 13:30.56 | tzanger | wizard2000: in Canada, for example, they drop the battery out altogether. If you have your meter across ring and tip you will see it hover around -8VDC for the duration of the call and when the remote end hangs up you'll see it drop to zero briefly before going back to -8V |
| 13:31.15 | wizard2000 | tzanger: the problem i have is getting it turned on |
| 13:31.28 | Zeeek | we can't with that here though |
| 13:31.30 | JerJer[mobile] | keep complaining until find someone with a clue |
| 13:31.35 | wizard2000 | no, i understand that |
| 13:31.38 | Zeeek | i/help/ |
| 13:31.54 | tzanger | wizard2000: battery reversal is also a common way to detect it. In this case the central office will actually reverse polarity for a brief amount of time but the end result is the same -- a very detectable signal from the CO telling you that the other end has hung up |
| 13:32.04 | arkaoss | i extracted 1.06.tar, and ran make install on my debian box.. but i can't tell if it "tried" to install the webcontrol panel.. or if it did, and it didn't link it properly to my actual webserver's configuration (installing this on a system that has other stuff already running on it) |
| 13:32.14 | Zeeek | the klu=dge is callprogress=yes I think? |
| 13:32.25 | wizard2000 | that only works in the US Zeeek |
| 13:32.31 | tzanger | wizard2000: far end disconnect supervision, battery reversal, battery removal, CPD (Calling party disconnect), etc. |
| 13:32.35 | Zeeek | then get back on the phone with BT |
| 13:32.48 | tzanger | asterisk can detect those on FXO ports with kewlstart (ugh I hate that term) signaling |
| 13:32.54 | JerJer[mobile] | lol |
| 13:32.56 | JerJer[mobile] | kewlstart |
| 13:33.04 | Zeeek | muhahaha |
| 13:33.16 | JerJer[mobile] | someone give me a k |
| 13:33.20 | JerJer[mobile] | :) |
| 13:33.32 | tzanger | if you can't get the telco to turn that shit on then you're scrogged -- how do your KSUs or PBXes (or fax machines) detect when the other end has hung up? |
| 13:33.37 | tzanger | JerJer[mobile]: bite me :-) |
| 13:33.54 | JerJer[mobile] | tzanger: my grilfriend bites me enough, sorry |
| 13:34.04 | wizard2000 | tzanger: they're not on analogue lines |
| 13:34.17 | JerJer[mobile] | then don't run analog |
| 13:34.39 | tzanger | wizard2000: that is probably the exact reason why |
| 13:34.42 | wizard2000 | JerJer[mobile]: i don't intend to |
| 13:34.44 | wizard2000 | this is not for me |
| 13:34.46 | tzanger | wizard2000: this isn't an asterisk-specific issue |
| 13:34.56 | JerJer[mobile] | telco issue |
| 13:34.58 | wizard2000 | i am writing an article for a UK computing magazine |
| 13:35.03 | Zeeek | it's for a train spotter mag? |
| 13:35.11 | tzanger | if your $18000 norstar system can't do it without digital lines, neither can Asterisk |
| 13:35.12 | wizard2000 | trying to promote asterisk |
| 13:35.18 | JerJer[mobile] | trainspotting |
| 13:35.18 | wizard2000 | its computer shopper UK |
| 13:35.29 | Zeeek | kewl |
| 13:35.32 | wizard2000 | biggest magazine in britain |
| 13:35.34 | tzanger | wizard2000: so promote it, but asterisk is not magical, it can't tell you when the far end's hung up if the telco doesn't tell it |
| 13:35.37 | Zeeek | fattest? |
| 13:35.42 | wizard2000 | 4 pages dedicated to asterisk |
| 13:35.46 | JerJer[mobile] | phatest |
| 13:35.56 | JerJer[mobile] | 4 whole pages? |
| 13:36.04 | tzanger | wizard2000: that's great news, but jerjer wasn't being an ass -- if the telco doesn't give you CPD, asterisk simply cannot tell you when the far end's gone |
| 13:36.12 | arkaoss | grrr.. that's more than the website's have.. |
| 13:36.16 | tzanger | wizard2000: NOW |
| 13:36.17 | Zeeek | the nation that drives on the left and has BT as a phone system |
| 13:36.20 | tzanger | wizard2000: having said that |
| 13:36.31 | tzanger | wizard2000: asterisk does have some form of magic in callprogress |
| 13:36.38 | tzanger | wizard2000: it's experimental, and it can cause problems |
| 13:36.51 | arkaoss | what's mean : unable to open management configruation manager.conf call managment disabled... |
| 13:37.01 | marak | does anyone know if digium cards make use of the sound card to control the audio of the calls |
| 13:37.04 | wizard2000 | tzanger: thank you |
| 13:37.12 | Zeeek | marak no sound card needed |
| 13:37.18 | tzanger | wizard2000: but this is where open source shines -- BT won't give you CPD on your analogue lines, so the Asterisk devs tried to give Asterisk the ability to "listen" to the audio stream to try and pick out various stages of call progress |
| 13:37.32 | tzanger | wizard2000: and as you've discovered, UK tones and so on are not all that well tested |
| 13:37.38 | Zeeek | I mentioned callprogress ten minutes ago |
| 13:37.40 | tzanger | wizard2000: simply because most of the developers aren't there |
| 13:37.42 | tzanger | Zeeek: I know |
| 13:37.45 | *** join/#asterisk slav_jb (~k@pirus.securax.be) |
| 13:37.46 | marak | zeeek: any idea why 2 users have horendous feedback in their headsets and no one else |
| 13:37.58 | Zeeek | ok - the record is set straight then :) |
| 13:38.03 | tzanger | wizard2000: but, if you're willing to spend some time you can hook up with a developer and help get UK call progress tones working better |
| 13:38.08 | wizard2000 | when i have more time i'm happy to do some work on it |
| 13:38.17 | Zeeek | marak not really |
| 13:38.21 | tzanger | wizard2000: that's again the beauty of open source and why people are using asterisk |
| 13:38.28 | wizard2000 | i am a coder btw, i write now and again freelance |
| 13:38.28 | Conductor | This is what i execute in the manager: "Action: originate Exten: 91 Context: sip Channel: SIP/99@asterisk Priority: 1" |
| 13:38.40 | Conductor | where asterisk is the asterisk server itself |
| 13:38.54 | tzanger | I myself have been working with Steve Kann on better jitter buffers and stuff... he's been doing all the hard work but I've been giving him feedback and providing a place to bounce ideas off of |
| 13:38.55 | JerJer[mobile] | until then: bitch at BT to add disconnect supervision, battery reversal, battery removal or whatever they call it to your line |
| 13:39.17 | Zeeek | tzanger is tyou head that hard? |
| 13:39.32 | tzanger | Zeeek: can you rephrase that in English, please? :-) |
| 13:39.42 | JerJer[mobile] | Conductor: and does exten => 91 call sip/99 ? |
| 13:39.45 | arkaoss | do i have to make manager sepretly??? that's nice to KNOW! |
| 13:39.47 | JerJer[mobile] | that would be a loop |
| 13:39.47 | Zeeek | tzanger izt das hed hardenst ? |
| 13:39.51 | Conductor | JerJer[mobile], no |
| 13:40.10 | JerJer[mobile] | sip/99@asterisk ? |
| 13:40.11 | Conductor | JerJer[mobile], the same with all other extensions |
| 13:40.16 | JerJer[mobile] | that is odd to me |
| 13:40.33 | tzanger | Zeeek: is my head hard? |
| 13:40.36 | Conductor | JerJer[mobile], this is a test extension |
| 13:40.44 | *** join/#asterisk RoyK (~roy@213.160.242.69) |
| 13:40.50 | JerJer[mobile] | no that is a channel |
| 13:40.52 | JerJer[mobile] | not extension |
| 13:40.56 | *** join/#asterisk afe ([Uvv35diBl@c-e616e055.123-1-64736c12.cust.bredbandsbolaget.se) |
| 13:41.04 | Zeeek | too late, WARNING No such comedic context |
| 13:41.07 | Conductor | JerJer[mobile], the 99 is a extension isn't it? |
| 13:41.27 | JerJer[mobile] | it is going to call 99 at the peer 'asterisk' |
| 13:41.32 | JerJer[mobile] | using sip |
| 13:41.42 | Conductor | JerJer[mobile], yes. |
| 13:41.44 | tzanger | Zeeek: heh |
| 13:41.49 | JerJer[mobile] | ok why? |
| 13:42.11 | JerJer[mobile] | how about |
| 13:42.22 | JerJer[mobile] | [99] |
| 13:42.22 | JerJer[mobile] | type=peer |
| 13:42.22 | JerJer[mobile] | host=1.2.3.4 |
| 13:42.29 | JerJer[mobile] | then channel: SIP/99 |
| 13:42.43 | Conductor | JerJer[mobile], in sip.conf? |
| 13:42.51 | JerJer[mobile] | the peer def, yes |
| 13:42.57 | *** join/#asterisk tekati (~captain@cpe-66-75-215-63.bak.res.rr.com) |
| 13:43.03 | JerJer[mobile] | channel: blah is part of the manager command |
| 13:43.40 | Conductor | JerJer[mobile], i will try that. thanks |
| 13:44.09 | JerJer[mobile] | how about keeping it simple first? |
| 13:44.17 | arkaoss | aaarg.. i thought i was... |
| 13:44.31 | arkaoss | what's command line to create an extension? |
| 13:44.44 | JerJer[mobile] | ~wiki |
| 13:44.58 | JerJer[mobile] | ~rtfm |
| 13:45.03 | jbot | methinks rtfm is read the f*cking manual... try asking me about "FAQ" |
| 13:45.20 | shepherd | ~faq |
| 13:45.21 | jbot | hmm... faq is The #debian FAQ is at http://www.linuks.mine.nu/debian-faq-wiki/ |
| 13:45.31 | JerJer[mobile] | lol stupid bot |
| 13:45.36 | Zeeek | ~seen clue |
| 13:45.46 | jbot | Zeeek: i haven't seen 'clue' |
| 13:45.52 | Zeeek | obviously not |
| 13:45.58 | JerJer[mobile] | ~seen a_clue |
| 13:45.59 | jbot | i haven't seen 'a_clue', JerJer[mobile] |
| 13:45.59 | shepherd | forget faq |
| 13:46.06 | shepherd | ~forget faq |
| 13:46.06 | jbot | shepherd: i forgot faq |
| 13:46.08 | Zeeek | ~got milk |
| 13:46.09 | jbot | ACTION chugs down a carton |
| 13:46.17 | Zeeek | ~got clue |
| 13:46.33 | tzanger | ~got phat beats |
| 13:46.57 | Zeeek | ~phish JerJer_account_inft |
| 13:47.02 | BrianR___ | jbot: eat a dick |
| 13:47.04 | jbot | ACTION eats a dick and falls over dead |
| 13:47.05 | JerJer[mobile] | ~seen yo_mama_tonight |
| 13:47.06 | jbot | JerJer[mobile]: i haven't seen 'yo_mama_tonight' |
| 13:47.12 | tuxinator_linux | he he he |
| 13:47.16 | arkaoss | Failed sending email :: PHP :: |
| 13:47.16 | arkaoss | DEBUG MODE |
| 13:47.17 | arkaoss | Line : 234 |
| 13:47.17 | arkaoss | File : emailer.php |
| 13:47.18 | tuxinator_linux | falls over dea |
| 13:47.20 | tuxinator_linux | dead |
| 13:47.31 | tzanger | don't make me break out the jbot fun |
| 13:47.36 | arkaoss | nop.. wanted to paste this http://www.voip-info.org/wiki-Asterisk+-+documentation+of+application+commands |
| 13:47.37 | JerJer[mobile] | lol |
| 13:47.49 | tzanger | ~seen my dick in 3 years |
| 13:47.50 | jbot | tzanger: i haven't seen 'my dick in 3 years' |
| 13:47.54 | Rival | anyone know a list of compliant modems? |
| 13:48.00 | tuxinator_linux | poor guy |
| 13:48.05 | JerJer[mobile] | Rival: there isn't any |
| 13:48.10 | Zeeek | there is a pretty big one of complaint ones though |
| 13:48.15 | tuxinator_linux | Rival: You still awake? |
| 13:48.23 | Rival | sup tux |
| 13:48.25 | Rival | ya |
| 13:48.28 | tzanger | ~seen that new Tarrintino film |
| 13:48.29 | jbot | tzanger: i haven't seen 'that new tarrintino film' |
| 13:48.33 | tuxinator_linux | Almost 7 AM here |
| 13:48.37 | Rival | 8:48am |
| 13:48.42 | tzanger | 8:44am here |
| 13:48.48 | Zeeek | 14:48 here |
| 13:48.51 | JerJer[mobile] | 7:48 here |
| 13:49.08 | Zeeek | wasim what time is it? |
| 13:49.10 | Conductor | JerJer[mobile], Ill try to explain my problem. i want to call extension 99 which is defined in extensions.conf |
| 13:49.14 | Zeeek | coppince? |
| 13:49.15 | tuxinator_linux | ~learn There is never enough time for sleep |
| 13:49.16 | jbot | tuxinator_linux: I think you lost me on that one |
| 13:49.37 | tzanger | ~seen jerjer ever help a newbie |
| 13:49.38 | jbot | i haven't seen 'jerjer ever help a newbie', tzanger |
| 13:49.41 | tzanger | :-) |
| 13:50.13 | clive- | lol |
| 13:50.14 | tzanger | wtf why's my nufone account suddenly suspended? :-) |
| 13:50.15 | Conductor | tzanger, you mean me? |
| 13:50.18 | clive- | he helped me:) |
| 13:50.20 | tuxinator_linux | I think I like this station http://www.shoutcast.com/sbin/tunein-station.pls?id=3033&filename=playlist.pls |
| 13:50.37 | tzanger | tuxinator_linux: what kind of music |
| 13:51.00 | tuxinator_linux | mostly dance |
| 13:51.06 | tuxinator_linux | techno |
| 13:51.13 | tzanger | tuxinator_linux: I listen ot the various beats on di.fm for that |
| 13:51.21 | tuxinator_linux | good programming music for me |
| 13:51.35 | tzanger | yup |
| 13:51.42 | tuxinator_linux | di.fm? |
| 13:52.12 | tuxinator_linux | I also like http://www.shoutcast.com/sbin/tunein-station.pls?id=2555&filename=playlist.pls |
| 13:53.16 | tzanger | yeah http://di.fm |
| 13:54.26 | Makenshi | i listen to no holds barred radio, provides the vital humor i need to get through the dullness of work :) |
| 13:55.18 | tzanger | :-) |
| 13:55.30 | *** join/#asterisk FABRIZIOXXX (FABRIZIO@81.208.26.86) |
| 13:55.34 | tuxinator_linux | oh ya, DI |
| 13:55.41 | tuxinator_linux | I used to listen to them all the time |
| 13:56.42 | JerJer[mobile] | i want to know how they afford the kind of bandwidth they burn |
| 13:57.23 | tzanger | JerJer[mobile]: agreed |
| 13:57.42 | tzanger | mind you the mixes I listen to are all 96kbps and amazingly good quality (I listen to it on my 5.1 receiver) for that bandwidth |
| 13:58.03 | tzanger | JerJer[mobile]: just remember you're burning through about the same at ulaw and charging pennies per minute |
| 13:58.58 | *** join/#asterisk delphiuk (~delphi@host217-44-239-113.range217-44.btcentralplus.com) |
| 13:59.22 | Conductor | @all: I want to use the manager to call an asterisk-extension. But when using Originate I always get a "loop" error. |
| 13:59.28 | tuxinator_linux | tzanger: crazy stuff |
| 14:00.01 | *** join/#asterisk jhiver (~jhiver@AStDenis-103-1-1-88.w81-249.abo.wanadoo.fr) |
| 14:00.06 | jhiver | hi all |
| 14:00.15 | Zeeek | salut |
| 14:00.25 | Essobi | Morning people. |
| 14:00.27 | jhiver | salut Zeeek |
| 14:00.52 | jhiver | I have a rather strange one way audio issue... calling party can hear me, but I can't hear them |
| 14:01.13 | tuxinator_linux | jhiver: not an issue, you'r just going deaf |
| 14:01.16 | Essobi | Conductor see dial local |
| 14:01.28 | jhiver | using SPA-3k behind NAT + asterisk box on a public IP address |
| 14:01.30 | jhiver | lol |
| 14:01.44 | tuxinator_linux | jhiver: microsoft-ish answer |
| 14:01.58 | jhiver | I've tried with a couple of different phones and also a couple of different ears :) |
| 14:02.27 | jhiver | oops sorry actually it's the reverse |
| 14:02.39 | jhiver | I can hear calling party, they can't hear me |
| 14:02.40 | jhiver | confusing |
| 14:02.44 | *** join/#asterisk MikeJ[Jayden] (~ircatjerr@65.170.43.34) |
| 14:03.06 | jhiver | so it's strange that the stream manages to get through NAT from the asterisk box but the reverse doesn't work |
| 14:03.15 | jhiver | any ideas on what can be causing this? |
| 14:04.36 | jhiver | I see :) |
| 14:04.45 | *** join/#asterisk dolson (~dana@Sudbury-HSE-ppp3978298.sympatico.ca) |
| 14:05.32 | dolson | is anyone good with tethereal filter syntax? I want to see all the sip traffic except for the OPTIONS method. I tried tethereal -R "sip.Method !='OPTIONS'" but that didn't work |
| 14:06.22 | *** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com) |
| 14:08.15 | *** join/#asterisk Hmmhesays (negative3k@66.173.103.108) |
| 14:10.08 | marak | just fyi the incredibly bad echo on the headset for the 1 user was not asterisk at all they had mic boost on since taking it off the calls are great |
| 14:10.24 | FABRIZIOXXX | hello all, i just received my new digium tdm400p with 2 FXO modules on it. To my suprise the plugs on the card seem to be the ones for isdn type jack and not standard phone jacks .. is this normal? I thought this card (as written on the website) could be used also for standard telephone lines. |
| 14:10.28 | marak | it also removed all the distortion |
| 14:11.32 | marak | does anyone know why sometimes when on a call that came in to the pbx and you press # key to transfer to another extension it does not work only plays the tones in your ear ? |
| 14:11.51 | *** join/#asterisk justinnnnnn (~justinnnn@c211-28-200-67.eburwd1.vic.optusnet.com.au) |
| 14:11.59 | Conductor | Essobi, thanks man. this is it... |
| 14:12.08 | Zeeek | FABRIO they are standard phone jacks |
| 14:12.14 | justinnnnnn | grrr txfax is still killing me :( |
| 14:12.34 | Zeeek | you can pplug a cable from a normal analogue phone line into it - in fact, that's what it's ofr |
| 14:12.53 | FABRIZIOXXX | Zeeek, if you say so .. i'll open the plastic and check right now.. |
| 14:12.55 | JerJer[mobile] | justinnnnnn: don't use analog |
| 14:13.09 | Zeeek | marak the dial application has to have the option : from CLI show application dial |
| 14:13.29 | Zeeek | FAB that what every TDM400P owner has plugged in to it |
| 14:13.41 | FABRIZIOXXX | ok |
| 14:13.45 | Zeeek | don't use fax, period |
| 14:14.09 | JerJer[mobile] | we use txfax on PRI quite a lot, without issue |
| 14:14.21 | Zeeek | FAB.*XXX the fit is a little tight (that's what she said) but it works |
| 14:14.33 | tzanger | I've found faxing to be very good generally, but the odd crapout (usually a SIGFPE) |
| 14:17.19 | marak | zeek: the -t and -T are identical ? |
| 14:17.56 | justinnnnnn | jerjer.. i cant get it going.. |
| 14:18.03 | justinnnnnn | rxfax works fine.. txfax makes it segfault |
| 14:18.04 | *** part/#asterisk gongoputch (~kseel@pcp01486721pcs.limstn01.de.comcast.net) |
| 14:18.10 | justinnnnnn | we need someone to fix her up :)) |
| 14:18.13 | justinnnnnn | reeaaal good |
| 14:18.46 | JerJer[mobile] | runnng the exact versions as documented? |
| 14:19.03 | marak | zeeek: this how i currently call extension exten => 1,1,Dial(iax2/2002,20,tr) |
| 14:19.09 | justinnnnnn | ive got cvs from like 15/01... |
| 14:19.26 | justinnnnnn | and the latest spand/libtif |
| 14:19.32 | justinnnnnn | ive tried allll the versions |
| 14:19.34 | JerJer[mobile] | damn last i knew there was only 12 months |
| 14:19.43 | justinnnnnn | 15/01/05 |
| 14:20.02 | JerJer[mobile] | i know, just giving you hell |
| 14:20.17 | drsperm | Has anyone ever used a SBE T1/PRI Card with * ? |
| 14:20.18 | JerJer[mobile] | run the exact versions of spand/libtiff that is documented |
| 14:20.39 | justinnnnnn | how do i reinstall both of them |
| 14:20.42 | justinnnnnn | without screwing it all up tho ? |
| 14:20.45 | justinnnnnn | that is the question |
| 14:20.51 | FABRIZIOXXX | sorry to disturb .. but i just checked the picture on the product sheet downloaded from the website and the plugs are completely different ..!! |
| 14:20.52 | JerJer[mobile] | rm -rf /boot ; reboot -n |
| 14:21.02 | justinnnnnn | k 1 sec... |
| 14:21.08 | JerJer[mobile] | funny |
| 14:21.12 | justinnnnnn | it will fix the fax u say ? this rm -rf |
| 14:21.34 | JerJer[mobile] | um er yeah it will solve all of your linux problems |
| 14:21.41 | JerJer[mobile] | right now |
| 14:21.50 | justinnnnnn | until work starts 2morow morning ? |
| 14:22.35 | JerJer[mobile] | everything will work great, until the bios POST completes :) |
| 14:26.38 | afe | FABRIZIOXXX: the contacts on the TDM400 are RJ45 (the same kind used for ISDN and Ethernet), and phones usually use RJ11 (smaller plug). |
| 14:26.46 | afe | it still works though |
| 14:27.11 | *** join/#asterisk eKo1 (~bernd@207.42.191.67) |
| 14:27.34 | FABRIZIOXXX | afe, so do i have to use isdn on these or can i plug in an RJ11 into these ..? |
| 14:28.04 | afe | FABRIZIOXXX: just plug in an RJ11. |
| 14:28.21 | FABRIZIOXXX | afe, really .. can it be done!? |
| 14:28.40 | afe | FABRIZIOXXX: That's how my TDM400 is connected |
| 14:29.17 | FABRIZIOXXX | ok great! Thank you very much for the info!! |
| 14:29.46 | afe | I guess they put RJ45 contacts there to make it easier to use regular patch cables if you want to use it in an office |
| 14:30.43 | eKo1 | Exactly, RJ45 ownz RJ11. |
| 14:31.03 | `Sauron | They're actually RJ48 |
| 14:31.05 | `Sauron | but who cares |
| 14:31.50 | *** part/#asterisk JerJer[mobile] (~jj@65.173.197.109) |
| 14:32.19 | *** join/#asterisk JerJer[mobile] (~jj@65.173.197.109) |
| 14:32.28 | *** join/#asterisk montoya (montoya@200.216.164.93) |
| 14:32.33 | *** join/#asterisk coppice (~chatzilla@134.195.17.210.dyn.pacific.net.hk) |
| 14:32.49 | montoya | alguem fala portugues aqui? |
| 14:32.59 | `Sauron | english, no portugese |
| 14:34.31 | montoya | i want to buy a X100P fax modem in brazil! |
| 14:34.56 | montoya | where is problem |
| 14:35.01 | eKo1 | Then god speed. |
| 14:35.53 | dolson | eBay |
| 14:35.56 | *** join/#asterisk shaZwaz (~adnans@202.59.71.209) |
| 14:38.12 | JerJer[mobile] | Caveat Emptor |
| 14:38.40 | montoya | what difference between sip and iax ? |
| 14:38.55 | mikegrb | two letters |
| 14:38.58 | JerJer[mobile] | s,p and x |
| 14:39.08 | mikegrb | JerJer[mobile]: I beat you! |
| 14:39.15 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.rr.com) |
| 14:39.15 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:39.21 | JerJer[mobile] | i guess a too |
| 14:39.27 | mikegrb | ja |
| 14:39.45 | *** join/#asterisk cbachman (~chatzilla@victory.ece.northwestern.edu) |
| 14:39.50 | clive- | has anyone here tried this iax ata called an ag-168ve ? |
| 14:39.57 | *** join/#asterisk mafoo (~foo@bombadil.mafoo.org.uk) |
| 14:40.02 | eKo1 | Who makes it? |
| 14:40.16 | Funbags | aanyone having issues with bv incoming this morning? |
| 14:40.24 | clive- | eko its got that pa168 chip in it, same as the ezzeephones etc |
| 14:41.11 | *** join/#asterisk tekati (~captain@cpe-66-75-215-63.bak.res.rr.com) |
| 14:41.27 | mafoo | ok -- simple question ... am i utterly misunderstanding something when I expect an external 56k modem to be the interface between an asterisk install and a POTS ? |
| 14:41.30 | `Sauron | funbags: Not that I know of. |
| 14:41.30 | spackle | hail cbachman, got * running on your Motorola AP? |
| 14:41.33 | eKo1 | This cepstral stuff is confusing. |
| 14:41.58 | coppice | mafoo: yes |
| 14:42.05 | eKo1 | mafoo: * is not a modem server. |
| 14:42.13 | cbachman | spackle, I'm close. I have it all compiled, but I'm apparently missing something. My SIP clients can register, but when I attempt to dial in, * segfaults. |
| 14:42.24 | mafoo | so i need the soundcard/phone card hybrid thing? |
| 14:42.31 | JerJer[mobile] | mafoo: hell no |
| 14:42.41 | eKo1 | You need an FXO card. |
| 14:42.44 | mafoo | yeah |
| 14:42.44 | mafoo | ok |
| 14:42.45 | JerJer[mobile] | cbachman: you are doing something seriously wrong then |
| 14:42.56 | spackle | cbachman, you probably need to do some stuff in modules.conf to dump some of the flotsam and jetsom. |
| 14:42.57 | drsperm | http://store.yahoo.com/asteriskpbx/noname.html |
| 14:43.03 | mafoo | cheers :) |
| 14:43.13 | `Sauron | funbags: Works for me right now |
| 14:43.32 | spackle | Bet JerJer could give us some hints on OpenWRT |
| 14:43.47 | cbachman | JerJer, yup, the client sends the Invite, and * dies either attempting or just after sending the "Trying" I haven't had a chance to check the client logs yet. |
| 14:43.48 | drsperm | spackle: which ver? |
| 14:43.52 | Funbags | its weird ... i see a sip ack from bv but asteriisk doesnt grab it |
| 14:44.06 | JerJer[mobile] | spackle: i've rolled my own |
| 14:44.11 | *** join/#asterisk viLeR (1000@ip-33-7.telesat.com.co) |
| 14:44.19 | `Sauron | you talking about from * -> BV, or from BV to *? |
| 14:44.23 | cbachman | spackle, actually, I am probably "missing" a module. I have it trimmed down to very very little. |
| 14:44.25 | spackle | JeJer, did you get voicemail working on yours? |
| 14:44.33 | Funbags | bv to * |
| 14:44.37 | JerJer[mobile] | hell no |
| 14:44.38 | `Sauron | hum, odd |
| 14:44.49 | `Sauron | reload *, so it re-registers? |
| 14:44.49 | JerJer[mobile] | why would i want to run voicemail on flash? |
| 14:44.51 | spackle | JerJer, good, that would scre me. |
| 14:44.53 | Funbags | * to bv works fine |
| 14:44.56 | spackle | er, scare me |
| 14:45.18 | `Sauron | or a sip reload |
| 14:45.53 | Funbags | yea just did |
| 14:45.56 | Funbags | it registers fine |
| 14:46.02 | JerJer[mobile] | if you are missing a module asterisk won't load or will bitch quite loudly |
| 14:46.19 | *** part/#asterisk Pinhole (~nuxi@lin-dsl-static-206-222-194-115.inetnebr.com) |
| 14:46.32 | *** join/#asterisk Pinhole (~nuxi@lin-dsl-static-206-222-194-115.inetnebr.com) |
| 14:48.19 | spackle | JerJer, why would you want to run Asterisk on a wireless router? |
| 14:48.22 | drsperm | I have a question if ya'll don't mind: If I have a central * server at a core location, where the pri's are located, then have several sites connected to that location via DS1's, can I join a "star" of * servers together? |
| 14:48.32 | cbachman | JerJer, what I've seen is segfaults, at least on the openwrt versions, although I can't remember which module I was missing which made it do that. |
| 14:48.52 | drsperm | So if the DS1's go down, each site can at least do interoffice comm? |
| 14:49.21 | JerJer[mobile] | missing a module wont' cause a seg |
| 14:49.23 | spackle | cbachman, my modules.conf is on pastebin: http://pastebin.ca/7015 |
| 14:49.49 | JerJer[mobile] | spackle: cheap CPE device |
| 14:50.04 | JerJer[mobile] | geek factor |
| 14:50.09 | JerJer[mobile] | cuz i can? |
| 14:50.16 | JerJer[mobile] | pick one |
| 14:50.28 | *** part/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
| 14:50.38 | cbachman | ah, it does actually send the Trying.... |
| 14:50.48 | spackle | JerJer, exactly, a cheap CPE device that was a pbx-in-a-can would be a neat coup don't you think? |
| 14:51.19 | JerJer[mobile] | cheap cpe! |
| 14:51.22 | cbachman | I'm with JerJer, CompUSA has a sale currently on the motorola's. They're less than $10 after the $60 or so in rebates. (high rebates, but they showed up fast for me on the last couple I bought) |
| 14:51.42 | JerJer[mobile] | motorolla's? |
| 14:52.23 | cbachman | JerJer, the wr850g is basically a linksys g model in terms of flash/ram. |
| 14:52.42 | JerJer[mobile] | but not based on the broadcom chipset |
| 14:52.48 | JerJer[mobile] | so openwrt won't cut it |
| 14:52.56 | cbachman | spackle, what modules do you have in /usr/lib/asterisk/modules? That modules.conf will load them all. |
| 14:53.16 | cbachman | JerJer, yes, that particular module (and it's gp twin) are both broadcom inside. |
| 14:54.24 | cbachman | In fact, they're so much the same that the ejtag software from sveasoft works with them. The only tricky part was the initial install (you need to have serial access, as documented on openwrt) |
| 14:55.06 | JerJer[mobile] | http://pastebin.ca/7017 |
| 14:55.14 | spackle | cbachman, here you go: http://pastebin.ca/7016 |
| 14:55.35 | JerJer[mobile] | wholy bloat batman |
| 14:55.54 | spackle | I had to make sure I identified the right IP addresses in the sip.conf, and IAX.conf, and maybe one more. |
| 14:56.14 | JerJer[mobile] | a make config option is sounding better and better everyday |
| 14:56.41 | cbachman | spackle, you probably don't need all of those. |
| 14:57.08 | JerJer[mobile] | um not probably |
| 14:57.16 | spackle | I know I don't. I was going to strat trimming it last night but it was 2:00 AM so I decided it was a project for another time. |
| 14:57.22 | JerJer[mobile] | more like most certainly don't |
| 14:57.44 | cbachman | JerJer, thanks, you have a few things that I left out, but I'll add them in and see if perhaps that helps me. |
