| 00:00.32 | ManxPower | voxzilla, voipsupply, atacomm |
| 00:00.45 | ManxPower | carrar, People keep writing articles about Asterisk |
| 00:00.59 | jedaustin | ManxPower: any you can walk into and compare phones? |
| 00:01.13 | ManxPower | jedaustin, That is an exersize left to the reader. |
| 00:02.09 | jedaustin | ManxPower: Im asking if chain stores carry ip phones |
| 00:02.20 | ManxPower | jedaustin, in our dreams. |
| 00:02.34 | mogorman | well bkw im going home ill talk to yall later |
| 00:02.35 | mogorman | dueces |
| 00:02.36 | *** join/#asterisk r0d3nt|m (anonymous@soveliss.luniac.com) |
| 00:02.37 | *** part/#asterisk mogorman (~mogorman@dhcp-162.digium.com) |
| 00:02.57 | ManxPower | jedaustin, The only VoIP devices that major stores sell are LOCKED to one provider. |
| 00:03.13 | *** join/#asterisk t3t (~t3t@cust018.mke.attron.net) |
| 00:03.46 | Qwell | heh, I was in circuit city yesterday, and they had vonage atas for $0.01 |
| 00:03.58 | jedaustin | ManxPower: figures.. loss leaders |
| 00:04.16 | cjk | anyone knows this ? |
| 00:04.17 | cjk | TFTP Error Code, Code: Not defined, Message: Phone RRQ quit |
| 00:04.31 | *** part/#asterisk t3t (~t3t@cust018.mke.attron.net) |
| 00:05.18 | *** join/#asterisk trig_hm (~jb@home.monkeypr0n.org) |
| 00:07.12 | carrar | ManxPower, yeah, it's getting crazy :) |
| 00:07.19 | carrar | ManxPower, anyone write a nice GUI yet? |
| 00:07.23 | mentat_ | so how hard are those ata to reconfigure? |
| 00:08.02 | *** join/#asterisk mstocco (~mstocco@adsl-223-194-118.mia.bellsouth.net) |
| 00:08.23 | ManxPower | mentat_, Anyone that knows how to reconfigure them isn't talking. |
| 00:08.36 | *** join/#asterisk Elshar (~Elshar@ip205-68.oregonfast.net) |
| 00:08.37 | ManxPower | But once the device gets to talk to the service provider....FORGET IT. |
| 00:09.49 | mentat_ | hmmm |
| 00:11.02 | *** join/#asterisk phrog123 (fmenard@109-75.tr.cgocable.ca) |
| 00:11.30 | phrog123 | is there any good reason why when I type show applications it says: There are no registered applications? |
| 00:11.34 | ManxPower | If you're a mentat, shouldn't you be telling us how to reconfigure them? |
| 00:11.53 | phrog123 | when I load asterisk with -vvvc it traces a bunch of .so's loading fine |
| 00:12.09 | ManxPower | phrog123, never seen that |
| 00:12.25 | mishehu | phrog123: possibly a deadlock |
| 00:12.44 | phrog123 | is there a problem with my build^ |
| 00:12.47 | phrog123 | is there a problem with my build? |
| 00:13.07 | jedaustin | Anyone had bad luck ordering from vegastream.com ? |
| 00:13.37 | mishehu | phrog123: donno. I've had it where one module gets caught in a deadlock, never all. |
| 00:14.22 | G0shen | I have been running 1.0.7 for 6 days without a crash |
| 00:14.52 | ManxPower | G0shen, be sure to report that via a bugnote |
| 00:15.09 | G0shen | hmm, does Asterisk keep two mpg123 run at all times? |
| 00:15.15 | *** join/#asterisk darwinscusp (~darwinscu@dsl081-114-184.dfw1.dsl.speakeasy.net) |
| 00:15.50 | *** join/#asterisk Darwin35 (~Darin@c-24-3-226-147.client.comcast.net) |
| 00:16.00 | mstocco | GOshen: if you has two music classes defined /etc/asterisk/musiconhold.conf |
| 00:16.23 | *** join/#asterisk Kumbang (~ecvs@167.205.24.4) |
| 00:16.37 | G0shen | I have two mpg123 processes running |
| 00:16.54 | G0shen | that started the same day I started asterisk 1.0.7 |
| 00:17.09 | G0shen | <PROTECTED> |
| 00:17.30 | phrog123 | so? |
| 00:17.37 | coolacid-p | I too have two mpg123 with only one classes running.. |
| 00:18.05 | phrog123 | is there any config file for controlling which applications register? |
| 00:18.05 | G0shen | just curious why there are two mpg123 processes running |
| 00:18.15 | G0shen | seems like it would only call them when it needed them |
| 00:18.17 | mstocco | G0shen: I have 7 but I have seven classes defined |
| 00:19.39 | phrog123 | I get a bunch of debug traces of the like pbx.c:1291 No Applicaion 'Wait' for extension... |
| 00:19.41 | *** join/#asterisk jey (~jey@221.219.230.31) |
| 00:19.51 | *** join/#asterisk Darwin35 (~Darin@c-24-3-226-147.client.comcast.net) |
| 00:21.39 | Darwin35 | Goshen each call starts a stream |
| 00:21.49 | Darwin35 | it seems |
| 00:23.12 | ManxPower | phrog123, you have a SERIOUSLY screwed up system. |
| 00:23.23 | coolacid-p | doesn't seem to do anything tho.. just checked TOP and all my mpg123 is not taking any cpu cycles |
| 00:24.23 | ManxPower | G0shen, Asterisk starts one mpg123 process per music on hold class. HOWEVER, mpg123 starts a second copy of itself when it needs to do buffering like with Asterisk. |
| 00:24.42 | ManxPower | I think there's an mpg3unbuffered option or something like it in musiconhold.conf.sample |
| 00:28.02 | G0shen | mpg123 takes .3 to .7 processor when playing, but nothing if not |
| 00:31.08 | phrog123 | are applications loaded i.e. seeing the .so show up on load, automatically registered, therefore automatically showing up in show applications? |
| 00:31.13 | *** join/#asterisk riksta (~rick@81-178-173-194.dsl.pipex.com) |
| 00:32.10 | ManxPower | If I ever meet the person that designed this Ekco brand kitchen timer I'm going to strangle them right then and there. |
| 00:32.27 | phrog123 | all those build warnings on kernel 2.4.29 related to spinlock.h, should I worry? |
| 00:32.58 | phrog123 | also on __set_64_bit_var? |
| 00:34.03 | *** join/#asterisk mrproper_ (~b@61.95.55.242) |
| 00:34.48 | mrproper_ | anyone got any idea why someone on a mobile gsm, recieves echo when i have a sip phone on handsfree but not when i pickup the handset |
| 00:37.31 | mentat_ | Anyone know if anyone makes a push2talk app like buzz2me? |
| 00:37.34 | mstocco | mrproper_: the sip/handsfree doesn't produce echo when you call to a landline? |
| 00:38.11 | mrproper_ | mrproper_: yes it does (sorry didnt have a land line to test until just then) |
| 00:38.42 | mstocco | mrproper_: what make/model of sip phone? |
| 00:38.46 | mrproper_ | actually after further testing it does it when speaking to internal sip phones as well. looks like the echo cancellation isnt working properly |
| 00:39.17 | mrproper_ | mstocco: tested it with grandstream budget tones and JR168S (cant see the brand of that phone) |
| 00:39.36 | eric_ | mrproper_: i think you can turn up the level of echo cancelling... and if your cpu is slower, it might have problems cancelling echoes |
| 00:40.12 | mrproper_ | eric_: at the moment its set to medium, so youd recommend high? |
| 00:40.48 | eric_ | ive never raised it... it could help |
| 00:40.56 | eric_ | but in my experience, speakerphone usually causes some echo |
| 00:41.33 | eric_ | has anyone gotten app_conference? |
| 00:41.40 | mrproper_ | eric_: this is shocking echo as in your talking to yourself with a delay |
| 00:44.26 | mstocco | mrproper_: my guess is the problem is at the phone (old firmware or shitty design) |
| 00:45.13 | mstocco | mrproper_: echo cancelling begins with 'training' at the beginning of the call |
| 00:46.40 | mstocco | mrproper_: if something changes throughout the call (switch to speakerphone) how does asterisk known to 'retrain' ? |
| 00:47.02 | mrproper_ | mstocco: no im talking about making the call from speakerphone, not switching to speakerphone |
| 00:48.25 | mstocco | mrproper_: almost all of the echo problems I have hunted down where at the point where the sound was still analog |
| 00:48.55 | mstocco | mrproper_: ie at the handset or in your case speaker phone |
| 00:49.04 | mstocco | mrproper: hope that helps |
| 00:51.46 | *** part/#asterisk eKo1 (~bernd@63.245.57.70) |
| 00:51.55 | *** join/#asterisk Tall-guy (tall-guy@hssxrg207-195-103-110.sasknet.sk.ca) |
| 00:52.21 | Tall-guy | any snom190 users here? |
| 00:52.37 | phrog123 | any intercom application that works with an IP phone ... I want to run it as a baby monitor |
| 00:53.21 | mstocco | mrproper_: http://www.cisco.com/en/US/tech/tk652/tk701/technologies_white_paper09186a00800d6b68.shtml |
| 00:54.13 | mrproper_ | mstocco: thanks for the info, i think its the crappy handsets, old firmware, ill try updating those first |
| 00:56.44 | PTG123 | hah |
| 00:56.46 | PTG123 | baby monitor |
| 00:57.59 | ManxPower | phrog123, You need this: |
| 00:58.07 | ManxPower | ~mailinglist |
| 00:58.08 | jbot | somebody said mailinglist was Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ |
| 00:58.29 | PTG123 | babymonitor would be easy |
| 00:58.34 | PTG123 | set phone to auto-answer in babys room |
| 00:58.37 | PTG123 | then just call it |
| 00:58.55 | eric_ | haha |
| 00:59.16 | eric_ | somewhat amusing... but practical |
| 01:00.07 | PTG123 | heh |
| 01:00.13 | Dr-Linux | i have MultiTech viop 8 port device and i have Asterisk server with dialogic card ... can i connect it to mutitech device ? |
| 01:00.16 | PTG123 | just put a cheap polycom in babys room with a good speakerphone |
| 01:00.33 | PTG123 | then you could even talk to baby |
| 01:01.02 | Dr-Linux | can asterisk server comunication with mutitech 8 port voip device ? |
| 01:01.17 | Dr-Linux | ManxPower ? |
| 01:01.19 | PTG123 | dont even know what that is mr linux :) |
| 01:07.46 | mstocco | Dr-Linux: MVP810 ? the website says it does SIP but that doesn't mean it will work |
| 01:08.24 | mstocco | Dr-Linux: looks like it was designed to extend an Avaya VoIP system |
| 01:09.23 | mstocco | Dr-Linux: 3000 for 8 phone lines... hrmm |
| 01:09.44 | Tall-guy | Hey, if asterisk sends a "Sip/2.0 407 Proxy authentication required" to my snom phone...does that mean "its required and you failed?" |
| 01:10.28 | *** join/#asterisk CosmicRay (~jgoerzen@2002:4545:7206:1:20e:a6ff:fe5c:55e1) |
| 01:14.02 | mstocco | Tall-guy: does the problem go away if you remove the secret line for that phone in sip.conf? |
| 01:14.03 | *** join/#asterisk cjk (~cjk@80.92.75.143) |
| 01:14.27 | cjk | i hate those tftp issues. is it possible that it is easyer to make sip pass the nat than tftp? |
| 01:14.35 | Tall-guy | <PROTECTED> |
| 01:15.01 | *** join/#asterisk IQ (~IQ@63-230-44-170.omah.qwest.net) |
| 01:15.05 | *** join/#asterisk __cmc__ (~cmc3c@pcp06581706pcs.nrockv01.md.comcast.net) |
| 01:17.19 | *** join/#asterisk cripito (~cripito@c-65-34-156-173.se.client2.attbi.com) |
| 01:17.48 | Trepalium | A bit off-topic, but does Fedora Core 3 have ksymoops in a package somewhere? |
| 01:19.15 | *** join/#asterisk didz_ (~omg@200.218.193.30) |
| 01:21.33 | *** join/#asterisk LeeColleton (~lc@dsl231-061-247.sea1.dsl.speakeasy.net) |
| 01:22.51 | *** join/#asterisk UnrealTear (UnrealTear@pool-70-23-129-251.ny325.east.verizon.net) |
| 01:23.32 | *** join/#asterisk mitcheloc (~mitchel@69-169-28-46.anhmca.adelphia.net) |
| 01:24.48 | *** join/#asterisk justnulling2 (justnullin@ool-18bc013e.dyn.optonline.net) |
| 01:25.05 | UnrealTear | I have setup an asterisk server at work and am trying to connect to it through my linksys router. I have tried to forward the ports to my phone but I still cannot place or recieve calls |
| 01:25.29 | UnrealTear | I get the following message on the asterisk server |
| 01:26.09 | ariel_ | UnrealTear, which router. Also make sure you have the asterisk server on the dmz |
| 01:26.15 | *** join/#asterisk _tekati_ (~captain@cpe-66-75-215-63.bak.res.rr.com) |
| 01:26.39 | justnulling2 | i can't connect to my asterisk server with sjphone but x-lite works fine, any idea? |
| 01:26.51 | UnrealTear | The asterisk server is not at my home. I have an IP phone at home trying to connect to the asterisk server at my work |
| 01:27.26 | mstocco | UnrealTear: what is the message on the server? |
| 01:27.57 | UnrealTear | Router is a linksys WRT54G |
| 01:28.10 | UnrealTear | message on the server is Mar 16 20:26:47 WARNING[21145]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call 003094c2-ace6003c-69ac938d-46f53567@192.168.1.124 for seqno 101 (Non-critical Response) |
| 01:28.41 | UnrealTear | Everything works at the office so its something with the router. Phone is set as the DMZ |
| 01:28.51 | mstocco | UnrealTear: are you forwarding UDP or TCP? |
| 01:29.08 | UnrealTear | ive tried both |
| 01:29.42 | ariel_ | UnrealTear, it's a sip phone? |
| 01:29.43 | UnrealTear | the phone is also set as the DMZ |
| 01:29.45 | UnrealTear | yes |
| 01:29.52 | UnrealTear | A Cisco 7960\ |
| 01:30.00 | ariel_ | did you setup externip= on the asterisk box? |
| 01:30.28 | ariel_ | also the Cisco needs to be able to send it's external ip address not it's internal. |
| 01:30.41 | UnrealTear | No I did not. |
| 01:31.03 | ariel_ | your going from a Asterisk box --- Nat ---Internet ----Nat --- Cisco This is called double nat and it's a problem. |
| 01:31.09 | UnrealTear | Thats what I was thinking also but I cant seem to give the phone a public IP behind the router as it tells me th IP is not on the network |
| 01:31.14 | ariel_ | It can work but you have to do some fancy settings. |
| 01:31.49 | UnrealTear | The asterisk box sees the Cisco phones private IP address |
| 01:32.02 | ariel_ | UnrealTear, you don't give the phone a public. There is an setting for the phone to put it's exterinal IP address which is the one that the router has. |
| 01:32.27 | UnrealTear | Where is that setting. I cannot find it. |
| 01:32.55 | ariel_ | UnrealTear, I don't remember where I have not used cisco phones in over a year. Too expensive and too hard to configure for me. |