| 14:58.44 | spackle | jerjer what are the modules that lead with pbx_ ? |
| 14:58.57 | *** join/#asterisk brettnem (~brettnem@user-0ccsr2l.cable.mindspring.com) |
| 14:59.05 | brettnem | good morning |
| 14:59.28 | cbachman | spackle, you need at least pbx_config if you are using extensions.conf |
| 14:59.29 | brettnem | hey anyone know a good program to do flow charting in? |
| 14:59.34 | drsperm | dia |
| 14:59.36 | brettnem | I'd like to make flow charts of my call flows |
| 14:59.38 | brettnem | ek |
| 14:59.42 | brettnem | I tried it in dia |
| 14:59.56 | brettnem | know of any that can script it?? :) |
| 15:00.48 | *** join/#asterisk X-Gen (~x-gen@rrba-146-120-223.telkomadsl.co.za) |
| 15:00.58 | Hmmhesays | dynamic flow charts based on the current calls er wat? |
| 15:01.10 | brettnem | nah.. well, that *would* be cool.. |
| 15:01.28 | brettnem | just a simple scripting.. if then else kinda language to a flow chart |
| 15:02.04 | Hmmhesays | umm.... ms paint? |
| 15:02.10 | bjohnson | what do the three columns of numbers in a traceroute represent? One must be the hop time but what are the others? |
| 15:02.13 | Darwin35 | ok that was fun I just loaded 6 freebsd boxes at once |
| 15:02.31 | X-Gen | Darwin35: u use ghost ? |
| 15:02.37 | brettnem | hmmm maybe graphvis! |
| 15:02.49 | *** join/#asterisk oej (~oej@63.83.135.35) |
| 15:02.56 | Darwin35 | no I have a hd cloner |
| 15:03.05 | Darwin35 | it clones unix drives |
| 15:03.08 | Hmmhesays | <chuckle>@brettnem |
| 15:03.10 | Darwin35 | its all hardware |
| 15:03.11 | spackle | 6 in one blow, ladies and gentlemen, we haveour new giant killer. |
| 15:03.28 | Darwin35 | it can do upto 20 drives at a time |
| 15:04.09 | Darwin35 | now I am loading asterisk on all them next |
| 15:04.18 | X-Gen | wow, u get mobo's with 20 SATA connectors on them |
| 15:04.27 | brettnem | Hmmhesays: why chuckling?? ;) |
| 15:04.44 | brettnem | X-Gen: I have lots of probs with SATA support.. but maybe that's my kernel. |
| 15:04.46 | Darwin35 | no this is a hardware unit made by a company |
| 15:04.47 | Hmmhesays | i'm really crying inside |
| 15:04.50 | brettnem | heh |
| 15:05.00 | Darwin35 | Drive x I think is the name |
| 15:05.04 | Hmmhesays | like a sad clown |
| 15:05.29 | Darwin35 | they make ide and scsi clone units |
| 15:05.48 | Darwin35 | we use both here bu I used a ide one today |
| 15:05.57 | *** join/#asterisk MuppetMaster (~muppetmas@a82-92-73-185.adsl.xs4all.nl) |
| 15:06.53 | Juggie | brettnem, SATA works here. |
| 15:07.13 | brettnem | I think libata may be interfering with my wct4xxp stuf |
| 15:07.25 | brettnem | I get noise on my calls sometimes.. |
| 15:07.45 | `Sauron | brettnem: Sorry, I farted. |
| 15:08.04 | brettnem | eek. I'm not THAT close |
| 15:08.06 | spackle | jerjer, you are loading chan_zap in modules.conf on WRT, do you have hardware for it? |
| 15:08.16 | Juggie | brettnem, check zttool for irqmisses |
| 15:08.26 | Juggie | chances are your board is sharing an irq with something else |
| 15:08.52 | brettnem | Juggie: it's not |
| 15:08.55 | brettnem | checked it |
| 15:09.21 | brettnem | however my eth card is sharing with libata.. which is why I'm suspicious |
| 15:09.32 | brettnem | using onboard e1000 and e100 too... |
| 15:09.40 | brettnem | it's sporatic too.. |
| 15:09.42 | Juggie | force the network to another irq |
| 15:09.50 | brettnem | Juggie: just how do you do that? |
| 15:09.59 | brettnem | like module parameters? |
| 15:10.29 | Juggie | bios |
| 15:10.40 | brettnem | bonus.. I think graphvis will do it: http://www.graphviz.org/Gallery/directed/datastruct.html |
| 15:10.51 | brettnem | hmm I don't remember that being a bios option |
| 15:10.58 | Juggie | ir get acpi working so u have more then 15 irq's |
| 15:11.10 | Juggie | but best way is to get into bios and start forcing irq's. |
| 15:11.32 | Juggie | when i setup a box not long ago, it put video,ethernet,hdd & zaptel on irq9 |
| 15:11.52 | brettnem | Juggie: I get noise when text scrolls in the CLI.. sounds like framebuffer eh? |
| 15:12.15 | brettnem | I don't 'think' I'm running any kind of FB.. <shrug> doesn't *look* like it.. |
| 15:12.18 | spackle | brettnem, forgive me for asking, but you aren't running X are you? |
| 15:12.24 | Juggie | no i dont run X |
| 15:12.30 | brettnem | nah.. not going over 3 |
| 15:12.33 | dfunnell | Can anyone help with a CAPI/ISDN problem I am having? Getting desperate! |
| 15:12.39 | eKo1 | Question: How can I use an E1 to receive calls on one number? |
| 15:12.42 | brettnem | dfunnell: you still up?! |
| 15:13.15 | dfunnell | brettnem: Gidday, have been to bed and up again. |
| 15:13.16 | brettnem | any other recommendations for noise.. ACPI is an interesting idea.. thats usually in bios right? |
| 15:13.17 | *** join/#asterisk afe ([hUwiRXKBt@c-e616e055.123-1-64736c12.cust.bredbandsbolaget.se) |
| 15:13.24 | cbachman | JerJer, thanks! It seems I was definitely missing some modules. You're right in that * should complain loudly, but perhaps the uclibc is missing things to do that. |
| 15:13.28 | Trepalium | APIC would be more useful than ACPI. |
| 15:13.29 | brettnem | dfunnell: ah. heh, same here |
| 15:13.48 | Trepalium | ACPI is mostly just power manglement. APIC would get you more interrupts. |
| 15:13.53 | brettnem | haha |
| 15:13.54 | Juggie | brettnem, check all your irq's |
| 15:14.02 | dfunnell | brettnem: Couldn't wait to get online! |
| 15:14.04 | brettnem | yeah yeah.. |
| 15:14.05 | junky[work] | http://pastebin.ca/7019 |
| 15:14.07 | spackle | cbachman, I plugged in JerJer's modules.conf and * starts fast now. Thanks JerJer |
| 15:14.09 | X-Gen | what would AIPC get u ? |
| 15:14.16 | Trepalium | IRQ 22. |
| 15:14.17 | brettnem | I didn't mean ACPI |
| 15:14.41 | boch | modulus_ hi |
| 15:15.05 | brettnem | doh |
| 15:15.17 | brettnem | I do have APIC on.. |
| 15:15.42 | *** join/#asterisk olivier_ (~olivier_@82.127.99.32) |
| 15:15.53 | brettnem | I've tried multiple hardware platforms, T1 cards..kernels.. distros.. |
| 15:16.08 | brettnem | bout to point my finger upstream.. lookout! |
| 15:16.21 | *** join/#asterisk freat (~freat@node-40242662.mdw.onnet.us.uu.net) |
| 15:16.24 | brettnem | gasp... graphvis is da bomb |
| 15:16.24 | freat | hello!!! |
| 15:16.43 | freat | company is having the first big conference call on asterisk... woo hoo quality is great! |
| 15:17.03 | brettnem | freat: have you used ICES? |
| 15:17.08 | cbachman | for future reference, it appears that res_features is the one I was missing. |
| 15:17.13 | freat | ?? |
| 15:17.16 | brettnem | stream it to the internet.. have them hook into it with winamp |
| 15:17.26 | freat | oooh nice! |
| 15:17.32 | brettnem | record it and email a transcript to the participants |
| 15:17.33 | Trepalium | heh. I should turn on APIC on my system too... |
| 15:17.36 | Trepalium | 11: 71644019 XT-PIC uhci_hcd, eth0, i915@pci:0000:00:02.0, wcfxo |
| 15:17.51 | freat | brettnem: is it a plugin for * ? |
| 15:18.12 | brettnem | I'm trying to write a manager daemon that will monitor for "Talking" activity in conference bridges and IM participants to let them know who is talking |
| 15:18.19 | brettnem | freat: show application ices I thnk |
| 15:18.26 | tzanger | whee |
| 15:18.32 | spackle | cbachman, don't know if you noticed, but jerjer doesn't load chan_sip either. |
| 15:18.42 | brettnem | you have to use a call file and have ices join the conf... should work nicely |
| 15:18.42 | tzanger | asterisk in a screen session will eventually block and stop all calls if you're in scrollback :-) |
| 15:19.02 | brettnem | tzanger: what's the cmd for scrollback in screen? I can't ever remember |
| 15:19.11 | tzanger | ^A[ |
| 15:19.21 | brettnem | thks |
| 15:19.24 | tzanger | I have mine set up that pageup/down works though |
| 15:19.31 | brettnem | keymap? |
| 15:19.31 | tzanger | er shift-pageup/down |
| 15:19.34 | tzanger | no |
| 15:19.45 | *** join/#asterisk loud (~ariel@null0.flapping.net) |
| 15:19.46 | tzanger | termcapinfo xterm ti@:te@ |
| 15:19.52 | tzanger | in .screenrc |
| 15:20.07 | tzanger | so technically yes, a keymap :-) |
| 15:20.23 | brettnem | cool... scrollback always gets me in screen |
| 15:20.37 | brettnem | so how do I determine if I'm using a FB. anyone know? |
| 15:20.44 | brettnem | er.. framebuffer |
| 15:21.04 | brettnem | cause my call noise issue really seems linked to CLI activity.. weird.. |
| 15:21.17 | brettnem | CLI accessed via ssh.. yadda yadda |
| 15:21.22 | sivana | brettnem: did you sleep at all yet? :) |
| 15:21.37 | brettnem | sivana: 4.5 hours. :) |
| 15:21.49 | tzanger | brettnem: is your main screen anything other than 80x25? |
| 15:21.52 | sivana | hehe |
| 15:22.09 | brettnem | tzanger: I don't know what's running on the console.. it's locked far far away.. can't remember.. |
| 15:22.13 | Trepalium | fbset -i should tell you. |
| 15:22.31 | brettnem | no such device.. that seems good, right? |
| 15:22.46 | Trepalium | Should be fine, then. |
| 15:22.56 | brettnem | sigh |
| 15:23.05 | brettnem | I wish I knew what caused these audio issues.. |
| 15:23.19 | cbachman | spackle, yes, but I'm interested in doing sip->IAX or sip->sip |
| 15:23.39 | brettnem | I'm not even sure if it's happening in asterisk.. may be "upstream".. |
| 15:23.47 | brettnem | anyone think RTCP could assist in debugging this? |
| 15:24.15 | spackle | cbachman, me to, I added chan_sip back in. I think JerJer really likes IAX. |
| 15:24.47 | brettnem | ok damnit.. who switched my left and right mouse buttons.. that just isn't right |
| 15:24.57 | spackle | cbachman, I can't figure out how he is managing to load chan_zap. Did it work for you? |
| 15:25.30 | *** join/#asterisk scrubb (~scrubb@OCI-19-41.onecall.net) |
| 15:26.07 | cbachman | spackle, I do as well, but for internal LAN usage sip has it's advantages, plus it's useful to be able to play with different options. |
| 15:27.02 | cbachman | spackle, I "believe" he's probably using it for timing, and it's his own. |
| 15:27.13 | *** part/#asterisk slav_jb (~k@pirus.securax.be) |
| 15:27.13 | spackle | cbachman, SIP is a bit easier to find. |
| 15:27.33 | cbachman | spackle, I have a spa-3000, so yes, definitely. |
| 15:27.56 | spackle | cbachman, so probably no hardware behind chan_zap? |
| 15:28.37 | cbachman | spackle, likely not. Although, some of the broadcom options have USB In them. Hmmm... I need to take a look at the dmesg. |
| 15:32.44 | spackle | cbachman, you need zaptel timing to do IAX trunking, I bet that's why he loads it |
| 15:33.48 | cbachman | spackle, yes, its why one of my machines has an X100P in it. |
| 15:36.54 | *** join/#asterisk pcm (~pcm@user-69-73-0-22.knology.net) |
| 15:36.54 | *** join/#asterisk Aviator (~ask@ip-65-111-77-10.customer.accelacom.net) |
| 15:37.40 | *** join/#asterisk JerJer[mobile] (~jj@feth100-fw.fament.net) |
| 15:40.13 | *** join/#asterisk carbon60 (~adam@gw.techsupport.ca) |
| 15:40.23 | carbon60 | Morning all. |
| 15:40.49 | carbon60 | Is there a simple way to restrict the number of simultaneous calls going to a SIP peer? |
| 15:40.50 | Goshen | Morning |
| 15:41.16 | *** join/#asterisk lung (~lung@24-148-96-186.ip.mhcable.com) |
| 15:42.30 | *** join/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
| 15:42.55 | MasterYoda | carbon60: you could use incoming and outgoing limit |
| 15:43.08 | carbon60 | MasterYoda: heh, just reading about that. |
| 15:43.11 | junky[work] | running asterisk -g (for valgrind) have any disavantages on performances? |
| 15:43.15 | MasterYoda | carbon60: or you could use setgroup and checkgroup, which I think are asterisk apps |
| 15:43.27 | *** join/#asterisk AgiNamu (~AgiNamu@216.230.151.230) |
| 15:43.29 | GMsoft | hey guys |
| 15:43.33 | MasterYoda | junky[work]: -g just makes asterisk core dump on crash |
| 15:43.35 | *** part/#asterisk X-Gen (~x-gen@rrba-146-120-223.telkomadsl.co.za) |
| 15:43.36 | AgiNamu | Do voicemail boxes have to be purely numeric? |
| 15:43.37 | GMsoft | how can I see which codec is used by a channel ? |
| 15:43.45 | GMsoft | I mean a VoIP channel of course :) |
| 15:43.54 | carbon60 | If a call goes to a SIP extension, then is transfered by that extension. Does Asterisk know the extension is now free? |
| 15:43.56 | Zeeek | Agi funny I just asked that a couple of days ago |
| 15:44.00 | MuppetMaster | GMSoft: From within an AGI, or at the CLI? |
| 15:44.00 | MasterYoda | GMsoft: sip show channels might tell you |
| 15:44.06 | junky[work] | MasterYoda: ya, but it has to deal with all memory traces, no ? any disavantages of that? |
| 15:44.08 | MasterYoda | GMsoft: not sure about with iax or mgcp channels |
| 15:44.09 | GMsoft | MuppetMaster: CLI |
| 15:44.21 | MuppetMaster | GMSoft: show channels |
| 15:44.25 | eKo1 | Say I have an e1 and I receive a call on one channel. If that channel is busy, can I make it use another, hopefully free, channel? |
| 15:44.27 | MuppetMaster | GMSoft: While the call is active. |
| 15:44.36 | AgiNamu | Zeeek: so what'd you find out? :) |
| 15:44.38 | MasterYoda | it's my cousin MuppetMaster |
| 15:44.47 | GMsoft | MuppetMaster: sip show channels does |
| 15:44.55 | GMsoft | but show channels doesnt |
| 15:45.00 | MuppetMaster | GMSoft: that is what I meant... ;) |
| 15:45.01 | Zeeek | nothing but I'd guess it's work except for dialing in with the cid as box name |
| 15:45.04 | MasterYoda | junky[work]: well the core dump is not actually done by asterisk, it is done by your operating system |
| 15:45.07 | *** part/#asterisk wizard2000 (~dang@fubar.arcbox.com) |
| 15:45.14 | GMsoft | MuppetMaster: ah k :) |
| 15:45.23 | GMsoft | MuppetMaster: no other way to see it ? |
| 15:45.39 | MasterYoda | eKo1: uhh, maybe, talk to your telco |
| 15:45.39 | *** join/#asterisk Moarth (~Moarth__@65.38.211.195) |
| 15:45.48 | AgiNamu | cause they prepend an 's' to the voicemailbox if it is to skip the password |
| 15:45.51 | MasterYoda | eKo1: they should be able to send you a call on the first available channel |
| 15:45.53 | AgiNamu | Which is a stupid-freaking-idea |
| 15:46.00 | AgiNamu | I mean, how hard is it to pass that as an argument? |
| 15:46.15 | AgiNamu | the lack of basic design insight sometimes is astounding :P |
| 15:46.22 | MuppetMaster | GMSoft: You can see it from within an AGI, using something like the phpagi.sourceforge.net classes. |
| 15:46.31 | MuppetMaster | GMSoft: But from the CLI I do not know of another way. |
| 15:46.42 | MasterYoda | AgiNamu: yeah voicemail could use a redisign, but why do you want alpha numeric mailboxes? |
| 15:46.43 | Zeeek | Agi organic design like a web site |
| 15:47.03 | Zeeek | shit just keeps getting added |
| 15:47.08 | GMsoft | MuppetMaster: ok thanks. I'll RTFM a bit :0- |
| 15:47.08 | Zeeek | which is good |
| 15:47.09 | Moarth | can a litter 8 port NeXXT switcher sopport a asterisk server and 37 stations? |
| 15:47.11 | GMsoft | :) |
| 15:47.29 | AgiNamu | MasterYoda, because my account codes are alphanumeric. |
| 15:47.41 | Zeeek | Agi I'd say try it and see |
| 15:47.49 | AgiNamu | Zeeek: good? as compared to having a solid design and specs? :) |
| 15:47.54 | freat | ahh |
| 15:48.01 | AgiNamu | seems more like organic hacking :P |
| 15:48.03 | Zeeek | it would make it hard to answer the prompt "box number?" |
| 15:48.18 | Moarth | Can any one help me on this? |
| 15:48.24 | MasterYoda | AgiNamu: I see, so just figure out what numbers corrispond the letters and your problem is solved |
| 15:48.28 | Zeeek | rephrase the question? |
| 15:48.35 | AgiNamu | no, my format is AAAXXXXX |
| 15:48.36 | *** join/#asterisk _omer (dfsdf@202.147.174.177) |
| 15:48.42 | AgiNamu | So I can prepend the 3-letter alphanumeric prefix |
| 15:49.05 | Zeeek | I'd guess it doesn't matter since everything is a string anyway |
| 15:49.12 | AgiNamu | yea |
| 15:49.19 | Moarth | i have 37 clients for asterisk connected to the server throughout a small 8 pot switcher.. can this switcher do the work... |
| 15:49.24 | Moarth | ? |
| 15:49.24 | Zeeek | unless you need to loop through the vmail boxes :) |
| 15:49.29 | AgiNamu | OK, here's a harder one... I think. How do you bill call forwarding? |
| 15:49.49 | Zeeek | a penny a time? |
| 15:49.50 | AgiNamu | I have an incoming call, so I have the CDR for it, and it's good. Then, the user is not available, so I call forward |
| 15:49.55 | MasterYoda | AgiNamu: or you could figure out what the 3 letters are as numbers on the phone, you could even write an agi script to make a channel var ${MAILBOX} to do it for you |
| 15:50.18 | AgiNamu | MasterYoda, yea, but people don't actually know their account codes |
| 15:50.45 | MasterYoda | AgiNamu: so, I still don't see the problem |
| 15:50.55 | MasterYoda | AgiNamu: where would you get their account code in the first place? |
| 15:51.01 | AgiNamu | Well, what if the account code starts with an S? |
| 15:51.11 | MasterYoda | AgiNamu: then you convert that s into a 7 |
| 15:51.24 | Zeeek | the mailbox is whatever is in mailbox= |
| 15:52.05 | AgiNamu | It switches on s and p |
| 15:52.19 | Zeeek | pass it s${account} |
| 15:52.20 | AgiNamu | i mean, switches on the extension passed in |
| 15:52.31 | AgiNamu | And if I don't want to skip the password? :) |
| 15:52.32 | Zeeek | should work |
| 15:52.40 | Zeeek | it won't |
| 15:52.48 | AgiNamu | why won't it? |
| 15:52.53 | Zeeek | why would it? |
| 15:52.58 | AgiNamu | The code does switch(*ext) { case 's'.... |
| 15:53.39 | Zeeek | oh I dsee now |
| 15:54.24 | AgiNamu | i can work around it. just wanted to note that it's a stupid design for absolutely no reason other than laziness/lack of design :) |
| 15:54.44 | AgiNamu | I'll probably take the account prefix and use that as a voicemai lcontext |
| 15:54.51 | AgiNamu | seems like that'll work. |
| 15:54.55 | *** join/#asterisk yaout (eric@CPE-65-30-220-56.wi.rr.com) |
| 15:55.56 | AgiNamu | so any thouights on how to get a second CDR? |
| 15:56.11 | JerJer[mobile] | register a second cdr backend |
| 15:56.25 | AgiNamu | no, I mean, one for the incoming call, and one for the outgoing call forwarding |
| 15:56.30 | boch | hey guys, is there a way to test the * stability? something like generate a large amount of calls? |
| 15:56.33 | JerJer[mobile] | that's not how it works |
| 15:56.34 | AgiNamu | because I need to bill the incoming call + the outgoing call |
| 15:56.35 | JerJer[mobile] | it is one call |
| 15:56.39 | JerJer[mobile] | then do that |
| 15:56.55 | *** join/#asterisk Goldenear (~goldenear@d193.dhcp212-198-200.noos.fr) |
| 15:57.04 | Goshen | boch: have us all call at one time? :) |
| 15:57.05 | JerJer[mobile] | we bill both ingress and egress off of one CDR |
| 15:57.10 | Beirdo | Hmm |
| 15:57.34 | boch | Goshen what? |
| 15:57.42 | JerJer[mobile] | for example someone has a toll-free number but wants it delivered to a PSTN number |
| 15:57.43 | Beirdo | is it a known issue that GSM calls to the VoiceMail app is borked, or do I have some pebcak issue here? |
| 15:57.49 | JerJer[mobile] | both legs of that call are billable |
| 15:57.56 | Goshen | "[08:56] <boch> hey guys, is there a way to test the * stability?" |
| 15:58.01 | boch | Goshen i mean, like hundreds of calls |
| 15:58.18 | AgiNamu | JerJer, how do you figure that out ? It's just one CDR |
| 15:58.26 | Goshen | there are over 200 people here, but I doubt you could get 1/10 to wake up :) |
| 15:58.38 | boch | Godsey lol, thats the problem |
| 15:58.40 | Zeeek | Goshen set off a firecracker |
| 15:58.46 | *** join/#asterisk mager (~noyb@ua-83-227-134-28.cust.bredbandsbolaget.se) |
| 15:59.06 | spackle | boch, search on astertest |
| 15:59.21 | AgiNamu | And the destination number isn't going to be right |
| 15:59.24 | boch | spackle ok, thanks |
| 15:59.37 | Goshen | Just got off the phone with broadvoice, they can use number portability now...but they are servicing their current clients first he said |
| 15:59.38 | AgiNamu | The first destination is the DID dialed (since some DIDs are billed differently) |
| 15:59.45 | AgiNamu | The second destination is the PSTN forward number |
| 15:59.51 | Goshen | which is 40-60 days for the LMP to kick in |
| 16:00.09 | *** join/#asterisk harryvv (~plato@S010600055d210201.vs.shawcable.net) |
| 16:00.22 | Goldenear | I've a simple (may be dumb) question : Why doesn't SIP doesn't always use the same UDP port for RTP ? what are the benfits of dynamic ports ? |
| 16:00.23 | Goshen | is Broadvoice sip only? |
| 16:00.38 | scrubb | yes |
| 16:00.53 | `Sauron | Yep |
| 16:01.04 | Goshen | I hate sip |
| 16:01.07 | boch | spackle astertest its all under construction (? |
| 16:01.10 | `Sauron | I think they run *, though.. so you'd think they could do iax |
| 16:01.43 | Goshen | Sauron: you would think |
| 16:01.58 | Goshen | perhaps is enough people request it |
| 16:02.11 | `Sauron | Sigh. |
| 16:02.12 | carbon60 | Anyone know how to disable "call waiting" or at least the tone, in Polycom SoundPoint IP500s? |
| 16:02.20 | `Sauron | All I want from them, is to have my number ported over. |
| 16:02.32 | JerJer[mobile] | Goldenear: because the committee decided it was better to randomly select a port |
| 16:02.43 | JerJer[mobile] | just to puck with edge deivces |
| 16:03.05 | JerJer[mobile] | `Sauron: broadvoice runs SER |
| 16:03.11 | Goldenear | JerJer[mobile]: edge devices ? |
| 16:03.12 | JerJer[mobile] | and TNT gateways |
| 16:03.14 | Goshen | Sauron: did you file your phone bill and sign the agreement with them yet? |
| 16:03.15 | `Sauron | hum |
| 16:03.17 | `Sauron | fair enough |
| 16:03.29 | Goshen | SER? |
| 16:03.40 | `Sauron | sip express router? |
| 16:04.04 | MuppetMaster | Sauron: Yes. |
| 16:04.35 | Goshen | `Sauron: http://www.broadvoice.com/support_faqs.html go to the portability section, they have a form to fill out and instructions for porting your number |
| 16:05.26 | `Sauron | Goshen: I did |
| 16:05.32 | `Sauron | well, I did the fax-in-the-pdf document |
| 16:05.45 | JerJer[mobile] | now u get to wait |
| 16:05.58 | JerJer[mobile] | until your telco decides they are ready to release your number |
| 16:06.13 | `Sauron | Yup |
| 16:06.22 | Goshen | JerJer: I wish you could do Salt Lake....I prefer IAX :) |
| 16:06.25 | JerJer[mobile] | make damn sure you do not cancel service with your current telco |
| 16:06.33 | `Sauron | I haven't |
| 16:06.40 | `Sauron | I've gone through the porting dance before |
| 16:06.42 | JerJer[mobile] | Goshen: give me a couple million and we'll become a CLEC out there |
| 16:06.46 | file | JerJer[mobile]: guess who got your spot at the pavilion |
| 16:06.50 | JerJer[mobile] | simple as that |
| 16:06.52 | `Sauron | I have an ATT number on my tmobile cellphone... ;) |
| 16:06.55 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 16:06.57 | JerJer[mobile] | file: gotocall ? |
| 16:07.16 | file | JerJer[mobile]: really? I thought it was switchvox, did somebody else drop out too? |
| 16:07.39 | JerJer[mobile] | duno |
| 16:07.49 | spackle | JerJer how long do you think it would take for you to make back "a couple million" for an investor? |
| 16:07.51 | file | they're really nice people though |
| 16:08.01 | Moc____ | hi all |
| 16:08.08 | JerJer[mobile] | sounded like u knew who it was |
| 16:08.12 | *** join/#asterisk pif (~pif@zenon.apartia.fr) |
| 16:08.24 | Goldenear | JerJer[mobile]: that's a weird decision from the committee... wouldn't always using port 5060 for SIP/signalisation and lets say port 8000 for RTP/voice be better ? (no more NAT issue) |
| 16:08.26 | `Sauron | only took att 4 hours to release my number |
| 16:08.37 | `Sauron | I could imagine SBC never wanting to release my home number |
| 16:08.38 | `Sauron | Grrr. |
| 16:08.40 | JerJer[mobile] | Goldenear: go complain to the IETF then |
| 16:09.05 | `Sauron | they kept trying ot tell my boss he couldn't move his number down the street to their new house, which is still served by the same CO |
| 16:09.30 | *** join/#asterisk inspired (mikael@213.197.167.61) |
| 16:09.45 | *** join/#asterisk Defraz (~t0tal@sonicwall.dcdi.net) |
| 16:10.06 | *** join/#asterisk GiabboO (~GiabboOo@host101-246.pool8173.interbusiness.it) |
| 16:10.20 | GiabboO | 1.0.5 on gentoo ebuild :) |
| 16:10.25 | *** join/#asterisk heison (~heison@w3.somanetworks.com) |
| 16:10.25 | GiabboO | masked as well lol |
| 16:10.26 | GiabboO | hi all |
| 16:10.43 | JerJer[mobile] | spackle: figure 5 to 7 year return on investment |
| 16:10.57 | JerJer[mobile] | maybe less if there is high demand |
| 16:11.25 | spackle | JerJer, do you see it slowing down? I don't think there is anything to suggest it at this point do you? |
| 16:11.25 | Poincare | Is there a quality problem with the Digium electronics? |
| 16:11.36 | Poincare | Got a 4th DOA... |
| 16:11.42 | tzanger | Poincare: all my digium equipment seems to wrk |
| 16:11.47 | tzanger | what do you mean by doa, exactly |
| 16:11.58 | JerJer[mobile] | Poincare: i find that very very very very very very very hard to believe |
| 16:12.09 | `Sauron | dead on arrival |
| 16:12.12 | tzanger | `Sauron: no shit |
| 16:12.13 | Poincare | Well, it's hard to believe for me to... |
| 16:12.14 | tzanger | I mean what kind of doa |
| 16:12.19 | spackle | Poincare, Ouch! Mine was shipped loosely in a fedex box, how did yours arrive? |
| 16:12.25 | Poincare | Got those 4 in 2 months.... |
| 16:12.33 | JerJer[mobile] | smells like you are running crap hardware |
| 16:12.36 | *** join/#asterisk intrin (intrin@c68.112.146.203.stc.mn.charter.com) |
| 16:12.39 | Poincare | The year before that no problems |
| 16:12.39 | JerJer[mobile] | ie motherboard |
| 16:12.59 | intrin | does nufone.net provide any eu numbers? |
| 16:13.01 | intrin | like uk |
| 16:13.17 | Poincare | Fine, then I got 4 crappy motherbords? |
| 16:13.35 | JerJer[mobile] | if they are all the same model, sure |
| 16:13.48 | Poincare | Nope 4 different brands/models |
| 16:13.54 | JerJer[mobile] | intrin: no |
| 16:14.06 | Poincare | BTW, replacement cards don't die (yet) |
| 16:14.15 | Zeeek | JerJer with a British accent... hmmmm |
| 16:14.28 | JerJer[mobile] | tea time anyone? |
| 16:14.37 | hajekd | is it possible to set default CALLERID? Looks like the default is "asterisk" ;-) |
| 16:14.37 | Zeeek | A little past, here actually |
| 16:14.40 | JerJer[mobile] | lol |
| 16:14.44 | intrin | know anywhere i can get one? |
| 16:14.52 | Zeeek | voiptalk does IAX |
| 16:15.05 | Zeeek | sipgate.co.uk SIP |
| 16:15.17 | *** join/#asterisk topping (~topping@dhcp024-210-082-196.columbus.rr.com) |
| 16:15.18 | Zeeek | both have UK DID |
| 16:15.28 | intrin | ty |
| 16:15.31 | Zeeek | yw |
| 16:15.48 | Zeeek | of course they aren't as goos as nufone |
| 16:16.05 | spackle | what kind of hardware does nufone run on? |
| 16:16.14 | Zeeek | they don't use hardware |
| 16:16.23 | Zeeek | it's all virgins in heaven |
| 16:16.59 | *** join/#asterisk kisu (~yo@218.237.126.167) |
| 16:18.03 | *** join/#asterisk ionix (~ionix@209.71.254.100) |
| 16:18.12 | JerJer[mobile] | spackle: Dell 1750s, TE410Ps, EdgeLink 100 (DS-3 mux), Plexsus 9000 (switch) |
| 16:18.32 | spackle | Do virgins have 3.3 or 5 volt PCI slots? |
| 16:18.38 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlgrv.pa.sed6.net) |
| 16:18.46 | eKo1 | They have both. |
| 16:18.47 | tzanger | what the fuck |
| 16:18.50 | tzanger | LEN is not working |
| 16:19.11 | tzanger | exten => s,n,GotoIf($[{$LEN{$ARG1}} = 10]?s,add1) |
| 16:19.34 | Zeeek | something about the placement of the brackets... |
| 16:19.44 | tzanger | $[$LEN{$ARG1}} = 10] is evaluating false for a 10-digit # |
| 16:19.45 | JerJer[mobile] | spackle: don't specificalyl remember... whatever the TE410P runs.... 5v? |
| 16:19.52 | Zeeek | tzanger |
| 16:20.04 | *** join/#asterisk t3t (~t3t@galley.pangalacticgargleblaster.com) |
| 16:20.14 | Zeeek | {$ARG1} or ${ARG1} ? |
| 16:20.20 | Goshen | intrin: voipuser.org |
| 16:20.20 | ionix | ${} |
| 16:20.22 | tzanger | oh for fuck sakes |
| 16:20.23 | *** join/#asterisk Signuts (~signuts@209.172.11.54) |
| 16:20.26 | Zeeek | yes, |
| 16:20.29 | ionix | is not valid |
| 16:20.33 | Zeeek | you did that in front of all of us |
| 16:20.44 | eKo1 | ha ha |
| 16:21.11 | Signuts | Hey all, got a question about transfering calls, say I want the user to hit the flash button, dial '#90' or something and then eb able to transfer the other line to the number they dial. Surely this is possible. |
| 16:21.26 | Zeeek | the entire channel is roaring with laughter, unprocessed for echo like in those bad horror movies |
| 16:21.34 | file | Signuts: through what hardware, an ATA or Digium TDM card? |
| 16:21.43 | spackle | JerJer, in a word, are you happy with Dell? |
| 16:21.57 | Zeeek | tzanger then has to cover his ears while Pscho like music gets louder over the echoing laughter |
| 16:22.28 | Signuts | file, i'm running Zaptal |
| 16:22.31 | Signuts | Digium |
| 16:22.47 | *** join/#asterisk _omer (dfsdf@202.147.174.177) |
| 16:22.51 | _omer | hello |
| 16:23.00 | Signuts | it's a 4 port T1 card, damn nice I might say =) |
| 16:23.19 | *** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
| 16:23.19 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 16:23.24 | file | Signuts: what are you using for phone I mean... where you want to transfer it |
| 16:23.36 | _omer | anyone....have some exp in Quintum gateways? |
| 16:24.16 | JerJer[mobile] | _omer: yes - selling them on ebay |
| 16:24.19 | JerJer[mobile] | for my customers |
| 16:24.33 | _omer | which series is it? |
| 16:24.35 | JerJer[mobile] | spackle: yes, dell has never let me down |
| 16:24.39 | JerJer[mobile] | _omer: all of them |
| 16:24.44 | JerJer[mobile] | they are all equally worthless |
| 16:24.46 | Signuts | file, oh I've got a regular software phone (sflphone) i'd liek to work with, but all that's important is a sip phone. Sipura 2000 (it's a box to convert RJ11 into VIOP) |
| 16:24.56 | _omer | Great to hear that :D |
| 16:25.01 | tzanger | Zeeek: hahahaha |
| 16:25.07 | bkw_ | ya know what... |
| 16:25.11 | bkw_ | MY FEET HURT |
| 16:25.15 | file | bkw_: so do mine |
| 16:25.16 | bkw_ | my legs HURT |
| 16:25.21 | _omer | JerJer: can you tell me something about AX series.... |
| 16:25.27 | tzanger | bkw_: sit down before you hurt yourself |
| 16:25.27 | file | Signuts: then look up the documentation on it for transferring, asterisk doesn't do it - your sipura does |
| 16:25.29 | JerJer[mobile] | no |
| 16:25.39 | _omer | I was expecting "yes" ;) |
| 16:25.40 | JerJer[mobile] | i just sold a few dozen on flebay |
| 16:25.54 | JerJer[mobile] | never heard of AX series |
| 16:26.16 | _omer | heard about "A" series? |
| 16:26.24 | MikeJ[Jayden] | BKW!.. |
| 16:26.25 | Signuts | ahh, ok. Great |
| 16:26.32 | MikeJ[Jayden] | where's the video.... |
| 16:27.20 | *** join/#asterisk AsteriskAR (~lolo@lineaAH241.velocom.com.ar) |
| 16:29.22 | *** join/#asterisk dfunnell (~dfunnell@port-222-152-55-43.fastadsl.net.nz) |
| 16:29.31 | tzanger | what video |
| 16:29.32 | tzanger | http://www.big-boys.com/articles/bumperhit.html |
| 16:29.34 | tzanger | that one? |
| 16:29.38 | tzanger | karma's a bitch, man |
| 16:29.51 | tzanger | bkw_: where the blue fuck is app_conference? |
| 16:30.10 | MikeJ[Jayden] | tzanger is colorful today :) |
| 16:30.23 | tzanger | MikeJ[Jayden]: you oughtta see what's coming out of my nose |
| 16:30.27 | tzanger | it's colourful too |
| 16:30.36 | MikeJ[Jayden] | nice |
| 16:31.00 | MikeJ[Jayden] | allright, who sepaks german ^^ |
| 16:31.46 | GiabboO | can anyone help me about this warning while loadin h3223 module: Warning, flexible rate not heavily tested! |
| 16:31.53 | GiabboO | h323 sorry |
| 16:31.57 | Beirdo | never discuss with idiots! They pull you on their level down and strike you then with experience |
| 16:32.03 | Beirdo | according to google |
| 16:32.30 | Beirdo | should give you the gist of it |
| 16:33.09 | *** join/#asterisk PakiPenguin (~uppal@202.176.230.225) |
| 16:33.19 | PakiPenguin | Hello Everyone |
| 16:33.51 | JerJer[mobile] | GiabboO: that has nothng to do wth h.323 |
| 16:33.57 | JerJer[mobile] | that is music on hold, i believe |
| 16:34.01 | *** join/#asterisk shaZwaz (~adnans@202.59.71.209) |
| 16:35.02 | *** join/#asterisk RoyK (~roy@host-81-191-147-248.bluecom.no) |
| 16:35.20 | Moc____ | yep, it the music on hold |
| 16:35.31 | Moc____ | wonder why digium mp3 was made like this in the first place |
| 16:35.43 | Moc____ | Anyone from voiceconduits arround / |
| 16:36.16 | *** join/#asterisk JerJer[mobile] (~jj@feth100-fw.fament.net) |
| 16:36.37 | spackle | GiabboO: Try using madplay for musiconhold, it works pretty well. |
| 16:37.09 | GiabboO | oh |
| 16:37.15 | GiabboO | lol |
| 16:37.15 | FaithX | anyone got zaphfc running? |
| 16:37.51 | GiabboO | spackle, madplay instead mpg123 ? |
| 16:38.08 | Darwin35 | madpla |
| 16:38.14 | FaithX | I like piping my favorite radio station from the internet to MOH |
| 16:38.28 | Darwin35 | did you go threw thcod and change mpg123 to mad play |
| 16:38.48 | *** join/#asterisk mogorman (~mogorman@dhcp-162.digium.com) |
| 16:39.06 | Zeeek | what d'you call astdb? Isn't it the database as in show database? |
| 16:52.23 | *** join/#asterisk jbot (ibot@apt.bot.TimRiker.active.supporter.pdpc) |
| 16:52.23 | *** topic/#asterisk is Asterisk: The Open Source PBX || 1.0.6 Released || Dev Conf 1PM CST MARCH 10th -> IAX2/guest@66.250.68.194/996 || ClueCon Dev Conf Aug 3rd - 5th |
| 16:53.13 | _omer | everyone I have a question about Quintum..... |
| 16:54.57 | AgiNamu | Is Speex really that aamzing? |
| 16:55.00 | AgiNamu | Compared to say, iLBC? |
| 16:55.05 | AgiNamu | for shitty connections, that is |
| 16:55.16 | *** join/#asterisk ManxPower (~eric@stirprop-s0-0-0-26.ndcr2.datasync.net) |
| 16:56.04 | AgiNamu | my phones and atas are getting iLBC support in a few days |
| 16:56.11 | AgiNamu | and i think Speex is up after that :) |
| 16:56.16 | PakiPenguin | AgiNamu, which one? |
| 16:57.15 | AgiNamu | Paki, PA168s |
| 16:57.16 | cypromis | AgiNamu: speex is smaller |
| 16:57.24 | AgiNamu | but odes it handle packet loss as well? |
| 16:58.06 | cypromis | don't think so |
| 16:58.31 | olivier_ | pri debug span 6 |
| 16:58.36 | olivier_ | oops |
| 17:00.39 | AgiNamu | What's the username in voicemail good for? |
| 17:00.43 | AgiNamu | can I safely leave it out? |
| 17:02.30 | *** join/#asterisk zigman (~zigman@irc.zigman.de) |
| 17:02.41 | bkw_ | SDLKFJWOEFIJSF |
| 17:02.42 | bkw_ | go faster |
| 17:02.52 | bkw_ | cvs checkout is taking AGES |
| 17:03.31 | goatmilk | bkw_: not at VON? |
| 17:04.14 | bkw_ | um yes |
| 17:04.34 | *** mode/#asterisk [+o DrRighteous] by bkw_ |
| 17:04.36 | *** mode/#asterisk [+o DrRighteous] by bkw_ |
| 17:04.40 | tzanger | bkw_: you use app_conference for the dev conf? |
| 17:04.50 | bkw_ | tzanger, not yet |
| 17:04.54 | tzanger | bkw_: still meetme? |
| 17:06.08 | tzanger | bkw_: what's your exten => 996 look like for calling up meetme? |
| 17:06.40 | Goshen | bkw_: I was curious about that too |
| 17:07.00 | Darwin35 | if madplayer works with * we should remove mpg123 and use madplaayer |
| 17:08.19 | tzanger | bkw_: I imagine ,idMxs |
| 17:08.29 | gr8nash | i tried the patch from BV and it didnt cleanly apply.. 2 files patched 5 didnt.. so my source is a tad scewd =) |
| 17:08.41 | gr8nash | as a newbie should i stay clear of cvs.. |
| 17:08.43 | gr8nash | ? |
| 17:08.52 | tzanger | gr8nash: for the most part yes |
| 17:09.01 | tzanger | use 1.0.6, get to love it and then embrace HEAD |
| 17:09.06 | gr8nash | i use cvs all the time with myth..but im a astrix newbie.. ok thanks |
| 17:09.36 | Goshen | gr8nash: let us know if 1.0.6 works with BV without the patch |
| 17:09.42 | `Sauron | ;) |
| 17:09.43 | gr8nash | ok will do |
| 17:09.48 | gr8nash | ill talk back in a few |
| 17:09.48 | `Sauron | goshen: It should |
| 17:10.00 | `Sauron | I'm using head (a few days behind) and can place/receive calls |
| 17:10.31 | Darwin35 | the patch is in 5 |
| 17:10.31 | Darwin35 | I just checked the files manual |
| 17:10.32 | `Sauron | if it's the ancient patch, then it's been in since .5 was -head |
| 17:10.44 | tzanger | I'm still not sure I understand the benefit of opening an empty (and optionally PINless) conference |
| 17:10.49 | `Sauron | but there might be a new patch, cuz they changed the sip invide auth thing |
| 17:11.21 | `Sauron | Goshen: you with BV? |
| 17:11.23 | spackle | I'm sorry, I was away, did you get your madplay questions answered? |
| 17:11.33 | Darwin35 | no |
| 17:11.36 | Goshen | Sauron: no, but I am looking at them becuase they can LMP my numbers |
| 17:11.44 | `Sauron | Ah. |
| 17:11.53 | Darwin35 | what did you do in * to get madplayer to work |
| 17:12.00 | *** join/#asterisk RoyK (~roy@host-81-191-147-248.bluecom.no) |
| 17:12.01 | bkw_ | nope |
| 17:12.02 | bkw_ | the one in the topic |
| 17:12.05 | spackle | Darwin, let me post my musiconhold.conf |
| 17:12.06 | Goshen | I don't want to sign up with them till they can, but they are processing their current clients first, which he said is taking 40-60 days |
| 17:12.08 | bkw_ | anthm got his chainsaw out.. and started hacking up meetme.. so now their is like two lines of meetme left |
| 17:12.10 | bkw_ | haha |
| 17:12.13 | `Sauron | wow |
| 17:12.18 | Darwin35 | did you change the src to point to mad player or link madplayer to mpg123 |
| 17:12.20 | bkw_ | |sm |
| 17:12.23 | tzanger | bkw_: what does your meetme exten look like? |
| 17:12.25 | tzanger | just sm? |
| 17:12.39 | `Sauron | I guess I have a few months to wait |
| 17:12.52 | tzanger | ahh so 'm' makes the user muted at start, not perma-muted |
| 17:12.52 | Nugget | I need to create a custom "telemarketers" music on hold queue and fill it up with angry death metal music. |
| 17:13.19 | spackle | Darwin35: here you go: http://pastebin.ca/7022 |
| 17:13.30 | Darwin35 | ok |
| 17:13.38 | `Sauron | Nugget: 24/7 rednex cotton eye joe |
| 17:13.44 | `Sauron | intermixed with some britney spears |
| 17:13.45 | tzanger | haha |
| 17:14.12 | spackle | Personally, I like apololyptica - Mettalica played by 4 cellos |
| 17:14.46 | `Sauron | hum |
| 17:15.07 | `Sauron | FCC mandated that the porting interval be 4 days |
| 17:15.07 | Nugget | oooh, better suggestion. |
| 17:15.20 | `Sauron | I guess BV's swamped with people |
| 17:15.24 | `Sauron | 60 days > 4 days |
| 17:15.29 | Darwin35 | cool |
| 17:16.28 | RoyK | ~seen kram |
| 17:16.33 | jbot | kram <~mark@kram.digium.sponsor.pdpc> was last seen on IRC in channel #asterisk, 3d 13h 39m 57s ago, saying: 'oh :)'. |
| 17:16.37 | Qwell | spackle++ |
| 17:16.49 | RoyK | famous last words.... |
| 17:17.06 | `Sauron | hehe |
| 17:17.16 | `Sauron | next after "Hold my beer, and check this out!" |
| 17:17.28 | tzanger | bkw_: how do you kick the last person entered out of the conf? |
| 17:17.39 | tzanger | I'm reading up on meetme on voipinfo and meetme kick doesn't seem to have a 'lastuser' option |
| 17:18.06 | jedirl | ANyone knows how to use the "silence" parameter of sox? |
| 17:18.10 | jedirl | it's quite cryptic |
| 17:18.43 | ManxPower | ~docs |
| 17:18.44 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 17:18.53 | `Sauron | tzanger: there's the "kick all users" option... ;) |
| 17:19.01 | tzanger | `Sauron: :-) |
| 17:19.11 | Qwell | `Sauron: also known as "hanging up"? |
| 17:19.30 | `Sauron | Qwell: If you hang up, the conference stays, and the other users can use it |
| 17:19.34 | Qwell | oh |
| 17:19.44 | Qwell | not like other conf lines I've used then. |
| 17:19.49 | Qwell | Is there an option to make it like that? |
| 17:20.29 | `Sauron | tzanger: You mean other than doing meetme kick <no> <channel> ? |
| 17:20.43 | tzanger | `Sauron: hmm maybe tha'ts all they're doing |
| 17:20.51 | `Sauron | Usage: meetme (un)lock|(un)mute|kick|list <confno> <usernumber> |
| 17:21.10 | tzanger | maybe a combo of getting the #of users and kicking user# that number |
| 17:21.10 | jedirl | noone knows? |
| 17:21.12 | jedirl | :( |
| 17:21.40 | tzanger | jedirl: I'd use the sox manpage for starters but no I personaly do not know |
| 17:21.58 | jedirl | I've read man sox |
| 17:22.09 | tzanger | so what's the silence prarameter docs say? |
| 17:22.49 | jedirl | it doesn't give any sample, and it doesn't give a complete semantic of each parameter |
| 17:23.07 | jedirl | so you must guess what "thresold", "above" and these are |
| 17:23.49 | *** join/#asterisk jbot (ibot@apt.bot.TimRiker.active.supporter.pdpc) |
| 17:23.49 | *** topic/#asterisk is Asterisk: The Open Source PBX || 1.0.6 Released || Dev Conf 1PM CST MARCH 10th -> IAX2/guest@66.250.68.194/996 || ClueCon Dev Conf Aug 3rd - 5th |
| 17:23.57 | jedirl | then my english is worse than I tought |
| 17:24.17 | tzanger | threshhold is in either dB or % of total audio |
| 17:25.10 | jedirl | silence 1 5 2% |
| 17:25.17 | jedirl | it doesn't remove any silence |
| 17:26.00 | *** join/#asterisk Verliba (~Miranda@250.1-14-84.ripe.coltfrance.com) |
| 17:26.49 | *** join/#asterisk crash3m (crash3m@crash3m.user) |
| 17:29.14 | *** join/#asterisk Ayano (~erik_leee@209.143.187.254) |
| 17:30.11 | *** join/#asterisk t3t (~t3t@galley.pangalacticgargleblaster.com) |
| 17:32.06 | Darwin35 | there is a new patch |
| 17:32.17 | Darwin35 | broadvoicesip2.txt |
| 17:32.27 | Darwin35 | it has to be patched |
| 17:32.33 | Darwin35 | for broadvoice |
| 17:33.37 | Darwin35 | I am fixing the patch now |
| 17:33.45 | Darwin35 | they made a bad ptch |
| 17:33.53 | Zeeek | anyone here use or hear of no-ip.com ? |
| 17:33.59 | Darwin35 | I use them |
| 17:34.04 | ionix | Anyone has an idea how I can use prepaid service with absolutetimeout when there are two or more calls at the same time for the same account ? |
| 17:34.08 | Darwin35 | they are a dynamic dns |
| 17:34.09 | *** join/#asterisk dalabera (~Dalabera@mail.pmrtechnologies.com) |
| 17:34.12 | Darwin35 | works great |
| 17:34.17 | Zeeek | Darwin35 are they better than dyndns? |
| 17:34.30 | Darwin35 | I never used dyndns |
| 17:34.41 | Darwin35 | I found no-ip first and went ith them |
| 17:34.44 | `Sauron | Darwin: Whats the new patch for? |
| 17:34.44 | Zeeek | ok, do they have 5 stable DNS a |
| 17:34.50 | Zeeek | ? |
| 17:34.51 | Darwin35 | have had them for 4 years no problems |
| 17:34.52 | JerJer[mobile] | ionix: that is a huge pain in the ass |
| 17:35.00 | ionix | yeh that's what I figured |
| 17:35.01 | Zeeek | ok Darwin, thx, it was for a customer |
| 17:35.11 | ionix | I want to do conferencing box with 1-800 dids |
| 17:35.24 | ionix | JerJer[mobile]: What approach did you use for Nufone ? |
| 17:35.36 | Darwin35 | the new patch is broadvoicesip2.txt and it patches chan_sip.c |
| 17:36.16 | Goshen | Darwin35: why doesn't Broadvoice just pass patches to asterisk develoupment for addition? |
| 17:36.34 | Goshen | it seems silly to make users patch their own server |
| 17:37.40 | *** join/#asterisk Ash (aaron@fudgecom.net) |
| 17:38.17 | `Sauron | because it's quicker for them to put a patch on a webpage, than beat up on some devloper to get the patch through |
| 17:38.39 | `Sauron | I'm guessing, anyways. |
| 17:38.45 | Goshen | I guess so |
| 17:39.24 | Goshen | Darwin35: are you going to add the patch to the tree then? or just send them back a better patch? |
| 17:40.27 | gr8nash | Hey guys i see i missed something.. i just compiled 1.0.6 havnt tested with BV |
| 17:40.48 | gr8nash | there is a newpatch that i need broadvoicesip2.txt? |
| 17:41.17 | Darwin35 | try to get it into the trrr |
| 17:41.20 | Darwin35 | tree |
| 17:41.38 | gr8nash | or do you want me to keep going patchless to see if it works with broadvoice |
| 17:41.47 | `Sauron | try patchless |
| 17:41.49 | `Sauron | first |
| 17:41.50 | gr8nash | k |
| 17:41.50 | Darwin35 | but I will also put out a better patch |
| 17:42.00 | Darwin35 | there are 11 new lines thus far |
| 17:43.13 | Sedorox | anyone else using FreeWorldTel Direct? (now link2voip)? |
| 17:44.30 | [Paul] | what's a free softphone that supports g729? |
| 17:44.36 | [Paul] | (SIP) |
| 17:45.02 | `Sauron | I doubt there's anything free that does 729, since 729 has to be licensed. |
| 17:45.20 | [Paul] | "There are some free for non-commerical experimental use implementations for G.729" |
| 17:45.25 | [Paul] | http://www.voip-info.org/wiki-ITU+G.729 |
| 17:45.48 | `Sauron | Paul: quit being pedantic :) |
| 17:45.49 | Goshen | is he talking about that intel thing? |
| 17:46.19 | [Paul] | itu |
| 17:47.07 | Goshen | sweet, I just found a contact juggling video over bittorent that is under creative commons license |
| 17:47.14 | Goshen | I always weanted to learn how to do that |
| 17:47.17 | *** join/#asterisk RoyK (~roy@host-81-191-147-248.bluecom.no) |
| 17:47.31 | Goshen | the ball thing that David Bowie does on Labrinth |
| 17:47.50 | `Sauron | oh, that's easy |
| 17:47.52 | [Paul] | the only sip softphone i found that uses g729 is Eyebeam ($60) |
| 17:47.56 | `Sauron | the 3 balls in a circle? |
| 17:48.56 | Goshen | the ball that rolls around on your hands that seems to defy gravity |
| 17:50.02 | Goshen | http://www.prodigem.com/torrents/details.php?id=304 |
| 17:53.45 | SexyKen | Guys I need a little help with queues. I have 3 queues setup. I want billing and sales queues to ring just one agent if he's available. If he's not, it should go to all other techs. |
| 18:04.37 | SexyKen | can anyone help me at least |
| 18:06.31 | *** join/#asterisk aminorex (~tony@c66.191.69.132.dul.mn.charter.com) |
| 18:06.39 | *** join/#asterisk Drel (~drel@dsl254-029-130.sea1.dsl.speakeasy.net) |
| 18:07.45 | BuckRogers | SexKen: ahh yes you know there is a whole section on that in VoIP Telephony with Asterick by Paul Mahler |
| 18:07.58 | Darwin35 | this patch is apain |
| 18:08.09 | BuckRogers | decent book, i recomend it strongly |
| 18:08.14 | Darwin35 | but I am close to having it done |
| 18:08.21 | *** join/#asterisk j0 (dan@S010600095b00a5a9.vc.shawcable.net) |
| 18:08.41 | BuckRogers | Carrier Grade VoIP by Collins is another good one |
| 18:09.29 | Goshen | Darwin35: people like you make asterisk what it is :) |
| 18:10.46 | BuckRogers | Goshen: yeah damn skippy, |
| 18:10.46 | *** join/#asterisk santiago (~santiago@63.245.86.95) |
| 18:10.54 | *** join/#asterisk RoyK (~roy@host-81-191-147-248.bluecom.no) [NETSPLIT VICTIM] |
| 18:10.54 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) [NETSPLIT VICTIM] |
| 18:10.54 | *** join/#asterisk ScaredyCat (~ScaredyCa@j25065.upc-j.chello.nl) [NETSPLIT VICTIM] |
| 18:10.54 | *** join/#asterisk ptblank (~MURDER1@68-169-176-29.lmdaca.adelphia.net) [NETSPLIT VICTIM] |
| 18:10.54 | *** join/#asterisk A-Tuin|work (~A-Tuin@nat.office.legend.net.uk) [NETSPLIT VICTIM] |
| 18:10.54 | *** join/#asterisk Qwell (~north@70-32-102-18.ontrca.adelphia.net) [NETSPLIT VICTIM] |
| 18:10.54 | *** join/#asterisk Beirdo (~gjhurlbu@beirdo.user) [NETSPLIT VICTIM] |
| 18:10.54 | *** join/#asterisk Gunnar (~gunnar@34.80-203-209.nextgentel.com) [NETSPLIT VICTIM] |
| 18:11.52 | BuckRogers | i had my first bad experince with asterisk realtime today |
| 18:12.27 | MuppetMaster | BuckRogers: What happened? |
| 18:12.36 | *** join/#asterisk Beirdo_ (~gjhurlbu@beirdo.user) |
| 18:12.37 | BuckRogers | MySql stoped functioning when we added a 6th perlmodule |
| 18:13.13 | Darwin35 | ok half this patch is in there half is not man |
| 18:13.23 | MuppetMaster | BuckRogers: So more of a MySQL problem than a RealTime problem? |
| 18:13.24 | BuckRogers | the perl -c compiler did not pick up a missing ";" |
| 18:13.38 | BuckRogers | so mysql got all sorts "F"ed up |
| 18:13.43 | BuckRogers | but its back in action now |
| 18:14.20 | gambolputty | Can * instruct a device to send out DTMF using application/dtmf-relay? |
| 18:14.50 | *** join/#asterisk Red_6 (~alex@m174.net81-66-29.noos.fr) |
| 18:14.58 | BuckRogers | gambo, make the device produce DTMF |
| 18:14.59 | BuckRogers | ? |
| 18:15.13 | BuckRogers | vi an IAX or Sip function? |
| 18:15.17 | gambolputty | sip |
| 18:15.49 | BuckRogers | Well thats a good one |
| 18:16.08 | BuckRogers | for the time being i dont know how but as with life nothing is impossible |
| 18:16.25 | BuckRogers | what would be the purpose for that? |
| 18:16.26 | CoaxD | is VoipJet a great place to go for VOIP services these days? (I helped the owner do some email routing stuff several months back. Wonder how he's done?) |
| 18:16.41 | *** part/#asterisk Red_6 (~alex@m174.net81-66-29.noos.fr) |
| 18:16.45 | spackle | I always say "No Bucks, No BuckRogers" |
| 18:17.08 | gambolputty | to instruct my sipura 3000 to generate DTMF itself in place of DTMF coming from a phone |
| 18:17.29 | BuckRogers | spackle:Y do you always say that |
| 18:18.20 | spackle | People I work for want high-tech, high-tech = buck rodgers, and high-tech can be expensive. |
| 18:18.46 | *** join/#asterisk Drel (~drel@dsl254-029-130.sea1.dsl.speakeasy.net) |
| 18:19.04 | Drel | I'm new to Asterisk and I've ordered an FXO PCI card, and am thinking that the first application I can try with Asterisk is building a fancy voicemail system for my house. Is there a good guide online for beginners who have an FXO but nothing else? I'd like to get an FXS card but they seem much more expensive and out of my budget for the moment :( |
| 18:19.07 | Darwin35 | ok I forget how do you diff 1 file againsed the same file in the cvs |
| 18:19.24 | Darwin35 | to make a patch |
| 18:20.17 | Mneumonic | hey does anyone know how to make my sipura spa-841 act like a traditional pbx, like if you hit 9 to get out it generates a dialtone after that? |
| 18:20.48 | Darwin35 | asterisk does not do tha from what I kow |
| 18:20.56 | Darwin35 | khow of |
| 18:21.00 | Darwin35 | grrr |
| 18:21.11 | Mneumonic | that sucks... i thought there was a way to do that... |
| 18:21.31 | Darwin35 | i have not heard of any but you might check the wiki |
| 18:21.38 | spackle | There is a way but it probably isn't worth the hassle. |
| 18:21.42 | Mneumonic | is there a name for what im looking to do? |
| 18:21.47 | *** join/#asterisk Nix (~Nix@dsl81-214-9283.adsl.ttnet.net.tr) |
| 18:21.52 | spackle | outside dial tone? |
| 18:21.56 | Mneumonic | gotcha |
| 18:22.16 | Darwin35 | ok I have a fully patched chan_sip.c |
| 18:22.29 | Darwin35 | now I have to diff it agained the cvs |
| 18:22.29 | CoaxD | Drel: Mneumonic: You can use DISA |
| 18:22.32 | *** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
| 18:22.32 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 18:22.36 | BuckRogers | i think hewants to instruct the ata from a remote location to produce DTMF signals or tones |
| 18:22.36 | Nix | hi guys |
| 18:22.37 | spackle | you can make '9' an extension to call, then play tone when it gets dialed |
| 18:23.01 | CoaxD | Drel: (in asterisk console, do 'show application DISA' |
| 18:23.02 | Drel | CoaxD: ? |
| 18:23.02 | CoaxD | ) |
| 18:23.21 | CoaxD | drel: It merely generates a dialtone when you dial the extension. You can add pin numbers, passwords, etc. if you like, but.. |
| 18:23.25 | Mneumonic | checkin out DISA now |
| 18:23.26 | BuckRogers | i was referring to what GAMBO wrote |
| 18:23.36 | Drel | CoaxD: You had my question confused with Mneumonic's, I think. |
| 18:23.41 | Darwin35 | ok I will diff when I get back |
| 18:23.43 | CoaxD | drel: On most PBX's, extension '9' is really just that. an extension. and it actually dials, internally. |
| 18:23.47 | CoaxD | Drel: Haha. Probably did |
| 18:23.47 | Darwin35 | and post a new patch |
| 18:23.55 | Darwin35 | unless someone want to do it now |
| 18:24.03 | Darwin35 | I have to go to the dr |
| 18:24.07 | CoaxD | All that stuff was meant for Mneumonic. heh |
| 18:24.08 | bjohnson | Darwin35: I think cvs diff will compare it to cvs |
| 18:24.11 | intrin | anyone program vb? |
| 18:24.13 | Mneumonic | i got it :) |
| 18:24.20 | Drel | CoaxD: Any tips for me? :) |
| 18:24.22 | Mneumonic | thanx guys |
| 18:24.23 | bjohnson | something like cvs diff > whatever.patch |
| 18:24.25 | Darwin35 | ok |
| 18:24.26 | CoaxD | Drel: Looking at your text |
| 18:24.36 | bjohnson | Darwin35: man cvs to confirm |
| 18:24.41 | CoaxD | Drel: I suggest setting up a sip phone first |
| 18:24.45 | Darwin35 | will when I get back |
| 18:24.47 | CoaxD | Drel: Make that work the way you want it |
| 18:25.00 | CoaxD | Drel: Then, once you go to set up a zap line, you'll know the concepts already |
| 18:25.34 | CoaxD | Drel: Asterisk does treat all interfaces identically.. with notable exceptions on initial configuration |
| 18:26.35 | Nix | CoaxD: that is not exactly true ;-) |
| 18:26.39 | Darwin35 | who here is using bv |
| 18:26.49 | Darwin35 | raise hands |
| 18:26.50 | *** join/#asterisk clint_ (~clint@snap.helixsystems.com) |
| 18:27.27 | DyOS | is the digium wildcard x100p the cheapest and best thing for the money I can buy to connect asterisk to an analag line? |
| 18:27.40 | Qwell | You can't get an x100p from Digium anymore |
| 18:28.27 | Darwin35 | get a tdm with 1 fxo port |
| 18:28.31 | CoaxD | Nix: Haha |
| 18:28.34 | Darwin35 | 125 |
| 18:28.41 | Drel | CoaxD: Most of that is over my head, but I will give it a shot :) |
| 18:28.42 | Nix | DyOS: cheapest yes. best no.. Digium stopped selling them anyway |
| 18:28.42 | CoaxD | Nix: Figures, you'd pop out of the bushes to say hello :) |
| 18:28.44 | *** join/#asterisk YoYo (~YoYo@pool-151-199-125-240.roa.east.verizon.net) |
| 18:29.06 | CoaxD | Drel: I just suggest learning how to set up asterisk some and then you'll be more comfortable setting up the zaptel interface |
| 18:29.21 | CoaxD | Drel: But if you're hell-bent on getting the zaptel interface set up right away, http://www.voip-info.org and search some |
| 18:29.21 | DyOS | they are on ebay |
| 18:29.26 | CoaxD | Nix: How you been, sir? |
| 18:29.33 | CoaxD | Nix: How goes the battle with app_prepaid? etc? |
| 18:29.34 | Nix | busy as hell |
| 18:29.36 | DyOS | i'm just wondering what i should buy to play around with without spending a bunch of money i just want to learn |