| 01:33.00 | ariel_ | But it's there. |
| 01:33.02 | *** join/#asterisk backfired (~asterisk@217.159.105.46) |
| 01:33.10 | UnrealTear | Yeah. Alright. Thank you for your help. I will keep looking |
| 01:33.41 | backfired | Hi everyone, does anyone know how I can get the SIP software for CISCO handsets? |
| 01:34.20 | mstocco | backfired: http://store.cisco.com :) |
| 01:35.07 | backfired | that link didn't work but thanks. |
| 01:36.15 | *** join/#asterisk convey (~chatzilla@208-216-127-234.cust.gti.net) |
| 01:38.49 | *** join/#asterisk lildivil (user@ool-18bc24d7.dyn.optonline.net) |
| 01:41.39 | convey | Has any one experienced this error message? ast_readaudio_callback: Failed to write frame |
| 01:52.51 | *** join/#asterisk The_Ball (~alex@static-100.35.240.220.dsl.comindico.com.au) |
| 01:58.10 | *** join/#asterisk carrar (tim@osburn.com) |
| 01:58.14 | *** join/#asterisk caesar2 (~igerl@ppp-62-245-210-208.mnet-online.de) |
| 01:58.55 | ManxPower | It's not my job to stand between the people talking to me, and the ones listening to me." -- Pego the Jerk |
| 01:59.09 | *** part/#asterisk didz_ (~omg@200.218.193.30) |
| 02:01.16 | *** join/#asterisk wankel (nobody@ohno.mrbill.net) |
| 02:02.56 | ariel_ | ManxPower, so how is your search for a job out of the states? |
| 02:04.01 | *** part/#asterisk Kumbang (~ecvs@167.205.24.4) |
| 02:09.07 | carrar | ManxPower! |
| 02:09.26 | carrar | ManxPower, find me a job in Japan! |
| 02:09.31 | carrar | Where are you looking? |
| 02:10.03 | *** join/#asterisk yxa (~void@203.118.40.42) |
| 02:10.19 | *** join/#asterisk Moc (~Moc@modemcable012.47-80-70.mc.videotron.ca) |
| 02:10.45 | Moc | hi all |
| 02:15.57 | *** join/#asterisk techie (gus@asterisk.horizonte.us) |
| 02:16.27 | caesar2 | hi! |
| 02:16.50 | caesar2 | carrar: what do you whant in Japan ? |
| 02:17.32 | carrar | yes |
| 02:17.34 | carrar | oh |
| 02:17.36 | carrar | well |
| 02:17.39 | carrar | anything |
| 02:17.44 | carrar | I am a Sr. Network Engineer |
| 02:17.48 | carrar | so that would be my first choice |
| 02:17.57 | caesar2 | ah... |
| 02:18.01 | carrar | 2nd would be Sr. Unix engineers (did that before networking) |
| 02:18.15 | caesar2 | why does overyone whant to leave the states ? |
| 02:18.16 | carrar | VoIP would be cool too |
| 02:18.56 | carrar | I've lived in other countries |
| 02:19.02 | carrar | it's now time for Asia! |
| 02:19.02 | caesar2 | where ? |
| 02:19.08 | carrar | Germany |
| 02:19.14 | caesar2 | right now ? |
| 02:19.17 | carrar | no |
| 02:19.20 | carrar | in Seattle now |
| 02:19.37 | *** join/#asterisk Grooby (~Grooby@ip24-250-126-171.dc.dc.cox.net) |
| 02:19.53 | caesar2 | I am i Germany and loved to go to the states... and CANT!!! no visa !! |
| 02:20.00 | carrar | that sucks |
| 02:20.05 | carrar | I lived in Berlin for a year |
| 02:20.11 | carrar | Sydney for 3 months |
| 02:20.13 | caesar2 | i'm in munic |
| 02:20.23 | carrar | right on |
| 02:20.29 | carrar | home of the first Pizza Hut |
| 02:20.31 | carrar | :) |
| 02:20.33 | carrar | in Germany |
| 02:20.34 | caesar2 | hehe |
| 02:20.40 | caesar2 | realy ? |
| 02:20.44 | caesar2 | cool |
| 02:20.47 | *** join/#asterisk elriah (~jfulcrum@adsl-068-209-198-242.sip.bhm.bellsouth.net) |
| 02:20.47 | carrar | yeah |
| 02:21.00 | carrar | I fly to munic one day just for Pizza |
| 02:21.03 | *** join/#asterisk convey (~chatzilla@208-216-127-234.cust.gti.net) |
| 02:21.05 | carrar | flew |
| 02:21.20 | caesar2 | made a trip from new mexco to new york last year... 8 weeks |
| 02:21.34 | elriah | Hi all, I've just installed my first asterisk system, it's running. I'm trying the iaxcomm softphone, but it says registration rejected. Anyone seen a how-to for this softphone? Maybe some sample sip.conf files? |
| 02:21.36 | caesar2 | come and visite me.. :-) |
| 02:21.37 | Beirdo | questions for the great minds... |
| 02:21.40 | Dr-Linux | MultiTech MVP810 is FXO or FXS ? |
| 02:22.03 | caesar2 | iax = no = sip |
| 02:22.08 | Beirdo | if I want to match any dialed extension and send to an extension, is using _. correct? |
| 02:22.20 | carrar | ok, time to go home |
| 02:22.23 | carrar | bbl |
| 02:22.40 | convey | Is there a setting to allow Asterisk to handel high call volumes? Like 100 users dialing in at once via SIP? I have been testing and I am getting a lot of Failed to write to frame errors and the calls drop. |
| 02:22.49 | mikegrb | Beirdo: ja |
| 02:22.52 | *** join/#asterisk NoCAT (NoCAT@c-24-9-32-2.client.comcast.net) |
| 02:23.01 | caesar2 | s |
| 02:23.04 | caesar2 | = |
| 02:23.05 | caesar2 | ? |
| 02:23.07 | Beirdo | just seems nasty :) |
| 02:23.43 | elriah | Ahh! Cool. iax.conf. |
| 02:23.53 | mstocco | elriah: not sip.conf but iax.conf |
| 02:24.00 | caesar2 | convey: why not use ser ? |
| 02:24.00 | elriah | Got it.. tnx |
| 02:24.03 | elriah | this is cool |
| 02:24.48 | convey | ceasar2: they are all going to a meetme in asterisk. Should I front end them with SER? |
| 02:26.18 | elriah | What's the preferred codec for softphones? GSM? |
| 02:26.21 | caesar2 | no don't think this would be better... just for norm. calls |
| 02:27.04 | convey | ceasar2: It is an application for a 100 party meetme. Asterisk seems to be having a problem keeping up with the incoming calls. |
| 02:27.05 | caesar2 | elriah: depend an what quality you want or what bandwidth you have |
| 02:27.16 | elriah | Lan? |
| 02:27.30 | caesar2 | use something uncompressed... |
| 02:27.34 | *** join/#asterisk brc__ (~brian@brc.base.supporter.pdpc) |
| 02:27.58 | elriah | Is there a chart somewhere comparing the codecs? |
| 02:28.01 | caesar2 | convey: don't know my maximum was 10 in 1 meetme |
| 02:28.23 | caesar2 | http://www.voip-info.org/wiki-Codecs |
| 02:28.36 | elriah | Thanks. |
| 02:28.40 | caesar2 | Some tables comparing different codecs: |
| 02:28.41 | caesar2 | Cisco.com: Voice Over IP - Per Call Bandwidth Consumption |
| 02:28.41 | caesar2 | http://www.cs.columbia.edu/~hgs/audio/codecs.html |
| 02:28.41 | caesar2 | http://www.speex.org/comparison.html |
| 02:28.41 | caesar2 | http://www.terracall.com/FAQs_white_1.aspx |
| 02:28.41 | caesar2 | http://www.signalogic.com/index.pl?page=codec_samples |
| 02:28.54 | elriah | I'm using iaxComm right now, is there a better softphone? |
| 02:30.23 | convey | ceasar2: My Max for meetme via sip is 184 |
| 02:30.30 | mstocco | elriah: the iax softphones (diax, iaxcomm) are built from the same iaxclient libs |
| 02:30.56 | elriah | Is sip better? Maybe the question should have been, what's the preferred softphone? |
| 02:30.58 | mstocco | elriah: firefly is a good iax client too |
| 02:31.14 | caesar2 | use iax! |
| 02:31.28 | PTG123 | better for unix or windows? |
| 02:31.31 | *** join/#asterisk roamer323 (~sing@67.71.60.211) |
| 02:31.59 | mstocco | elriah: iax is the best for crossing NAT firewalls |
| 02:32.18 | PTG123 | elric: what platform? |
| 02:32.50 | Beirdo | iax is good, mmkay? :) |
| 02:41.41 | caesar2 | good nigth! time to sleep |
| 02:42.17 | IQ | elriah: sip is not better but its better to use sip softphone - atleast thats what I do :) |
| 02:43.04 | bannerman | IQ: Can you recommend any particular softphone, sip or otherwise? |
| 02:43.18 | Beirdo | time to go buy an SPA-3000 |
| 02:43.18 | Beirdo | :) |
| 02:43.34 | Beirdo | X-lite |
| 02:43.35 | IQ | Beirdo: I ordered one yesterday : |
| 02:43.38 | bannerman | IQ: My phones from ariaVoice came with the wrong power supplies yesterday, and I need them tomorrow... the power supplies were 500ma instead of 300ma, and reverse polarity, all the phones I bought are dead. |
| 02:43.41 | IQ | bannerman: X-Lite |
| 02:43.48 | bannerman | X-lite it is |
| 02:43.55 | Beirdo | I'm getting one in just under an hour, but I gotta get on the subway to go meet him :) |
| 02:44.38 | bannerman | lol, Beirdo, sounds kinda exciting! psst, hey kid.. wanna buy an SPA-3000? |
| 02:44.40 | IQ | bannerman: I dont think having 500ma is bad but wrong polarity is NO NO NO |
| 02:45.10 | mstocco | X-Lite for windows but not for a Mac; use SJPhone under OS-X :) |
| 02:45.26 | bannerman | IQ: yeah, I dunno really anything about power supplies.. but the last phone.. before I plugged it into the wrong power supplies, which ariaVoice had sent.. I tried it on my test power supply, and it worked good.. plugged it into the other power supply, then thought to check the power supply type |
| 02:45.28 | Eight | mstocco: ya? I ended up with X-lite on the mac. |
| 02:45.31 | IQ | Beirdo: where r u buying ur 3000 from? |
| 02:45.32 | PTG123 | ~seen czero |
| 02:45.35 | jbot | czero <~me@CPE0090f800c5b0-CM001225704b6e.cpe.net.cable.rogers.com> was last seen on IRC in channel #asterisk, 51d 11m 13s ago, saying: 'blah bateris in my wireles mouse are dead :)'. |
| 02:45.40 | Eight | mstocco: SJPhone never actually worked for me. |
| 02:46.11 | PTG123 | Xpro actually is awesome |
| 02:46.16 | PTG123 | eyebeam their newest phone has a few issues |
| 02:46.21 | bannerman | ariaVoice was real good about it though, they guy new about it before I called him, he sounded sick heh, he said he was proactively shipping new phones to everyone who had purchased phones from them in the last two weeks |
| 02:46.41 | *** join/#asterisk cjk (~cjk@80.92.75.143) [NETSPLIT VICTIM] |
| 02:46.41 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) [NETSPLIT VICTIM] |
| 02:46.41 | *** join/#asterisk A-Tuin|work (~A-Tuin@nat.office.legend.net.uk) [NETSPLIT VICTIM] |
| 02:46.41 | *** join/#asterisk Gunnar (~gunnar@34.80-203-209.nextgentel.com) [NETSPLIT VICTIM] |
| 02:46.42 | *** join/#asterisk ScaredyCat (~ScaredyCa@j25065.upc-j.chello.nl) [NETSPLIT VICTIM] |
| 02:46.42 | *** join/#asterisk Qwell (~north@70-32-102-18.ontrca.adelphia.net) [NETSPLIT VICTIM] |
| 02:46.45 | mstocco | Eight: their lastest will, just that X-Lite under OS-X is a pig (for me) |
| 02:46.49 | IQ | bannerman: its not your fault coz they were shipped with wrong powersupply.... but I dont think they'll be able to send new ones on time |
| 02:48.03 | NoCAT | whats the most t1s you can have on a pci card? |
| 02:48.50 | *** join/#asterisk easydone (~notdone@eksel.demon.nl) |
| 02:49.16 | mstocco | Eight: SJPhone Build 1.30.256b (Nov 24 2004)... the previous client didn't register correctly on asterisk |
| 02:49.36 | IQ | NoCAT: I think it depends upon what u r going to do with the Card |
| 02:50.08 | IQ | NoCAT: You can easily find a 4xT1 PCI card but then what are you planning to do with this card |
| 02:50.24 | NoCAT | voip wifi |
| 02:50.31 | dersteer | then u have to ask yourself if T1's make sence |
| 02:50.36 | IQ | NoCAT: depending upon the processing power this card has onboard it can do so much... |
| 02:50.38 | dersteer | u can get a fractional t3 cheaper :) |
| 02:51.02 | IQ | NoCAT: voip wifi? Why u need T1 for ? |
| 02:51.27 | NoCAT | how do you interface a fractional t3 to your pc with asterisk? |
| 02:52.00 | IQ | dersteer: never used a T3 PCI card on a machine :( ... |
| 02:52.10 | NoCAT | iq i don't need a t1, i want more then a t1 actually the cost of t1s is too high, it would be nice is asterisk supported t3 cards |
| 02:52.15 | wildcard0 | offtopic political plug -> http://www.dvdr-core.org/ <-- friend of mine getting sued by the MPAA. post, bitch, etc </offtopic> |
| 02:52.39 | IQ | dersteer: what T3 card you have experienced? |
| 02:53.05 | dersteer | was just a idea off the hip.... I know 4 T1's cost more than a fractional T3 |
| 02:53.27 | IQ | NoCAT: never used T3 PCI - no experience |
| 02:53.28 | wildcard0 | dersteer, depends on the location/provider |
| 02:54.16 | IQ | we have fiber at work |
| 02:54.35 | dersteer | fiber rocks :) |
| 02:54.52 | dersteer | cable company offeres fiber speeds up to 1 gigabit :) |
| 02:55.00 | dersteer | its a odd cable company here |
| 02:55.11 | IQ | where are you located and what cable company :) ? |
| 02:55.20 | dersteer | Big Rapids, Michigan |
| 02:55.25 | dersteer | charter cable |
| 02:55.43 | *** join/#asterisk r0d3nt|m (~RatMan@adsl-67-118-87-170.dsl.lsan03.pacbell.net) |
| 02:55.47 | *** join/#asterisk slePP (~slepp@S01060040f48412ad.ed.shawcable.net) |
| 02:55.54 | IQ | this is tooooooo good - - - how much they charge for this |
| 02:55.56 | dersteer | http://www.charter-business.com/FiberBasedInternet.cfm |
| 02:56.00 | dersteer | here is a link |
| 02:56.09 | IQ | lets check the link |
| 02:56.11 | *** join/#asterisk r0d3nt|m (anonymous@soveliss.luniac.com) |
| 02:56.48 | IQ | dersteer: do u know how much they charge for 1GB? |
| 02:56.52 | dersteer | :) |
| 02:56.53 | dersteer | a ton |
| 02:56.54 | dersteer | lol |
| 02:56.57 | dersteer | they laughed at me |
| 02:56.58 | IQ | lool |
| 02:57.11 | dersteer | I'm looking at 100 megabit |
| 02:57.39 | IQ | 100M less than $1000/month ? |
| 02:57.44 | dersteer | yes |
| 02:58.07 | dersteer | not by much... lol... |
| 02:58.11 | dersteer | 899 I got quoted |
| 02:58.25 | IQ | better than 899.95 |
| 02:58.33 | dersteer | their fiber runs right by me here :) |
| 02:58.59 | dersteer | not 20 feet away :) |
| 02:59.11 | IQ | maybe I should move close to your place... how is the apartment rent :) |
| 03:00.12 | dersteer | u didn't believe me did u IQ |
| 03:00.14 | dersteer | lol |
| 03:00.26 | IQ | yes I did |
| 03:00.58 | dersteer | a truely awesome cable company |