| 18:29.45 | Drel | DyOS: You can pick up grey market clones of the Wilcard X100P for around $13 w/ shipping on Ebay. I'm embarassingly poor at the moment, so I got one that way to start exploring Asterisk. |
| 18:29.46 | Nix | all the prepaid stuff works very nicely now |
| 18:29.46 | DyOS | what do you guys recommend for the money |
| 18:29.58 | CoaxD | DyOS: Thats the beauty of Asterisk.. Dont need to spend any money to test it out ;) |
| 18:30.03 | Nix | we have finished adding multilevel support this week |
| 18:30.11 | Nix | 5 levels of resellers etc |
| 18:30.14 | CoaxD | Nix: Ooooooh |
| 18:30.22 | Nix | just doing the PHP to manage it all now |
| 18:30.24 | CoaxD | Nix: You know, I patched it to support longer account numbers a while back |
| 18:30.28 | DyOS | as far as a analag port adapter goes though....what do you guys recommend for best quality and money |
| 18:30.30 | Nix | really? |
| 18:30.32 | CoaxD | Nix: Yeah |
| 18:30.40 | topping | DyOS: i just bought a junk (winmodem, apparently) FXO card off ebay for $10 last night |
| 18:30.45 | Nix | we changed everything to float also. instead of integer |
| 18:30.50 | CoaxD | Nix: I didn't actually end up USING it, but I did patch it.. |
| 18:30.57 | Nix | send me the patch |
| 18:31.06 | CoaxD | Nix: I didn't like the short account numbers.. easier to hack. |
| 18:31.09 | Drel | What is the advantage/disadvantage of using an FXS card compared to an ATA? |
| 18:31.12 | CoaxD | Nix: Hmm. I wonder if I stil have that around.. |
| 18:31.14 | Nix | ahh |
| 18:31.16 | DyOS | so an analog adadpter for asterisk the technical term is an fx0 card right? |
| 18:31.19 | Drel | (hopes to god that isn't item #1 in the asterisk FAQ) |
| 18:31.20 | Nix | ok.. I know what you are talking about |
| 18:31.27 | bjohnson | DyOS: I recommend not buying any hardware to start out |
| 18:31.28 | Nix | thats easy to change |
| 18:31.37 | CoaxD | Nix: Sorta :) |
| 18:31.41 | Nix | lol |
| 18:31.43 | topping | DyOS: http://cgi.ebay.co.uk/ws/eBayISAPI.dll?ViewItem&item=5756419834 |
| 18:31.55 | Nugget | the disadvantage is that the zaptel drivers are kinda flaky, especially if you don't use linux. the advantage is that you get four ports with no additional power supply and no space beyond the asterisk server itself. |
| 18:31.56 | topping | apparently not good for production but good to play around with |
| 18:32.21 | Nugget | another advantage is that you help out digium. |
| 18:32.25 | bjohnson | well .. you can play around with it |
| 18:32.27 | *** join/#asterisk Gh0sty (~Ghosty@ip-81-11-201-165.dsl.scarlet.be) |
| 18:32.37 | bjohnson | but why not just go VOIP provider for the same $$ |
| 18:32.48 | *** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) |
| 18:32.52 | *** join/#asterisk RoyK (~roy@host-81-191-147-248.bluecom.no) |
| 18:33.18 | DyOS | i want to start setting up asterisk adn offering it as a solution for my small business customers |
| 18:33.23 | bjohnson | there are even some free options |
| 18:33.24 | DyOS | right now i'm just trying to learn it |
| 18:33.40 | DyOS | but most customers are not goign to want voip provider because of qualoity and reliability |
| 18:33.53 | Darwin35 | DyOS no competeing withme |
| 18:33.56 | Darwin35 | grr |
| 18:33.58 | topping | hehe |
| 18:34.02 | bjohnson | echo is always a problem with fxo/analog lines |
| 18:34.04 | topping | yah me too |
| 18:34.06 | topping | ;) |
| 18:34.11 | Nix | we plan to make the prepaid code work with FreeRADIUS and gnugk soon too. It already works with Asterisk and Aefirion of course |
| 18:34.30 | *** join/#asterisk abatista (~Ariel@dsl-20-177.cofs.net) |
| 18:34.31 | Darwin35 | we have calling cards woring |
| 18:34.33 | Darwin35 | yes |
| 18:34.44 | Darwin35 | BKW we love you |
| 18:34.48 | bjohnson | DyOS: best system for small office is a mix |
| 18:35.07 | *** join/#asterisk calvinhp (~calvinhp@rrcs-24-123-25-236.central.biz.rr.com) |
| 18:35.12 | DyOS | what do you mean by mix? |
| 18:35.18 | Darwin35 | I have 15/2025/35/45 line pbx setups |
| 18:35.21 | JerJer[mobile] | nx: bleh |
| 18:35.27 | JerJer[mobile] | nix |
| 18:35.38 | Darwin35 | I have 15/20/25/35/45 line pbx setups we are starting to sell |
| 18:35.47 | *** join/#asterisk Tili (~Tili@202-133-65-131-dialup.sat.net.pk) |
| 18:35.48 | Darwin35 | all voip |
| 18:36.01 | gr8nash | Darwin Do VOIP work for critical systems like a medium size buisiness |
| 18:36.05 | gr8nash | ? |
| 18:36.19 | Darwin35 | but working on a ver with the 4port tdm fxo in it if needed |
| 18:36.23 | gr8nash | my company is ~45 phones |
| 18:36.43 | Darwin35 | with iax providers yes |
| 18:36.50 | Darwin35 | with sip providers no |
| 18:36.55 | gr8nash | iax... |
| 18:36.58 | Darwin35 | sip still has many issues |
| 18:37.03 | GiabboO | i'm tryin to set up asterisk to read users sip and iax into a mysql db, can anyone tell me where must i fill information about the db? |
| 18:37.04 | gr8nash | hmmm new acronmy |
| 18:37.35 | jontow | gr8nash; IAX is good stuff.. do your research on that one :) |
| 18:37.38 | GiabboO | i insered it on sip.conf iax.conf extconfig.conf |
| 18:37.40 | Darwin35 | inter asterisk exchange |
| 18:37.58 | gr8nash | jontow ok ill read alittle |
| 18:38.27 | Darwin35 | well off to the dr bbiab |
| 18:38.36 | *** join/#asterisk slePP (~slepp@S01060040f48412ad.ed.shawcable.net) |
| 18:38.45 | GiabboO | Darwin35 can i pvt ? |
| 18:38.58 | Darwin35 | off to the dr running late |
| 18:39.02 | Darwin35 | when I get back |
| 18:39.06 | Darwin35 | in 2 hours |
| 18:39.23 | GiabboO | anyone can help me with asterisk + mysql ? |
| 18:39.43 | Hmmhesays | GiabboO probably |
| 18:40.04 | bjohnson | DyOS: I mean some voip and some analog lines |
| 18:40.04 | Hmmhesays | but i'm not a mind reader |
| 18:40.21 | GiabboO | Hmmhesays can i pvt ? |
| 18:40.35 | Hmmhesays | i suppose |
| 18:40.50 | bjohnson | DyOS: maybe voip for outgoing and failover to pstn if problems .. or some voip and then pstn, then voip again |
| 18:41.25 | Nugget | unable to load module chan_esp.so |
| 18:42.03 | bjohnson | DyOS: for incoming .. maybe analog for incoming or analog forwarded to a voip DID .. or a voip DID forwarded to analog if VOIP connection not available |
| 18:42.36 | *** join/#asterisk afe ([qH2ZrhT2I@c-e616e055.123-1-64736c12.cust.bredbandsbolaget.se) |
| 18:42.59 | *** join/#asterisk obsidian-studios (~obsidian-@c-66-177-188-197.se.client2.attbi.com) |
| 18:43.31 | obsidian-studios | greetings, I got some funkiness going on with an existing * deployment |
| 18:43.41 | machinehd | The Cisco 7960 has 6 lines and I notice they all try registering. I haven't yet found how to disable the extra lines. Any know? |
| 18:43.57 | jontow | machinehd; SIPmacaddr.cnf on your tftp server. |
| 18:44.13 | jontow | obviously, replace macaddr with the MAC address of the phone in question |
| 18:44.15 | obsidian-studios | I did some upgrades recently software wise, *, zaptel, zapata, and for some reason * is only showing activity on zap 1 and no 2-4? |
| 18:44.22 | jontow | lineN_... should be fixed/removed/etc :) |
| 18:44.59 | obsidian-studios | since the configs have not changed I am thinking it is something to do with the upgrades? |
| 18:45.09 | machinehd | jontow, I wasn't sure if I should remove those lines from the .cnf file. Thanks |
| 18:45.27 | obsidian-studios | outbound everything works, but inbound I only get calls on zap1 |
| 18:45.29 | jontow | machinehd; keep the main 3 at least.. (i think they're in sets of 3..) |
| 18:45.39 | jontow | obsidian; grouping must have changed somehow |
| 18:45.46 | obsidian-studios | that's what I am thinking |
| 18:45.46 | jontow | obsidian; do you use stable or HEAD? |
| 18:46.06 | machinehd | jontow, ok, would just setting the username to blank disable registration attempts? |
| 18:46.10 | obsidian-studios | I never had grouping correct? and I use the version in the ~x86 tree of gentoo, which tends to be a bit behind at times |
| 18:46.21 | machinehd | guess I should just try it and see :) |
| 18:46.43 | obsidian-studios | I am thinking that grouping may have been corrected or modified and my incorrect config or something is now causing the problem, when it did not before? |
| 18:48.03 | obsidian-studios | I am using * 1.0.5-r1, zaptel 1.0.4, ad zapata 1.0.1 |
| 18:48.12 | GiabboO | can anyone help me to set up asterisk to read iax and sip users on a mysql db ? |
| 18:49.53 | GiabboO | im not sure if its right to insert db info into sip.conf and iax.conf or use res_mysql.conf |
| 18:50.16 | GiabboO | nobody use mysql for sip and iax? |
| 18:50.18 | *** join/#asterisk Tili (~Tili@202-133-65-238-dialup.sat.net.pk) |
| 18:52.16 | *** join/#asterisk t3t (~t3t@galley.pangalacticgargleblaster.com) |
| 18:54.19 | obsidian-studios | did something with groups change? |
| 18:54.35 | obsidian-studios | when messing around with that it seems to effect things some what? |
| 18:54.39 | MikeJ[Jayden] | giabbo0, http://www.voip-info.org/wiki-Asterisk+Realtime |
| 18:54.49 | Jones | it worked before you broke it :-D |
| 18:55.06 | *** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) |
| 18:56.11 | GiabboO | MikeJ[Jayden], do i need to make res_config_mysql.so also if im using app_addon_sql_mysql.so ? |
| 18:56.16 | *** join/#asterisk allanon (~allanon@c-24-18-177-201.client.comcast.net) |
| 18:56.53 | obsidian-studios | if you mean me yes it was working before I broke |
| 18:59.46 | spackle | Is anyone using one of the legacy-toVOIP boxes from citel? |
| 19:00.26 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
| 19:01.57 | *** join/#asterisk SexyKen (~sexyken@c-67-161-5-149.client.comcast.net) |
| 19:02.14 | GiabboO | :( |
| 19:02.16 | SexyKen | Anyone know why I'd get this error for one extension and not the rest: Mar 8 12:53:47 WARNING[4378]: chan_agent.c:1548 __login_exec: Extension '1305@default' is not valid for automatic login of agent '1305' |
| 19:02.51 | GiabboO | must I use app_addon_sql_mysql.so or res_mysql_config.so to make mysql working ? |
| 19:03.53 | Drel | Is anyone here using the DTA-310 with Asterisk? I have this left over from being a (now wiser) early adopter of Packet8 ;) I'd love to be able to use it with Asterisk, and have found some directions on doing so. The one thing I can't track down is the v12.34 firmware to use with it. |
| 19:04.41 | *** join/#asterisk jalsot (~tamas@abacus.eworldcom.hu) |
| 19:05.55 | *** join/#asterisk rav3n (rav3n@user-2ivff3o.dsl.mindspring.com) |
| 19:08.09 | spackle | SexyKen, typo, extension is duplicated, or contexts are messed up? |
| 19:09.05 | *** join/#asterisk redder86 (~lee@gateway.howardsilvan.com) |
| 19:09.08 | *** part/#asterisk obsidian-studios (~obsidian-@c-66-177-188-197.se.client2.attbi.com) |
| 19:09.56 | redder86 | Anyone here using Linux QoS and kernel 2.4.29? I'm getting unresolved symbol problems when loading the QoS modules. |
| 19:11.45 | *** join/#asterisk sosico (~oeo@199.75.106.64) |
| 19:11.55 | sosico | hello |
| 19:12.05 | sosico | I'm having problems registering polycom soundpoint 500 to Asterisk |
| 19:12.07 | sosico | any ideas? |
| 19:12.21 | *** join/#asterisk ruiner (ruiner@ruiner.netslacking.net) |
| 19:12.49 | *** join/#asterisk gezick (gezick@sartre.ispvip.biz) |
| 19:12.53 | *** join/#asterisk fyrfaktry (~fyr@ip68-106-201-109.pn.at.cox.net) |
| 19:13.06 | gezick | how should i go about dealing with an error like Mar 8 13:11:16 WARNING[12968]: app_dial.c:1006 dial_exec: Had to drop call because I couldn't make SIP/polycom1-cb55 compatible with IAX2/66.234.228.160:4569/ |
| 19:13.20 | *** part/#asterisk fyrfaktry (~fyr@ip68-106-201-109.pn.at.cox.net) |
| 19:13.22 | nestAr | sosico: that's fairly vauge |
| 19:13.27 | sosico | vauge? |
| 19:13.27 | nestAr | might want to give more details |
| 19:13.28 | sosico | what is that |
| 19:13.30 | gezick | i'm assuming that the phone is trying to send gsm and my * server won't like it? |
| 19:13.33 | sosico | oh yes |
| 19:13.47 | sosico | ok....I download the firmware from http://www.voip-info.org/wiki-Polycom+Phones |
| 19:13.56 | nestAr | sorry.. vague |
| 19:13.57 | nestAr | :) |
| 19:13.59 | sosico | :) |
| 19:14.10 | nestAr | was it not a SIP phone to start with? |
| 19:14.11 | sosico | and I configured the phone from the Web |
| 19:14.21 | sosico | it is a polycom Soundpoint 5000 |
| 19:14.22 | sosico | 500 |
| 19:14.23 | sosico | sorry |
| 19:14.42 | nestAr | well, if it was a 500 with SIP already.. no need to install firmware |
| 19:14.44 | *** join/#asterisk MuppetMaster (~muppetmas@a82-92-73-185.adsl.xs4all.nl) |
| 19:14.47 | nestAr | should work out of the box. |
| 19:14.57 | CleanerX | can someone do a dns lookup for me? |
| 19:15.10 | ruiner | CleanerX: sure |
| 19:15.12 | CleanerX | I'm behind nat and need to get an ip |
| 19:15.21 | topping | CleanerX: go |
| 19:15.27 | sosico | I donwloaded the firmware from http://www.freedomphones.net/polycom/files/ |
| 19:15.36 | *** join/#asterisk jalsot (~tamas@195.56.44.83) |
| 19:15.39 | nestAr | sosico: the WEB interface is pretty complex.. i'd say it's probably a typo.. |
| 19:15.42 | sosico | it didn't have the SIP firmare |
| 19:15.50 | topping | CleanerX: what's the address? |
| 19:16.08 | sosico | I configured all the ip addresses to point to the asterisk server |
| 19:16.12 | *** part/#asterisk redder86 (~lee@gateway.howardsilvan.com) |
| 19:16.32 | sosico | and every time I get the same message chan_sip.c:7691 handle_request: Registration from '"oas" <sip:10.10.5.45@10.10.5.45>' failed for '10.10.12.54' |
| 19:16.49 | sosico | I was reading that sometimes polycom phones do not like to be configured as type=friend |
| 19:16.52 | nestAr | do you have cas defined in your asterisk sip.conf? |
| 19:16.57 | sosico | yes |
| 19:17.03 | nestAr | did you reload? |
| 19:17.06 | sosico | yes |
| 19:17.06 | topping | CleanerX: http://www.dnsstuff.com/ is good too |
| 19:17.08 | sosico | several times |
| 19:17.16 | SexyKen | Anyone know why I'd get this error for one extension and not the rest: Mar 8 12:53:47 WARNING[4378]: chan_agent.c:1548 __login_exec: Extension '1305@default' is not valid for automatic login of agent '1305' |
| 19:17.17 | sosico | I used type=friend |
| 19:17.19 | nestAr | sounds like password typo somewhere in the middle |
| 19:17.20 | gezick | sosico: if you've followed everything in the wiki and that you've found online and that's still happening, try the following |
| 19:17.29 | gezick | hard-reset the phone |
| 19:17.33 | gezick | go to menu |
| 19:17.38 | eKo1 | Hmm...for some reason, * is keeping sip channels where the number passed was a wrong number. |
| 19:17.41 | gezick | settings |
| 19:17.42 | nestAr | i use friend for all of my polycom phones. |
| 19:17.47 | gezick | Reset user settings |
| 19:17.52 | gezick | and reset all the user settings |
| 19:17.59 | gezick | i did that, and just like magic it decided to work |
| 19:18.06 | sosico | ok I'll try that right now |
| 19:18.17 | sosico | is it gonna get all the configuration back from the ftp server? |
| 19:18.22 | gezick | yeah |
| 19:18.26 | nestAr | yes |
| 19:18.39 | sosico | great I will try it right away thanks |
| 19:18.43 | gezick | i did that not 20 min ago, and now the phone is recieving calls into it |
| 19:18.44 | sosico | I'll let you know how it went |
| 19:18.46 | sosico | brb |
| 19:19.04 | nestAr | every registration problem i've had with polycom phones has been me fatfingering the passwords or not reloading asterisk |
| 19:19.07 | nestAr | ;) |
| 19:19.14 | gezick | but it isn't making calls correctly. it's just doing ulaw (i think) and * can convert it to IAX to get out |
| 19:19.28 | gr8nash | can someone point out where i can RTFM. =) |
| 19:19.42 | gr8nash | or the best docs for a newbie |
| 19:19.45 | nestAr | ~docs |
| 19:19.47 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 19:19.50 | RoyK | ~lart digium |
| 19:20.03 | nestAr | i'd go to http://www.voip-info.org/wiki-Asterisk |
| 19:20.04 | gr8nash | jbot thank you |
| 19:20.04 | jbot | gr8nash: no worries |
| 19:20.16 | nestAr | it's been the most useful to me overall |
| 19:21.18 | gezick | how should i go about dealing with an error like this: "Mar 8 13:20:18 WARNING[12968]: channel.c:2115 ast_channel_make_compatible: No path to translate from SIP/polycom1-f3e5(256) to IAX2/66.234.228.160:4569/2(4) |
| 19:22.01 | GiabboO | uff |
| 19:22.06 | GiabboO | i read documentations |
| 19:22.14 | GiabboO | but my asterisk + mysql wont work |
| 19:22.39 | *** join/#asterisk j0 (dan@S010600095b00a5a9.vc.shawcable.net) |
| 19:22.42 | opus___ | hello |
| 19:23.00 | opus___ | giabbo - i had the same problem |
| 19:23.07 | GiabboO | hi opus___ |
| 19:23.07 | opus___ | giabbo which route did you take |
| 19:23.19 | GiabboO | do you use mysql for users storing ? |
| 19:23.28 | opus___ | No, i'm having problems |
| 19:23.29 | JerJer[mobile] | gezick: how about adding in a G.729 codec |
| 19:23.33 | JerJer[mobile] | or not usng G.729 |
| 19:23.46 | opus___ | I noticed that the build script used -O6 also.. |
| 19:23.59 | topping | GiabboO, opus___: it works on FC3 if that helps at all |
| 19:24.01 | opus___ | giabbo0 - i used the real time way |
| 19:24.10 | GiabboO | opus___, what modules are u loading to do it? app_addon_sql_mysql.so or res_mysql_config.conf ? |
| 19:24.26 | GiabboO | ops, res_mysql_config.so |
| 19:24.35 | opus___ | where did it fail? |
| 19:24.38 | *** part/#asterisk crash3m (crash3m@crash3m.user) |
| 19:25.09 | GiabboO | i thought that i only need to load add_addong_sql_mysql instead of res_mysql_conigf.so |
| 19:25.18 | JerJer[mobile] | i simply do not understand why so many people are hell bent on forcing asterisk to depend on a database to operate |
| 19:25.32 | GiabboO | res_mysql_config.so looks like old way to interface asterisk with mysql |
| 19:25.38 | Nugget | JerJer[mobile]: especially mysql. :) |
| 19:25.45 | eKo1 | JerJer[mobile]: I don't think that's the reason. |
| 19:25.53 | slePP | JerJer[mobile]: so they can have silly front ends that their managers can figure out |
| 19:25.57 | GiabboO | JerJer[mobile]: :) |
| 19:26.00 | Nugget | "how can I make asterisk more complex and less reliable? I know! mysql!" |
| 19:26.04 | opus___ | giabbo where did it fail |
| 19:26.06 | JerJer[mobile] | slePP: we have front ends |
| 19:26.06 | GiabboO | muahah |
| 19:26.08 | eKo1 | They want extra functionality/services. |
| 19:26.08 | GiabboO | LoOoL |
| 19:26.17 | JerJer[mobile] | ala GUIs |
| 19:26.24 | slePP | JerJer[mobile]: "silly" front ends |
| 19:26.27 | slePP | ie: php |
| 19:26.30 | JerJer[mobile] | sure |
| 19:26.32 | JerJer[mobile] | all day long |
| 19:26.32 | eKo1 | Who knows. |
| 19:26.34 | GiabboO | opus___ i have that modules on /usr/lib/asterisk/modules |
| 19:26.37 | GiabboO | i installed the addons too |
| 19:26.41 | GiabboO | but nothing work :( |
| 19:26.42 | *** join/#asterisk glikely (~glikely@S010600a0d1d8824b.cg.shawcable.net) |
| 19:27.01 | opus___ | nothing work? where did it fail? |
| 19:27.04 | GiabboO | last addon dont have res_mysql_config.so so i though its no more needed |
| 19:27.08 | GiabboO | i dont know! |
| 19:27.16 | GiabboO | i dont know where it log about db access |
| 19:27.19 | spackle | We don't need no stinkin' functionality and front ends! |
| 19:27.22 | GiabboO | called out by the extconfig.conf |
| 19:27.56 | GiabboO | anyway, stop, its 8.30pm here in italy, and ac milan is starting a match against manchester |
| 19:27.56 | eKo1 | The only thing I need a DB for with * is for storinng CDR data. |
| 19:28.00 | GiabboO | i go home and watch the match |
| 19:28.07 | GiabboO | CDR working here |
| 19:28.11 | GiabboO | u need to load the module |
| 19:28.31 | GiabboO | and set up properly the conf file cdr_mysql.conf |
| 19:28.37 | eKo1 | You know MU will win that match. |
| 19:29.10 | GiabboO | the module you need is cdr_addon_mysql.so opus___ |
| 19:29.13 | eKo1 | hah! I left everyone speechless |
| 19:29.17 | GiabboO | ok ppl |
| 19:29.23 | GiabboO | thx for help :P |
| 19:29.27 | GiabboO | i go cya all |
| 19:32.34 | slePP | no one ever seems to read anything on pastebin |
| 19:33.04 | slePP | or maybe new stuff should be bright pink |
| 19:33.46 | eKo1 | Ah! Corrupted download. |
| 19:33.59 | eKo1 | ·$$%"·$·@#~|@ |
| 19:34.01 | topping | slePP: what's in there that's relevant? |
| 19:34.11 | slePP | http://pastebin.ca/draw.php |
| 19:34.25 | slePP | http://pastebin.ca/news.php -- the request for comments bit |
| 19:35.29 | *** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net) |
| 19:36.53 | gr8nash | Broadvoice users.. is the accountID the same as Username? |
| 19:37.00 | slePP | oh well :> k. off i go to find coffee. |
| 19:39.25 | *** part/#asterisk glikely (~glikely@S010600a0d1d8824b.cg.shawcable.net) |
| 19:40.37 | tuxinator_linux | ~kiwa |
| 19:40.46 | tuxinator_linux | ~weather kiwa |
| 19:40.51 | *** join/#asterisk rephorm (~rephorm@ip67-95-13-62.z13-95-67.customer.algx.net) |
| 19:44.33 | spackle | ~weather dsm |
| 19:44.51 | *** join/#asterisk KrimHum (~barry@mercury.santabarbararealty.net) |
| 19:44.54 | topping | ~weather kdsm |
| 19:45.07 | KrimHum | ~weather sba |
| 19:45.36 | riksta | is anyone interested in helping me with the ongoing development of ADM http://adm.hamnett.org |
| 19:45.41 | ruiner | ~weather lnk |
| 19:45.44 | afe | ~weather ESCM |
| 19:45.52 | afe | ~weather ESSA |
| 19:45.56 | ManxPower | ~docs |
| 19:45.57 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 19:46.10 | ManxPower | There, at least jbot knows SOMETHING useful |
| 19:46.16 | afe | indeed :) |
| 19:46.17 | topping | ICAO codes for american airports start with a 'K' folks |
| 19:46.29 | KrimHum | ~weather kp44 |
| 19:46.40 | ManxPower | Perhaps you could wank the bot in private? |
| 19:46.41 | spackle | ~weather kdmx |
| 19:46.45 | topping | riksta: what's it written in? |
| 19:46.52 | riksta | gtk2-perl |
| 19:46.58 | riksta | for now |
| 19:47.08 | KrimHum | ~weather ksba |
| 19:47.14 | ruiner | ~weather klnk |
| 19:47.25 | topping | weather ~kink |
| 19:47.33 | riksta | duh |
| 19:47.38 | *** join/#asterisk bzzzz (bzzz@84.217.4.200) |
| 19:47.48 | topping | ~weather kink |
| 19:47.53 | Juggie | is there anyway to transfer pri->pri and get asterisk out of the conversation? |
| 19:47.56 | *** join/#asterisk insomni (~insomni@82.192.163.226) |
| 19:48.02 | afe | on http://www.voip-info.org/wiki-Monitor+setup+sample there's an example on how to monitor outgoing calls. |
| 19:48.19 | afe | Would it be possible to make * mail the file automatically from the dialplan? |
| 19:48.41 | topping | riksta: i've been considering something similar in java |
| 19:49.00 | topping | tis the reason i ask... prefer to help someone already started if possible |
| 19:49.11 | rephorm | on the wiki, it says that the 's' extension currenty isn't supported from realtime extensions tables. is that still true? |
| 19:49.40 | *** join/#asterisk shmaltz (~chatzilla@69.28.255.210) |
| 19:49.46 | riksta | topping: cool, i was considering re-writing it in java, so it'll work in windows, how are your skills? |
| 19:50.03 | topping | http://www.dentaku.org will give you some idea |
| 19:50.17 | *** join/#asterisk miguellinux (~miguellin@200.47.223.190) |
| 19:50.24 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlgrv.pa.sed6.net) |
| 19:51.08 | topping | riksta: i'm currently trying to write a SOAP client module for asterisk to talk to an ESB |
| 19:51.11 | riksta | topping: oh, this looks cool |
| 19:51.16 | rephorm | afe: after recording you can run a script on the CALLFILENAME (system app, i believe) |
| 19:51.20 | riksta | topping: yeah i've used SOAP a few times |
| 19:51.37 | topping | i want to make asterisk a part of an enterprise service bus |
| 19:51.38 | riksta | topping: would you be interested in working together on ADM then? |
| 19:51.50 | topping | sure! if it's in java that would be good! |
| 19:51.57 | riksta | topping: yeah i agree |
| 19:51.59 | topping | i also use maven |
| 19:52.00 | *** join/#asterisk MicH323 (~micosat@host-84-9-63-27.bulldogdsl.com) |
| 19:52.08 | topping | hope that's not a dealbreaker |
| 19:52.09 | riksta | i only started it in gtk2-perl as a project to learn about the perl bindings for gtk2 |
| 19:52.09 | topping | lol |
| 19:52.29 | topping | riksta: how much java experience do you have? |
| 19:52.54 | afe | rephorm, I was thinking of something like that - but my scripting skills are somewhat restricted so I was hoping for an example somewhere :) |
| 19:52.57 | gr8nash | anyone got a second to explain the last step .. (adding BV to the dial plan.) i dont know what im looking for/at i have the wiki stuff in front.. but im alittle confused still |
| 19:53.12 | riksta | topping: not a huge amount, enough for the basics of OO, threads, sockets etc |
| 19:53.27 | afe | rephorm, also, could it not be a problem if the in and out files are not yet mixed |
| 19:53.29 | riksta | topping: at the moment, really, ADM is pretty much very simple.....but effective |
| 19:53.42 | topping | riksta: check out http://www.muleumo.org |
| 19:53.48 | riksta | topping: do you have an instant messenger address |
| 19:53.50 | topping | that's what i'm wiring asterisk into |
| 19:53.57 | riksta | looking |
| 19:54.29 | riksta | ahh yeah that is interesting |
| 19:54.42 | topping | so asterisk just creates events on the bus and interested clients can subscribe to them if they want |
| 19:54.49 | riksta | yeah |
| 19:54.49 | topping | it's all asynchronous |
| 19:55.03 | riksta | i like it |
| 19:55.14 | topping | well, it's happening yo... :-) |
| 19:55.23 | riksta | can i PM |
| 19:55.29 | topping | of course |
| 20:01.30 | bzzzz | Hi everyone looking for help with H323 channel |
| 20:01.48 | bzzzz | it builds fine but all calls are rejected |
| 20:03.13 | *** join/#asterisk lilwookie (~bender@modemcable215.87-81-70.mc.videotron.ca) |
| 20:04.03 | lilwookie | hi all.. quick questio what channel bank/card would be best for * and 100+ FXS ports? |
| 20:04.03 | Moc____ | Is nufone down ? |
| 20:04.23 | JerJer[mobile] | Adtran TA750 to start |
| 20:04.37 | Moc____ | JerJer[mobile]: , having some problem / |
| 20:04.43 | JerJer[mobile] | not a 100 channels, but still damn good |
| 20:05.06 | JerJer[mobile] | Moc____: first off we do our own terminaton |
| 20:05.18 | spackle | bzzzz, which h323 implemenetation? |
| 20:05.19 | Moc____ | I got International call going throught this morning, but not anymore |
| 20:05.20 | lilwookie | JerJer, I was looking @ the 750's what type of interface would I need for *, does it work over Ethernet also? or I need a TDM? |
| 20:05.39 | JerJer[mobile] | Moc____: second why would you lie and say we send calls out VoIP, when u have no clue about our operation? |
| 20:05.46 | bzzzz | spackle: H323 (not OH323) |
| 20:05.51 | Moc____ | JerJer[mobile]: look, I personally dont care, all I know is call aint bidirectional, so it kind of weird from a TDM network |
| 20:06.06 | inspired | how do I measure the amount of PRI channels in use on my server? |
| 20:06.10 | JerJer[mobile] | show channels |
| 20:06.25 | inspired | possible to get it into mrtg or somethimg similar? |
| 20:06.26 | Moc____ | JerJer[mobile]: I didnt said you were, I said they it seem like you do it |
| 20:06.33 | lilwookie | <---- lines with nufone work bothways :) |
| 20:06.43 | Moc____ | since I get the same audio problem from you and Voiceconduits |
| 20:06.48 | Moc____ | but not any other |
| 20:07.00 | JerJer[mobile] | Moc____: so that magcally means we are sendng our calls IP? |
| 20:07.13 | JerJer[mobile] | perhaps its the same underlying carrier? |
| 20:07.28 | Moc____ | like I just said, I didnt said you were, I said it seem... |
| 20:07.56 | JerJer[mobile] | we have no control what these little ass countries do with their telephone services |
| 20:08.55 | *** join/#asterisk feklee (feklee@genba.ffii.org) |
| 20:09.10 | JerJer[mobile] | and why are you complaining to others and not NuFone support? |
| 20:09.16 | JerJer[mobile] | i haven't seen a call trouble report from you |
| 20:09.23 | feklee | How can I find out who called at a certain time? I.e. in which log file do I have to look? |
| 20:09.37 | ionix | cdr |
| 20:09.38 | ionix | heh |
| 20:09.47 | Moc____ | JerJer[mobile]: my message to you got the answer, we are fine, it your problem |
| 20:09.53 | JerJer[mobile] | cat /var/log/asterisk/cdr-csv/Master.csv | grep <whatever u are looknig for> |
| 20:10.03 | JerJer[mobile] | its not our problem |
| 20:10.22 | Moc____ | then I talked to someone in the conf about my voip experience, and he said he had similar problem |
| 20:10.38 | JerJer[mobile] | we've got thousands of sasisfied customers |
| 20:10.51 | Moc____ | im satisfied when it work too.. |
| 20:11.12 | JerJer[mobile] | so then complain to nufone support, not some random person in a conference call |
| 20:11.17 | Moc____ | like International throught nufone work for me this morning, it doesn't anymore :/ |
| 20:11.24 | JerJer[mobile] | it doesn't |
| 20:11.32 | JerJer[mobile] | funny i just hung up with Japan |
| 20:11.36 | JerJer[mobile] | and eariler I called UK |
| 20:11.51 | Moc____ | Ive called 3 time UK this morning just fine !! I find this weird |
| 20:12.00 | Moc____ | I keep getting invalid context now |
| 20:12.02 | *** join/#asterisk empire667 (~user1@h71032.upc-h.chello.nl) |
| 20:12.04 | ionix | nufone works for me in Japan now... |
| 20:12.12 | JerJer[mobile] | then don't put a context in a type=peer |
| 20:12.29 | JerJer[mobile] | and send us 011+countrycode+number |
| 20:13.19 | lilwookie | so what sort of configuration would be best on a 750? can it support only one T1 |
| 20:13.22 | feklee | Again: How can I find out who called at a certain time? I.e. in which log file do I have to look? |
| 20:13.43 | Moc____ | JerJer[mobile]: no context and I send 011.. weird |
| 20:13.50 | Moc____ | urm wait.. |
| 20:14.05 | JerJer[mobile] | (14:09:08) JerJer[mobile]: cat /var/log/asterisk/cdr-csv/Master.csv | grep <whatever u are looknig for> |
| 20:14.23 | ruiner | ^^ That's for you feklee |
| 20:15.02 | JerJer[mobile] | if u don't want to send us 011, then tell me and i'll put you in a different context on this end |
| 20:15.04 | Moc____ | this is weird, how come it worked this morning |
| 20:15.07 | JerJer[mobile] | it is not that dfficult |
| 20:15.17 | Moc____ | yea I didnt put 011 !! |
| 20:15.17 | feklee | ruiner: JerJer[mobile]: Thanks. I was confused by the ongoing discussion (sorry for chiming in). |
| 20:15.20 | *** join/#asterisk santiago (~santiago@63.245.86.95) |
| 20:15.27 | Moc____ | brb gota call the user |
| 20:17.27 | feklee | ruiner: JerJer[mobile]: Weird. The file is incomplete: The last registered call is from yesterday morning. There have been several calls afterwards. What may be the explanation? |
| 20:18.28 | feklee | Hm, I've an idea ... |
| 20:19.02 | lilwookie | JerJer[mobile], thanks.. I am looking to hook up an apt building's cat3 into a Asterisk and move it out to a service like nufone, what would you recomend please? :) |
| 20:20.04 | Moc____ | ok this work now thanks |
| 20:20.41 | feklee | ruiner: JerJer[mobile]: AFAICS, I found the reason: I was running Asterisk as root for some time and restarted it later to be run as user "asterisk". This user, however, didn't have write permissions to "Master.csv". Sucks, but oh well... |
| 20:20.55 | feklee | I'm too curious who made that call. |
| 20:21.36 | feklee | s/who/to find out who/ |
| 20:22.12 | opus___ | what exactly does type=peer do to an entry |
| 20:22.20 | AgiNamu | Is digium's FTP site fucked up right now? |
| 20:22.29 | AgiNamu | trying to download the registration and g729 codec |
| 20:22.34 | j0 | is it possible to load balance between 2 peers? or to use 1 incase another goes down? |
| 20:23.41 | Moc____ | j0: sure |
| 20:24.06 | j0 | ok.. just wanted to confirm before i looked into it further.. thanks :) |
| 20:24.13 | Moc____ | you can make a variable and just switch between 0 and 1, or just use the same provider, and put your second provider in a different priority |
| 20:24.29 | j0 | ok |
| 20:24.49 | *** join/#asterisk jero (~boo@199.243.85.90) |
| 20:24.52 | jero | hello |
| 20:26.39 | Moc____ | hi |
| 20:26.56 | jero | hi moc :) |
| 20:27.42 | *** join/#asterisk newpers (newpers@ip24-56-8-180.ph.ph.cox.net) |
| 20:27.47 | jero | Can anyone recommend an open source web-based GUI for asterisk configuration ? |
| 20:27.49 | gezick | i'm having trouble making outgoing calls with a polycom phone, is there anywhere that anyone can think of to point me that has information beyond what is on the wiki? |
| 20:28.05 | Moc____ | gezick: what kind of problem É |
| 20:28.17 | jontow | jero; i hear AMP is nice.. never used it though. |
| 20:28.34 | *** join/#asterisk file (~file@sjcc30x31.sjccnet.com) |
| 20:28.53 | JunK-C | jero: simply wait for * 1.2 |
| 20:29.02 | zipp | JunK-C, explain? |
| 20:29.03 | jero | I can't :) |
| 20:29.16 | RoyK | ~lart digium |
| 20:29.23 | file | soooo many people in the Digium booth |
| 20:29.39 | gezick | Moc____: chan_sip.c:1829 sip_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 4/2) |
| 20:29.59 | gezick | Moc____: and channel.c:1691 ast_set_write_format: Unable to find a path from ulaw to g729 |
| 20:29.59 | JunK-C | zipp: explain what? |
| 20:30.04 | JunK-C | file: how many ppl? |
| 20:30.15 | *** join/#asterisk loud (~ariel@null0.flapping.net) |
| 20:30.30 | file | 15 |
| 20:30.37 | zipp | JunK-C, I know 1.1(1.2 stable) has ara, which is nice |
| 20:30.43 | zipp | what about a web GUI in 1.2? |
| 20:31.17 | JunK-C | great. |
| 20:31.28 | jero | when is 1.2.0 due to ? |
| 20:31.36 | Goshen | When I forward my PTSN phone to my cellphone and someone calls it, my cellphone shows an arrow next to the caller id showing it is a forwarded call |
| 20:31.46 | *** join/#asterisk abatista (~Ariel@dsl-20-177.cofs.net) |
| 20:31.55 | Goshen | is there any way to set that forward flag when I make a call...say through NuFone? |
| 20:32.10 | Moc____ | gezick: ha easy ;) |
| 20:32.11 | Goshen | its nice to know to answer a certain way when you are answering the forwarded call |
| 20:32.12 | JunK-C | zipp: ive heard 1.2 is suppose to has a web GUI. |
| 20:32.19 | zipp | JunK-C, ah, ok |
| 20:32.22 | JunK-C | dunno. 6-8 months. |
| 20:32.26 | jero | k :) |
| 20:32.41 | AgiNamu | voicemail... app.c:599 ast_play_and_record: No audio available on IAX2/TLF11004@TLF11004/5?? |
| 20:32.41 | gezick | Moc____: is it just as simple as installing the g729 codec from digium? |
| 20:32.47 | AgiNamu | any clue what that means? |
| 20:32.49 | Moc____ | gezick: your phone want g729 (weird..) you have disallow=all allow=ulaw in your sip entry ? |
| 20:33.08 | gezick | disallow=all |
| 20:33.08 | gezick | allow=ulaw ; dtmfmode=inband only works with ulaw or alaw! |
| 20:33.08 | gezick | allow=gsm |
| 20:33.09 | Goshen | I think I will post that question to thie mailing list |
| 20:33.10 | Moc____ | gezick: it 1 Disable g729, or 1 install and buy the g729 liscence |
| 20:33.25 | Moc____ | gezick: that is global ? or just in your SIP entry for the phone ? |
| 20:33.34 | gezick | just the one for the phone |
| 20:33.58 | shmaltz | tzafrir? |
| 20:34.02 | Moc____ | where did you get your polycom config file ? |
| 20:34.10 | Moc____ | by default is ulaw... |
| 20:34.13 | gezick | the wiki |
| 20:34.26 | gezick | the one that uses the two part (user and peer) config |
| 20:34.29 | AgiNamu | is there a problem with g729 clients and voicemail? |
| 20:34.33 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
| 20:34.44 | SexyKen | Hey guys I can't seem to dial internatinoally with my polyCOM sp ip600 -- it doesn't even make it to the server. I hit dial and it gives afast busy tone but nothing comes up on asterisks end |
| 20:34.49 | shmaltz | tzafrir_home, you around? |
| 20:34.56 | shmaltz | ~seen tzafrir |
| 20:34.58 | jbot | tzafrir <~tzafrir@62.90.10.53> was last seen on IRC in channel #asterisk, 1d 3h 53m 39s ago, saying: 'ManxPower, it is even something I have considered. My latest * packages have sqlite support'. |
| 20:35.17 | shmaltz | ~tzafrir_home |
| 20:35.30 | shmaltz | ~seen tzafrir_home |
| 20:35.32 | jbot | tzafrir_home is currently on #asterisk #debian |
| 20:35.39 | Moc____ | gezick: I mean the polycom FTP config file ? |
| 20:36.21 | *** join/#asterisk denon (denon@synapse.subneural.net) |
| 20:36.21 | *** mode/#asterisk [+o denon] by ChanServ |
| 20:36.31 | gezick | Moc____: krisk.org/asterisk/pmcom |
| 20:36.39 | gezick | er... /asterisk/pcom |
| 20:37.27 | Moc____ | get the untouch config file |
| 20:37.27 | gezick | when i try to make an outgoing call, i get the following |
| 20:37.56 | *** join/#asterisk clive-- (~pirch@rrba-146-103-206.telkomadsl.co.za) |
| 20:38.04 | gezick | Executing Dial("SIP/polycom1-2c4b","IAX2/username:password@gateway/1NXXNXXXXXX") in new stack |
| 20:38.12 | *** join/#asterisk eipi (~eipi@100-172-114-200.fibertel.com.ar) |
| 20:38.14 | gezick | called username@gateway/number |
| 20:38.16 | gezick | Call accepted by 66.234.228.160 (format ulaw) |
| 20:38.17 | gezick | <PROTECTED> |
| 20:38.17 | gezick | <PROTECTED> |
| 20:38.17 | gezick | <PROTECTED> |
| 20:38.37 | gezick | and then |
| 20:38.38 | gezick | <PROTECTED> |
| 20:38.38 | gezick | Mar 8 14:34:38 NOTICE[12968]: channel.c:1691 ast_set_write_format: Unable to find a path from ulaw to g729 |
| 20:38.50 | gezick | and then |
| 20:38.50 | gezick | Mar 8 14:34:38 WARNING[12968]: chan_sip.c:1829 sip_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 4/4) |
| 20:38.54 | AgiNamu | anyone know what the play and record no audio voicemail error is due to? |
| 20:38.58 | SexyKen | How do I dial internationally with my polycom |
| 20:38.59 | JerJer[mobile] | don't flood the channel |
| 20:39.00 | JerJer[mobile] | gezick: |
| 20:39.00 | gezick | will just keep repeating until i break the call |
| 20:39.16 | JerJer[mobile] | either fire up the G.729 codec or do not use G.729 |
| 20:39.20 | JerJer[mobile] | very simple |
| 20:39.21 | Moc____ | gezick: pasterbin.ca please |
| 20:39.26 | shido6 | back |
| 20:39.26 | Moc____ | gezick: http://www.freedomphones.net/polycom/files/ |
| 20:39.31 | JerJer[mobile] | front |
| 20:39.32 | shido6 | that was kewl... |
| 20:39.36 | Moc____ | get the SoundPointIP_SIP_1_4_1.zip |
| 20:39.39 | shido6 | cold out there |
| 20:39.44 | shido6 | lucky bastard |
| 20:39.51 | gezick | JerJer[mobile]: i've got disallow=all and allow=gsm in the global section now |
| 20:39.52 | Moc____ | and dont modify nothing except what you need changed. |
| 20:40.02 | SexyKen | HELLO/ |
| 20:40.09 | AgiNamu | So... Go2Call is big at the Asterisk Pavilion at VON, but they don't even mention Asterisk on their site :P |
| 20:40.13 | Moc____ | gezick: your polycom sip entry need to have disallow=all allow=ulaw |
| 20:40.23 | gezick | it has it as well |
| 20:40.43 | JerJer[mobile] | shido6: only 49 here today - was 60 something yesterday |
| 20:40.47 | *** join/#asterisk lyroy (~lyroy@picachou.csaffluents.qc.ca) |
| 20:40.57 | JerJer[mobile] | gezick: somebody is asking for G>729 and u cannot provide it |
| 20:41.03 | afe | anyone knows why Windows Media Player (10) reports an error when playing waw49 files generated by asterisk? |
| 20:41.05 | Moc____ | it -15oC here ;) |
| 20:41.17 | jero | hopefully not snowing anymore :p |
| 20:41.37 | Moc____ | JerJer[mobile]: it his Polycom phone, he got a modified polycom config from someone who probably force g729 ! |
| 20:42.12 | jero | anyone can recommend a good SIP phone, almost as robust as a Nortel office phone ? |
| 20:42.18 | *** join/#asterisk Nix (~Nix@81.213.125.220) |
| 20:42.22 | spackle | Is anyone using one of the legacy PBX-to-VOIP boxes from citel? |
| 20:42.34 | JunK-C | jero: polycom SIP500 |
| 20:42.35 | gezick | Moc____: i was looking for that, but just in sip.conf... now looking in ipmid.cfg i see <preferences voice.codecPref.G711Mu="2" voice.codecPref.G711A="" voice\ |
| 20:42.35 | gezick | .codecPref.G729AB="1"/> |
| 20:42.36 | Moc____ | Jero, the best phone on the market are the polycom rightnow |
| 20:42.53 | shido6 | IP 500 polycoms are $195.00 |
| 20:42.59 | jero | okay, do you know a dealer in montreal ? |
| 20:43.00 | Moc____ | gezick: there you go...use the original damn config.. |
| 20:43.09 | spackle | Jero: the Uniden UIP200 phones "feel" like a nortel phone. |
| 20:43.12 | AgiNamu | Is there a way to get access to the wav file from asterisk after a voicemail record? I want to do MP3 support. |
| 20:43.21 | jero | thanks for the info |
| 20:43.23 | Moc____ | jero, you got Nortel IP phones ? |
| 20:43.39 | jero | who didnt :) |
| 20:43.44 | jero | sorry |
| 20:43.50 | jero | nortel meridian phones |
| 20:43.53 | *** join/#asterisk Jason357 (~m00@adsl-68-77-94-87.dsl.ipltin.ameritech.net) |
| 20:43.55 | jero | you know the ones everyone has |
| 20:43.56 | Jason357 | Hello |
| 20:44.06 | SexyKen | Christ, I wish my fucking phone would just fucking dial internationally |
| 20:44.08 | SexyKen | THis is bullshit |
| 20:44.12 | Moc____ | jero, there is a nortel meridian phone Channel bank for 3k$ on voipsupply ;) |
| 20:44.12 | Jason357 | Has anyone ported * to other *NIX platforms? |
| 20:44.18 | Jason357 | I want to run it on NetBSD |
| 20:44.20 | jero | lol |
| 20:44.29 | Moc____ | SexyKen: you normally need to ask your provider for it |
| 20:44.32 | jero | Jason357: it runs on a variety of systems other than linux |
| 20:44.47 | jero | even on win32 afaik |
| 20:44.47 | SexyKen | •Moc• My phone wont even make it to the server. |
| 20:45.04 | SexyKen | It gives a fast busy signal when I press the dial button...never even tries to send the dial to asterisk |
| 20:45.08 | jero | :) |
| 20:45.17 | Moc____ | SexyKen: what type of phone ? |
| 20:45.30 | SexyKen | Polycom SP IP600 |
| 20:45.36 | *** join/#asterisk file (~file@sjcc30x31.sjccnet.com) |
| 20:45.37 | spackle | Moc, that's the thing I'm asking about - from citel. |
| 20:45.40 | Moc____ | SexyKen: you got your dialplan right / |
| 20:45.53 | Jason357 | jero, is NetBSD one of them? |
| 20:45.54 | SexyKen | •Moc• Let me show you |
| 20:46.27 | gezick | Moc____: someone might want to change the wiki, because the files that i used for config are the ones that are listed on the wiki page for the polycom phones |
| 20:46.32 | SexyKen | <PROTECTED> |
| 20:46.32 | SexyKen | <PROTECTED> |
| 20:47.03 | Moc____ | SexyKen: you do 9 to do international call ? |
| 20:47.07 | Jason357 | "Runs Solid on NetBSD" |
| 20:47.18 | SexyKen | •Moc• No what do you mean? |
| 20:48.08 | Moc____ | just to be sure |
| 20:48.22 | SexyKen | •Moc• Does the dial plan look right? |
| 20:48.22 | Moc____ | SexyKen: Polycom use a feature to see if that extention is valid on the PBX |
| 20:48.30 | Moc____ | try sip debug on your phone IP and try again |
| 20:48.34 | Moc____ | yes it look fine |
| 20:48.41 | SexyKen | Try sip debug on my phone? |
| 20:48.46 | *** join/#asterisk jsolares (~jsolares@200.30.141.85) |
| 20:48.56 | Moc____ | SexyKen: sip debug peer PHONEPEER |
| 20:49.08 | Moc____ | in asterisk, and try to dial your 011 # from the phone |
| 20:49.25 | jero | sflphone is going mac/win32 this summer |
| 20:49.25 | Moc____ | im sure it query #, but you dont have it configured correctly in asterisk |
| 20:50.08 | Moc____ | now gota finish testing all those providers for this nice little QoS list |
| 20:50.41 | tzanger | Moc____: eh? |
| 20:50.47 | SexyKen | Moc - 1.200/1.200 24.191.236.37 D N 255.255.255.255 1060 Unmonitored |
| 20:50.49 | jero | moc and JunKy you like the polycom ip500 ? |
| 20:50.51 | SexyKen | What would I eneter for PHONEPEER |
| 20:52.35 | SexyKen | sip debug peer 67.161.5.149:5060 |
| 20:52.35 | SexyKen | No such peer '67.161.5.149:5060' |
| 20:52.39 | SexyKen | ? |
| 20:53.15 | ruiner | maybe try without the port? |
| 20:53.19 | SexyKen | I did that too |
| 20:53.56 | *** join/#asterisk redtech (~reddog@redtech.user) |
| 20:54.02 | Juggie | has anyone played with * 2b channel transfer? |
| 20:54.17 | jontow | revision 1.54.2.1 |
| 20:54.17 | jontow | date: 2005/01/17 00:37:22; author: russell; state: Exp; lines: +1 -1 |
| 20:54.17 | jontow | Fix ringback on outbound skinny calls (bug #3295) |
| 20:54.21 | jontow | heh! damn yeah.. been waiting for that one :) |
| 20:54.33 | Moc____ | jero they are the best phone |
| 20:54.45 | *** part/#asterisk clint_ (~clint@snap.helixsystems.com) |
| 20:54.45 | JunK-C | Juggie: see http://bugs.digium.com/bug_view_page.php?bug_id=0003554 , maybe that would help. |
| 20:54.47 | SexyKen | Moc - any idea why I cant debug the peer |
| 20:55.03 | Moc____ | SexyKen the name of the phone, or to use the ip use sip debug ip xx.xx.xx.xx.... |
| 20:55.19 | SexyKen | Thanks :-) |
| 20:55.20 | SexyKen | That worked |
| 20:55.53 | AgiNamu | Dammit... I can't record voicemail :( |
| 20:56.48 | *** join/#asterisk newpers (newpers@ip24-56-8-180.ph.ph.cox.net) |
| 20:56.51 | ruiner | damn work...have to do other stuff before i can work on my asterisk box again |
| 20:56.53 | *** join/#asterisk zotz (~zotz@24.231.32.191) |
| 20:57.00 | ruiner | and i think i've figured out some of the problems |
| 20:57.01 | AgiNamu | Mar 8 15:56:47 WARNING[1800]: app.c:599 ast_play_and_record: No audio available on IAX2/TLF11004@TLF11004/10?? |
| 20:57.05 | SexyKen | Is there a less verbosity than 0? |
| 20:57.08 | jero | Moc: are the keys ergonomic/ the handset sits well in the cradle ? |
| 20:57.09 | *** join/#asterisk Gronker (~Gronker2@adsl-220-75-161.ags.bellsouth.net) |
| 20:57.27 | AgiNamu | SexyKen: sure, turn off warnings errors notices, etc. |
| 20:57.31 | Juggie | junk, i'm reading that, i'm compiling the latest cvs, i'm just wondering if anyone knows anything more about it. |
| 20:57.43 | Moc____ | jero the IP600 yea, IP500 is alittle less good, but they have to justify the price ;) |
| 20:57.44 | AgiNamu | which are next ot useless. I get a bloody warning when a dialed user is busy :P |
| 20:58.07 | JunK-C | Juggie: ive downloaded the pdf, but dont have time to read it :( |
| 20:58.09 | jero | I use snom 190s, |
| 20:58.24 | AgiNamu | and NOTICES for IAX internals that you absolutely should not care about |
| 20:58.30 | jero | their software is good, the design is cool but the ergonomics sucks |
| 20:59.16 | jero | even the grandstreams bt100 have a better handset cradle :) |
| 20:59.34 | ruiner | if i want to update asterisk by getting the newer tarball and compiling, will a make isntall overwrite my existing config files? |
| 20:59.43 | mikegrb | no |
| 20:59.49 | SexyKen | When I try to dial it returns address incomplete |
| 21:03.20 | Juggie | JunK-C, i am just looking to find out how i should implement it in asterisk, do i use cmd Transfer or cmd Dial, etc... |
| 21:03.30 | Juggie | does it require a certain switch type |
| 21:03.33 | Juggie | things like that |
| 21:04.07 | *** join/#asterisk Smythe (~Smythe@spock.cbcag.edu) |
| 21:05.09 | SexyKen | Moc you there? |
| 21:05.15 | SexyKen | Asterisk returns this: From: "Ken Sandell" <sip:ken@66.55.69.242>;tag=46BA2414-11DD4E1D |
| 21:05.16 | SexyKen | To: <sip:011447870699479@66.55.69.242;user=phone>;tag=as259d340c |
| 21:05.29 | JunK-C | i never see the need to the Transfer application. |
| 21:05.34 | JunK-C | always use Dial. |
| 21:05.54 | SexyKen | It says to sip: shouldn't it say to iax? |
| 21:06.06 | *** join/#asterisk arkaoss (~Saber@69.37.244.103.adsl.snet.net) |
| 21:06.22 | Juggie | JunK-C, me either, but i am wondering if its required for this patch |
| 21:06.29 | *** join/#asterisk Nwallins (~chatzilla@c-24-99-119-75.atl.client2.attbi.com) |
| 21:06.34 | *** part/#asterisk Nwallins (~chatzilla@c-24-99-119-75.atl.client2.attbi.com) |
| 21:07.13 | AgiNamu | voicemail uses local time? |
| 21:07.16 | AgiNamu | sigh... |
| 21:07.24 | Nugget | you can set the user's timezone in voicemail.conf |
| 21:07.40 | AgiNamu | and it'll autocorrect everything? |
| 21:07.58 | Smythe | hi, all! has anyone had experience tying a Mitel SX-200 light to Asterisk? I've got them talking, but can't get the number dial to be passed through from the Mitel -> Asterisk, and can't get Asterisk to dial through the T1 to the Mitel.......... |
| 21:08.05 | Nugget | I'm not sure what you mean by "autocorrect" |
| 21:08.18 | Nugget | as long as your server's timezone is set properly there's nothing to correct. |
| 21:08.28 | *** join/#asterisk R3DB0x (nobody@66.142.28.36) |
| 21:09.00 | *** join/#asterisk Inv_arp (junya@adsl-8-232-168.mia.bellsouth.net) |
| 21:09.31 | *** join/#asterisk dfunnell (~dfunnell@port-222-152-55-43.fastadsl.net.nz) |
| 21:10.09 | Nugget | it's not like the voicemails didn't come in when it says. it's just a matter of presentation |
| 21:10.11 | *** join/#asterisk lesouvage (~lesouvage@cc341200-a.assen1.dr.home.nl) |
| 21:10.50 | *** join/#asterisk afrosheen (~chatzilla@txprotoa2.august.net) |
| 21:12.00 | lesouvage | I converted the sounds (in dutch) from wav to gsm. They work in a way but much to slow and low. What is the proper kHz rate for asterisk? |
| 21:13.19 | newpers | 8 |
| 21:13.21 | newpers | i think |
| 21:13.52 | *** part/#asterisk arkaoss (~Saber@69.37.244.103.adsl.snet.net) |
| 21:14.43 | Darwin35 | Lucy I'm a homo. what for super |
| 21:15.24 | Darwin35 | 8000 |
| 21:17.23 | lesouvage | darwin35/newpers: thanks I convert again and let you know. |
| 21:20.06 | modulus_ | anyone here know what an IVR is? |
| 21:20.38 | jontow | modullus.. thats not a very well formed question, for this forum :) |
| 21:21.06 | afrosheen | ivr? uhmmm intravenous rat |
| 21:21.32 | jontow | well.. if you'd like stupid interpretations.. what about inverted rectum? |
| 21:21.56 | jontow | i like to think its a set of voice prompts with menu choices selected via standard DTMF type keypresses on a telephone |
| 21:22.05 | jontow | but maybe im narrow-minded :) |
| 21:24.27 | afrosheen | yeah we know what an ivr is, now what's the question :) |
| 21:25.20 | ruiner | how come when i try to use the use the transfer command while i'm calling into my asterisk box it says there is no call to transfer |
| 21:25.37 | ruiner | but i have an active sip channel |
| 21:25.52 | modulus_ | how do you dial out via that channel? |
| 21:26.14 | ruiner | i have not yet attempted outbound dials. i'm still really green here |
| 21:26.21 | rvhi | hi, how safe is extention reload command? |
| 21:26.36 | modulus_ | T = caller can transfer |
| 21:26.36 | rvhi | if you are in the middle of a call, would it break the system? |
| 21:26.51 | modulus_ | t = callee can transfer |
| 21:26.51 | modulus_ | rvhi, nope |
| 21:27.12 | ruiner | i've just got everything setup and i'm calling in. i get the demo sound but when i dial 600 to do the echo test nothing happens, so i was trying to manually transfer it to extension 600 |
| 21:27.23 | rvhi | e.g. if you are on prio 3 play a file, if you change prio 4 during the play |
| 21:27.27 | *** join/#asterisk ionix (~ionix@209.71.254.100) |
| 21:27.31 | rvhi | it won't be a problem? how? |
| 21:27.59 | rvhi | does it read all steps for a call in the begining? |
| 21:28.56 | modulus_ | yeah so you've all heard of ivr? |
| 21:29.19 | afrosheen | yeah |
| 21:29.22 | modulus_ | cool |
| 21:29.24 | modulus_ | what is it? |
| 21:29.31 | modulus_ | just a menu system? |
| 21:29.38 | afrosheen | it's part of asterisk, answers incoming calls, routes based on grouping, etc |
| 21:29.51 | modulus_ | sounds buzz word ish |
| 21:30.00 | modulus_ | another tla |
| 21:30.06 | afrosheen | naw |
| 21:30.13 | afrosheen | it's a critical component |
| 21:30.33 | modulus_ | # wc -l extensions.conf |
| 21:30.33 | modulus_ | <PROTECTED> |
| 21:30.37 | modulus_ | ick that's too large |
| 21:31.13 | nestAr | lol |
| 21:31.19 | *** join/#asterisk mafkees (~michiel@mafkees.xs4all.nl) |
| 21:31.27 | nestAr | mine is 473 |
| 21:31.28 | nestAr | :( |
| 21:31.30 | mafkees | heya all :) |
| 21:31.55 | Darwin35 | some one call me |
| 21:32.04 | mafkees | Thanks for the excelent product :) |
| 21:32.08 | JerJer[mobile] | mine is 202 |
| 21:32.39 | modulus_ | imma start cleaning my extensions.conf |
| 21:32.43 | mafkees | Can someone tell me where I can post a sample working config for a dutch pstn provider |
| 21:33.08 | Pinhole | mine is 398 |
| 21:34.41 | Pinhole | I've heard you can drastically reduce the size of extensions.conf with #include |
| 21:35.12 | afrosheen | yeah |
| 21:35.15 | Darwin35 | I was going to ask about that |
| 21:35.18 | afrosheen | that way you can bloat your includes instead |
| 21:35.28 | mafkees | lol afrosheen |
| 21:35.30 | Darwin35 | to put macros in 1 file and extensioons in the other |
| 21:36.29 | Pinhole | I think you can actually get it down to one line. |
| 21:37.31 | Pinhole | Or, you should be able to do it in two lines: [default]\nexten => _.,1,AGI(extensions.agi) |
| 21:37.51 | ruiner | anyone here have a Cisco SmartNet account? |
| 21:37.59 | *** join/#asterisk santiago (~santiago@63.245.86.95) |
| 21:38.41 | `Sauron | ruiner: Does it matter? |
| 21:38.56 | ruiner | yeah...I need an IOS image and don't have an account |
| 21:39.06 | jontow | mines 493 lines :/ |
| 21:39.06 | `Sauron | Bummer, isn't it? |
| 21:39.09 | jontow | for just a test pbx, hehe |
| 21:39.12 | ruiner | indeed |
| 21:39.18 | ruiner | my IOS image doesn't support voice |
| 21:39.24 | afrosheen | ruiner: ios image for what device |
| 21:39.30 | `Sauron | get the $9.99 smartnet account |
| 21:39.30 | SexyKen | I really need some help with international dialing. |
| 21:39.33 | ruiner | afrosheen: 3640 |
| 21:39.51 | afrosheen | ruiner: I have a cisco cd around here somewhere with all kinds of firmware on it..hang on |
| 21:40.06 | *** join/#asterisk bandrew (~Snak@c-67-184-114-237.client.comcast.net) |
| 21:40.14 | ruiner | need the enterprise feature set for 12.3(4)T11 |
| 21:40.31 | ruiner | c3640-js-mz.123-4.T11.bin |
| 21:40.51 | ruiner | `Sauron: $9.99/month/year? |
| 21:40.52 | jontow | hmm.. we have a 3640 voice gateway here somewhere |
| 21:40.55 | afrosheen | there's also a little freeware app for finding cisco stuff |
| 21:40.59 | `Sauron | year |