| 03:01.11 | dersteer | I'm working on a fixed wireless ISP |
| 03:01.27 | IQ | for locals? |
| 03:01.47 | Shido6 | talk to jerjer dersteer thats right up his ally |
| 03:01.57 | dersteer | already have :) |
| 03:02.10 | dersteer | picked his brain for all wireless knowledge |
| 03:02.12 | dersteer | :) |
| 03:02.28 | IQ | do we need brain? |
| 03:02.36 | mgth | dersteer: I work for a company that has 3 wisps under it |
| 03:02.36 | cripito | :D |
| 03:02.57 | dersteer | I have a 7 mile link going right now :) |
| 03:03.34 | mgth | they do a city hopping thing where they start at their offices and get about 100 miles out with like 10-15 stops |
| 03:03.36 | IQ | dersteer: do I have to believe everything you say :) ? |
| 03:03.45 | dersteer | nope :) |
| 03:04.11 | IQ | do you need a license to do that? |
| 03:04.14 | dersteer | nope |
| 03:04.21 | dersteer | its license free band |
| 03:04.26 | dersteer | all u have to do is buy equipment |
| 03:04.34 | mikegrb | dersteer: it is you not you |
| 03:04.46 | mikegrb | if what you have to say isn't worth you typing shit out then don't say it |
| 03:04.50 | mikegrb | er, you not u |
| 03:04.59 | IQ | looks like a good business... under 100k ? |
| 03:05.40 | dersteer | lol... no money yet |
| 03:05.44 | dersteer | waiting for real access |
| 03:05.55 | dersteer | I'm using cable modems right now |
| 03:06.14 | dersteer | the 7 mile link is to buddys house... I'm right on the end of the cable run |
| 03:06.49 | IQ | what is the equipment cost if you want to cover 10 mile radius? |
| 03:07.30 | dersteer | depending on exactly what u needed... not including the tower itself less than 2000 |
| 03:07.39 | dersteer | and thats using good stuff |
| 03:07.47 | IQ | just $2000 |
| 03:07.48 | mikegrb | dersteer: it is you not u |
| 03:07.53 | mikegrb | if what you have to say isn't worth you typing shit out then don't say it |
| 03:08.18 | dersteer | mikegrb: 2 words... fuck off :) |
| 03:08.30 | mikegrb | dersteer: 2 words... no |
| 03:08.44 | dersteer | another added my my iggy :) |
| 03:10.14 | riksta | it's nice to be able to fucking swear in a channel for a change :) |
| 03:10.30 | slePP | no! |
| 03:10.32 | slePP | no swearing! |
| 03:10.33 | brc_ | SLEPP! |
| 03:10.35 | brc_ | dude |
| 03:10.40 | slePP | wut? |
| 03:10.40 | brc_ | I was just looking for you |
| 03:10.44 | slePP | oi |
| 03:10.45 | slePP | damn it |
| 03:10.47 | riksta | hehe slePP you swear on the phone i've heard ya :P |
| 03:10.48 | brc_ | muhahahah! |
| 03:11.11 | riksta | is that "peers" or "peers" :) |
| 03:11.20 | slePP | a bit o' both |
| 03:11.27 | riksta | thought so :P |
| 03:12.37 | *** join/#asterisk soundguy (~soundguy@zeus.blendtek.com.au) |
| 03:12.39 | *** join/#asterisk TheEmperor (~mattn@203.121.47.100) |
| 03:14.39 | *** join/#asterisk azzazz|n (~beard@203.14.171.182) |
| 03:15.23 | Druken | slepp, swear? NOOOOOOOO, never...... |
| 03:15.42 | *** join/#asterisk heison (~heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
| 03:17.51 | *** join/#asterisk jey (~jey@221.219.230.31) |
| 03:18.02 | Shido6 | how do u register asterisk for deltathree,iconnecthere inbound numbers? |
| 03:18.11 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 03:18.40 | Druken | using a register line in your sip.conf ? |
| 03:19.00 | Shido6 | no |
| 03:19.02 | Shido6 | that didnt work |
| 03:20.59 | TheEmperor | hi guys.I am always getting confused with FXS/FXO ports. The FXO ports are the ones which you plug your analog line into right? |
| 03:22.21 | *** join/#asterisk NormAst (NormAst@toronto-HSE-ppp3972479.sympatico.ca) |
| 03:23.33 | loud | yes |
| 03:24.18 | Druken | TheEmperor: O = oh shit, i need to pay for it... S = shit phones i plug into it |
| 03:25.10 | dersteer | Druken: good way of remembering :) |
| 03:25.27 | Druken | i guess |
| 03:25.36 | Druken | as long as it works :) |
| 03:26.08 | convey | I am getting a 404 on my broadvoice registration. |
| 03:26.19 | *** join/#asterisk carlosh (~carlosh@203-96-159-89.paradise.net.nz) |
| 03:26.21 | convey | anyone else experience this? |
| 03:26.30 | dersteer | convey: what hardware? |
| 03:26.36 | dersteer | u using a softphone? |
| 03:26.51 | carlosh | g'day all |
| 03:27.02 | convey | derser: just a simple registration from my sip.conf |
| 03:27.25 | *** part/#asterisk __cmc__ (~cmc3c@pcp06581706pcs.nrockv01.md.comcast.net) |
| 03:27.28 | dersteer | its up here... if that helps ya? |
| 03:27.42 | convey | dersteer: it worked two hours ago and I reloaded and now it is down :( |
| 03:28.04 | dersteer | u on dca proxy? |
| 03:28.12 | convey | dersteer: Yes |
| 03:28.12 | carlosh | anyone played with Asterisk@home and AMP ? |
| 03:28.16 | dersteer | its up here |
| 03:28.18 | dersteer | same one as me |
| 03:28.31 | convey | dersteer: thanks I will keep tweaking it |
| 03:29.11 | *** join/#asterisk mog_home (~mog_home@146.229.183.147) |
| 03:29.42 | carlosh | question2: anyne gets better Status than 100ms with budgetoneIp phones? |
| 03:30.00 | carlosh | on a LAN ..? |
| 03:32.06 | dersteer | 100ms... thats awful |
| 03:32.17 | dersteer | note to self... no budgetoneip service :) |
| 03:32.39 | Sedorox | I don't have problems with mine... |
| 03:33.28 | Sedorox | I have "unmonitored" for mine.. |
| 03:33.43 | Druken | 100ms on a lan? |
| 03:34.08 | Druken | wtf... you got a mouse running the wheel running your switch ? |
| 03:34.35 | dersteer | damn lazy mouse |
| 03:34.37 | dersteer | run run |
| 03:34.41 | dersteer | lol |
| 03:34.46 | Druken | that's horrible... i get less than that to connections across the country |
| 03:34.58 | carlosh | i meant those budgetone Ip phones... I have a chinese other, and I get 100+ ms status on same LAN, I blame the low speed of IP phone hardware... |
| 03:34.59 | Druken | and even those i think aren't the greatest |
| 03:35.14 | pigpen | Hey...can I assign specific codec's per extention? (server side) |
| 03:35.27 | Druken | pigpen: yes |
| 03:35.29 | mstocco | pigpen: yes |
| 03:35.47 | *** join/#asterisk atmel (~vlad@ip68-4-101-199.oc.oc.cox.net) |
| 03:35.48 | pigpen | cool! |
| 03:35.49 | pigpen | thanks. |
| 03:35.51 | carlosh | pigpen: yes you can, disallow all, then allow each of the ones you want enabled, gsm will take the lead.. |
| 03:36.04 | pigpen | but per extention? |
| 03:36.13 | pigpen | isn't that global? |
| 03:36.16 | carlosh | pigpen: yes, per extension.. |
| 03:36.16 | Druken | extension == user |
| 03:36.25 | pigpen | cool. |
| 03:36.27 | cpatry | just as a curiosity, whats ur asterisk uptime? |
| 03:36.31 | pigpen | thanks again? |
| 03:36.36 | Druken | not nearly enough? |
| 03:37.07 | carlosh | who here using budgetone or any other Ip phone.. kind of right now ? |
| 03:37.19 | carlosh | on LANs.. |
| 03:37.47 | mstocco | carlosh: I have two.. lemme check |
| 03:39.06 | carlosh | I'm trying one based on the pa1688 chipset.. not bad phones.. I wonder if budgetone uses same chipset..? |
| 03:39.34 | Sedorox | carlosh: I have a BT100... no problems |
| 03:40.00 | carlosh | what's status on same LAN ? |
| 03:40.11 | Sedorox | "unmonitored" |
| 03:40.15 | Sedorox | thats all that shows up |
| 03:40.33 | carlosh | could you change it so it shows delay ? |
| 03:40.37 | Sedorox | brandon/brandon 10.100.23.33 D 255.255.255.255 5060 Unmonitored |
| 03:40.40 | Sedorox | dunno how to do it... |
| 03:40.55 | carlosh | go to IAX or SIP .conf, to the definition of brandon.. |
| 03:41.28 | carlosh | and add a line: qualify=yes |
| 03:41.53 | carlosh | or qualify=400 ; this last is for miliseconds.. |
| 03:41.54 | Sedorox | so thats what that does... |
| 03:42.00 | carlosh | yep.. |
| 03:42.14 | Sedorox | brandon/brandon 10.100.23.33 D 255.255.255.255 5060 OK (9 ms) |
| 03:42.26 | Sedorox | the phone is 10mbit... server is 100mbit... goes through a wirelessrouter... |
| 03:42.53 | carlosh | wow.. |
| 03:42.59 | carlosh | damn good... |
| 03:43.13 | MikeJ[Jayden] | hey, anyone know who does toronto, ca did's? |
| 03:43.15 | Sedorox | should be less then that I think |
| 03:43.23 | Sedorox | (and the server and the phone are on the same side of the router) |
| 03:43.29 | *** join/#asterisk Andrew- (admin@216-dsl.pdstower02.wireless.precisionds.com) |
| 03:43.30 | Sedorox | brandon/brandon 10.100.23.33 D 255.255.255.255 5060 OK (4 ms) |
| 03:43.31 | Sedorox | thats better |
| 03:43.39 | carlosh | anyone using pa1688 chipset based IP Phones?.. |
| 03:43.55 | carlosh | I get 1ms from softphone (linux - gnomemeeting) |
| 03:44.12 | Andrew- | where can i find screenshots and such of features |
| 03:44.54 | jontow | carlosh; what soundcard do you use on your linux machines? |
| 03:45.14 | carlosh | hmm let me check.. |
| 03:46.01 | carlosh | it says ATI IXP (compaq laptop) |
| 03:46.27 | jontow | aha. |
| 03:47.31 | NormAst | mikeJ[Jayden]: www.tollfreetelecom.ca |
| 03:47.47 | NormAst | Toronto DID's |
| 03:47.48 | *** join/#asterisk kimc (~kimc@pcp04039944pcs.wbrmfd01.mi.comcast.net) |
| 03:49.52 | *** join/#asterisk t3t (~t3t@galley.pangalacticgargleblaster.com) |
| 03:52.16 | MikeJ[Jayden] | anyone else... I have a friend doing a community project in toronto, hopefully we can find somone to donate 1 did with unlimited inbound |
| 03:54.19 | MikeJ[Jayden] | sedorox, is that a sipura box? |
| 03:55.08 | Sedorox | no |
| 03:55.12 | Sedorox | bt100 |
| 03:55.32 | mishehu | blah. |
| 03:56.14 | NormAst | MikeJ[Jayden] What kind of Community project? |
| 03:56.36 | eric_ | anyone know who does Montreal, Canada DID's, besides unlimitel ? |
| 03:56.48 | eric_ | they seem like the only one |
| 03:56.57 | NormAst | eric_: I do toronto.. Barrie, hamilton.. |
| 03:57.00 | mishehu | this is odd. I try to make an iax2 call between 2 * boxes, and when I have iax2 debug enabled on both, only the calling box shows any debug output, the other one doesn't, and the call does not go thru. a tcpdump though shows that there is an auth request, but that it's returned as inval for some reason (md5/rsa auth)... both keys are initialized. |
| 03:57.04 | MikeJ[Jayden] | I am not totally clear on it, as it is my friends project, not mine, but it involves signs at local points of interest, where people call in and get info |
| 03:57.14 | MikeJ[Jayden] | history, arts, ect... |
| 03:57.46 | MikeJ[Jayden] | it is already rolled out in a couple areas of toronto, although I could not tell you what they are |
| 03:57.51 | Shido6 | who does canada? |
| 03:58.07 | MikeJ[Jayden] | canadians :) |
| 03:58.17 | NormAst | MikeJ[Jayden]. Might be interest if I can do a bit of Advertising... few seconds of audio. |
| 03:58.58 | heison | shido6: voctel.net does Canada |
| 03:59.02 | MikeJ[Jayden] | he said they would not do audio advertising on this, but would be willing to entertain somthing in the print material or website |
| 03:59.14 | NormAst | Cool.. |
| 03:59.48 | MikeJ[Jayden] | are you interested in sponsoring? |
| 04:00.12 | MikeJ[Jayden] | I can certainly pass your info along to him |
| 04:00.37 | NormAst | MikeJ: We will see if we can work something out |
| 04:09.02 | *** join/#asterisk stevecayona (~neondog@adsl-065-006-194-171.sip.pns.bellsouth.net) |
| 04:09.21 | *** join/#asterisk tmccrary (~tmccrary@d14-69-216-83.try.wideopenwest.com) |
| 04:09.44 | eric_ | heison: voctel.net gives me an under construction website... |
| 04:09.47 | tmccrary | Does anyone know of a good HOW-TO on asterisk? |
| 04:10.03 | eric_ | tmccrary: check out www.voip-info.org |
| 04:10.17 | eric_ | they have tons of resources... i dont remember specifically seeing a how-to |
| 04:10.36 | stevecayona | http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current_v1/docs-html/book1.html is a good starting place |
| 04:10.51 | tmccrary | thank you! |
| 04:12.28 | *** join/#asterisk leolarrel (~leolarrel@220-130-142-43.HINET-IP.hinet.net) |
| 04:12.39 | leolarrel | Hi, |
| 04:12.58 | *** part/#asterisk stevecayona (~neondog@adsl-065-006-194-171.sip.pns.bellsouth.net) |
| 04:13.27 | leolarrel | Now I porting asterisk into mipsel .... |
| 04:14.31 | Essobi | Wow I got chan_h323 to core -head! WEEEEEEEE! |
| 04:16.26 | leolarrel | Have any plan to re-design Makefile ? |
| 04:16.31 | *** join/#asterisk soundguy (~soundguy@zeus.blendtek.com.au) |
| 04:18.10 | Essobi | ast_channel_walk_locked: Avoiding initial deadlock for 'H323 WEEEE DEALOCKS ABOUND! |
| 04:19.55 | Essobi | Deadlocks even. |
| 04:20.44 | heison | eric_: if that's the case, check back in a few days... they are still very new |
| 04:29.10 | eric_ | heison: How'd u hear about them? |
| 04:33.31 | *** join/#asterisk rious (~rious@adsl-69-208-72-102.dsl.klmzmi.ameritech.net) |
| 04:33.36 | *** join/#asterisk Koshatul (~evangelio@inf-203-132-65-157.bne.ipnetworks.net.au) |
| 04:34.45 | mishehu | hmm is it me or does astdb seem to get either corrupt or filled with stale data after a while? |
| 04:35.04 | *** join/#asterisk NK123 (~p645@cpe-024-163-078-012.nc.rr.com) |
| 04:35.31 | *** join/#asterisk j0hn (~john@dsl-1.130.240.220.lns02-wick-bne.dsl.comindico.com.au) |
| 04:35.52 | Damin | drumkilla? |
| 04:35.57 | drumkilla | pong |
| 04:37.14 | mishehu | ping pong king kong wing wong sing song |
| 04:37.41 | NK123 | does any 1 know abt the govarion v400p E4 . Does it support R2 MFC |