| 21:41.13 | bzzzz | Has anyone ever used video with H323 or OH323 ? |
| 21:41.19 | jontow | i don't really know what it does though.. i suspect its trunking into this arelnet box |
| 21:41.24 | bandrew | Hi folks, I'm trying to convince my boss to let me set up asterisk as our new PBX and have a couple questions. If you are using asterisk with analog phone lines, what do you do if there is a power outage? |
| 21:41.45 | `Sauron | bandrew: Buy an UPS |
| 21:42.05 | *** join/#asterisk zapa (zapa@200.92.145.186) |
| 21:42.05 | afrosheen | ruiner: http://www.gold-software.com/IOSHunter-file1808.html |
| 21:42.11 | Pinhole | bandrew, call the power company |
| 21:42.19 | bzzzz | What about video from H323 to SIP ? |
| 21:42.43 | bandrew | heh, thanks. What hardware do I need to buy to get asterisk to work with four analog phone lines? |
| 21:42.46 | *** join/#asterisk PTG123 (~PTG123@ip68-106-24-139.ph.ph.cox.net) |
| 21:43.17 | `Sauron | the 4 port FXO card |
| 21:43.20 | afrosheen | ruiner: or here http://www.brothersoft.com/Internet_Search_Tools_IOSHunter_20542.html |
| 21:43.35 | bandrew | Sauron, how much are those? |
| 21:43.37 | ruiner | afrosheen: thanks |
| 21:43.38 | afrosheen | bandrew: tdm400 with all fxs ports I believe |
| 21:43.48 | `Sauron | bandrew: Don't remember |
| 21:43.53 | `Sauron | couple hundred maybe |
| 21:44.05 | `Sauron | no dude those are fxo ports |
| 21:44.05 | `Sauron | :) |
| 21:44.07 | bandrew | Is that card the only thing I need besides the phones? |
| 21:44.13 | afrosheen | yeah |
| 21:44.14 | `Sauron | well |
| 21:44.17 | `Sauron | ethernet wiring |
| 21:44.18 | afrosheen | and a server |
| 21:44.22 | jontow | what are you going to use for phones? |
| 21:44.24 | `Sauron | probably a switch |
| 21:44.31 | afrosheen | electricity.. |
| 21:44.32 | *** join/#asterisk IOscanner (~IOscanner@c-67-162-251-133.client.comcast.net) |
| 21:44.42 | jontow | standard hardware type VoIP sets? |
| 21:44.42 | `Sauron | you'll need 5 lbs of copper |
| 21:44.46 | `Sauron | and 3.5 lbs of plastic |
| 21:44.48 | ruiner | `Sauron: does the $9.99 SmartNet account give you file access? |
| 21:44.49 | SexyKen | Anyone know why dialing international numbers on my polycom would give me a fast busy signal and not show anything in the Asterisk cli? |
| 21:44.52 | jontow | duct tape.. lots and lots of duct tape. |
| 21:44.56 | bandrew | right. I don't know what I'm going to use for the phones. So these fxs cards, they have four inputs and how many outputs? Is there a phone hub of some sort I need to get? |
| 21:44.57 | `Sauron | ruiner: yes |
| 21:45.04 | ruiner | URL? |
| 21:45.09 | jontow | fxo cards !#%*) |
| 21:45.13 | jontow | fxs are to plug phones into |
| 21:45.14 | `Sauron | ruiner: look at voip-info |
| 21:45.17 | jontow | fxo are to plug lines into |
| 21:45.20 | `Sauron | bandrew |
| 21:45.24 | `Sauron | you'll need the tdm400 |
| 21:45.27 | jontow | make SURE you have that straight.. |
| 21:45.28 | `Sauron | 4 fxo modules |
| 21:45.37 | `Sauron | X number of SPA-841 phones |
| 21:45.41 | `Sauron | a server |
| 21:45.42 | afrosheen | bandrew: ok let's rewind here |
| 21:45.47 | jontow | otherwise its gonna be really boring and it will fail miserably :) |
| 21:45.51 | afrosheen | bandrew: how do you want calling to be handled |
| 21:45.52 | `Sauron | and enough ethernet cabling to connect it all together |
| 21:46.09 | Pinhole | probably need a real phone for when the power goes out. |
| 21:46.11 | ruiner | `Sauron: thanks |
| 21:46.13 | bandrew | afrosheen: I don't know what you mean :) |
| 21:46.14 | afrosheen | bandrew: will you have 4 phones internally that dial out through the internet or use actual phone lines? |
| 21:46.25 | *** join/#asterisk waddy (waddy@66.90.92.190) |
| 21:46.28 | jontow | afro; 4 fxo ports.. 4 analog lines |
| 21:46.32 | bandrew | We'll have probably six phones. We want to use actual phone lines |
| 21:46.34 | jontow | afro; and X voip phones |
| 21:46.43 | afrosheen | ah ok |
| 21:46.55 | `Sauron | bandrew: then you'll need 6 spa841's |
| 21:46.59 | bandrew | It would be great to have a phone for in case the power goes out -- it would put my boss at ease |
| 21:47.01 | afrosheen | so yeah, a tdm400 |
| 21:47.02 | `Sauron | or any other voip phone of your choice |
| 21:47.07 | eipi | anyone tried a tdm02b on pci 2.1? works? |
| 21:47.20 | bandrew | Sauron, but I don't want to use voip. I want to use analog phones. |
| 21:47.23 | jontow | bandrew; your best bet.. set it up.. play with it :) |
| 21:47.25 | afrosheen | maybe a second tdm with a single fxo port for 911 rollover/emergency to pstn |
| 21:47.37 | `Sauron | bandrew: Internally in your company you want to use analog phones? |
| 21:47.53 | bandrew | We want to use phones that will work if the power goes out |
| 21:48.04 | `Sauron | Hehn. |
| 21:48.15 | afrosheen | bandrew: how often does the power go out |
| 21:48.15 | `Sauron | then go spend $50k on a nortel meridian system and an UPS |
| 21:48.17 | bandrew | Damn I clearly don't understand this stuff |
| 21:48.42 | jontow | oh nm |
| 21:48.45 | bandrew | afrosheen: I know it's stupid. I've got a boss that looks for every little problem he can find though |
| 21:48.49 | jontow | its a norstar.. the meridian is the one thats in use |
| 21:49.00 | `Sauron | Okay |
| 21:49.01 | `Sauron | bandrew |
| 21:49.04 | `Sauron | this is how it works |
| 21:49.15 | `Sauron | from the CO, you have 4 analog lines coming in |
| 21:49.21 | bandrew | what's the CO ha |
| 21:49.27 | bandrew | from the phone company? |
| 21:49.28 | afrosheen | central office |
| 21:49.30 | `Sauron | Central OFfice |
| 21:49.32 | `Sauron | phone company |
| 21:49.36 | bandrew | yay, okay |
| 21:49.38 | bandrew | proceed |
| 21:49.47 | `Sauron | from there, you use a tdm400 with 4 FXO daughter cards |
| 21:49.59 | `Sauron | that lets the analog lines connect to the server that runs asterisk |
| 21:50.03 | afrosheen | which is technically a single pci card with four ports |
| 21:50.07 | bandrew | so I have to install 5 pci cards in my PC? |
| 21:50.10 | `Sauron | no |
| 21:50.13 | bandrew | oh okay |
| 21:50.13 | bandrew | got it |
| 21:50.16 | *** join/#asterisk dnum (~dnum@adsl-68-73-197-64.dsl.sfldmi.ameritech.net) |
| 21:50.22 | `Sauron | 1 pci card, with 4 itty bitty cards that sit ontop of it |
| 21:50.27 | heison | does anyone know what the market share is for Cisco on IP telephony equipment for Q3 2004? |
| 21:50.31 | `Sauron | now |
| 21:50.34 | afrosheen | heison: too much |
| 21:50.38 | mafkees | heison: a lot |
| 21:50.41 | afrosheen | lol |
| 21:50.49 | mafkees | whehehe |
| 21:50.55 | heison | apparently not as much as Avaya |
| 21:50.58 | `Sauron | now comes the interesting part |
| 21:51.00 | mafkees | but the phones are great |
| 21:51.01 | afrosheen | reckon I'm a polycom man mahself |
| 21:51.06 | bandrew | Does the TDM400 include those FXO daughter cards? |
| 21:51.10 | `Sauron | Nope |
| 21:51.14 | mafkees | bandrew: no |
| 21:51.17 | dnum | Anybody willing to help me solve some Asterisk configuration problems? |
| 21:51.19 | afrosheen | bandrew: IF you order it that way |
| 21:51.27 | `Sauron | $125 for the tdm400, and approx. $75 for each FXO card |
| 21:51.29 | afrosheen | the tdm400 is flexible |
| 21:51.32 | `Sauron | WEll |
| 21:51.50 | `Sauron | they're priced seperately, but you can buy the "tdm400 + 4 fxo card" combo |
| 21:52.16 | afrosheen | yep |
| 21:52.22 | `Sauron | Now |
| 21:52.28 | mafkees | is there a way to buy them without using the yahoo shopping cart ? |
| 21:52.47 | afrosheen | ebay maybe |
| 21:52.50 | `Sauron | yeah, somewhere |
| 21:52.54 | `Sauron | http://www.digium.com/index.php?menu=wildcard_tdm400p2 |
| 21:53.00 | `Sauron | TDM04B - 4-port FXO bundle |
| 21:53.04 | dnum | Anybody know where I can go to ask questions about setting up Asterisk? |
| 21:53.12 | mafkees | dnum: ask |
| 21:53.12 | MikeJ[Jayden] | here |
| 21:53.28 | MikeJ[Jayden] | ~docs |
| 21:53.29 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 21:53.30 | afrosheen | dnum: first rule of irc, don't ask to ask |
| 21:53.34 | `Sauron | bandrew, next comes your office phones |
| 21:53.43 | `Sauron | this is where it gets tricky |
| 21:53.56 | `Sauron | you can either buy phones that are powered seperatetly, from a wall wart |
| 21:54.02 | MikeJ[Jayden] | dnum, read that stuff ^^, it will help, it is probably a better idea to do a little reading if you are new to this stuff |
| 21:54.05 | `Sauron | or phones that can use POE (power-over-ethernet) |
| 21:54.46 | mafkees | I have a etiquette question about the wiki. Can I just create a new page with sample configs for a dutch voip/pstn provider ? or do I have to make that a subpage of another ? |
| 21:54.48 | afrosheen | POE phones are generally more expensive and require backend resources like powered switches right? |
| 21:54.49 | `Sauron | if you get POE phones, you can just put one UPS where the switch is, and if you lose power, the phones stay up until the switch dies |
| 21:54.50 | heison | does anyone have access to statistics of VoIP usage in the US? |
| 21:54.59 | dnum | ok... it is really weird... I can call from my land line to my VOIP, and the sound works, but when I call from VOIP to landline, as soon as I answer, the sound stops working. One the VOIP side, if I press hold, the landline will hear the Asterisk hold music. |
| 21:55.23 | afrosheen | dnum: firewall |
| 21:55.26 | `Sauron | if you don't get POE phones, you'll have to figure some other way of keeping power to the phones when the mains power goes out |
| 21:55.28 | dnum | But again, when I call from landline to VOIP, everything works ok |
| 21:55.46 | dnum | I am running a firewall, but it does not say it is blocking anything |
| 21:56.05 | afrosheen | `Sauron: we have little ups' in each cube here |
| 21:56.12 | ScythelX | saying and doing are two seperate things |
| 21:56.16 | mafkees | afrosheen: nice " |
| 21:56.23 | `Sauron | afrosheen: imho, that's retarded, but if it works for you, that's great. :) |
| 21:56.23 | mafkees | s/"/!/ |
| 21:56.45 | afrosheen | `Sauron: they're for the phones and the pc's, each are 4 port and 2 cubes share each one |
| 21:56.56 | mafkees | we use laptops only, and have a ups powering our * box and cisco switch |
| 21:56.58 | afrosheen | so it's actually smart :) |
| 21:57.07 | dnum | This is my setup: kphone at 192.168.1.100, Asterisk at public IP, connected to ISP through DSL |
| 21:57.08 | mafkees | if power fails, we fire up our soft phone |
| 21:57.11 | *** join/#asterisk lesouvage (~lesouvage@cc341200-a.assen1.dr.home.nl) |
| 21:57.17 | afrosheen | oh no not kphone |
| 21:57.23 | lesouvage | darwin35/newpers: I have converted the files to 8000 and now there are working correct. |
| 21:57.30 | mafkees | install minisip |
| 21:57.33 | mafkees | ;) |
| 21:57.47 | bandrew | Sauron, that's great, thanks so much |
| 21:58.03 | Darwin35 | cool |
| 21:58.18 | bandrew | Is that SPA841 phone power over ethernet? |
| 21:58.21 | afrosheen | bandrew: I highly recommend the polycom 600 phones if you're going to use POE |
| 21:58.22 | lesouvage | darwin35/newpers: I have now a dutch speaking asterisk |
| 21:58.30 | BoRiS | bandrew: no |
| 21:58.36 | dnum | afrosheen: why would it work when I call from landline to VOIP, but not VOIP to landline? |
| 21:58.38 | Darwin35 | Boris |
| 21:58.41 | Darwin35 | whats up |
| 21:58.43 | BoRiS | Darwin!!!!!!! |
| 21:58.44 | BoRiS | Hey |
| 21:58.47 | bandrew | afrosheen: what do you like about the polycom? |
| 21:59.01 | mafkees | POE works great on Cisco too. Cisco switch for the power supply and cisco 7905 phones is our setup |
| 21:59.03 | afrosheen | bandrew: price, features, configurability, works with * |
| 21:59.12 | BoRiS | Just reinstall * from scratch and setting up a new dial plan.... you? |
| 21:59.14 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@l02m-12-87.d4.club-internet.fr) |
| 21:59.17 | `Sauron | hum |
| 21:59.22 | PoWeRKiLL | hi |
| 21:59.24 | `Sauron | the polycom 600 isn't bad |
| 21:59.30 | PoWeRKiLL | how to call a vonage subscriber via asterisk ? |
| 21:59.34 | Darwin35 | boris I have a extenisons file for you |
| 21:59.35 | afrosheen | we have a few of them, mostly have the 500's though |
| 21:59.40 | `Sauron | powerkill: dial their number |
| 21:59.43 | mafkees | `Sauron: they are hard to get in EU |
| 21:59.50 | afrosheen | the screen on the 600 is a beaut |
| 22:00.00 | `Sauron | mafkees: order online? :) |
| 22:00.11 | `Sauron | http://www.telephonyware.com/telephonyware/tw00043.html?mv_pc=A00015 |
| 22:00.18 | mafkees | yeah, but then the shipping will be double the phone price |
| 22:00.24 | `Sauron | eh |
| 22:00.31 | bandrew | One more question. So there's some kind of cable sending the phone signals out from the TDM400, right? What does that go to? Are all the phones dasiychaned or is there some kind of hub? |
| 22:00.36 | `Sauron | can't cost over $50 to ship a phone to the EU from here |
| 22:00.38 | *** part/#asterisk dnum (~dnum@adsl-68-73-197-64.dsl.sfldmi.ameritech.net) |
| 22:00.53 | afrosheen | bandrew: the phones plug into ethernet like your computers do |
| 22:00.57 | `Sauron | bandrew: the signal from the phone company to the tdm400 goes through regular phone wiring |
| 22:01.10 | `Sauron | the server has a network card in it |
| 22:01.15 | `Sauron | that plugs into a switch |
| 22:01.22 | `Sauron | all phones plug into said switch |
| 22:01.28 | afrosheen | yep |
| 22:01.36 | bandrew | So the phones and the server all plug into the same switch? Okay. Can I just buy a standard router or something? |
| 22:01.48 | bandrew | All the other computers on the network plug into this switch too, right? |
| 22:01.48 | mafkees | sure |
| 22:01.53 | `Sauron | buy a switch, not router |
| 22:01.57 | `Sauron | unless you need a router |
| 22:01.58 | afrosheen | no router, just a simple switch, maybe one with QoS or vlan support |
| 22:01.59 | ScythelX | anyone use odbcstorage for voicemail? |
| 22:02.26 | mafkees | but if you want to be sure that voip gets priority on your network get a switch that supports vlans/qos |
| 22:02.28 | Qwell | Say you have two extension 3000's. You have two external IPs that could possibly call you. If .101 calls, you want it to go to 3000a, if .102 calls, you want it to go to 3000b. Is that possible? |
| 22:02.36 | Qwell | ^^ assuming I'm understanding my boss right |
| 22:02.54 | afrosheen | mafkees: hence my suggestion :) |
| 22:02.59 | mafkees | :) |
| 22:03.14 | mafkees | was pressing enter while reading your line :) |
| 22:03.24 | afrosheen | man when we built our system, we upgraded our firewall and half the network to gigabit ,and we have zero issues with calls |
| 22:03.34 | `Sauron | same here |
| 22:03.47 | mafkees | still getting used to dvorak key layout. my typing speed is not up to the old yet |
| 22:03.51 | `Sauron | there's something about having dark fiber all across town that just makes me warm and fuzzy |
| 22:04.03 | bandrew | Ok. Is there any delay or weird eccentricities with using a phone system like this? |
| 22:04.03 | afrosheen | `Sauron: you must live in plano |
| 22:04.05 | afrosheen | :) |
| 22:04.11 | `Sauron | austin, tx |
| 22:04.17 | afrosheen | bandrew: depends on your netowrk traffic really |
| 22:04.27 | mafkees | and network quality |
| 22:04.31 | afrosheen | yep |
| 22:04.40 | CoaxD | bandrew: "like this". Like what? |
| 22:04.43 | mafkees | a cheap hub can mess up things real bad |
| 22:04.44 | bandrew | Also my boss wants to know if there is a cordless VOIP phone that works with asterisk |
| 22:04.53 | mafkees | there are several |
| 22:04.53 | `Sauron | we have the largest private network in town |
| 22:04.58 | bandrew | afrosheen: You mean local network traffic and quality, right? |
| 22:04.58 | `Sauron | bandrew: therer are several. |
| 22:05.07 | CoaxD | bandrew: sec |
| 22:05.13 | afrosheen | bandrew: there are some wifi phones but most of them suck..there are other ways to handle this though |
| 22:05.21 | `Sauron | They rate from "works really well, but is a pain to set up" to "uhh.... why?" |
| 22:05.25 | afrosheen | bandrew: yeah, local net traffic |
| 22:05.41 | bandrew | can anyone recommend a good switch with vlan and QoS? |
| 22:05.50 | ScythelX | build one |
| 22:05.54 | `Sauron | I like the 7920, but they're a pain, because of the central config/code download stuff |
| 22:05.54 | bandrew | god, this is going to be such a fun, geeky project |
| 22:05.55 | afrosheen | bandrew: do you want 'good' or 'cheap' or 'good and cheap' |
| 22:06.03 | mafkees | cisco or 3com |
| 22:06.05 | bandrew | sauron, what brand? |
| 22:06.08 | *** join/#asterisk bgmncwj (~chatzilla@c-67-175-65-122.client.comcast.net) |
| 22:06.17 | `Sauron | bandrew: good, cheap, pick one. |
| 22:06.27 | bandrew | i pick good |
| 22:06.29 | spackle | bandrew: Nortel switches |
| 22:06.30 | mafkees | `Sauron: I happen to like the idea of tftp setups |
| 22:06.46 | `Sauron | mafkees: I do as well, but it's a pain if you're onesy-twosey-ing it |
| 22:06.55 | mafkees | uhhuh |
| 22:06.55 | `Sauron | bandrew: we use exclusively cisco gear here |
| 22:07.03 | afrosheen | bandrew: cisco, layer 2 switch |
| 22:07.12 | CoaxD | bandrew: You really just want an ATA, man |
| 22:07.13 | mafkees | Cisco rocks, but is expensive |
| 22:07.14 | `Sauron | 7940/7960 desk phones, 7920 wifi phones, and 2560 poe switches |
| 22:07.16 | CoaxD | bandrew: Like a Sipura 2000 |
| 22:07.21 | `Sauron | err |
| 22:07.23 | CoaxD | bandrew: That way, you can plug a cordless phone into it |
| 22:07.26 | `Sauron | 3560 poe switches |
| 22:07.31 | `Sauron | the 2560 is a router |
| 22:07.33 | CoaxD | Mmmm. POE |
| 22:07.45 | rvhi | does 7920 wifi support sip? |
| 22:07.50 | `Sauron | yup |
| 22:07.53 | mafkees | yes |
| 22:07.55 | `Sauron | you have to do code upgrade on it |
| 22:08.00 | bandrew | holy number overload |
| 22:08.05 | rvhi | didn't show it on the web site |
| 22:08.07 | CoaxD | bandrew: You'll get used to it |
| 22:08.08 | bandrew | i guess i asked for it |
| 22:08.20 | ScythelX | hello all I'm setting up voicemail to read messages from the database in blob format - I have the table setup now do I edit extconfig.conf to point asterisk to start using the database to store its information? odbcstorage => odbc,mysql1,voicemessages isnt working |
| 22:08.25 | CoaxD | bandrew: Basically you're hearing numbers that are just model numbers of specific brands |
| 22:08.35 | `Sauron | the nice thing about the 7920, is it can do WPA authentication/encryption |
| 22:08.45 | `Sauron | bandrew: what coaxd said |
| 22:08.46 | CoaxD | bandrew: and the folks who are "seasoned voice over ip users" just learn them 1 at a time and they talk about the ones they are familiar with |
| 22:08.48 | `Sauron | ;) |
| 22:08.51 | mafkees | ScythelX: don't. file storage is more scalable |
| 22:08.51 | *** join/#asterisk Cherebrum (~jgarland@216.32.77.10) |
| 22:08.58 | bgmncwj | hi everybody, I've got a problem I'm hoping somebody can help me with. I installed asterisk@home, but none of the voicemail prompts will play. Any ideas? |
| 22:09.18 | CoaxD | bgmncwj: I would 'set debug 5' on the asterisk console, and watch what happens |
| 22:09.31 | CoaxD | bgmncwj: I would also check that /var/lib/asterisk/sounds/* exists |
| 22:09.37 | `Sauron | what version of *? |
| 22:09.39 | ScythelX | mafkees: well if I have 5 asterisk boxes that ser load balances connections do you suggest a network file server? then mount the voicemail dir? |
| 22:09.41 | *** join/#asterisk pfn (500@adsl-69-107-210-254.dsl.pltn13.pacbell.net) |
| 22:09.43 | CoaxD | bgmncwj: and /var/lib/asterisk/sounds/voicemail specifically |
| 22:09.47 | bandrew | alright, well you guys have answered all my questions, thanks so much for everything |
| 22:09.51 | `Sauron | I think in old (old) versions, you have to do answer() before the voicemail stuff |
| 22:09.56 | Cherebrum | I just got a pair of video sip phones today and I can't get them to register with my asterisk server. They will register with my 1.0.6 asterisk server but not my CVS-HEAD-03/07/05-10:23:38 anyone have any ideas.. remember... VIDEO PHONES! whee! |
| 22:09.58 | bgmncwj | CoaxD: Yeah, I've checked both of those, and there aren't any debug errors returned (everything looks good) |
| 22:10.03 | CoaxD | bandrew: You'll have more questions. guaranteed. |
| 22:10.08 | `Sauron | hehe |
| 22:10.09 | CoaxD | bandrew: But you're asking good questions |
| 22:10.13 | CoaxD | bandrew: If i might make a suggestion |
| 22:10.16 | mafkees | ScythelX: yes. Storing files in mysql blob is really a killer for your database server |
| 22:10.22 | CoaxD | bandrew: Visit this site: http://www.voip-info.org |
| 22:10.30 | bgmncwj | `Sauron: CVS from 3/7/05 |
| 22:10.40 | CoaxD | bgmncwj: Um, I would guarantee you there are errors happening |
| 22:10.43 | `Sauron | CoaxD: You did it. Now he'll stay up all night reading the site... |
| 22:10.44 | `Sauron | :) |
| 22:10.55 | mafkees | ;) |
| 22:11.03 | CoaxD | Sauron: Hehe. Just like every single one of us does as we get familiar with voip and asterisk |
| 22:11.04 | `Sauron | I remember staying up until 4am when I got my spa1001 |
| 22:11.10 | `Sauron | Yup |
| 22:11.13 | bandrew | ok CoaxD. will do |
| 22:11.14 | CoaxD | Sauron: See?? Same deal! *g* |
| 22:11.18 | `Sauron | and 2am every following day for a week |
| 22:11.25 | mafkees | same here |
| 22:11.28 | CoaxD | sauron: Same deal for me, only that was almost a year ago. hehe |
| 22:11.29 | `Sauron | then I crashed, overslept horribly for work |
| 22:11.37 | `Sauron | and got exited about the gumstix |
| 22:11.41 | `Sauron | which kept me up last week |
| 22:11.42 | ScythelX | mafkees: I'm still interested on the odbcstorage though, do you know how to implement it / is it a different voicemail application in asterisk or is it something as easy as extconfig.conf |
| 22:11.43 | mafkees | messed up social life to get my brain in sync with it |
| 22:11.44 | mafkees | :) |
| 22:11.48 | afrosheen | bandrew: go read up at amp.voxbox.ca while you're at it, it's a nice interface/config tool for asterisk, also google for asterisk@home |
| 22:11.51 | CoaxD | yeah, thats a common problem |
| 22:11.53 | `Sauron | once BV ports my number, I'll be up late with * again |
| 22:12.04 | CoaxD | one of these days, i'm going to have to rewrite my PBX config from scratch |
| 22:12.09 | bgmncwj | CoaxD: well that might be a problem /var/lib/asterisk/sounds/voicemail exists, with just a default directory under it. |
| 22:12.11 | CoaxD | I have at *least* a zillion contexts |
| 22:12.12 | `Sauron | and then I'll try to build the voipstix |
| 22:12.15 | `Sauron | Mmm. |
| 22:12.27 | CoaxD | bgmncwj: You need to be able to see what asterisk is doing |
| 22:12.36 | CoaxD | bgmncwj: My guess? You'll see errors at a debug level of 5 |
| 22:12.40 | bgmncwj | CoaxD: and then just the mailbox folders. |
| 22:12.41 | `Sauron | anyway |
| 22:12.42 | mafkees | ScythelX: I have no experience with * voicemail storage in mysql. but I did some applications in php with mysql blob store and in the end I recoded them all to use filesystem storage |
| 22:12.44 | `Sauron | time to go home from work |
| 22:12.49 | mafkees | works 30000% better |
| 22:13.16 | Cherebrum | did anyone ever work on video for meetme conferences? |
| 22:13.29 | rvhi | ok i still can't find any information from cisco web site about supporting sip on 7920 |
| 22:13.37 | rvhi | http://www.cisco.com/en/US/partner/products/hw/phones/ps379/products_qanda_item09186a00801739cd.shtml |
| 22:13.41 | ScythelX | I figured a network mount across all the servers would be better |
| 22:13.41 | mafkees | have fun `Sauron |
| 22:13.49 | rvhi | Q. Which protocol does the Cisco Wireless IP Phone 7920 use? |
| 22:13.49 | rvhi | A. The Cisco Wireless IP Phone 7920 is compatible with Cisco CallManager Version 3.2 and later, using SCCP. |
| 22:14.07 | bgmncwj | CoaxD: I'm at debug level 5, and it just states: executing Answer... exec Wait... exec VoiceMailMain... Playing 'vm-login'... |
| 22:14.17 | Qwell | can 7920 do sip? |
| 22:14.20 | mafkees | rvhi, they offer a sip firmware image to resellers |
| 22:14.43 | CoaxD | bgmncw: Do you hear any sound whatsoever? |
| 22:14.46 | rvhi | any idea where i can it? |
| 22:14.48 | ScythelX | rvhi: get a smartnet contract then you have access to download images from cisco |
| 22:14.50 | rvhi | get it? |
| 22:14.57 | rvhi | i have smartnet |
| 22:15.06 | rvhi | but no image on web site |
| 22:15.11 | mafkees | I've run chan_sccp for 2 weeks with my 7905 and I can say the sip image is WAY better |
| 22:15.18 | Cherebrum | www.cisco.com click "login" |
| 22:15.38 | rvhi | <PROTECTED> |
| 22:15.38 | rvhi | NOTE: |
| 22:15.38 | rvhi | There are currently no files for this type. |
| 22:15.40 | bgmncwj | CoaxD: Nope. Not from anything automated. I can call between clients (both softphones) and it works fine though. |
| 22:15.59 | CoaxD | bgmncwj: Okay. Yeah, there's something whacked with your asterisk |
| 22:16.25 | bgmncwj | CoaxD: sound I have sound files under that /var/lib/asterisk/sounds/voicemail folder? |
| 22:16.31 | Silik0n | any APX8000 + sip gurus around? |
| 22:16.39 | Juggie | cd asterisk-sounds;make install perhaps? |
| 22:16.43 | Juggie | did u install that package? |
| 22:16.54 | CoaxD | bgmncwj: No |
| 22:17.13 | Juggie | download the asterisk-sounds for what ever version you are running |
| 22:17.14 | Juggie | and install |
| 22:17.19 | CoaxD | bgmncwj: that is where messages are stored, etc |
| 22:17.21 | bandrew | holy hell that 3560 switch is 4 grand. Is there a more affordable switch you guys can recommend, like a couple hundred? |
| 22:17.25 | bgmncwj | Juggie: Yeah, I've done that twice actually. |
| 22:17.44 | CoaxD | bgmncwj: does /var/lib/asterisk/sounds have a bunch of vm-* files in it? |
| 22:17.49 | Cherebrum | I have a cisco partner account |
| 22:18.11 | *** join/#asterisk CosmicRay (~jgoerzen@2002:4463:7269:1:20e:a6ff:fe66:c5a3) |
| 22:18.15 | moonwick | um, yeah, you can always get a cheap, unmanaged switch |
| 22:18.17 | *** join/#asterisk W1thdraw (~Withdraw@ip68-5-125-44.oc.oc.cox.net) |
| 22:18.24 | bgmncwj | CoaxD: yeah there's quite a few there. |
| 22:18.37 | CoaxD | bgmncwj: Okay, well, something's DEFINITELY whacked with your * then |
| 22:18.47 | CoaxD | bgmncwj: Because its playing the sounds.. or at least it thinks it i |
| 22:18.47 | CoaxD | s |
| 22:19.09 | CosmicRay | Hello. I am experienced with linux but new to asterisk and PBXs... to get a basic experimental setup, with some SIP softphones only, would people suggest installing amportal, or slogging through the config files? |
| 22:19.29 | CoaxD | cosmicray: Learn how to do it by hand |
| 22:19.48 | CoaxD | cosmicray: Then, programs that help manage will be much easier to fix when they break something :) |
| 22:19.55 | CosmicRay | CoaxD: heh |
| 22:20.07 | CosmicRay | CoaxD: well, from what I gather, amportal cannot use an existing config file |
| 22:20.14 | CosmicRay | so if I want to switch to it later, I lose anything I've done |
| 22:20.32 | CosmicRay | is amportal generally considered to be a quality program? |
| 22:20.44 | CoaxD | CosmicRay: Never used it |
| 22:20.44 | *** join/#asterisk menger (~menger@static-88.243.240.220.dsl.comindico.com.au) |
| 22:20.54 | CoaxD | CosmicRay: Learn the syntax. You'll be better off. |
| 22:20.54 | bgmncwj | CoaxD: Yeah, that's what I was thinking too. Alright, well I'll try and install Asterisk@Home again and see if that fixes it. If not I guess I'll just install from scratch. |
| 22:20.56 | ScythelX | time to switch asterisk over to freebsd now |
| 22:21.05 | CoaxD | What the hell is this "Asterisk@Home" thing |
| 22:21.24 | CosmicRay | CoaxD: http://asteriskathome.sourceforge.net/ |
| 22:21.36 | CosmicRay | that's where I heard of asterisk management portal to begin with |
| 22:21.54 | CosmicRay | nice article about getting going with asterisk here: http://techdatapros.com/asterisk/ |
| 22:22.00 | lesouvage | CosmicRay: Try Xorcom rapid. I had it up and running within half an hour. It's a dedicated debian asterisk distribution that will arease everything from your harddisk during the instalation proces. |
| 22:22.06 | Primer | Chad Schwartz |
| 22:22.18 | CoaxD | CosmicRay: Ahh |
| 22:22.23 | CoaxD | Primer: Daniel Ceregatti |
| 22:22.27 | Primer | heh |
| 22:22.34 | bgmncwj | CoaxD: Yeah, that's the site. It's been on a couple of news sites recently. |
| 22:22.34 | CosmicRay | well, I can't use these dedicated distros for several reasons.... 1) I don't want to dedicate a PC to this task, and 2) my hardware is non-x86. |
| 22:22.39 | Primer | fancy seeing you here |
| 22:22.45 | CoaxD | Primer: Heh yeah :) |
| 22:22.46 | CosmicRay | I want to install asterisk on my existing debian/alpha system, basically. |
| 22:22.53 | CoaxD | Primer; I'm here quite a bit lately. Dunno why |
| 22:23.07 | KalD|Work | how does Dial() detect when a user hits #? does it listen for the dtmf tones or does it get a DTMF event from the channel? |
| 22:23.12 | PTG123 | ~seen czero |
| 22:23.13 | jbot | czero <~me@CPE0090f800c5b0-CM001225704b6e.cpe.net.cable.rogers.com> was last seen on IRC in channel #asterisk, 42d 19h 48m 51s ago, saying: 'blah bateris in my wireles mouse are dead :)'. |
| 22:23.13 | CosmicRay | lesouvage: I wonder if they publish source .debs that I cuold use... |
| 22:24.02 | mafkees | ScythelX: I run * on openbsd |
| 22:24.19 | lesouvage | CosmicRay: you could take a trial version of vmware and make a virtual pc to install the dedicated distribution on. For trial it's ok, it worked for me. |
| 22:24.22 | mafkees | all works great, cept meetme and iax trunking |
| 22:24.40 | CosmicRay | lesouvage: really, the timing is accurate enough to do it in vmware? wow. |
| 22:24.42 | mafkees | they need a timing device, but that has not been ported to openbsd yet |
| 22:24.45 | mafkees | :( |
| 22:24.52 | CosmicRay | lesouvage: I have a full vmware license already, might just have to try that. |
| 22:24.54 | *** join/#asterisk ReVoK (~ReVoK@82.224.60.46) |
| 22:24.56 | ReVoK | hi |
| 22:25.00 | PTG123 | vmware ROCKS |
| 22:25.14 | KalD|Work | bochs is good too |
| 22:25.15 | mafkees | and kills your server |
| 22:25.18 | CosmicRay | well, it is a mostly good implementation of an existing concept :-) |
| 22:25.21 | mafkees | bochs is way better |
| 22:25.31 | Nugget | way better in what respect? |
| 22:25.32 | CosmicRay | PTG123: I have had multiple issues with its networking and its support for an amd64 host |
| 22:25.46 | CosmicRay | mafkees: if "10 times slower" means "Way better", ok :-) |
| 22:26.01 | mafkees | vmware is just to slow for me |
| 22:26.04 | CosmicRay | oh yeah. |
| 22:26.05 | CosmicRay | <PROTECTED> |
| 22:26.09 | Nugget | and bochs is faster for you? |
| 22:26.13 | mafkees | yes |
| 22:26.13 | Nugget | I find that hard to believe |
| 22:26.16 | bsdfreak | whoa |
| 22:26.20 | bsdfreak | what teh fock |
| 22:26.24 | bsdfreak | extra chewy. |
| 22:26.39 | setjmp | how's going bsd? |
| 22:26.45 | lesouvage | CosmicRay: the nice thing about vmware is the snapshot feature. If you messup the asterisk server it's restored in seconds. |
| 22:27.00 | *** join/#asterisk goatmilk (~goatmilk@130-127-45-50.chouse.resnet.clemson.edu) |
| 22:27.09 | mafkees | lesouvage: I use rdiff-backup for that :) |
| 22:27.25 | ReVoK | hey guys i have a couple of basic questions ;) someone can help ? |
| 22:27.32 | CosmicRay | mafkees: rdiff-backup is great. ben is awesome. |
| 22:27.41 | KalD|Work | ReVoK, 10 PRINT HELLO 20 GOTO 10 |
| 22:27.43 | afrosheen | ReVoK: don't ask to ask, just go |
| 22:27.43 | spackle | has the Linksys PAP2 been hacked? |
| 22:27.49 | CosmicRay | mafkees: I use it as much as possible. I even added hacks to support OSX finder info |
| 22:27.55 | KalD|Work | ReVoK, does that help? =) we can all help you unless you really want to know BASIC |
| 22:27.57 | mafkees | bsdfreak: do you have any idea when zaprtc will be ported from freebsd to openbsd ? |
| 22:28.03 | ReVoK | sorry :x |
| 22:28.10 | *** join/#asterisk harryvv (~plato@S010600055d210201.vs.shawcable.net) |
| 22:28.25 | KalD|Work | ReVoK, what's your q? =) |
| 22:28.27 | ReVoK | well i can make call with sip between two Sjphone trough asterisk |
| 22:28.32 | ReVoK | but with h323 |
| 22:28.38 | *** join/#asterisk jets (~jetsn@guardian.pmt.org) |
| 22:28.38 | ReVoK | it doesn't work |
| 22:28.43 | mafkees | CosmicRay: nice :) I use it everywhere |
| 22:28.57 | ReVoK | it says, phone xxxx called |
| 22:29.04 | KalD|Work | ReVoK, are you calling thru a gateway? |
| 22:29.07 | ReVoK | and that no one answered |
| 22:29.10 | ReVoK | sure i am |
| 22:29.13 | ScythelX | mafkees: any issues - I orig tried on freebsd but back then there wasnt *any* support for zaptel |
| 22:29.16 | setjmp | I can't seem to get SJphone to work with it at all :( Though definately newbee on asterisk, and making ground :) |
| 22:29.33 | KalD|Work | ReVoK, this is an incoming call to a client or asterisk or? |
| 22:29.33 | setjmp | ScythelX: is now (FreeBSD) |
| 22:29.58 | mafkees | ScythelX: :(( my home machine is running openbsd 3.6 and * works great on it. |
| 22:30.02 | ReVoK | a sjphone calling another Sjphone |
| 22:30.05 | setjmp | Though still not as complete as linux (need to fix that :) ) |
| 22:30.16 | mafkees | but all the stuff that needs a timer is a nono for me |
| 22:30.22 | KalD|Work | ReVoK, sjphone -> * -> sjphone ? |
| 22:30.26 | ReVoK | yep |
| 22:30.43 | KalD|Work | ReVoK, sjphone [sip] -> * [h323] -> sjphone ? |
| 22:31.12 | mafkees | is the zaptel stuff supported on freedsd 5 ? |
| 22:31.13 | ReVoK | KalD|Work, Sjphone[h323] => * => Sjphone[h323] |
| 22:31.28 | KalD|Work | ReVoK, who prints the mesg? * or sjphone? |
| 22:31.35 | ReVoK | asterisk |
| 22:31.54 | ReVoK | Sjphone continue to ring in the wind |
| 22:31.58 | mafkees | would be great to see the zaptel stuff on openbsd |
| 22:32.03 | KalD|Work | ReVoK, do you have a context in oh323.conf? (or h323.conf) ? and in that context do you have exten => s,1,Answer ? |
| 22:32.12 | shepherd | just use linux :) |
| 22:32.16 | *** join/#asterisk [Paul] (~paul@80.100.33.108) |
| 22:32.18 | ScythelX | I dont like linux that much |
| 22:32.29 | shepherd | why? |
| 22:32.33 | Nugget | it's nice to be able to avoid linux when possible. |
| 22:32.34 | mafkees | that way you can implement a realtime failover setup for your PBX |
| 22:32.36 | shepherd | there is hardly any difference |
| 22:32.46 | setjmp | me either |
| 22:32.47 | mafkees | shepherd: that is NOT true |
| 22:32.53 | ScythelX | thats a debate |
| 22:32.59 | lesouvage | CosmicRay: If you try Xorcom and music on hold isn't working uncomment noload => chan_oss.so and noload => chan_alsa.so in modules.conf and restart asterisk from rapad-menu. That worked for me. |
| 22:33.01 | setjmp | (linux). Good R&D , though not the layout I need |
| 22:33.19 | ReVoK | KalD|Work, well i have an extansion yes and exten => 1001,1,dial(H323/1001) |
| 22:33.22 | lesouvage | rapad-menu=rapid-menu |
| 22:33.35 | mafkees | with linux you are stuck to the mess called iptables |
| 22:33.50 | [Paul] | has anyone a license key for eyebeam? |
| 22:33.57 | ScythelX | http://people.freebsd.org/~murray/bsd_flier.html |
| 22:33.58 | KalD|Work | ReVoK, ok - and you are registered w/ 1001 from on sjphone client? and do you see on the console that it is reaching this? |
| 22:34.01 | setjmp | FreeBSD has ipfilter, and IPFW, though they need configuration too, hehe |
| 22:34.02 | *** join/#asterisk denon (denon@synapse.subneural.net) |
| 22:34.02 | *** mode/#asterisk [+o denon] by ChanServ |
| 22:34.08 | *** join/#asterisk zarloc (~carlo@62.211.183.181) |
| 22:34.22 | KalD|Work | ~seen kram |
| 22:34.24 | jbot | kram <~mark@kram.digium.sponsor.pdpc> was last seen on IRC in channel #asterisk, 3d 18h 57m 48s ago, saying: 'oh :)'. |
| 22:34.28 | mafkees | setjmp: does freebsd has carp and pf altq stuff ? |
| 22:34.31 | shepherd | he's at von |
| 22:34.32 | *** part/#asterisk zarloc (~carlo@62.211.183.181) |
| 22:34.32 | KalD|Work | damn... |
| 22:34.38 | ReVoK | KalD|Work, we can reg with h323..? |
| 22:34.41 | spackle | ~von |
| 22:34.43 | Nugget | [Paul]: I have two copies of eyebeam. I suggest just buying x-pro, though. most of the gunk in eyebeam isn't functional through asterisk. |
| 22:35.06 | setjmp | carp in perl? |
| 22:35.09 | shepherd | if you want firewalling then just put a fbsd box infront of asterisk :) |
| 22:35.11 | KalD|Work | ReVoK, do this: exten => 1001,1,Dial(h323/1001@other.clients.ip.address,,r) and see if that works |
| 22:35.21 | shepherd | it needs all the cpu it can get anyways |
| 22:35.23 | mafkees | no, carp as ip failover |
| 22:35.47 | setjmp | ahh, not familiar with that. Once in a while I have a good look at linux, though remain FreeBSD. |
| 22:36.17 | setjmp | Not to knock linux. It's come a long long way since redhat was an easy own on efnet. |
| 22:36.22 | *** join/#asterisk Red_6 (~alex@m174.net81-66-29.noos.fr) |
| 22:36.25 | ReVoK | KalD|Work, asterisk shouldn't see in h323.conf the ip adress? |
| 22:36.25 | Red_6 | hi all |
| 22:36.28 | KalD|Work | hey guys - is there a way to pass vendor specific information thru IAX to a PRI line? |
| 22:36.32 | mafkees | setjmp: http://www.openbsd.org/faq/faq6.html#CARP |
| 22:36.45 | setjmp | Cool, will look :) |
| 22:36.52 | mafkees | it rocks |
| 22:37.06 | mafkees | it is kindda like VRRP from Cisco |
| 22:37.31 | KalD|Work | ReVoK, do you have the sjphone registered w/ *? Can you verify that * is calling the 2nd client? Have you tried turning on auto-answer on the 2nd client? |
| 22:37.47 | mafkees | but better. Daniel Hartmeier is one of the key developers. So it's at least as good as pf :) |
| 22:37.51 | KalD|Work | ReVoK, maybe - i always have the clients registered |
| 22:37.56 | setjmp | interesting. |
| 22:38.00 | Nugget | I love pf+altq. |
| 22:38.09 | Nugget | freebsd has carp in head, too. |
| 22:38.28 | mafkees | Nugget: ah, didn't know that. nice to know |
| 22:38.29 | setjmp | wouldn't surprise me. : |
| 22:38.30 | setjmp | :) |
| 22:38.39 | setjmp | head as in 6-CURRENT? |
| 22:38.46 | ReVoK | KalD|Work, u 'r using Sjphone too ?, cause in Sjphone u can only say to use H323 through a h323 gateway |
| 22:38.55 | mafkees | now let's hope it gets into netbsd |
| 22:39.16 | mafkees | would be spiffy to run it on 2 Nokia Communicator phones ;) |
| 22:39.20 | setjmp | probably will. |
| 22:39.24 | KalD|Work | ReVoK, yeah and that gateway is * right? Do you see if the sjphone client is registered w/ *? |
| 22:39.35 | eipi | anyone tried a tdm02b on pci 2.1? works? |
| 22:39.43 | setjmp | Familiar with the asterisk port maintainer on FreeBSD? |
| 22:39.53 | mafkees | real time failover between 2 Nokia communicator phones, now how is that ;) |
| 22:39.55 | ReVoK | nop in the consol i never see Sjphone(h323) reged with * |
| 22:40.15 | mafkees | setjmp: nope, I just installed the cvs version of * by hand |
| 22:40.40 | mafkees | cd asterisk && gmake && gmake install && gmake samples |
| 22:40.42 | setjmp | About tempted. Seems the port isn't that far along. Has zaptel and such.. But still 1.0.3 |
| 22:40.46 | mafkees | that worked great here |
| 22:40.58 | Nugget | yeah, on my freebsd asterisk box I just build from stable cvs. |
| 22:41.09 | Nugget | I'm not sure what the port does, but apparently I don't need it. :) |
| 22:41.19 | mafkees | me neither |
| 22:41.34 | mafkees | would love to see the zaptel working tho. |
| 22:41.50 | setjmp | Yeah, the build seems to work well. |
| 22:41.52 | mafkees | iax trunking with work seems nice for my home * box |
| 22:42.01 | setjmp | did on a couple different linux over the weekend. |
| 22:42.22 | mafkees | on Debian GNU/Linux all works great |
| 22:42.29 | setjmp | Just started playing with * friday, but going to be hard to stop now :-D |
| 22:42.29 | ReVoK | KalD|Work, are u using a GateKeeper? |
| 22:42.36 | mafkees | but I dont trust my public ip boxes to Linux |
| 22:42.43 | *** part/#asterisk bgmncwj (~chatzilla@c-67-175-65-122.client.comcast.net) |
| 22:42.52 | harryvv | Well my SAR com chief is interested to a point on asterisk. We had a emergency search activation and lots of people were calling the SAR building and no one was answering the phone :) a * box could have routed there calls to those people who were important. |
| 22:43.07 | setjmp | think I will 1.0.6 here . on RELENG_5 |
| 22:43.53 | *** join/#asterisk jbot (ibot@apt.bot.TimRiker.active.supporter.pdpc) |
| 22:43.53 | *** topic/#asterisk is Asterisk: The Open Source PBX || 1.0.6 Released || Dev Conf 1PM CST MARCH 10th -> IAX2/guest@66.250.68.194/996 || ClueCon Dev Conf Aug 3rd - 5th |
| 22:43.56 | harryvv | afro, what about zap you do not like |
| 22:44.08 | Nugget | no, I don't need zaptel for work. |
| 22:44.14 | mafkees | afrosheen: me too. but it's the only thing that supports timing for iax trunking and meetme |
| 22:44.17 | harryvv | only voip then |
| 22:44.22 | setjmp | what ver of FreeBSD Nugget? |
| 22:44.28 | Nugget | 5-STABLE |
| 22:44.53 | setjmp | cool. Think I'll grab 1.0.6 and give it a shot :) |
| 22:44.55 | harryvv | Anyone have a list of voip services in the pacific nw? |
| 22:45.01 | afrosheen | harryvv: oh, the fact that I am forced to reboot my * server every few days due to driver issues with zaptel |
| 22:45.21 | harryvv | afro really. Which zap cards? |
| 22:45.28 | afrosheen | harryvv: two tdm400's |
| 22:45.46 | mafkees | linux 2.6 ? |
| 22:45.47 | afrosheen | the second card starts acting crazy after a few days, static, popping, digital noise, not answering lines, etc. |
| 22:45.47 | harryvv | I cannot recall a time where I had to reboot my server for the x100p |
| 22:45.49 | afrosheen | 2.4.x |
| 22:46.10 | mafkees | SMP ? |
| 22:46.12 | Nugget | my tdm400p stops working every once in a while. |
| 22:46.18 | Nugget | I have to powercycle the machine to un wedge it. |
| 22:46.24 | Nugget | maybe once every two months |
| 22:46.30 | pimpwell | if my asterisk box has a cdr account, is their a way to give restricted CDR users to see what there status is? |
| 22:46.40 | pimpwell | CDR account = serice |
| 22:46.43 | pimpwell | service* |
| 22:46.45 | mafkees | there are some probs with the APIC stuff on SMP services |
| 22:46.59 | mafkees | s/services/servers/ |
| 22:47.04 | afrosheen | no smp |
| 22:47.08 | pimpwell | ya I'm shot right now, 1 sec |
| 22:47.09 | harryvv | nugget is there a way for a remote asterisk box to detect and reboot it? or mabey some other software on the system to detect and reboot it? |
| 22:47.21 | afrosheen | it's just a driver problem with the cards, I googled it and others have the same problem, always with 2 cards in the machine |
| 22:47.36 | Nugget | no, because it can't be rebooted. the card doesn't initialize unless it's a full powercycle boot. |
| 22:47.44 | mafkees | afrosheen: are they sharing resources ? |
| 22:47.48 | pimpwell | if my asterisk box does CDR, is their a way to give my customers access to the manager interface for real time information on the calls they placed without giving them access to everyone elses? |
| 22:47.48 | afrosheen | nope |
| 22:47.50 | harryvv | nugget you mean cold reboot |
| 22:47.53 | afrosheen | nothing common, trust me :) |
| 22:48.05 | afrosheen | common as in there's a solution |
| 22:48.09 | *** part/#asterisk Red_6 (~alex@m174.net81-66-29.noos.fr) |
| 22:48.11 | Nugget | was I unclear? |
| 22:49.15 | KalD|Work | afk |
| 22:49.17 | mafkees | I see no probs on my P4 with intel 865 motherboard here |
| 22:49.18 | harryvv | so two cards are a issue then. |
| 22:49.35 | mafkees | HT is disabled in bios |
| 22:49.48 | CoaxD | P4 sucks. You should run asterisk on a 386. |
| 22:49.49 | mafkees | as soon as I enable it, things go crazy |
| 22:50.05 | harryvv | hyperthreading |
| 22:50.11 | mafkees | CoaxD: XeoNs work great here |
| 22:50.12 | JunK-C | CoaxD: huh? |
| 22:50.29 | CoaxD | mafkees: How much load you put on 'em? |
| 22:50.34 | CoaxD | Junk-C: I'm in a mood. Don't mind me. |
| 22:50.34 | afrosheen | mafkees: how many tdm cards do you have |
| 22:50.35 | mafkees | as long as there is no zaptel hardware in it |
| 22:50.39 | harryvv | mak, i have a opteron here that unfortunatly is doubling as my asterisk box. |
| 22:50.48 | Damin | Opteron rocks.. |
| 22:50.49 | Damin | ROCKS! |
| 22:50.49 | mafkees | afrosheen: 3 |
| 22:50.58 | JunK-C | ya, opteron is powerful. |
| 22:51.00 | harryvv | damin its working smoothly |
| 22:51.05 | afrosheen | harryvv: hah, we have an amd64 3200 that's just a firewall, talk about wasted hardware |
| 22:51.06 | Damin | ROCKS! :) |
| 22:51.14 | mafkees | erm, 2 and one X100p |
| 22:51.14 | harryvv | afrosheen, hehehe |
| 22:51.14 | JunK-C | my agi machine is a dual xeon, i wonder if i should take an opteron. |
| 22:51.21 | CoaxD | afrosheen: But the thing is, stability accounts for something too |
| 22:51.27 | harryvv | afro, just put up a cheaper box for that. |
| 22:51.30 | CoaxD | afrosheen: And some of the older hardware out there just can't be as stable as some of the newer stuff |
| 22:51.31 | afrosheen | CoaxD: yeah it's solid as a rock |
| 22:51.32 | shmaltz | pimpwell, what do you use to display your CDR? |
| 22:51.39 | JunK-C | afrosheen: run distributed.net on it, im sure nugget gonna loves that :P |
| 22:51.46 | CoaxD | afrosheen: I can't quite explain it. But. That brand new P200MMX was fucking _stable_. |
| 22:51.51 | Nugget | yay! |
| 22:51.54 | afrosheen | hah |
| 22:51.55 | CoaxD | afrosheen: but after a few years of sitting on the shelves... the parts lose it |
| 22:51.56 | harryvv | so who here has * and digium boxes in a production enviroment and working well for say 6 months? |
| 22:52.00 | JunK-C | hhehehe |
| 22:52.03 | afrosheen | capacitors age |
| 22:52.04 | Damin | CoaxD: My P133 rocks! ROCKS! :) |
| 22:52.05 | harryvv | digium hardware i mean |
| 22:52.11 | CoaxD | Damin: Yeah, for sure |
| 22:52.12 | *** part/#asterisk Pinhole (~nuxi@lin-dsl-static-206-222-194-115.inetnebr.com) |
| 22:52.21 | CoaxD | afrosheen: They do do that, yes. But i think there's more than that |
| 22:52.23 | JunK-C | harryvv: not me. |
| 22:52.30 | afrosheen | harryvv: we're coming up on our first month and can't wait to get rid of our tdm cards |
| 22:52.32 | Damin | CoaxD: Except that as we speak, one of my techs is over at my house replacing a drive in it that failed. ;) |
| 22:52.36 | CoaxD | afrosheen: I think that software moves in a specific direction, and even tho the software still supports that old hardware, its just not TESTED there as much |
| 22:52.37 | JunK-C | but isnt trouble with hardware. |
| 22:52.44 | CoaxD | Damin: Hahahahaha |
| 22:52.48 | Nugget | <PROTECTED> |
| 22:52.53 | mafkees | I dont like the tdm cards either |
| 22:52.54 | harryvv | afrosheen, well what other substitute hardware is there? |
| 22:53.06 | CoaxD | mafkees: I have one.. good card |
| 22:53.13 | CoaxD | mafkees: its just 1 FXS interface tho |
| 22:53.18 | JunK-C | harryvv: whats ur uptime btw? |
| 22:53.19 | bandrew | If you want to use POE phones, does your network switch need to support POE? |
| 22:53.21 | harryvv | nugget, is that asterisk or your linux box uptime. |
| 22:53.27 | CoaxD | bandrew: No |
| 22:53.28 | Nugget | it's not a linux box. |
| 22:53.34 | Nugget | and it doesn't run asterisk. |
| 22:53.34 | mafkees | harryvv: we just trash the tdm hardware and get the PSTN provider to deliver an IAX2 trunk |
| 22:53.36 | CoaxD | bandrew: You can do POE manually too |
| 22:53.40 | mafkees | that works way better |
| 22:53.45 | CoaxD | bandrew: Just that it gets just as icky in the server room then |
| 22:53.45 | shmaltz | bandrew, how do you think the juice will get on the wire? |
| 22:53.52 | Juggie | my primary * box has been up 135 days |
| 22:53.56 | afrosheen | harryvv: it's just for incoming, we're going to a straight iax trunk to a provider with did's |
| 22:53.59 | CoaxD | bandrew: because you have to plug in the AC adaptors to the unused pairs |
| 22:54.04 | Juggie | would be up longer, only i kernel paniced it |
| 22:54.20 | CoaxD | bandrew: The thing is, it is just better to go with a POE switch if you really care to |
| 22:54.29 | harryvv | afrosheen yea I worry about putting a system with digium into a production enviroment then it goes down. |
| 22:54.39 | CoaxD | afrosheen: Have you worked with POE? |
| 22:54.45 | bandrew | Oh ok. So do POE switches start at a reasonable price? Can you get one for five hundred? |
| 22:54.52 | afrosheen | harryvv: our * server itself has been a champ and everyone loves the new phones |
| 22:54.58 | harryvv | cisco switches with the xl provide poe |
| 22:55.04 | CoaxD | bandrew: I'm sure you can. But it really matters how many ports you need |
| 22:55.08 | afrosheen | we have a few 2500xl's here |
| 22:55.13 | Damin | harryvv: Dude.. at VON, Intel has a booth and they are demonistrating carrier class Blade Servers using the ATCA specification.. FREAKING ROCK! Totally Redundant... |
| 22:55.14 | CoaxD | afrosheen: The reason I ask - is i'm trying to figure out how they regulate power on a POE switch |
| 22:55.15 | bandrew | I only need about 16 ports |
| 22:55.16 | harryvv | afrosheen great :) and no digium hardware? |
| 22:55.24 | Damin | harrvy: 16 servers in 7 U of rack space.. |
| 22:55.29 | CoaxD | afrosheen: I.e. some devices need 9v/1.5a, some devices need 5v/2a, some need 18v/.5a, etc. |
| 22:55.29 | Damin | harcy: Totally redundant.. |
| 22:55.30 | bandrew | All the switches my searches are turning up are like 4 grand |
| 22:55.32 | afrosheen | harryvv: once we are sure the iax trunk is perfect, then yes |
| 22:55.48 | harryvv | afrosheen whats the cost of the iax trunk? |
| 22:55.55 | CoaxD | afrosheen: Let alone the power polarity differences, blah |
| 22:55.56 | afrosheen | CoaxD: that's a valid point and the main reason we avoided using POE |
| 22:56.10 | harryvv | damin could not make von this time. |
| 22:56.12 | CoaxD | afrosheen: Well, i'm just wondering if POE switches have the capability of providing those wierdly different power requirements |
| 22:56.18 | afrosheen | CoaxD: you get one POE switch then you're stuck with specific phones |
| 22:56.20 | Juggie | i dont know what POE switch we have in our closet here, but most are auto sensing |
| 22:56.23 | mafkees | CoaxD: no |
| 22:56.25 | CoaxD | afrosheen: AHHHH |
| 22:56.25 | Juggie | i've used ours with cisco and mitel phones |
| 22:56.27 | harryvv | I want to start selling some asterisk boxes but need some phones. |
| 22:56.28 | pimpwell | if my asterisk box does CDR, is their a way to give my customers access to the manager interface for real time information on the calls they placed without giving them access to everyone elses? |
| 22:56.30 | Juggie | and it powers both no problem |
| 22:56.32 | mafkees | they only support one standard |
| 22:56.36 | CoaxD | *very* lame |
| 22:56.43 | CoaxD | (but useful if you care enough about that sort of thing) |
| 22:56.54 | afrosheen | we had a netgear poe adapter blow up one of our ciscos |
| 22:56.57 | CoaxD | the main reason for it is that you can provide backup power easily |
| 22:57.01 | afrosheen | zzzzzshshsh boom |
| 22:57.06 | CoaxD | afrosheen: *doh* |
| 22:57.14 | *** join/#asterisk ozJames79 (~james@CPE20320889-1842-1.gex.ncable.net.au) |
| 22:57.15 | afrosheen | that's what RMA is for baby |
| 22:57.19 | harryvv | afrosheen, what distro are you running asterisk ? |
| 22:57.20 | mafkees | that's the reason why I stick to Cisco |
| 22:57.20 | CoaxD | afrosheen: Haha. |
| 22:57.27 | zipp | pimpwell, write a manager proxy :) |
| 22:57.29 | afrosheen | harryvv: our iax trunk is cheap |
| 22:57.30 | CoaxD | harryvv: Why would that matter? |
| 22:57.38 | shepherd | coax: why not use a channel bank ? |
| 22:57.42 | shmaltz | anybody know if EMT64 from intel make any difference on asterisk? |
| 22:57.51 | shepherd | and analog phones :) |
| 22:57.51 | afrosheen | harryvv: we're dealing with some locals, txlink.net I believe..our did's are only 1.50 each |
| 22:58.03 | harryvv | afrosheen, about the price of a biz line? |
| 22:58.05 | harryvv | shees |
| 22:58.07 | zipp | afrosheen, where are you at? |
| 22:58.12 | afrosheen | dallas/richardson |
| 22:58.16 | zipp | SA here |
| 22:58.20 | afrosheen | cool |
| 22:58.26 | zipp | used to live on 75 near mockingbird :) |
| 22:58.28 | CoaxD | shepherd: cant exactly use a channel bank with sip phones |
| 22:58.37 | pimpwell | zipp: you serious? |
| 22:58.38 | afrosheen | zipp: good part of town |
| 22:58.39 | ozJames79 | Hey guys wondering if anyone could give me some pointers with * i have Cisco ATAs connected to my box but every now and then (not all the time) i get Registration failed messages get maybe 2 -4 then it stops and its all good again... any ideas? |
| 22:58.51 | harryvv | afrosheen okay well our SAR building does have Wan over dish to a local ISP and the building does have lan in it. |
| 22:59.00 | CoaxD | ozJames: Sounds like network issues |
| 22:59.14 | zipp | pimpwell, there will be one soon, I am working on a perl manager proxy |
| 22:59.19 | CoaxD | ozJames: not that i use ATA186's |
| 22:59.28 | mafkees | ozJames79: I had the same till I replaced my network switch |
| 22:59.35 | pimpwell | so currently manager interface is only to see everything? |
| 22:59.47 | ozJames79 | Thanks Guys i will look ay my network |
| 22:59.48 | afrosheen | a clean network is crucial for voip |
| 22:59.49 | zipp | afrosheen, 1.50 per month per incoming did, how many simeltaneous calls? |
| 22:59.50 | pimpwell | and can't filter certain data |
| 22:59.51 | *** join/#asterisk fafnir (~hello@tdds-gw.Moscow.gldn.net) |
| 22:59.59 | afrosheen | zipp: unlimited |
| 23:00.07 | afrosheen | but we pay bulk for outgoing minutes |
| 23:00.21 | zipp | afrosheen, come on, voicepulse connect is 7.5 per month 4 concurrent incoming calls |
| 23:00.25 | mafkees | how much per minute ? |
| 23:00.30 | mafkees | I pay 1 cent |
| 23:00.35 | afrosheen | 2 cents I think |