| 04:37.47 | Damin | Hey.. read my latest update to the 1.0.7rc bugnote.. I'm willing to verify that this doesn't affect stable if someone can help me get shit setup on my devel box.. |
| 04:38.52 | Damin | drumkilla: Also.. related to http://bugs.digium.com/bug_view_page.php?bug_id=0003681 I posted an updated patch for stable.. |
| 04:39.02 | *** join/#asterisk mrproper_ (~b@61.95.55.242) |
| 04:39.30 | drumkilla | Damin: I committed 3681 |
| 04:39.36 | Damin | drumkilla: Rock! |
| 04:39.36 | mrproper_ | anyone here using asterisk-oh323? if so what version of asterisk are you using and what versions of oh323 did you end up using? |
| 04:40.12 | drumkilla | Damin: as for the agent stuff, anthm wrote out a description on how to reproduce it for cvs head |
| 04:40.13 | tmccrary | What kind of support does Asterisk have for LDAP? Or am I going to have to implement that myself |
| 04:40.37 | leolarrel | Have any plan to re-design Makefile ? |
| 04:40.57 | NK123 | Does the digium TE410p support R2 MFC? |
| 04:42.41 | Damin | drumkilla: I may be thick, but I'm not versed enough w/ agents to understand how to set it UP. :) |
| 04:43.12 | drumkilla | ha ... it's all good |
| 04:43.21 | drumkilla | I'd try to help, but I'm sick and have class in 8 hours |
| 04:43.25 | drumkilla | so I'm going to pass out |
| 04:43.31 | drumkilla | thank you for helping me out, though :) |
| 04:43.39 | drumkilla | like I said, there is no way for me to test all of this stuff |
| 04:43.52 | Damin | drumkilla: Anyone else that could help? :) |
| 04:44.13 | drumkilla | ha, I'm sure someone in this channel knows what to do :) |
| 04:44.27 | Damin | drumkilla: Well.. I'm available to help.... |
| 04:45.01 | drumkilla | awesome, well if you don't figure out the agents tonight, maybe we can get it tomorrow |
| 04:45.30 | drumkilla | but I'm fairly confident that it's not a 1.0 issue |
| 04:45.41 | drumkilla | because the person reporting it said that a previous version was working fine |
| 04:46.00 | drumkilla | that's usually a good sign |
| 04:46.01 | mrproper_ | anyone know what the better version of asterisk to use with h323 channels, and should i use the asterisk h323 channels or the oh323 |
| 04:46.18 | drumkilla | mrproper_: cvs head is better for h323 |
| 04:47.45 | mrproper_ | drumkilla: im having major issues compiling oh323 and the asterisk h323 channels (currently im using Asterisk CVS-HEAD-02/16/05-00:21:34) |
| 04:48.07 | drumkilla | I don't use them |
| 04:48.07 | JunK-U | mrproper_: have ya read the infos related to that? |
| 04:48.10 | drumkilla | bedtime for me ... |
| 04:48.17 | JunK-U | drumkilla: see ya buddy. |
| 04:48.59 | mrproper_ | JunK-U: yeah i have read them, still getting strange errors, for example using the oh323 latest release (0.7) i get errors relating to callerid when compiling and from googling that tells me to go back to using asterisk stable |
| 04:49.40 | JunK-U | im not using h323, so i cant help neither. |
| 04:50.18 | Trepalium | Do any of the zaptel drivers have proper reference counting under Linux kernel 2.6? I see no references to try_module_get in either the released version or the CVS version. :-( |
| 04:56.52 | mrproper_ | does anyone know if you can get sip audio+video to go across a h323 gateway? |
| 04:57.48 | azzazz|n | anyone like to share their .confz working with hop-1002 handsets? |
| 05:03.58 | eric_ | anyone know of any canadian toll free did providers? |
| 05:04.40 | tmccrary | What kind of support does Asterisk have for LDAP? Or am I going to have to implement that myself |
| 05:06.02 | NK123 | Does govarion v400p E4 support R2 MFC? |
| 05:06.04 | Trepalium | http://www.voip-info.org/wiki-Asterisk+cmd+LDAPget |
| 05:07.06 | *** join/#asterisk mickm (~mickm@220-245-98-72-qld-pppoe.tpgi.com.au) |
| 05:07.22 | tmccrary | Thank you1 |
| 05:08.22 | *** part/#asterisk tmccrary (~tmccrary@d14-69-216-83.try.wideopenwest.com) |
| 05:08.32 | Trepalium | I'm not sure if that's what you're looking for, though... |
| 05:08.42 | Trepalium | Oops... Too late. |
| 05:11.59 | *** join/#asterisk leolarrel (~leolarrel@220-130-142-43.HINET-IP.hinet.net) |
| 05:14.59 | eric_ | anyone have problems with x-lite and the # key? |
| 05:15.10 | eric_ | it seems to replace # for sip:% ... |
| 05:25.44 | IQ | why would someone use IAXy ? |
| 05:26.03 | Qwell | IQ: Why not? |
| 05:26.15 | IQ | Qwell: I asked first :) |
| 05:26.29 | Qwell | because SIP+NAT=suck |
| 05:27.27 | IQ | Qwell: I love * and IAX... its just this device I'm talking about... ifNAT is not the issue? |
| 05:27.42 | Qwell | in that case, dunno |
| 05:27.46 | Qwell | I'm sure there are plenty of uses |
| 05:27.53 | Qwell | I don't own one, so...couldn't tell you |
| 05:28.39 | *** join/#asterisk justinnnn (~dsf@solid.mpa.net.au) |
| 05:28.39 | IQ | it supports G711 only |
| 05:28.42 | justinnnn | hey ppls |
| 05:28.47 | justinnnn | anyone here played with 7914's ??? |
| 05:28.55 | justinnnn | i couldnt get the second one coming up with all the speed dials :( |
| 05:29.21 | *** join/#asterisk |Vulture| (~Vulture@c-24-129-110-157.se.client2.attbi.com) |
| 05:29.36 | |Vulture| | hey guys anyone use a Polycom SoundStation IP 4000 EX before? |
| 05:30.55 | *** join/#asterisk Mike (~mike@201.135.48.217) |
| 05:31.02 | Mike | someone knows the correct path for Mar 16 17:19:27 WARNING[2278]: libh323_linux_x86_r.so.1.12.2: cannot open shared object file: No such file or directory |
| 05:31.57 | *** join/#asterisk IcePenguin (~G4Master8@207-118-186-210.dyn.centurytel.net) |
| 05:32.07 | NK123 | Does govarion v400p E4/digium te410p support R2 MFC? |
| 05:32.21 | IcePenguin | Sounds like * has a big following... 282 users :D |
| 05:32.37 | *** join/#asterisk Secretive (~polarisx@c-67-161-5-149.client.comcast.net) |
| 05:32.49 | Secretive | Anyone know a good DID Provider for LD, Local & Toll-Free |
| 05:33.10 | IcePenguin | Okay... total newbie question |
| 05:33.24 | IcePenguin | Can asterisk take incoming calls and route them to PBX phones? |
| 05:33.36 | mstocco | IcePenguin: yes |
| 05:33.50 | IcePenguin | It is fairly easy to set up? |
| 05:34.25 | mstocco | IcePenguin: how well do you know your legacy PBX? |
| 05:34.30 | IcePenguin | pffy |
| 05:34.44 | IcePenguin | this would be a first time thing |
| 05:34.47 | IcePenguin | something to play with |
| 05:35.16 | |Vulture| | you prolly need a channel banl |
| 05:35.22 | *** join/#asterisk elriah (~jfulcrum@adsl-068-209-198-242.sip.bhm.bellsouth.net) |
| 05:35.28 | *** join/#asterisk terrapen_ (~cjs@cpe-66-25-86-139.satx.res.rr.com) |
| 05:35.37 | mstocco | IcePenguin: does your legacy PBX use T1? |
| 05:35.40 | terrapen_ | howdy |
| 05:36.12 | IcePenguin | mstocco... i would be using this in a home/small business setting. |
| 05:36.16 | IcePenguin | small install |
| 05:36.37 | elriah | Hi all - I just built my first asterisk box tonight, I'm using the iaxcomm softphone to test. It's a 1GHz Via Epia MS10000 with 512MB of RAM and a 40GB HD. I've only tried with iaxcomm softphone, but it seems If I have more than 2 or 3 connections the audio quality goes to heck. Any suggestions? think it could be the softphone? |
| 05:37.36 | elriah | It's defaulting to the GSM codec. |
| 05:37.39 | terrapen_ | what os is your client? |
| 05:38.03 | mstocco | IcePenguin: big or small doesn't matter, what kind of PBX do you have now? |
| 05:38.08 | IcePenguin | mstocco... can asterisk take incoming calls over POTS? |
| 05:38.12 | IcePenguin | using a modem? |
| 05:38.17 | `Sauron | nope |
| 05:38.22 | `Sauron | need a x100p or tdm400 |
| 05:38.31 | *** part/#asterisk t3t (~t3t@galley.pangalacticgargleblaster.com) |
| 05:38.35 | mstocco | IcePenguin: yes, get a card from Digium |
| 05:39.03 | |Vulture| | hey guys anyone use a Polycom SoundStation IP 4000 EX before? |
| 05:39.08 | IcePenguin | Digium has modem cards? |
| 05:39.11 | mstocco | IcePenguin: the tdm400 |
| 05:40.13 | IcePenguin | okay... so I can interface with my standard rj11 POTS with the tdm400? |
| 05:40.18 | `Sauron | yes |
| 05:40.30 | Trepalium | Yes. That's what the TDM400 is for. |
| 05:40.52 | IcePenguin | excellent |
| 05:40.57 | Trepalium | Along with interfacing with analog phones, if you want. |
| 05:40.58 | IcePenguin | thanks much guys |
| 05:41.00 | `Sauron | well, it's also for connecting phones to * |
| 05:41.19 | IcePenguin | i can have a input tdm400 and an output tdm400? |
| 05:41.24 | mstocco | yeah, its a modular card |
| 05:41.53 | IcePenguin | very excellent |
| 05:41.56 | IcePenguin | thanks so much guys |
| 05:42.09 | Trepalium | The TDM card takes 4 lines, what type depends on what modules are on it. FXO for POST lines, FXS for analog phones. |
| 05:42.32 | Trepalium | er, POTS lines. |
| 05:42.37 | mstocco | :0 |
| 05:42.46 | Secretive | Who is everyone using for VOIP IAX termination / DID |
| 05:44.14 | CoaxD | Secretive: And you're asking this at 11:43pm on a wednesday ngiht? :) |
| 05:44.25 | CoaxD | Secretive; I use NuFone. others might give you other info |
| 05:44.30 | *** part/#asterisk IcePenguin (~G4Master8@207-118-186-210.dyn.centurytel.net) |
| 05:44.45 | bkw_ | LIIIIIIIIIIIIIIIIIIIIIIIIIIIIIIIARS |
| 05:44.56 | leolarrel | ? |
| 05:44.56 | CoaxD | bkw: Hello, sweet thang |
| 05:45.03 | |Vulture| | bkw_: is teh ghey |
| 05:45.04 | |Vulture| | :P |
| 05:45.14 | CoaxD | Vulture: It aint only bkw! |
| 05:45.16 | bkw_ | http://www2.foxsearchlight.com/napoleondynamite/ <-- fucking waste of a good DVD |
| 05:45.19 | |Vulture| | sup man? |
| 05:45.20 | bkw_ | stupid |
| 05:45.22 | bkw_ | stupid movie |
| 05:45.24 | bkw_ | totally stupid |
| 05:45.33 | bkw_ | so stupid.. I think I'll go shoot myself now |
| 05:45.34 | |Vulture| | bkw_: down boy down |
| 05:45.45 | CoaxD | Vulture: (I'm not gay. but, for the purposes of this conversation.. you know) |
| 05:45.49 | niZon | bkw_: agreed. |
| 05:45.50 | leolarrel | have anybody do prot astrisk to No-x86 platform ? |
| 05:45.54 | |Vulture| | bkw_: lemme cut off your head with a sword first so I can steal your powers |
| 05:45.59 | bkw_ | leolarrel, it already works on my ppc |
| 05:46.00 | niZon | terrible movie |
| 05:46.01 | |Vulture| | CoaxD: of course! |
| 05:46.14 | leolarrel | thank you bkw |
| 05:46.47 | leolarrel | is use cross-conple or into ppc linux and native compile ? bkw |
| 05:46.50 | bkw_ | IIIIIIIIIIIIIIIIIIIIIIIIIIIIIIIIIIIBOOK |
| 05:46.53 | bkw_ | OSX baby |
| 05:47.03 | |Vulture| | bkw_: god.... have mercy |
| 05:47.21 | bkw_ | now that that monkey is off my back |
| 05:47.24 | leolarrel | into OS X and native compile ? |
| 05:48.18 | Trepalium | Okay, now I have probably pissed off a developer by reopening the bug I submitted yesterday. :-/ |
| 05:48.33 | CoaxD | bkw: Aww yeah babe |
| 05:48.36 | CoaxD | bkw: Yer makin' me hot |
| 05:48.37 | leolarrel | I am porting astrisk into mipsel , I must use 'cross-compile" to porting |
| 05:48.58 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) |
| 05:49.05 | CoaxD | leolarrel: Probably wont be too hard to port * to a mips processor. |
| 05:49.16 | leolarrel | Makefile have bug very mush |
| 05:49.24 | CoaxD | leolarrel: Anything the processor doesn't support, you can support with compiler tricks |
| 05:49.27 | |Vulture| | rewired one of my offices today.. that was a bitch and half |
| 05:49.34 | |Vulture| | 22 runs |
| 05:49.43 | CoaxD | leolarrel: Well, its all in the #defines everywhere in .c files that probably need to be added to or changed |
| 05:50.05 | leolarrel | well , thank you. |
| 05:50.13 | CoaxD | plus, you'll be hard pressed to get zaptel to work there.. |
| 05:50.14 | leolarrel | thank you , CoaxD |
| 05:50.24 | CoaxD | leolarrel: Sure! Although i'm not quite sure what you're thanking me for? |
| 05:50.29 | *** join/#asterisk KirkL (~me@c-24-22-57-111.client.comcast.net) |
| 05:50.40 | CoaxD | leolarrel: In any event, I wish you luck! |
| 05:52.51 | leolarrel | in Makefile , have very much XXX=$(shell if [ -f /usr/include/zaptel/yyyy.h ] then echo "xxx.so") but , this do not work in 'cross-compile' envariment |
| 05:55.17 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 05:56.45 | *** join/#asterisk leolarrel (~leolarrel@220-130-142-43.HINET-IP.hinet.net) |
| 05:57.38 | *** join/#asterisk yertle (yertle@ip68-6-98-122.sb.sd.cox.net) |
| 05:57.48 | bkw_ | ok |
| 05:57.49 | bkw_ | i'm back |
| 05:57.59 | justinnnn | anyone played with 7914's ??? |
| 05:58.12 | bkw_ | 7914 works with SCCP only |
| 05:58.16 | justinnnn | ya |
| 05:58.20 | bkw_ | and no |
| 05:58.23 | justinnnn | ive got a 7960 with 2 7914's attached.. with sccp |
| 05:58.25 | justinnnn | working 100% |
| 05:58.30 | bkw_ | yay for you |
| 05:58.33 | justinnnn | except the second 7914 wont load up the speed dials |
| 05:58.38 | bkw_ | haha |
| 05:58.40 | bkw_ | two of them |
| 05:58.41 | bkw_ | IC |
| 05:58.43 | bkw_ | no clue |
| 05:58.45 | bkw_ | send me some and i'll test |
| 05:58.48 | justinnnn | :P |
| 05:58.53 | justinnnn | there stupid |
| 05:58.55 | bkw_ | and he thinks i'm kidding |
| 05:59.00 | justinnnn | in call manager u get the lines |
| 05:59.01 | leolarrel | bkw , your asterisk is into OS X , and native compile ? |
| 05:59.04 | justinnnn | with asterisk u get 'speed dials' |
| 05:59.08 | bkw_ | leolarrel, yes |
| 05:59.09 | justinnnn | how crap |
| 05:59.13 | leolarrel | thank you bkw |
| 05:59.28 | bkw_ | the only thing holding back alot of stuff in asterisk is people sending me lots of hardware for testing ya know |