| 23:00.44 | mafkees | but that's in EU |
| 23:00.48 | harryvv | afrosheen, I would hopfully be interested in locating a local iaxtrunk provider in the bellingham to seattle coridor. See the whole idea is to make the asterisk box as reliable as possible. no worries about hardware failure or driver issues. |
| 23:00.57 | mafkees | calls to US cost me 4 cent a minute |
| 23:01.15 | harryvv | iax.cc is is located in miami? |
| 23:01.25 | mafkees | harryvv: do you need meetme or iax trunking ? |
| 23:01.31 | zipp | mafkees, why don't you just IAX to the us and us nufone or something? |
| 23:01.34 | afrosheen | zipp: call txlink and ask them prices :) |
| 23:01.49 | afrosheen | but again, 1.50 just covers each single DID incoming |
| 23:01.51 | zipp | afrosheen, is this iax termination? |
| 23:01.59 | mafkees | zipp: I'm looking into nikotel now, but that's only sip :( |
| 23:02.04 | afrosheen | yeah iax trunk from here to there, they provide termination |
| 23:02.16 | zipp | mafkees, nufone++ |
| 23:02.16 | afrosheen | they're big asterisk fans |
| 23:02.24 | mafkees | afrosheen: trunk or registration ? |
| 23:02.28 | afrosheen | trunk |
| 23:02.35 | afrosheen | we register locally on our * box |
| 23:02.36 | mafkees | zipp: will look at nuphone tomorrow |
| 23:02.52 | CoaxD | nuFone |
| 23:02.53 | mafkees | I'm on a bottle of wine now and the time is 1 AM |
| 23:02.55 | CoaxD | <F> |
| 23:02.55 | mafkees | ;) |
| 23:03.06 | mafkees | not going to mess with my configs now |
| 23:03.07 | mafkees | ;) |
| 23:03.43 | *** join/#asterisk Nukemizer (~Nuke@65.103.231.133) |
| 23:03.45 | mafkees | afrosheen: you have realtime failover for your * box ? |
| 23:04.10 | afrosheen | mafkees: of course, that's why we keep pstn lines and a tdm card |
| 23:04.25 | zipp | afrosheen, what is the company again? |
| 23:04.36 | afrosheen | but our sla with our ISP is deadly for them, so we don't worry too much about our connection going down |
| 23:04.47 | afrosheen | zipp: txlink.net |
| 23:04.50 | *** join/#asterisk fafnir (~hello@tdds-gw.Moscow.gldn.net) |
| 23:04.55 | mafkees | afrosheen: what about hardware failure ? |
| 23:04.56 | zipp | no, what does your company do |
| 23:05.06 | harryvv | mafk, not right now but the idea of the call going from bellingham to another state or across the country may no go over to well with the sar counsel. In two years somone broke or accessed the 911 valt bunker between our city and bremerton and cut the fiber optic cable cutting off all 911 service exccept one line and a majority of the calls could not leave the area. So the idea of a service outside the area may not go over well with t |
| 23:05.06 | harryvv | he city and sar counsel. |
| 23:05.28 | afrosheen | zipp: texasprototypes.com <--that's us |
| 23:05.32 | harryvv | FBI is investigating that one. |
| 23:05.59 | Juggie | harryvv, strikeing phone company workers cut the two fiber optic links (primary & backup) that serve the entire province of newfoundland in canada. |
| 23:06.02 | mafkees | http://www.covide.net |
| 23:06.04 | mafkees | that's me |
| 23:06.09 | Juggie | cut 500k people off fomr the rest of the world for like 12 hours |
| 23:06.14 | Nukemizer | I am SO happy I have to sing the praise of Digium Support -- After three weeks of trouble shooting I got a tech at Digium who really worked at solving my problem. The result is that my PRI works And I could not be happier :) *GRIN*GRIN* THANK YOU DIGIUM |
| 23:06.38 | Beirdo | Juggie: I hope they went to jail for a long time |
| 23:06.53 | CoaxD | Nukemizer: What was the problem? |
| 23:06.53 | *** join/#asterisk rustyryan (rustyryan@cpe-65-29-77-113.indy.res.rr.com) |
| 23:06.55 | Juggie | Beirdo, they could never prove who it was |
| 23:06.57 | Beirdo | and are there really 500k people in Newfoundland, or are you counting the seals too? |
| 23:07.06 | afrosheen | oh man 500k people couldn't call each other and talk about how cold it is |
| 23:07.08 | Juggie | but they knew it was someone who was on strike at the time |
| 23:07.17 | Juggie | Beirdo, yes there are :) 250k in st. john's and area. |
| 23:07.23 | harryvv | juggie wow did the rcmp find out who did it? |
| 23:07.27 | Juggie | no. |
| 23:07.32 | Beirdo | afrosheen: no, they could call each other, just not the rest of the world |
| 23:07.33 | afrosheen | it was beavers |
| 23:07.39 | Juggie | a group of people worked together |
| 23:07.44 | CoaxD | afrosheen: I hate those little bastards |
| 23:07.47 | Juggie | primary and back up went out at almost the same time |
| 23:07.55 | Juggie | and they were cut kilmoetres apart |
| 23:07.55 | CoaxD | afrosheen: My voip network is always being hindered by beavers |
| 23:08.00 | afrosheen | I know a guy that has a stuffed beaver he got on a hunting trip |
| 23:08.01 | Juggie | and they dug into the ground to cut them |
| 23:08.10 | Juggie | so they knew where the lines were |
| 23:08.15 | harryvv | juggie, now i suspect there are cameras and perimiter detectors on the outside of that valt? |
| 23:08.27 | Juggie | harryvv, they didnt go to any valt |
| 23:08.32 | Juggie | they just dug into the ground and cut it |
| 23:08.33 | *** join/#asterisk ManxPower (~eric@ip-209-16-83-10.i-55.com) |
| 23:08.38 | harryvv | it was in one of the main co offices? |
| 23:08.43 | harryvv | wow |
| 23:08.46 | Juggie | no DUH |
| 23:08.48 | mafkees | they should be hanged in public |
| 23:08.49 | Juggie | holy crap |
| 23:08.51 | Juggie | smarten up |
| 23:08.55 | Juggie | newfoundland = island |
| 23:09.04 | Juggie | big ocean inbetween |
| 23:09.07 | Juggie | it and canada |
| 23:09.13 | afrosheen | and icebergs |
| 23:09.18 | Juggie | its about 4hrs ferry from north sydney |
| 23:09.26 | epoch | I didn't even know newfoundland had phones! ;) |
| 23:09.31 | harryvv | the diagrams of where the cables are laid is usually only accessable by those techs? |
| 23:09.31 | Nukemizer | lol |
| 23:09.34 | mafkees | lol epoch |
| 23:09.42 | CoaxD | Nukemizer: ?? |
| 23:09.42 | Juggie | someone KNEW where to dig for and cut both the cables |
| 23:09.45 | Juggie | at the same time almost |
| 23:09.51 | Juggie | even tho they were km's apart |
| 23:09.53 | CoaxD | Nukemizer: (Answer my question, lamer!) |
| 23:09.59 | mafkees | Juggie: blame Al Quada |
| 23:10.03 | Juggie | so clearly it was a planned attack 99.9% chance by the telephony company workers who were on strike |
| 23:10.12 | Nukemizer | ouch sorry Coaxd.. missed that :) |
| 23:10.13 | harryvv | ohh sure |
| 23:10.18 | afrosheen | they had to fire up the backup smoke stacks |
| 23:10.22 | Nukemizer | pri would not come up and stay up |
| 23:10.31 | afrosheen | 3 clouds...that means they're hungry |
| 23:10.31 | Juggie | hah, they actually had microwave as a backup :P |
| 23:10.42 | Nukemizer | CoaxD: cylcing through the B channels |
| 23:10.46 | CoaxD | Nukemizer: And the problem was in driverland? |
| 23:10.54 | jets | hahaha our phone company's aerial drops are always being nibbled on by squirrels |
| 23:10.58 | jets | squirrels suck. |
| 23:11.00 | Nukemizer | CoaxD: Turns out Dell is not a good platform to build * on |
| 23:11.14 | CoaxD | Nukemizer: *lol* |
| 23:11.14 | afrosheen | dell = teh suxorz |
| 23:11.17 | shmaltz | I like this about windblows: |
| 23:11.18 | shmaltz | http://www.livejournal.com/community/cube_dwellers/101345.html?#cutid1 |
| 23:11.20 | Juggie | Nukemizer, i have two dells, both work fine. |
| 23:11.24 | zipp | Nukemizer, how do you figure? |
| 23:11.27 | CoaxD | Nukemizer: Crappy card problem is very much more likely :) |
| 23:11.35 | jets | nukemizer: i've ran * on dell's |
| 23:11.41 | Nukemizer | Dell and Crap are one in the same.. who knew ? |
| 23:11.43 | *** join/#asterisk bjohnson (~bjohnson@ip226-181.tor.istop.com) |
| 23:11.48 | xkev | if I'm in a while loop, how will n+101 be handled? |
| 23:11.49 | afrosheen | dell is french for turd |
| 23:11.52 | zipp | that is a bit to broad to state |
| 23:11.52 | afrosheen | ;) |
| 23:11.55 | Juggie | i have * on two dells with no problem |
| 23:12.03 | mafkees | Dells sound like vacuum cleaners |
| 23:12.03 | CoaxD | juggie: probably not with T1 cards |
| 23:12.10 | Juggie | wrong |
| 23:12.10 | afrosheen | I'll say it this way: there are good dells and bad dells |
| 23:12.14 | CoaxD | juggie: oh. ok. |
| 23:12.17 | Nukemizer | Digium says not all Dell but some Dell servers have trouble |
| 23:12.23 | Juggie | one has a 1 span card, and one has a 4 span card. |
| 23:12.24 | afrosheen | but you never know which dell is good or bad |
| 23:12.25 | Nukemizer | mine are in that pile |
| 23:12.38 | CoaxD | nukemizer: What is the specific prob |
| 23:12.43 | CoaxD | nukemizer: DId they say? |
| 23:13.07 | mafkees | Dells are bad in interrupt routing |
| 23:13.11 | CoaxD | Ah. |
| 23:13.16 | CoaxD | Yeah. That'll do it in a heartbeat |
| 23:13.17 | JerJer[mobile] | Nukemizer: that's news to me - we exclusively run dell hardware and have never had any issues |
| 23:13.23 | Nukemizer | CoaxD; no did not say or should say could not pinput other thatn loopback tests proved to be unsuccessful |
| 23:13.41 | afrosheen | JerJer[mobile]: you bought the expensive ones right? |
| 23:13.42 | CoaxD | Nukemizer: gotcha. yeah. |
| 23:14.00 | JerJer[mobile] | define expensive |
| 23:14.07 | afrosheen | JerJer[mobile]: raid arrays, smp, etc |
| 23:14.09 | Juggie | once you get the card on its own irq i've had no problems. |
| 23:14.12 | jets | JerJer: do you run 26xx/27xx series? |
| 23:14.13 | JerJer[mobile] | um no |
| 23:14.13 | afrosheen | make that scsi raid arrays |
| 23:14.20 | mafkees | I've seen a lot of stack corruption on Dell hardware |
| 23:14.38 | mafkees | mosly caused by wrong interrupt assignment |
| 23:14.40 | Juggie | theres no reason to run a huge raid array on a pbx |
| 23:14.44 | Nukemizer | TRied MANY MANY settings and many 3 techs at Digium attempted to fix |
| 23:15.00 | Juggie | i would run two drives in raid1 |
| 23:15.02 | Juggie | and that would be it |
| 23:15.12 | afrosheen | who said anything about huge? |
| 23:15.25 | Nukemizer | Tech finally made me try a fourth PC and that did it |
| 23:15.36 | mafkees | raid1 is fine for a pbx |
| 23:15.36 | JerJer[mobile] | i've got 400sc's runnng TDM400P |
| 23:15.43 | mafkees | in my opinion |
| 23:15.47 | harryvv | I have worked with a ton of dell hardware. We only had one issue and it was a bad bach of desktops that had failing onboard vidio electronics on it. Other then that fairly good system. BTW thay dont put these systems together in texas anymore? |
| 23:15.48 | JerJer[mobile] | and i've got 1750s running TE410P |
| 23:16.11 | JerJer[mobile] | even got a laptop with a docking station running a TDM400P card |
| 23:16.11 | Nukemizer | Dell 600SC was the machine in case that is of value to anyone |
| 23:16.20 | JerJer[mobile] | absolutey no trouble whatsoever |
| 23:16.23 | mafkees | the 2U dells have major heath problems |
| 23:16.34 | mafkees | seen a lot of them melt down |
| 23:16.39 | afrosheen | mafkees: I hope our 1u here doesn't get any hotter |
| 23:16.49 | *** join/#asterisk nel (~oeo@199.75.113.128) |
| 23:16.51 | CoaxD | mafkees: I still run PowerEdge 350's.. Those are the 1U's. They're finicky about what RAM you put in 'em, btu thats about it. Great boxen |
| 23:16.53 | Nukemizer | 600SC = (Somewhat Crappy) |
| 23:16.53 | zipp | I buy the cheapest athlon pc's I can find |
| 23:16.55 | nel | hello |
| 23:17.06 | nel | I need help registering a polycom soundpoint ip 500 on asterisk |
| 23:17.08 | mafkees | our colocating provider has a seperate room for Dell servers |
| 23:17.13 | JerJer[mobile] | and i think the latest system was a 750sc with a TE405P |
| 23:17.14 | afrosheen | nel: ok what's the problem |
| 23:17.17 | nel | I followed the instructions on the wiki |
| 23:17.20 | mafkees | with extra fire prevention systems |
| 23:17.32 | nel | and I still got the message chan_sip.c:7691 handle_request: Registration from '"enc" <sip:10.10.5.45@10.10.5.45>' failed for '10.10.12.54' |
| 23:17.37 | nel | the phone does not register |
| 23:17.44 | nel | I tried calling ext 500 |
| 23:17.49 | nel | and it doesn't work either |
| 23:17.50 | *** join/#asterisk hermie (~nick@24.236.167.53.bay.mi.chartermi.net) |
| 23:17.54 | nel | for outgoing or incoming calls |
| 23:17.58 | nel | I just hear dialtone |
| 23:17.58 | JerJer[mobile] | your sip.conf is not rght |
| 23:18.03 | afrosheen | yep |
| 23:18.08 | JerJer[mobile] | or the phone is not sending the right username and/or secret |
| 23:18.13 | JerJer[mobile] | very simple |
| 23:18.17 | nel | is there any way to see that |
| 23:18.49 | *** join/#asterisk jbot (ibot@apt.bot.TimRiker.active.supporter.pdpc) |
| 23:18.49 | *** topic/#asterisk is Asterisk: The Open Source PBX || 1.0.6 Released || Dev Conf 1PM CST MARCH 10th -> IAX2/guest@66.250.68.194/996 || ClueCon Dev Conf Aug 3rd - 5th |
| 23:18.57 | Sed_dinner | mafkees: they dont' make hardware anymore, do they? |
| 23:19.20 | KalD|Work | ~bzflag |
| 23:19.21 | jbot | hmm... bzflag is a 3D internet multiplayer multiplatform (linux, irix, win32, etc) opensource opengl Battlezone capture the flag game of the same name that you must try at http://BZFlag.org/ or a continual development project with periodic gaming interuptions. See also TimRiker |
| 23:19.24 | nel | I just see NOTICE[2093]: chan_sip.c:7691 handle_request: Registration from '"enc" <sip:10.10.5.45@10.10.5.45>' failed for '10.10.12.54' |
| 23:19.26 | mafkees | Sedorox: no, they went out-of-business |
| 23:19.36 | Sedorox | hmm.. I thought they just changed stuff... |
| 23:19.37 | Sedorox | weird |
| 23:19.55 | KalD|Work | hmm... wonder if Tim put that in there =) |
| 23:19.59 | nel | is there anyway to see more mesages |
| 23:20.08 | mafkees | jbot: try wesnoth |
| 23:20.19 | Nukemizer | To the topic earlier, are you sure it was fiber and not string that got cut ? or perhaps the rope that is used for to pull the ferry ? |
| 23:21.04 | nel | this is my sip.conf file entry for the phone, can I paste it? |
| 23:21.07 | Juggie | Nukemizer, quite sure, everything on the island runs off those two fiber lines... |
| 23:21.09 | Juggie | everything was down |
| 23:21.24 | Juggie | interac, bank machines, internet, all phone service etc. |
| 23:21.25 | KalD|Work | will any of the digium products work on Sparc/AXP systems? |
| 23:21.45 | xkev | kal, if you run linux on a sparc64, you have larger problems :) |
| 23:21.56 | nel | JerJer have you used polycom Ip 500 ? |
| 23:21.58 | xkev | I bet you'll find issues with the 32-bit userland/64-bit kernel |
| 23:22.06 | JerJer[mobile] | nel: some |
| 23:22.17 | nel | you never had the problem I'm having |
| 23:22.18 | nel | ? |
| 23:22.25 | xkev | nel, i have extensive polycom exp. |
| 23:22.35 | justinnnn | t31.c:66: unknown field `s_regs' specified in initializer |
| 23:22.37 | Smythe | can anyone give me a hint as to why I can't dial through to my Mitel PBX? I can receive incoming calls - connected with a T1 |
| 23:22.37 | xkev | nel, you have the wrong username or password |
| 23:22.40 | justinnnn | whats that mean.. when i try install spandsp ? |
| 23:23.04 | Juggie | Smythe, i am connected to a mitel switch of some sort and it works fine |
| 23:23.09 | xkev | [100] secret=foobar... set 100 and foobar in the phone. |
| 23:23.13 | Juggie | dont ask me what it is. |
| 23:23.32 | Juggie | i just took the settings they gave me and it worked. |
| 23:23.39 | xkev | nel I can send you my polycom config snippets and sip.conf |
| 23:23.55 | Smythe | I didn't have anyone give me settings, which is part of the problem...... |
| 23:24.13 | Juggie | Smythe, then you need to find the coding and sync on the port you are on |
| 23:24.16 | nel | xkev: do you use type=friend on sip.conf or the peer and user approach? |
| 23:24.18 | Juggie | and switchtype |
| 23:24.23 | xkev | friend |
| 23:24.25 | *** join/#asterisk firestrm (~pcott@s206-116-63-85.bc.hsia.telus.net) |
| 23:24.27 | Smythe | it is like the Miteldoesn't detect the call originating from Asterisk |
| 23:24.32 | harryvv | juggie there is alot of mitel phones in the bc gov offices |
| 23:24.35 | justinnnn | anyone ? |
| 23:24.43 | nel | xkev: that would be great, could you please send it to sosico@gmail.com ? |
| 23:24.45 | Juggie | harryvv, i have mitel phones on my desk |
| 23:24.48 | nel | or dcc |
| 23:24.53 | nel | whatever is easier for you |
| 23:24.56 | xkev | [foo]; username=foo; secret=bar; insecure=no; type=friend |
| 23:25.18 | Smythe | can I have the coding and sync wrong, and still receive calls? |
| 23:25.29 | xkev | nel sending |
| 23:25.32 | harryvv | So someone explain to me what is inside a standard pbx. Is it propriatory phone based electronics hardware PS and Battery? I have not looked inside one before. |
| 23:25.34 | xkev | I can't dcc from here |
| 23:25.49 | nel | could you send it to sosico@gmail.com ? |
| 23:25.51 | nel | please? |
| 23:25.53 | Juggie | Smythe, does receiving calls work? |
| 23:25.55 | firestrm | if i have a machie with a tdm400 with 2 fxs 2fxo and i make samples that should get a demo runing on it right out of the box right? or is there some additional config i eed to do? I just get fastbusy whe tryig to dial extensions from the fxs port |
| 23:25.58 | Smythe | yes |
| 23:26.10 | Juggie | no alarms on your pri? |
| 23:26.17 | Smythe | works nicely, after a lot of tinkering |
| 23:26.26 | Juggie | what happens when you try to dialout |
| 23:26.31 | xkev | nel sending |
| 23:26.49 | harryvv | is there a wiki that can send messages to say a cell phone or other network device when error messages do occure? |
| 23:27.00 | Smythe | I dialout, and asterisk shows the channel, but I don't see anything in the log on the Mitel |
| 23:27.14 | Juggie | what do you hear on the sip phone? |
| 23:27.41 | xkev | ooh someone has a mitel? |
| 23:27.48 | mafkees | I'm off, time to get some sleep so I won't be wasted on the CEBIT in germany tomorrow |
| 23:27.51 | xkev | how is it? |
| 23:27.57 | Smythe | the ip phone rings |
| 23:28.13 | Juggie | xkev, i have the 5055 getting 5220's soon |
| 23:28.15 | Smythe | but nothing on the mitel side |
| 23:28.23 | PTG123 | ok i bought a polycomm ip500 and a cisco 7960 today :) woo hoo |
| 23:28.23 | PTG123 | heh |
| 23:28.31 | Juggie | Smythe, the ip phone rings when u call incomming? |
| 23:28.41 | Juggie | what about when u try and dial out over the pri |
| 23:28.47 | Darwin35 | ok the broadvoice patch does not work |
| 23:28.48 | mafkees | we're at HAL 5, stand F18 |
| 23:28.54 | Darwin35 | it screws thigns up |
| 23:29.06 | mafkees | Covide BV ;) |
| 23:29.26 | mafkees | see you all there |
| 23:29.28 | mafkees | whehehehe |
| 23:29.37 | Smythe | mitel - > ast works fine ast - > mitel rings on ip phone side, but nothing on mitel |
| 23:29.46 | *** join/#asterisk gongoputch (~kseel@pcp01486721pcs.limstn01.de.comcast.net) |
| 23:30.29 | Darwin35 | frick |
| 23:30.33 | Juggie | Smythe, show me console logs from a dial, and your extensions.conf paste them to pastebin.ca |
| 23:30.39 | mafkees | we will be demonstrating a webbased groupware application with asterisk integration there |
| 23:30.41 | Juggie | and hurry because i'm leaving. |
| 23:31.23 | mafkees | toodles |
| 23:31.53 | KalD|Work | Smythe, could it be the number format? have you tried unknown? |
| 23:32.15 | gr8nash | just throwing it out there. anyone know why i would get a 404 not found from astrix when i try and connect to broadvoice |
| 23:33.05 | gr8nash | that was in the "sip debug" |
| 23:33.18 | *** join/#asterisk lesouvage (~lesouvage@cc341200-a.assen1.dr.home.nl) |
| 23:34.08 | Smythe | ok - its at http://pastebin.ca/7043 |
| 23:34.41 | lesouvage | On http://www.ctitec.com/asterisk/sounds.htm there seems to be a listing in written of all the sound files. When I go there the domain seems to be available. Does anybody of you knows another link where I can download the list and text of the sounds files? |
| 23:35.02 | nel | what firmware do you use on the Polycom 500? I'm using SoundPoint_IP_bootrom_2.4.1_w-SIP.zip |
| 23:36.29 | Juggie | Smythe, i need to know those global variables |
| 23:36.35 | Juggie | i also need to see what happens after the dial |
| 23:37.06 | Smythe | KaID - I have have to plead ignorance - what do you mean by number format? I have created an ext 1221, which dials to the same ext on the mitel - no wildcards |
| 23:37.15 | *** join/#asterisk j0 (~dan@S010600095b00a5a9.vc.shawcable.net) |
| 23:37.31 | Mavvie | who did here yesterday talk about Pingtel? |
| 23:37.39 | firestrm | help!!, cant get tdm to dial extensios.. i get fastbusy on the first dtmf digit.. ive loaded the samples (make samples) it answers incoming zap (fxo) no problem and i ca dial ito the fxs extension.. |
| 23:37.41 | Juggie | also, what are you trying to accomplish here ${EXTEN:${TRUNKMSD} |
| 23:37.59 | KalD|Work | Smythe, look in /etc/asterisk/zapata.conf and specifically at the pridialplan= line |
| 23:38.50 | Juggie | Smythe, you are going to have to respond quicker or i am going to leave. |
| 23:38.52 | *** join/#asterisk syslod (~yurplsl@65.114.0.198) |
| 23:39.29 | Smythe | I updated http://pastebin.ca/7044 with the globals - this is the out of the box config, and so I think that is the number of digits to strip, which is set to 0. |
| 23:39.54 | syslod | Hello |
| 23:40.14 | Smythe | KaID - this is a T1, not a pri, unless that can still effect it |
| 23:40.30 | KalD|Work | Smythe, same diff in this regard |
| 23:40.48 | Juggie | Smythe, is group2 the proper group of lines? |
| 23:40.55 | Juggie | if you only have 24 it should be group one. |
| 23:40.59 | syslod | Anyone seen anything on getting busy and unavailbe working with multiline phones? |
| 23:41.13 | Juggie | change that Dial to be Dial(Zap/g1/${EXTEN}) |
| 23:41.15 | Juggie | and try |
| 23:41.31 | Smythe | I have it going to a specific channel until I knew it was working |
| 23:41.55 | Smythe | I will set it to g1 to see what happens |
| 23:42.13 | KalD|Work | Smythe, how many cards do you have in the box? Just 1? use g1 then |
| 23:42.14 | *** join/#asterisk syamajala (~syamajala@h0030bd1e9b96.ne.client2.attbi.com) |
| 23:42.19 | syamajala | hi |
| 23:42.23 | Smythe | just the one |
| 23:42.38 | syamajala | just found the asterisk site |
| 23:42.49 | syamajala | and i am really interested |
| 23:43.00 | KalD|Work | syamajala, welcome |
| 23:43.24 | harryvv | I wonder if netsurfer got run over by a truck |
| 23:43.38 | syamajala | i don't really know much about phone systems |
| 23:44.17 | syamajala | my mom does |
| 23:44.22 | syamajala | but she doesn't know anything about linux |
| 23:44.58 | syamajala | my parents own a doctor's office and they would like to replace their phone system |
| 23:45.20 | harryvv | syam you are in the right channel |
| 23:45.30 | syamajala | they were thinking about getting phonetree but she said that asterisk seems to do everything phonetree does |
| 23:45.46 | harryvv | what is phonetree |
| 23:45.47 | Smythe | Juggie - no go - the ast console shows nothing after it dials - strange |
| 23:46.10 | Juggie | Smythe, try the prilocaldialplan then, i have to go. its in /etc/asterisk/zapata.conf |
| 23:46.31 | insomni | syamajala, do you know anything about linux? |
| 23:46.35 | syamajala | yeah |
| 23:46.37 | Juggie | syamajala, your looking at 2-3k for server, and phones, depending on how many lines you need. |
| 23:46.47 | harryvv | sounds as if phonetree is another voip provider for consumers. |
| 23:46.47 | Juggie | more if you need to pay someone for setup. |
| 23:47.08 | syamajala | well i got a bunch of systems i could use for a server |
| 23:47.25 | Juggie | p3+512mbram+ |
| 23:47.30 | KalD|Work | syamajala, what kind of phone connection do they have now? |
| 23:48.03 | syamajala | i need to talk to her |
| 23:48.16 | Juggie | likely just a few analog lines. |
| 23:48.22 | syamajala | i know they have 1 analog line |
| 23:48.24 | KalD|Work | syamajala, more specifically what kind of connection to the telco - and what kind of connection from the current phone system to the hand sets |
| 23:48.32 | Smythe | so use prilocaldialplan=line? |
| 23:48.44 | *** join/#asterisk r0d3nt|m (anonymous@soveliss.luniac.com) |
| 23:48.46 | Juggie | Smythe, no try the different settings in there to see if any work. |
| 23:48.47 | KalD|Work | Smythe, try pridialplan=unknown |
| 23:48.54 | Smythe | cool |
| 23:49.15 | *** join/#asterisk mamcinty (~mamcinty@adsl-068-209-174-113.sip.int.bellsouth.net) |
| 23:49.15 | Juggie | you can try turning in pri debug |
| 23:49.16 | KalD|Work | Smythe, then do a stop now and reload asterisk |
| 23:49.18 | Juggie | to see whats going on |
| 23:49.37 | syamajala | i'll come back when she isn't busy |
| 23:49.42 | syamajala | later |
| 23:49.46 | *** part/#asterisk syamajala (~syamajala@h0030bd1e9b96.ne.client2.attbi.com) |
| 23:49.47 | KalD|Work | syamajala, later |
| 23:50.49 | Smythe | where do I go to turn on pri debug? |
| 23:50.58 | KalD|Work | Smythe, pri debug span 1 |
| 23:51.03 | KalD|Work | Smythe, from the console |
| 23:51.21 | *** join/#asterisk Grooby (~Grooby@ip24-250-126-171.dc.dc.cox.net) |
| 23:51.42 | Smythe | "No PRI running on span 1" |
| 23:52.01 | KalD|Work | pri debug span 2 ? |
| 23:53.03 | Smythe | nope - but like I said this is a T1, not a pri - unless ast views them the same |
| 23:53.23 | Smythe | I think it just dislikes me - it's a consipracy |
| 23:53.42 | KalD|Work | Smythe, what does zap show channels show? |
| 23:54.05 | r0d3nt|m | Who has a working AutoDialer ? |
| 23:54.17 | *** join/#asterisk puppet (puppet@1-1-3-3b.ox.mlm.bostream.se) |
| 23:54.31 | Smythe | shows 24 channels with my jamestest contect |
| 23:54.36 | KalD|Work | Smythe, afk be back innna few |
| 23:54.38 | Smythe | context - can't type |
| 23:54.44 | puppet | Hi all :) |
| 23:54.48 | r0d3nt|m | Asterisk-GUI & ViciDial is an abortion. |
| 23:55.05 | r0d3nt|m | Anyone have a working AutoDialer/Predictive Dialer ?? |
| 23:55.15 | harryvv | anyone by chance know where i can get xlite beta for linux. Seems my email has yet to be answered by xten to obtain a copy. |
| 23:55.47 | KalD|Work | Smythe, ok what hardware do you have in the box? a digium card or? |
| 23:57.34 | Hmmhesays | why the hell do people need gui interfaces |
| 23:57.49 | Smythe | yes - digum - trying to find the model - but its is the single span t1/pri card |
| 23:57.56 | *** join/#asterisk mandreko (mandreko@12-222-3-81.client.insightBB.com) |
| 23:58.13 | KalD|Work | Smythe, pastebin your /etc/zaptel.conf |
| 23:58.51 | gr8nash | Hmmhesays i believe i am qualified to answer that.. cause this is HARD stuff when you first get into it |
| 23:58.53 | harryvv | bizzare i decompressed a tar file and got this after each file "implausibly old time stamp 1969-12-31 16:00:00 |
| 23:58.53 | harryvv | " |
| 23:59.22 | Smythe | k |
| 23:59.28 | Hmmhesays | gr8nash: not if you can read |
| 23:59.35 | gr8nash | =/ |
| 23:59.42 | Hmmhesays | i can see how if you illiterate it might be tough to muck through it |