| 05:59.42 | bkw_ | I like hardware |
| 05:59.49 | leolarrel | ^^ |
| 05:59.51 | Beirdo | OK, now to setup the SPA3000 |
| 05:59.54 | brc_ | bkq |
| 06:00.14 | bkw_ | ya send me one of those too |
| 06:00.16 | bkw_ | whois bkw.org |
| 06:00.25 | bkw_ | valid shipping info there boyz and girlz |
| 06:00.39 | yertle | snicker |
| 06:01.13 | bkw_ | hey can't blame a girl for trying now can ya |
| 06:01.22 | Qwell | g-g-g-irl?! |
| 06:01.32 | brc_ | what, surprised? |
| 06:01.45 | Qwell | not really |
| 06:02.30 | Qwell | just has a manly voice...and name... |
| 06:02.47 | Qwell | and, we all know there are no girls on irc |
| 06:03.00 | bkw_ | actually their is |
| 06:03.05 | Qwell | yeah, two |
| 06:03.17 | brc_ | there are |
| 06:03.18 | Trepalium | What's that saying. IRC, where men are men, women are men, and 13 year old girls are FBI agents, who are men. |
| 06:03.19 | yertle | is would be singular, so one |
| 06:03.26 | Qwell | yertle: heh |
| 06:03.36 | Qwell | Trepalium: ;] |
| 06:03.48 | brc_ | freebsd roxorz~!! |
| 06:04.13 | brc_ | no really |
| 06:04.16 | brc_ | I just installed it |
| 06:04.18 | brc_ | really nice |
| 06:04.34 | brc_ | ports is a little odd though with all these make files |
| 06:04.37 | *** join/#asterisk clive- (~pirch@rndf-146-22-79.telkomadsl.co.za) |
| 06:05.14 | |Vulture| | can anyone recommend a good headset between 50-100 for telemarketers... using IP500 phones |
| 06:05.27 | Qwell | telemarketers...vulture...it fits |
| 06:05.30 | brc_ | plantronics |
| 06:05.30 | yertle | grrr |
| 06:05.41 | brc_ | don't bother with anything else |
| 06:05.43 | |Vulture| | lol |
| 06:05.46 | |Vulture| | yea I know |
| 06:05.49 | |Vulture| | but which one |
| 06:05.52 | |Vulture| | they have voice tubles |
| 06:05.58 | brc_ | depends on what style they prefer |
| 06:05.59 | |Vulture| | noise-canceling regular ones |
| 06:06.03 | |Vulture| | single ear, dual |
| 06:06.05 | leolarrel | wait ... |
| 06:06.11 | brc_ | dual ear are awful |
| 06:06.15 | brc_ | erm |
| 06:06.19 | |Vulture| | brc_: oky so single ear... |
| 06:06.23 | yertle | they are? |
| 06:06.33 | |Vulture| | brc_: whats the diff between voice tube and regular? |
| 06:06.44 | *** part/#asterisk leolarrel (~leolarrel@220-130-142-43.HINET-IP.hinet.net) |
| 06:06.50 | brc_ | not sure, I've always used and seen voice tube, work great |
| 06:06.57 | Qwell | |Vulture|: for telemarketers, I'd get the most uncomfortable set I could find |
| 06:07.14 | yertle | nod |
| 06:07.14 | Qwell | If they're gonna call me at dinner, they better be uncomfortable. :p |
| 06:07.22 | brc_ | voice tube is where the mic is up in the speaker part, and there's a clear plastic stiff tube that goes down to near your mouth |
| 06:08.18 | brc_ | you might want to get one of each style and let everybody pick, but if you're just gonna get one get the standard over the head style |
| 06:09.09 | Trepalium | Qwell: I hear the ones that have three inch screws that must be drilled into the telemarketer's head work well for that. |
| 06:09.14 | |Vulture| | Qwell: lol not those type of telemarketers |
| 06:09.17 | yertle | and provide electric shocks too |
| 06:09.17 | Qwell | Trepalium: whatever works |
| 06:09.18 | Qwell | meh |
| 06:09.23 | Qwell | telemarketers are telemarketers :P |
| 06:09.34 | |Vulture| | they call businesses for commercial cleaning |
| 06:09.39 | |Vulture| | they don't call homes |
| 06:09.43 | yertle | oh so they're the ones that call my work |
| 06:09.47 | |Vulture| | yup |
| 06:09.53 | Qwell | If they're gonna call me at work during lunch... |
| 06:10.41 | |Vulture| | nah they don't work enough to do that LOL |
| 06:11.00 | NK123 | Does govarion v400p E4/digium te410p support R2 MFC? |
| 06:12.08 | brc_ | no clue, ask them |
| 06:13.16 | NK123 | ok |
| 06:13.16 | bkw_ | sleepy time for me |
| 06:13.20 | bkw_ | HAVE FUN GUYS |
| 06:13.39 | carrar | ok!! |
| 06:13.47 | carrar | <FUN> |
| 06:18.06 | *** join/#asterisk joaovianna (naturalvoi@node-40247a6a.ewr.onnet.us.uu.net) |
| 06:18.38 | joaovianna | Any brazilian ? |
| 06:19.11 | brc_ | cobryce, DUDE! |
| 06:20.26 | Beirdo | ah dammit |
| 06:20.33 | Beirdo | I forget my password on the spa2100 |
| 06:21.13 | Beirdo | OK, there's the user password |
| 06:21.35 | *** join/#asterisk RestLessGemini (~umairbari@202.142.189.86) |
| 06:27.10 | *** join/#asterisk Inv_arp (junya@adsl-8-232-25.mia.bellsouth.net) |
| 06:27.34 | terrapen_ | holy shit, what a crazy 24 hours |
| 06:27.40 | terrapen_ | i almost died last night, like, for real |
| 06:28.35 | Inv_arp | terrapen: how? |
| 06:32.57 | cobryce | brc_: .... |
| 06:33.03 | brc_ | I don't either much |
| 06:34.35 | PTG123 | Hey anyone know how i assign a specific proxy to a specific line on a cisci 7960? |
| 06:35.36 | *** join/#asterisk Mazda-MX5 (~leolarrel@220-130-142-43.HINET-IP.hinet.net) |
| 06:36.19 | terrapen_ | i went to the shooting range with a friend |
| 06:36.29 | |Vulture| | this can't go anywhere good |
| 06:36.32 | terrapen_ | and then i went to his condo later that day to clean our weapons |
| 06:36.43 | *** join/#asterisk booyeah7 (~afdas@cpe-24-175-29-253.houston.res.rr.com) |
| 06:36.46 | terrapen_ | and we were sitting on the ground on his concrete-walled porch, cleaning our handguns |
| 06:36.46 | *** join/#asterisk marc32344 (~marc32344@69-90-241-15.dsl.teksavvy.com) |
| 06:36.54 | terrapen_ | sitting right next to each other |
| 06:37.06 | terrapen_ | my head was down and i was focused on cleaning my M1911 |
| 06:37.23 | booyeah7 | is there a way to create a queue programmatically, or motify its attributes programmaticly? |
| 06:37.30 | booyeah7 | *modify |
| 06:37.35 | carrar | TG123, proxy2_address: "x.x.x.x" |
| 06:37.35 | terrapen_ | all of the sudden, he accidentally chambers a (hollow-point) round in his .45 |
| 06:37.39 | carrar | TG123, proxy3_address: "x.x.x.x" |
| 06:37.40 | carrar | etc... |
| 06:37.45 | terrapen_ | and the slide rides forward and the hammer falls |
| 06:37.48 | yertle | how did he manage that |
| 06:37.55 | |Vulture| | wtf were you doing with bullets in the clip? |
| 06:37.59 | terrapen_ | and the round goes through his leg |
| 06:37.59 | |Vulture| | shit... |
| 06:38.04 | |Vulture| | omg |
| 06:38.06 | terrapen_ | and off a thick metal flower pot |
| 06:38.10 | terrapen_ | and zings all around the porch |
| 06:38.11 | booyeah7 | you mean .40 |
| 06:38.15 | |Vulture| | bad wound? |
| 06:38.16 | *** join/#asterisk edifis (~nchachan@202.56.199.66) |
| 06:38.17 | booyeah7 | you talking about the dea one? |
| 06:38.18 | |Vulture| | ACP? |
| 06:38.35 | terrapen_ | had his weapon been turned the other way (ie., if he was left-handed) i would be a dead man, brains on the wall |
| 06:38.43 | terrapen_ | Springfield .45 |
| 06:38.50 | terrapen_ | a M1911 clone is what he had |
| 06:38.56 | |Vulture| | terrapen: thats gunna live with you for awhile |
| 06:39.04 | terrapen_ | fuck if i know why he was trying to clean a handgun with rounds in the clip |
| 06:39.06 | yertle | ears ringing a bit? |
| 06:39.23 | |Vulture| | terrapen: did you just kinda sit ther for a sec. dazed? |
| 06:39.24 | terrapen_ | the first thing i learned about weapons in the Army was to drop magazine and clear the chamber every fucking time you finish shooting |
| 06:39.26 | edifis | Hi, i need some help to set up a TDM400 card |
| 06:39.35 | terrapen_ | yertle: holy shit you better believe... |
| 06:39.41 | terrapen_ | i was deaf in my left ear for 10min |
| 06:39.46 | yertle | heh |
| 06:39.47 | |Vulture| | terrapen: drop the mag, and slide the rail 2 times.... |
| 06:39.54 | |Vulture| | terrapen: you knwo that, why didn't he |
| 06:40.01 | terrapen_ | this happened 28 hours ago and i still can barely hear |
| 06:40.06 | terrapen_ | vulture, he knew it |
| 06:40.13 | terrapen_ | he was a little stoned, i think |
| 06:40.20 | |Vulture| | terrapen: not a good combo |
| 06:40.28 | terrapen_ | after it happened, i look over at him |
| 06:40.36 | terrapen_ | the round had grazed his hand |
| 06:40.40 | terrapen_ | i thought it had gone through |
| 06:40.45 | *** join/#asterisk Inv_arp (junya@adsl-3-245-254.mia.bellsouth.net) |
| 06:40.47 | terrapen_ | then i saw his pants leg bloody |
| 06:40.56 | terrapen_ | so i grabbed the knife from his pocket and cut his pants off |
| 06:41.07 | terrapen_ | and did first aid as best as i could remember (from the army) |
| 06:41.13 | terrapen_ | it was un-fucking-real |
| 06:41.15 | |Vulture| | terrapen: was it a bad wound? |
| 06:41.59 | Grooby | sup peeps |
| 06:42.05 | Grooby | what's this new jitter patch I read |
| 06:42.05 | terrapen_ | welll |
| 06:42.10 | terrapen_ | he totally lucked out |
| 06:42.16 | terrapen_ | first off, the bullet did not expand |
| 06:42.25 | terrapen_ | which is amazing, considering it was a hollow point |
| 06:42.39 | terrapen_ | then it missed his femural artery |
| 06:42.54 | terrapen_ | and the main nerve that runs down the leg |
| 06:43.04 | *** part/#asterisk edifis (~nchachan@202.56.199.66) |
| 06:43.08 | terrapen_ | it hit his femur right wheere it expands before the knee joint |
| 06:43.09 | yertle | jeez |
| 06:43.11 | terrapen_ | and took out 30% of it |
| 06:43.48 | *** join/#asterisk edifis (~nchachan@202.56.199.66) |
| 06:44.03 | terrapen_ | i was fine until today |
| 06:44.07 | terrapen_ | ive had a rough day |
| 06:44.12 | terrapen_ | it started to sink in this afternoon |
| 06:45.09 | terrapen_ | so, kids, make sure you clear your fucking weapons after shooting them and do not fuck around with a weapon that you do not know (100%) is cleared and safe |
| 06:45.31 | *** join/#asterisk denon (denon@synapse.subneural.net) |
| 06:45.31 | *** mode/#asterisk [+o denon] by ChanServ |
| 06:45.36 | yertle | crazy |
| 06:45.39 | terrapen_ | yeah |
| 06:47.04 | edifis | Hi can anybody help me here. Having problems installing a TDM400 digium card |
| 06:47.28 | cobryce | go ahead edifis |
| 06:47.45 | edifis | i have a TDM400 card with one FXO module |
| 06:48.06 | edifis | i configured the zapata and zaptel config files |
| 06:48.26 | edifis | in ztcfg -vv , it says |
| 06:48.28 | edifis | Channel 01: FXS Kewlstart (Default) (Slaves: 01) |
| 06:48.29 | edifis | 1 channels configured. |
| 06:48.41 | edifis | zttool also shows the card and ok |
| 06:48.52 | edifis | when i sart asterisk, it says |
| 06:49.08 | edifis | Mar 17 01:15:59 WARNING[13587]: chan_zap.c:771 zt_open: Unable to specify channel 1: No such device |
| 06:49.08 | edifis | Mar 17 01:15:59 ERROR[13587]: chan_zap.c:6208 mkintf: Unable to open channel 1: No such device |
| 06:49.08 | edifis | here = 0, tmp->channel = 1, channel = 1 |
| 06:49.08 | edifis | Mar 17 01:15:59 ERROR[13587]: chan_zap.c:9155 setup_zap: Unable to register channel '1' |
| 06:49.08 | edifis | Mar 17 01:15:59 WARNING[13587]: loader.c:345 ast_load_resource: chan_zap.so: load_module failed, returning -1 |
| 06:49.10 | edifis | <PROTECTED> |
| 06:49.12 | edifis | <PROTECTED> |
| 06:49.14 | edifis | Mar 17 01:15:59 WARNING[13587]: loader.c:440 load_modules: Loading module chan_zap.so failed! |
| 06:49.46 | edifis | so I am stuck here |
| 06:50.01 | edifis | Am using asterisk-1.0.6 |
| 06:51.10 | Mazda-MX5 | chan_zap.so is in /usr/lib/asterisk/modules ? |
| 06:51.10 | *** join/#asterisk criptos (~criptos@201.129.126.24) |
| 06:51.23 | edifis | one sec checking |
| 06:51.47 | edifis | yes its there |
| 06:51.57 | |Vulture| | wow... edifis... pastebin.ca |
| 06:52.50 | edifis | whats that? |
| 06:53.10 | Mazda-MX5 | channel 1 failed ... |
| 06:53.16 | *** join/#asterisk pranav (pranav@202-149-48-208.broadband.isp.exatt.net) |
| 06:53.39 | Mazda-MX5 | I do not know why it hapeen , sorry edifis |
| 06:53.45 | |Vulture| | ztcfg -vvv |
| 06:53.46 | Mazda-MX5 | I do not know why it hapeen ,too . sorry edifis |
| 06:53.46 | *** join/#asterisk terracon (~tc@CPE0050da608e99-CM0012254076d6.cpe.net.cable.rogers.com) |
| 06:53.49 | edifis | thanx anyway |
| 06:53.49 | |Vulture| | do that edifis |
| 06:53.51 | pranav | hello everyone |
| 06:54.03 | Mazda-MX5 | hello |
| 06:54.20 | pranav | can someone tell me hoe to register the X-Lite soft phone with asterisk |
| 06:54.20 | edifis | [root@localhost asterisk-1.0.6]# ztcfg -vv |
| 06:54.20 | edifis | Zaptel Configuration |
| 06:54.20 | edifis | ====================== |
| 06:54.20 | edifis | Channel map: |
| 06:54.20 | edifis | Channel 01: FXS Kewlstart (Default) (Slaves: 01) |
| 06:54.20 | edifis | 1 channels configured. |
| 06:55.41 | pranav | actually i want to use it for FWD , can i use it directly without connecting to asterisk |
| 06:55.43 | |Vulture| | edifis: is it FXO or FXS? |
| 06:55.53 | edifis | FXO |
| 06:56.09 | edifis | so fxs signalling |
| 06:56.25 | |Vulture| | X100P? |
| 06:56.38 | pranav | Mazda-MX5:any idea? |
| 06:56.42 | edifis | TDM400 with 1 FXO module |
| 06:56.47 | joaovianna | pranav: You have to create a entry in you sip.conf |
| 06:57.14 | Mazda-MX5 | I am newbie of asterisk , too . sorry. |
| 06:57.16 | pranav | ya but can i use it to connect to FWD without using asterisk |
| 06:57.17 | |Vulture| | edifis: go to pastebin.ca and paste your zaptel.conf and zapata.conf |
| 06:57.27 | edifis | ok |
| 06:58.17 | edifis | doing that but how would that help? |
| 06:58.47 | |Vulture| | then I can see what you config is and see where the problem is |
| 06:58.57 | PTG123 | do any good ip phones have a backlit display? |
| 06:59.03 | |Vulture| | looks like zapata.conf is configured wrong |
| 06:59.31 | edifis | ok sure |
| 06:59.33 | pranav | joaovianna:i'll have to make entry in sip.conf as well as extensions.conf |
| 07:01.08 | pranav | but my main aim is to connect to FWD and asterisk not, my network is behind NAT so with SIP it wont work |
| 07:01.23 | Mazda-MX5 | what is doing that codecs/gsm/*.so ? |
| 07:01.40 | joaovianna | Yes, sinse you have a entry in your sip.conf you can see your asterisk registering your software and then you create a extension for it in your extension.conf |
| 07:01.46 | *** part/#asterisk yertle (yertle@ip68-6-98-122.sb.sd.cox.net) |
| 07:02.04 | pranav | but i wont be able to dial to FWD |
| 07:02.07 | Mazda-MX5 | whay the 'make install' not install this *.so file ? |
| 07:02.44 | pranav | i have configured the sipura phone with asterisk so i know the configurations |
| 07:02.56 | pranav | but i am not abe to connect with FWD |
| 07:04.03 | joaovianna | I have a softfhone and a sip adapter behing nat, connected with fwd / naturalvoice and it is working... |
| 07:04.23 | pranav | which soft phone are u using? |
| 07:05.00 | edifis | hi Vulture , pasted - its here http://pastebin.ca/7615 |
| 07:05.06 | joaovianna | Xten softphone |
| 07:05.15 | pranav | i am using the same |
| 07:05.38 | Mazda-MX5 | I am porting asterisk to do IP-BPX |
| 07:05.40 | pranav | can u psate ur configurations if u dont mind? |
| 07:05.42 | joaovianna | Can you check if your asterisk have your softphone as peer ? |
| 07:06.55 | pranav | no its not registered with sip |
| 07:07.08 | *** join/#asterisk tessier (~treed@210.245.115.7) |
| 07:07.31 | pranav | when i give sip show peers , it does not show the soft phone registration |
| 07:07.48 | |Vulture| | edifis: try using http://pastebin.ca/7616 |
| 07:07.53 | |Vulture| | jsut change your context |
| 07:07.57 | edifis | ok sure |
| 07:08.11 | joaovianna | check in your xten configuration (F8) if you have your * IP (192.168.xxx.xxx) as sip server. |
| 07:09.02 | Mazda-MX5 | what is asterisk/codecs/gsm/*.so for ? why it not install by "make install" ?? |
| 07:09.20 | Mazda-MX5 | what is asterisk/codecs/gsm/*.so for ? why it not ibe nstaled l by "make install" ?? |
| 07:09.29 | Mazda-MX5 | Orz |
| 07:09.42 | joaovianna | pranav: Put your asterisk in debug mode to see what happening (asterisk -vvvvvvvvvvvvvvc) |
| 07:09.47 | Mazda-MX5 | what is asterisk/codecs/gsm/*.so for ? why it not be installed by "make install" ?? |
| 07:09.55 | |Vulture| | Mazda-MX5: shut the hell up |
| 07:11.23 | edifis | Nope no change |
| 07:12.03 | justinnnn | ppls |
| 07:12.15 | Mazda-MX5 | sorry... |
| 07:12.18 | pranav | see in the system settings-network-outbound proxy i have entered my asterisk ip |
| 07:12.52 | |Vulture| | edifis: pastebin your debug |
| 07:12.57 | *** join/#asterisk marc32344 (~marc32344@69-90-241-15.dsl.teksavvy.com) |
| 07:13.02 | edifis | same. no change |
| 07:13.18 | |Vulture| | pastebin the whole error I want to see it as one |
| 07:13.26 | |Vulture| | make sure your in at least verbose 3 |
| 07:13.31 | edifis | http://pastebin.ca/7617 |
| 07:13.35 | pranav | see in the system settings-network-network-default-sip proxy -asterisk ip address |
| 07:13.38 | edifis | ok one sec |
| 07:13.44 | edifis | done |
| 07:13.56 | booyeah7 | anyone use pgsql with asterisk? |
| 07:14.14 | booyeah7 | also is it possible to programmaticlly add/change queues ? |
| 07:14.23 | booyeah7 | other than changing the config then reloading |
| 07:14.35 | Inv_arp | booyeah7: yep thru agi |
| 07:15.06 | booyeah7 | i know how to dynamically add/remove members and show queue status |
| 07:15.30 | booyeah7 | but i cant see how i could change a queue from roundrobin to ringall |
| 07:15.44 | pranav | joaovianna:are u there |
| 07:15.49 | |Vulture| | edifis: and a ztcfg -vvv shows it as all configed? |
| 07:16.24 | booyeah7 | i dont believe that is possible in agi |
| 07:16.40 | edifis | yes |
| 07:16.47 | edifis | zttool also says ok |
| 07:17.01 | booyeah7 | the only way i can think of is modifying the asterisk objects in memory |
| 07:17.27 | *** join/#asterisk mesi (~player@dsl-082-083-132-146.arcor-ip.net) |
| 07:18.33 | pranav | hello anyone familiar with the X-ten or X-Lite phone configurations |
| 07:18.37 | *** join/#asterisk pascals (~248d34d6@ip503c8584.speed.planet.nl) |
| 07:18.58 | |Vulture| | edifis: very strange |
| 07:19.11 | booyeah7 | pranav: yes |
| 07:19.26 | pascals | good morning. |
| 07:19.49 | pranav | can u tell me what entries we have to make in the sip.conf and extensions.conf |
| 07:20.47 | SexyKen | Anyone know anything about Polycom XML? |
| 07:21.09 | *** join/#asterisk ACiDV (~joel@122-68-181.dr.cgocable.ca) |
| 07:21.10 | edifis | thanx anyway Vulture |
| 07:21.17 | pascals | SexyKen: ask me next wednessday :) |
| 07:21.25 | *** part/#asterisk edifis (~nchachan@202.56.199.66) |
| 07:21.29 | pranav | booyeah7: see inside the phone settings i have set the ip address of my asterisk server |
| 07:21.35 | wildcard0 | when you find out next wednesday, send it on to me also :) |
| 07:21.53 | pranav | but how to register it with asterisk |
| 07:21.58 | SexyKen | pascals, ? |
| 07:22.03 | pascals | I'll be playing with one or two ip500's next monday |
| 07:22.12 | terrapen_ | sexyken, what about the XML |
| 07:22.16 | pascals | And tuesday is my day off |
| 07:22.26 | *** join/#asterisk Himeko (~himeko@S01060040ca128fc3.ed.shawcable.net) |
| 07:22.30 | ACiDV | With IAX debug, I can see message error like : No such context/extensions but dont give more detail like the transmitted/received context/extension. Does exist a way to do this ? |
| 07:22.33 | booyeah7 | pranav: ok |
| 07:23.17 | *** join/#asterisk shantanoo (~shantanoo@shantanoo.user) |
| 07:23.17 | booyeah7 | pranav: you running sip debug? |
| 07:23.20 | shantanoo | hi!!!!!!! |
| 07:23.21 | SexyKen | terrapen, I want to know if I can get queue information like how many people are waiting, agent info, etc. |
| 07:23.35 | Mazda-MX5 | hi.. |
| 07:23.54 | shantanoo | i will need help after some time :) |
| 07:24.02 | shantanoo | rt now cvsupping |
| 07:24.04 | shantanoo | :) |
| 07:24.28 | shantanoo | any specific points i should take care of while I compile? |
| 07:25.00 | pranav | booyeah7 : yes sip debug is enabled |
| 07:25.28 | SwedMiro | anyone but me that have experienced memory leak with asterisk@home? |
| 07:25.48 | Mazda-MX5 | shantanoo > asterisk need openssl |
| 07:26.07 | SexyKen | terrapen, Are you there? |
| 07:26.17 | *** join/#asterisk r0d3nt|m (anonymous@soveliss.luniac.com) |
| 07:26.17 | Mazda-MX5 | shantanoo > and ncurses library |
| 07:26.23 | shantanoo | Mazda-MX5: openssl is there |
| 07:26.36 | shantanoo | ncurses is there |
| 07:27.05 | elric | might need OpenH323 |
| 07:27.11 | Mazda-MX5 | Make use -O6, maybe you want to change |
| 07:27.19 | elric | if you want H323 |
| 07:27.42 | shantanoo | will be using on intranet. without telephony cards |
| 07:27.57 | elric | shantanoo, make sure you have kernel headers |
| 07:28.07 | shantanoo | :-O |
| 07:28.14 | pascals | I don't quite understand the s extension - isn't that a wildcard extension I can use for all incomming calls? |
| 07:28.30 | elric | * will need kernel headers to compile |
| 07:28.53 | brc_ | pascals, the extension s is gone to if a call goes into a context but isn't sent to a specific extension |
| 07:28.56 | shantanoo | kernel-devel-2.6.9-5.EL <-- does this include kernel headers? |
| 07:28.59 | brc_ | so, yes |
| 07:29.00 | shantanoo | on RHEL |
| 07:29.20 | elric | check /usr/src |
| 07:29.33 | pascals | brc_: but not if the call is incomming from the pstn? |
| 07:29.37 | ACiDV | pascals you can use exten => _X.,1,etc... for a wildcard |
| 07:29.57 | shantanoo | there is something over there is /usr/src |
| 07:30.00 | elric | i havent used redhat since 5.2 - 6 days |
| 07:30.07 | pascals | I am using bristuff, and it doesn't start ringing any phones |
| 07:30.19 | shantanoo | 2.6.9-5.EL-i686 <--- dir in /usr/src |
| 07:30.27 | pascals | But with a specific msn, it will ring the phones |
| 07:30.40 | shantanoo | so i hope it works |
| 07:30.44 | elric | make a softlink to linux |
| 07:30.47 | pascals | So the call must already have been answered by * for it to use the s extension? |
| 07:31.23 | brc_ | no |
| 07:31.24 | brc_ | no |
| 07:31.28 | shantanoo | /usr/src/linux --> /usr/src/kernels/2.6.9-5.EL-i686 |
| 07:31.32 | shantanoo | ? |
| 07:31.36 | elric | yes |
| 07:31.42 | shantanoo | ok |
| 07:31.46 | wildcard0 | pascals, no. often when you're using zap devices (non-pri), you'll get an incoming call but not to a specific extension. you specify the context and it'll go to the s exten in that |
| 07:31.47 | brc_ | pascals, you got tdm hardware? |
| 07:32.06 | pascals | brc_ no beronet HFC-4s card |
| 07:32.12 | brc_ | what wildcard0 said |
| 07:33.37 | pascals | wildcard0: Doesn't seem to do that - the zapata has the context 'extern' in it, and it does go there, but * says: |
| 07:34.18 | pascals | <PROTECTED> |
| 07:34.35 | brc_ | uhm |
| 07:34.35 | brc_ | dude |
| 07:34.55 | brc_ | that's because it's being sent to 0104514890 in the extern context, and that extension does not exist |
| 07:35.07 | wildcard0 | wait. the hfc-4s is an isdn card, isn't it? |
| 07:35.12 | terrapen_ | pascals, you have to add an extension that will match that number |
| 07:35.13 | pascals | Yes, it is. |
| 07:35.15 | elric | pascals, make an extension in extern context |
| 07:35.16 | pascals | isdn |
| 07:35.20 | wildcard0 | then you don't need the s exten |
| 07:35.23 | brc_ | pascals, s is only for calls that ARE NOT SENT TO AN EXTENSION |
| 07:35.26 | wildcard0 | you need to create a 0104514890 exten |
| 07:35.32 | brc_ | wildcard0, no reason to remove it |
| 07:35.37 | brc_ | no |
| 07:35.42 | brc_ | what he wants is a pattern match |
| 07:35.48 | terrapen_ | yep |
| 07:35.55 | terrapen_ | 01NXXXXXXX |
| 07:35.55 | wildcard0 | yes. he can use a _X. or something also |
| 07:35.59 | terrapen_ | err |
| 07:36.00 | terrapen_ | yeah |
| 07:36.10 | wildcard0 | depending if there are additional dids on that line |
| 07:36.18 | pascals | I completely mis understood s. |
| 07:36.22 | *** join/#asterisk mbaron (~mbaron@AVelizy-154-1-37-253.w82-124.abo.wanadoo.fr) |
| 07:36.24 | terrapen_ | brc: unless that is the only number he ever wants to call ;-) |
| 07:36.32 | brc_ | exten => _X.,GoTo(Internal-phones,4890,1) |
| 07:36.41 | brc_ | terracon, seems to be what he's saying |
| 07:36.51 | brc_ | it's for incoming, not outbound |
| 07:37.02 | terrapen_ | who the heck is terracon anyway? |
| 07:37.04 | pascals | there are more dids, yes. |
| 07:37.08 | terrapen_ | i know everyones nick completer matches him |
| 07:37.09 | brc_ | how should I know |
| 07:37.15 | terrapen_ | and i dont mind |
| 07:37.18 | terrapen_ | but i wonder who he/she is |
| 07:37.21 | terrapen_ | never sen him active |
| 07:37.22 | brc_ | kick him |
| 07:37.25 | terrapen_ | heh |
| 07:37.29 | terrapen_ | im o-less |
| 07:37.29 | wildcard0 | pascals, then you can direct each did to a separate phone...or use matching and direct groups to a specific phone... |
| 07:37.30 | brc_ | I'll do it later |
| 07:37.35 | wildcard0 | pascals, asterisk is very flexable that way |
| 07:37.45 | brc_ | flexible |
| 07:37.47 | terrapen_ | oh shit, |
| 07:37.55 | terrapen_ | i totally forgot to get my payphone and bring it home |
| 07:37.59 | wildcard0 | er ya. ignore my typos |
| 07:38.00 | brc_ | payphone? |
| 07:38.01 | brc_ | eh? |
| 07:38.06 | terrapen_ | the excitement of the shooting made me forget it |
| 07:38.15 | brc_ | you shot your payphone? |
| 07:38.15 | terrapen_ | yeah, brc did you hear, i bought a payphoen on ebay |
| 07:38.15 | pascals | Yes, I am still trying to understand some of the basic concepts of asterisk, but I love it. |
| 07:38.25 | brc_ | ~docs |
| 07:38.26 | 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 |
| 07:38.32 | terrapen_ | and im hooking it to an IAXy and a wireless bridge |
| 07:38.40 | elric | we need ~betterdocs |
| 07:38.47 | terrapen_ | brc, no i didn't shoot the payphone, i almost died in a shooting last night |
| 07:39.13 | pascals | :O |
| 07:39.26 | Mazda-MX5 | what is gsm codec ? |
| 07:39.33 | elric | terrapen_, where are you situated? |
| 07:39.40 | pascals | Not good! Where do you live, terrapen? |
| 07:39.43 | terrapen_ | brc, my friend and i were cleaning our weapons on his porch and his .45 accidentally discharged and shot through his leg and narrowly missed my head |
| 07:39.49 | terrapen_ | im in san antonio, tx |
| 07:39.55 | mesi | Mazda: Well, the codec used with mobiles. |
| 07:40.15 | Mazda-MX5 | Oh~~ I see. tahnk you, mesi |
| 07:40.21 | pascals | Jikes! |
| 07:40.26 | elric | cleaning loaded weapons :( |
| 07:40.30 | terrapen_ | yes, it was crazy |
| 07:40.39 | terrapen_ | elric, i have no idea why he didn't drop mag and clear |
| 07:40.49 | Eight | terrapen_: weapons? That implies you needed a more general term than 'firearms'... pray tell what was outside the realm of 'firearm'? |
| 07:40.51 | terrapen_ | elric, he's very knowledgable about guns |
| 07:40.57 | terrapen_ | i think he was stoned though, last night... |
| 07:40.59 | elric | terrapen_, maybe he is too confident |
| 07:41.01 | elric | ah |
| 07:41.05 | elric | that could be it. |
| 07:41.10 | terrapen_ | eight, sorry, i'm an ex-Army guy |
| 07:41.18 | pascals | intoxication and guns... hmm not a good idea |
| 07:41.24 | terrapen_ | "It's not a gun, it's a weapon" |
| 07:41.53 | elric | just like my fists terrapen_ :P |
| 07:41.54 | terrapen_ | he was too confident, i think |
| 07:41.54 | brc_ | where does the payphone come in? |
| 07:42.08 | terrapen_ | well the payphone arrived at work two days ago |
| 07:42.13 | terrapen_ | and i was going to bring it home |
| 07:42.19 | Eight | terrapen_: There's something to be said for specificity. Your bayonet is a weapon too. Doesn't mean it'll do you much good in a firefight =p |
| 07:42.21 | terrapen_ | but i forgot because of the excitement |
| 07:42.22 | pascals | I think I would *always* check what the others are doing around guns. |
| 07:42.38 | brc_ | pascals, obviously |
| 07:42.39 | terrapen_ | i'm dying to open the payphone up and figure out how the hell the software on the CD-ROm works and how it interfaces with the phone |
| 07:42.47 | terrapen_ | pascals, that was my fault... |
| 07:43.01 | brc_ | I don't see any reason to think it was |
| 07:43.10 | terrapen_ | eight, we also call rifles and pistols |
| 07:43.18 | terrapen_ | but usually people in the army just call them weapons |
| 07:43.28 | terrapen_ | "Clear your goddamned weapons before coming in my armory!" |
| 07:43.32 | pascals | I don't have a gun (somewhat harder to get than stoned in the netherlands), the only comparison i have is diving |
| 07:43.47 | elric | are there many * based jobs opening up in the USA? i wouldnt mind visiting and if I have work while I am over its just the icing on the cake |
| 07:43.50 | pascals | I check my buddy because I don't want to die from him having faulty equipment |
| 07:43.55 | terrapen_ | elric: nope |
| 07:44.08 | terrapen_ | elric, not unless you want to try and run your own consulting biz |
| 07:44.09 | elric | oh |
| 07:44.17 | brc_ | what he said |
| 07:44.17 | terrapen_ | very few people are hiring an asterisk admin |
| 07:44.22 | elric | i just want to come for a year and travel. |
| 07:44.24 | terrapen_ | maybe that will happen 5 years from now |
| 07:44.30 | brc_ | don't count on it |
| 07:44.32 | terrapen_ | elric, then forget about working |
| 07:44.40 | terrapen_ | come and travel and enjoy yourself |
| 07:44.43 | terrapen_ | come to Texas! |
| 07:45.03 | terrapen_ | brc, i would bet that it will happen when * gains traction in the corp. world |
| 07:45.19 | elric | shall do. one of my friends is in houston and she is someone i definitely want to see again ;) |
| 07:45.25 | terrapen_ | and i bet that will happen within 5 yrs |
| 07:45.36 | terrapen_ | elric, where are you? |
| 07:45.49 | elric | terrapen_, Sydney, australia |
| 07:45.56 | terrapen_ | ahhh |
| 07:46.14 | brc_ | terracon, don't count on it |
| 07:46.21 | terrapen_ | i'm going to a job fair tomorrow to apply for a job working in Antarctica |
| 07:46.26 | terrapen_ | brc, why not? |
| 07:46.32 | brc_ | antarctia? |
| 07:46.33 | brc_ | why! |
| 07:46.42 | terrapen_ | brc, "Because it's there" |
| 07:46.49 | terrapen_ | i'm an adventurer by heart |
| 07:46.56 | elric | first of all IAX2 needs recognition so people start making hardware for it. |
| 07:46.58 | terrapen_ | i love going crazy places and doing crazy things |
| 07:47.01 | terrapen_ | i love the snow |
| 07:47.09 | brc_ | elric, BHWHHAHAHAHAHHAHAHHAHAH |
| 07:47.21 | elric | brc_? |
| 07:47.28 | wildcard0 | an adventurer? don't forget the rope |
| 07:47.28 | pascals | Why does IAX2 need that? |
| 07:47.29 | terrapen_ | brc, why do you doubt this? |
| 07:47.29 | brc_ | not-going-to-happen |
| 07:47.46 | brc_ | sip is gonna be around for a longgg longggg time |
| 07:47.52 | terrapen_ | I don't doubt it |
| 07:47.54 | wildcard0 | oh ya. sip will rule the world for long |
| 07:47.57 | pascals | It works perfectly with SIP, asterisk doesn't need iax recognition to be very good. |
| 07:48.10 | terrapen_ | but the NAT transversing qualities of IAX will make it appealing to manufacturers i think |
| 07:48.11 | elric | pascals, IAX is easier to use though |
| 07:48.19 | brc_ | terracon, doubt it. seriously doubt it. |
| 07:48.32 | terrapen_ | brc, we should bet on this :) |
| 07:48.35 | terrapen_ | i wont bet on IAX |
| 07:48.36 | brc_ | manuf's don't care, don't care at all. they just make what there is a market for |
| 07:48.39 | terrapen_ | but i will bet on Asterisk |
| 07:48.51 | elric | brc_, a NAT within a NAT and with SIP you have no choice. it doesnt work |
| 07:49.01 | elric | or well i couldnt get it to |
| 07:49.11 | terrapen_ | there will be a market for IP phones for home users, i bet |
| 07:49.17 | terrapen_ | and when this happens, IAX will be a must |
| 07:49.20 | brc_ | double nat is a broken network architecture |
| 07:49.27 | terrapen_ | because home users will want to use things like vonage |
| 07:49.31 | terrapen_ | with their IP phones |
| 07:49.32 | elric | brc_, ofcourse |
| 07:49.34 | clive- | I can vouch that NAt and sip is a huge headache |
| 07:49.37 | terrapen_ | maybe i am smoking crack |
| 07:49.37 | brc_ | I can't see why a home user would ever have double nat |
| 07:49.43 | elric | but I had not control over it |
| 07:49.45 | Eight | terracon: except, everyone puts the ATA outside the NAT, and lets it do QoS too. |
| 07:49.52 | wildcard0 | you could bridge instead of double nat |
| 07:49.52 | brc_ | exactly |
| 07:49.57 | pascals | Asterisk can receive faxes, right? |
| 07:50.01 | clive- | its worth getting iax just to avoid all that nat hassle, even if its not as good as sip |
| 07:50.03 | brc_ | pascals, define receive |
| 07:50.07 | Eight | pascals: ya, but there might be some specifics to work out. |
| 07:50.12 | terrapen_ | eight, yep |
| 07:50.16 | brc_ | Eight, hold on there |
| 07:50.24 | terrapen_ | but i see IP phones becoming popular for new houses |
| 07:50.30 | brc_ | pascals, by receive do you mean decode? |
| 07:50.31 | pascals | Someone sends a fax, asterisk sends it as an email attachement somewhere |
| 07:50.34 | brc_ | no |
| 07:50.39 | wildcard0 | rxfax works except when it crashes asterisk :) |
| 07:50.41 | brc_ | it can not |
| 07:50.48 | pascals | Okay. |
| 07:50.56 | brc_ | there's something called rxfax (spandsp) which can be added |
| 07:50.57 | terrapen_ | i know that when i build a house, it will have all-Cisco phones |
| 07:51.17 | elric | when I have a house |
| 07:51.21 | brc_ | but...sometimes it crashes fwihh |
| 07:51.23 | elric | it will have no phones |
| 07:51.29 | pascals | wildcard0: so rxfax is not very stable |
| 07:51.34 | terrapen_ | i take that back |
| 07:51.38 | terrapen_ | my house will have all Cisco |
| 07:51.41 | brc_ | cisco? |
| 07:51.41 | terrapen_ | except for my payphone |
| 07:51.45 | brc_ | *BELCH* |
| 07:51.45 | terrapen_ | which will go in the living room |
| 07:51.49 | terrapen_ | or out by the pool :) |
| 07:51.54 | terrapen_ | hell yeah! |
| 07:51.57 | wildcard0 | pascals, i haven't had the best of luck with it. YMMV |
| 07:52.01 | brc_ | *BARF* |
| 07:52.06 | brc_ | they aren't all that |
| 07:52.06 | terrapen_ | wha-evah! :) |
| 07:52.17 | terrapen_ | i like it more than i like my pcom ip500 |
| 07:52.23 | brc_ | why |
| 07:52.24 | bonez39 | what command to show up time? |
| 07:52.30 | brc_ | uptime |
| 07:52.32 | terrapen_ | its solid as fuck all and the configuration is way easier |
| 07:52.35 | bonez39 | thanks |
| 07:52.41 | wildcard0 | that was a tough one |
| 07:52.43 | mesi | uptime shows the load average, too :-) |
| 07:52.44 | terrapen_ | (though, admittedly, firmware upgrade is a bitch and a half) |
| 07:53.14 | wildcard0 | pcom upgrades are a pain also |
| 07:53.19 | terrapen_ | i should start upgrading people's cisco fw for $$$ |
| 07:53.27 | terrapen_ | wildcard, it was pretty easy for me |
| 07:53.36 | terrapen_ | but the XML configs for pcom were insane |
| 07:53.41 | terrapen_ | not impossible at all |
| 07:53.43 | terrapen_ | not super-hard |
| 07:53.45 | SexyKen | Doesn't anyone know how in the world to use the xml services on Polycom phones? |
| 07:53.55 | terrapen_ | but it was not clear/obvious like the cisco configs |
| 07:54.33 | terrapen_ | brc, im not dissing the polycom phones |
| 07:54.36 | terrapen_ | they *are* very nice |
| 07:54.43 | terrapen_ | certainly adequate for most any use |
| 07:54.49 | SexyKen | terrapen, do you know how to use the xml browser? |
| 07:55.13 | terrapen_ | but if you use one in an environment like our company has, it probably will not last 10 years |
| 07:55.19 | terrapen_ | sexy, didn't even know they had one |
| 07:55.25 | SexyKen | Oh. |
| 07:55.27 | *** join/#asterisk florz (~florz@2001:1a50:503c:0:0:0:0:1) |
| 07:55.28 | SexyKen | Yea, they do. |
| 07:55.29 | terrapen_ | the cisco has one |
| 07:55.31 | terrapen_ | but i never tried it |
| 07:55.36 | SexyKen | So does the polycom |
| 07:55.38 | terrapen_ | sexy, phone apps are overrated |
| 07:55.50 | Beirdo | yay |
| 07:55.56 | terrapen_ | what do you want to do with it, ken? |
| 07:55.56 | SexyKen | •terrapen• I could use it for queue information. |
| 07:55.58 | Beirdo | SPA-3000 now configured :) |
| 07:55.59 | brc_ | way over rated |
| 07:56.06 | wildcard0 | terrapen, you have docs on that pcom xml stuff? |
| 07:56.13 | brc_ | florz, ipv6? |
| 07:56.18 | SexyKen | I want queue info. How many people are in which queues and which agents are logged in. |
| 07:56.23 | terrapen_ | wildcard, i only have the docs that i got from the pcom website...on the config files |
| 07:56.32 | terrapen_ | nothing on the XML browser shit, if it really does exist |
| 07:56.33 | florz | brc_: ? |
| 07:56.43 | brc_ | are you using ipv6? |
| 07:56.45 | wildcard0 | i have those. i was hoping there were secret docs too :) |
| 07:56.46 | brc_ | "* florz (~florz@2001:1a50:503c:0:0:0:0:1) has joined #asterisk" |
| 07:56.54 | terrapen_ | im tempted to drive ot the office to pick up my payphone |
| 07:57.05 | terrapen_ | but wouldn't it look a little strange, carrying a payphone out of a building at 2AM? |
| 07:57.20 | florz | brc_: Nope, that's an IPv4 address. Or a domain name? Dunno ... :-> |
| 07:57.21 | terrapen_ | "seriously officer, i bought this on ebay!" |
| 07:57.40 | pascals | Just put it in a box, terrapen |
| 07:57.44 | terrapen_ | i have had way too many encounters with the police department in the last 24 hours :P |
| 07:57.47 | wildcard0 | or in a car |
| 07:57.52 | pascals | Or wrap it in something |
| 07:57.54 | terrapen_ | pascals, it is ridiculously heavy |
| 07:57.54 | wildcard0 | like...more than 0? |
| 07:57.58 | terrapen_ | and large enough already |
| 07:58.09 | terrapen_ | wild, see earlier conversation re: the shooting |
| 07:58.16 | wildcard0 | ah |
| 07:58.20 | elric | terrapen_, steal a telco uniform |
| 07:58.24 | rvhi | if i insert a new line in extensions.conf, i have to renumber everything below, is there a better way? |
| 07:58.37 | rvhi | using stable |
| 07:58.46 | terrapen_ | elric, heh |
| 07:58.59 | terrapen_ | i should have bought that Bell System hardhat auction |
| 07:59.23 | brc_ | rvhi, cvs head? or stable |
| 07:59.32 | brc_ | oh |
| 07:59.35 | brc_ | not a better way |
| 07:59.40 | brc_ | unless you use VIM |
| 07:59.49 | rvhi | i use vim |
| 08:00.01 | brc_ | get the ReNumber plugin from vim.org |
| 08:00.02 | brc_ | it RULES |
| 08:00.05 | rvhi | what can i do? |
| 08:00.07 | terrapen_ | i trained my dog to renumber. |
| 08:00.12 | pascals | Why would I see this all the time for an iaxclient based phone all the time: chan_iax2.c:3915 register_verify: No registration for peer 'pascal2' (from 192.168.103.12) |
| 08:00.33 | rvhi | does it ever do 101+? |
| 08:00.48 | brc_ | http://www.vim.org/scripts/script.php?script_id=156 |
| 08:01.07 | florz | brc_: you got any questions or anything? Sorry, but I don't get why you asked ... :-) |
| 08:01.15 | brc_ | just currious |
| 08:01.43 | elric | florz, i have questions. |
| 08:01.45 | rvhi | brc_, thx |
| 08:01.48 | elric | lots of them |
| 08:02.16 | elric | like how do you reverse dns an ipv6 address? |
| 08:02.45 | elric | why hasnt everyone started using ipv6 yet? |
| 08:03.01 | elric | will my computer run faster if i use ipv6? |
| 08:03.02 | florz | elric: Did you see any domain name in my /whois? =:-) |
| 08:03.02 | brc_ | elric, dude, "florz brc_: Nope, that's an IPv4 address. Or a domain name? Dunno ... :->" |
| 08:03.27 | Leland | florz: nice ipv6 addy :D |
| 08:04.04 | *** join/#asterisk Silik0n (~krice@rso.suspicious.org) |
| 08:04.10 | florz | Leland: ? |
| 08:04.29 | RaYmAn-Bx | it shouldn't be a problem getting reverse dns for ipv6..(Granted there are the old and the new way of doing them and some irc servers support one while others support the other, same with providers) |
| 08:04.48 | terrapen_ | ipv6, you've got to be kidding me |
| 08:04.57 | terrapen_ | why should i implement ipv6 on my network? |
| 08:05.11 | wildcard0 | so you can have a billion computers at home |
| 08:05.13 | elric | it makes your routers run faster. |
| 08:05.16 | elric | :| |
| 08:05.17 | brc_ | so you won't ever ever never have to worry about sip and nat |
| 08:05.18 | terrapen_ | heh |
| 08:05.35 | terrapen_ | i run m0n0wall at home |
| 08:05.38 | brc_ | so do I |
| 08:05.39 | RaYmAn-Bx | it's pretty much useless until you can get native ipv6 connectivity everywhere :/ |
| 08:05.40 | brc_ | it rules |
| 08:05.40 | wildcard0 | so you can lose the ability to multihome using bgp |
| 08:05.41 | terrapen_ | so no ipv6 for me |
| 08:05.42 | elric | m0n0 rules |
| 08:05.45 | Silik0n | way less routing overhead on a native v6 network |
| 08:05.52 | brc_ | RaYmAn-Bx, you can't do that until many people are using it |
| 08:05.57 | elric | terrapen_, what have you embedded it on? |
| 08:05.59 | terrapen_ | i sure hope they do a Ralink driver for FreeBSD quickly |
| 08:06.01 | elric | soekris boards? |
| 08:06.03 | RaYmAn-Bx | tunneling all your traffic through random places is just slow |
| 08:06.03 | terrapen_ | and that m0n0 picks it up |
| 08:06.07 | brc_ | old pc's |
| 08:06.09 | terrapen_ | elric: yup, net4801 |
| 08:06.14 | florz | brc_: Instead you can worry about IPv6 support in the devices and software. At least you don't have to worry that there will be less to worry about ... =:-) |
| 08:06.14 | brc_ | much cheaper |
| 08:06.17 | RaYmAn-Bx | brc_: well, or when isp decide to change |
| 08:06.21 | Silik0n | i have monowall or openbsd on net4501s |
| 08:06.26 | terrapen_ | brc, indeed, but so much noisier :) |
| 08:06.31 | Silik0n | brc stfu |
| 08:06.35 | elric | terrapen_, nice, i am thinking of getting a soekris board with sangoma |
| 08:06.36 | brc_ | and they won't do that till software and routers support it |
| 08:06.38 | brc_ | Silik0n!!!!!! |
| 08:06.41 | terrapen_ | silikon, how do those work for you? |
| 08:06.43 | elric | e1 interface |
| 08:06.46 | RaYmAn-Bx | afaik there are quite a few isp's in Japan that provide native ipv6 on adsl for example |
| 08:06.47 | terrapen_ | elric, hold off |
| 08:06.51 | terrapen_ | elric, buy a WRAP from me |
| 08:06.54 | Silik0n | terracon just fine... |
| 08:06.55 | terrapen_ | i will be selling them soon |
| 08:07.00 | elric | ah |
| 08:07.08 | florz | BTW, my IPv6 connection is native DSL dialin, no such thing as a tunnel =:-) |
| 08:07.10 | Silik0n | terracon i have them in customer locations... no problems |
| 08:07.13 | pascals | What is the best way to connect a bunch of analog phones? SIP converters or FX? card in the asterisk server? |
| 08:07.25 | terrapen_ | define "a bunch" |
| 08:07.26 | elric | the buisness I started we plan to sell embedded firewalls for cheap as chips |
| 08:07.33 | pascals | I have three or four I need to connect, among them, a fax. |
| 08:07.46 | Silik0n | get a tdm4xx |
| 08:07.50 | Mazda-RX8 | Can bind the asterisk eith gnomemeeting |
| 08:07.53 | Mazda-RX8 | Can bind the asterisk eith gnomemeeting?? |
| 08:07.53 | criptos | pascals fxo ports.. |
| 08:07.56 | terrapen_ | don't hold your breath on the fax |
| 08:07.58 | criptos | buch means more than 8 :) |
| 08:08.22 | pascals | terrapen why? |
| 08:08.35 | terrapen_ | pascals, very codec dependent |
| 08:08.36 | pascals | Is it going to stop working properly? |
| 08:08.42 | Silik0n | fax +sip/iax == ass |
| 08:08.52 | terrapen_ | remember most codecs are designed for doing voice |
| 08:08.58 | terrapen_ | most asterisk codecs anyway |
| 08:09.05 | terrapen_ | s/asterisk/voip/ |
| 08:09.13 | wildcard0 | the -only- good foip i've used at all is t.38 |
| 08:09.23 | terrapen_ | you are asking for trouble hooking a fax up |
| 08:09.24 | Silik0n | supposedly you can do faxx passthruover 711 but it doesnt work that well |
| 08:09.26 | terrapen_ | save yourself the headache |
| 08:09.27 | wildcard0 | everything else has come outlooking like crap for me |
| 08:09.28 | terrapen_ | hook it to POTS |
| 08:09.36 | pascals | I can buy the fxo hardware if that will solve it. |
| 08:09.45 | terrapen_ | no. |
| 08:09.45 | Silik0n | pascals then do that |
| 08:09.51 | wildcard0 | pascals, look into t.38 if you're interested in fax over ip |
| 08:09.56 | Silik0n | buy the fxo/fxs hardware itworks fine |
| 08:10.01 | pascals | Doesn't have to be 'over ip' |
| 08:10.25 | wildcard0 | then just plug the pots line into the fax machine |
| 08:10.27 | Silik0n | andits not over IP if you get a TDM31 or is it 13? |
| 08:10.29 | wildcard0 | leave asterisk out of it |
| 08:10.30 | wildcard0 | heh |
| 08:10.36 | pascals | The hardlines are there already, the asterisk server is replacing some shitty ptt provided thing |
| 08:10.38 | terrapen_ | yes, i agree |
| 08:10.47 | Silik0n | hmmm "nonsmoking room" at the radiso.... |
| 08:11.00 | Silik0n | i think i'm bout to fire one up |
| 08:11.09 | *** join/#asterisk Alexis (~alexis@www.trim.it) |
| 08:11.19 | Alexis | hello :) |
| 08:11.26 | *** join/#asterisk zoa (~zoa@pirus.securax.be) |
| 08:11.34 | pascals | Even when no transcoding is done from the incomming isdn line to the fxo device, it will still fail? |
| 08:11.41 | *** join/#asterisk MrY (MrX@adsl-63-200-37-144.dsl.snfc21.pacbell.net) |
| 08:11.48 | MrY | hi all |
| 08:12.01 | rvhi | is dialplan command case sensitive? |
| 08:12.16 | terrapen_ | well its bedtime for me |
| 08:12.21 | terrapen_ | i will play with the payphone at work |
| 08:12.22 | wildcard0 | pascals, faxes are just -really- sensitive to latency |
| 08:12.38 | pascals | Ah. Oh, well. |
| 08:12.45 | terrapen_ | i hope the payphone software does not require a modem |
| 08:12.48 | pascals | I can leave it outside the asterisk loop. |
| 08:12.50 | terrapen_ | because that will be gnarly |
| 08:12.54 | terrapen_ | im sure it does, though |
| 08:13.08 | wildcard0 | get an acousic modem |
| 08:13.08 | wildcard0 | hehe |
| 08:13.15 | wildcard0 | i remember doing that forever ago |
| 08:13.16 | terrapen_ | i run mostly Macs |
| 08:13.23 | terrapen_ | i dont want to deal with DOS payphone software |
| 08:13.29 | *** join/#asterisk DEVILoper (~x@202.5.145.50) |
| 08:13.31 | terrapen_ | or win16 software |
| 08:13.35 | terrapen_ | but i will probably have to |
| 08:13.43 | pascals | terrapen rewrite in Tcl :) |
| 08:13.54 | MrY | if i have a linux box w/ asterisk installed... assuming i also have X100P card, is that enough for voip calls route thru asterisk server and make outgoing calls via pstn line connect to x100p card? |
| 08:13.54 | DEVILoper | Hi ALL |
| 08:13.57 | booyeah7 | anyone run ztdummy on old kernels? |
| 08:13.59 | pascals | The command set can't be that big |
| 08:14.39 | wildcard0 | MrY, yes |
| 08:14.41 | SwedMiro | Question.... Is ulaw or alaw the best quality? |
| 08:14.47 | terrapen_ | pascals, hahah |
| 08:14.51 | terrapen_ | it would be a neat hack though |
| 08:14.56 | DEVILoper | Swed: NO |
| 08:14.57 | terrapen_ | to rewrite in Perl |
| 08:15.05 | MrY | wildcard0: thanks. |
| 08:15.05 | pascals | Oh, no, Not perl |
| 08:15.06 | wildcard0 | hahaha |
| 08:15.08 | terrapen_ | im planning on submitting the whole project as a /. article when done |
| 08:15.11 | wildcard0 | in a single regex |
| 08:15.15 | terrapen_ | and that would make it neater |
| 08:15.27 | SwedMiro | Question..ehh..my question was which one of them had the best quality |
| 08:15.29 | terrapen_ | "Yes, I had to write my own software to control the payphone." |
| 08:15.42 | pascals | Perl would make the code only readable by you ;-) |
| 08:15.56 | terrapen_ | whatever :P |
| 08:15.59 | MrY | i'm planing to buy some Nortel, ip phone, any ideas if these phones are compatible w/ asterisk? |
| 08:16.03 | wildcard0 | perl is a write only languag |
| 08:16.04 | wildcard0 | e |
| 08:16.05 | terrapen_ | i write purty purl |
| 08:16.22 | wildcard0 | MrY, i'd avoid nortel like the spanish flu |
| 08:16.29 | MrY | i see. |
| 08:16.35 | wildcard0 | it's compatible with itself |
| 08:16.38 | Beirdo | why would it require a modem, terrapen? |
| 08:17.00 | terrapen_ | beirdo, im not sure that it does...but how would software communicate with a payphone? |
| 08:17.04 | terrapen_ | it came with no cables |
| 08:17.16 | terrapen_ | i havent even opened the phone up yet |
| 08:17.17 | pascals | wildcard0: write only indeed ;) |
| 08:17.22 | terrapen_ | i will open it tomorrow and post pictures |
| 08:17.24 | DEVILoper | I am using experimenting * with my external modem iam able to establish call <Voip client> <-----*----> PSTN .Voip Client able to hear a person on PSTN but PSTn guy can t hear of Voip CLient any clue ??? |
| 08:17.38 | Beirdo | what software? |
| 08:17.39 | pascals | terrapen_ likely out of band, above 8Khz |
| 08:17.53 | Beirdo | for coin deposits, it uses tones |
| 08:18.01 | Beirdo | at least the older ones did |
| 08:18.12 | *** join/#asterisk hanhoong (~hanhoong@218.111.48.230) |
| 08:18.15 | terrapen_ | well, it came with a CD-ROM |
| 08:18.16 | pascals | Beirdo, the newer ones too - just out of band, afaik |
| 08:18.24 | terrapen_ | and some keys |
| 08:18.25 | Beirdo | :) |
| 08:18.26 | MrY | wildcard0: i guess those phones are not good... any idea where I can buy some copy of softphones knowing it would work asterisk? i want my lady in France to call me here in the States via the internet using my server... |
| 08:18.31 | terrapen_ | and thats it |
| 08:18.46 | Beirdo | put in the CDROM and see what's on it |
| 08:18.55 | terrapen_ | its at work |
| 08:18.58 | terrapen_ | i will, tomorrow |
| 08:19.03 | hanhoong | hi everyone |
| 08:19.03 | Beirdo | ahhh |
| 08:19.04 | wildcard0 | MrY, there are a few free ones...iaxcomm, xten lite |
| 08:19.04 | terrapen_ | i'll do a little post on my website |
| 08:19.10 | terrapen_ | and tell the channel about it |
| 08:19.38 | terrapen_ | http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=26261&item=3878461107 |
| 08:19.41 | terrapen_ | thats the phone i bought |
| 08:19.55 | wildcard0 | nice |
| 08:20.00 | wildcard0 | i remember those |
| 08:20.14 | wildcard0 | we use to put a paperclip in the phone and ground it... |
| 08:20.24 | terrapen_ | elcotel? |
| 08:20.29 | hanhoong | i need help here ! ! couldnt install asterisk on suse 9.1... |
| 08:20.31 | terrapen_ | series 5? |
| 08:20.43 | terrapen_ | this is one of the most top of the line payphones |
| 08:20.43 | wildcard0 | the look like the old bell ones in new york |
| 08:20.55 | wildcard0 | hanhoong, try harder! |
| 08:21.08 | wildcard0 | it looks fun and cool :) |
| 08:21.30 | hanhoong | i'm just newbie here... any advice from u guys ... |
| 08:21.34 | MrY | thanks wildcard0. |
| 08:22.02 | wildcard0 | hanhoong, ask more specific questions |
| 08:23.00 | hanhoong | i was following the steps provided by asterisk website |
| 08:23.16 | wildcard0 | url? |
| 08:23.32 | hanhoong | trying to do a cvs but returned no such command ... |
| 08:23.40 | hanhoong | is my suse not fully installed |
| 08:23.42 | hanhoong | ? |
| 08:23.59 | wildcard0 | sounds like you need to install the cvs program |
| 08:24.06 | Inv_arp | hanhoong: install cvs... yum install cvs |
| 08:24.16 | Inv_arp | hmm does suse use yum |
| 08:24.20 | wildcard0 | no idea |
| 08:24.28 | pascals | hanhoong: suse doesn't install cvs by default if you don't select development at install time |
| 08:24.32 | wildcard0 | i haven't used suse in years |
| 08:24.37 | wildcard0 | yast2 ? |
| 08:24.45 | hanhoong | yes yast |
| 08:24.54 | pascals | yast2 or yast. both work. |
| 08:25.17 | Beirdo | OK, anyone here have a SPA-3000? |
| 08:25.19 | pascals | Just start yast2, install remove software, search cvs. |
| 08:25.20 | hanhoong | hmm... maybe should i try an easier distribution ... not suse |
| 08:25.29 | pascals | suse is simple, dead simple. |
| 08:25.54 | pascals | Yast is in your start menu, under system |
| 08:26.24 | hanhoong | yes .. i think i tried but doesnt return any result |
| 08:26.36 | pascals | suse 9.2 comes with *working* asterisk packages. |
| 08:26.51 | wildcard0 | that makes it easier |
| 08:27.16 | pascals | hanhoong did you search for cvs or CVS? |
| 08:27.33 | hanhoong | CVS |
| 08:27.36 | pascals | try cvs |
| 08:27.39 | pascals | lower case |
| 08:27.53 | hanhoong | ohh ... case sensitive yeah... ok got it.... |
| 08:27.57 | pascals | Most, if not all, packages are lowercase |
| 08:28.00 | terrapen_ | hey guys check this out |
| 08:28.01 | terrapen_ | https://www.payphone.com/shop/instructions/Series5Manual.pdf |
| 08:28.08 | terrapen_ | thats the manual for my payphone |
| 08:28.28 | hanhoong | thanls |
| 08:28.32 | hanhoong | thanks |
| 08:28.35 | Beirdo | cool |
| 08:29.00 | terrapen_ | fuck. no programming info |
| 08:29.03 | terrapen_ | dammut |
| 08:29.38 | pascals | terrapen_ you could call them :) |
| 08:29.52 | terrapen_ | well i bought mine from someone else |
| 08:29.58 | terrapen_ | ok, going to bed now |
| 08:29.59 | terrapen_ | night all |
| 08:30.03 | pascals | later |
| 08:30.44 | hanhoong | but i do have the files from asterisk... .tar |
| 08:31.16 | hanhoong | downloaded via web , not through cvs , are they the same |
| 08:32.01 | clive- | what does raw_hangup mean ? |
| 08:32.17 | *** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
| 08:33.00 | *** join/#asterisk Verliba (~Miranda@250.1-14-84.ripe.coltfrance.com) |
| 08:39.54 | *** join/#asterisk cgeek (~cgeek@pl6.lawrence.org.uk) |
| 08:50.38 | elric | alright this is really weird |
| 08:50.59 | Zeeek | you're just noticing that? |
| 08:50.59 | elric | the zap channel wont stop ringing |
| 08:51.05 | elric | when a call comes in |
| 08:51.18 | Zeeek | UK ? |
| 08:51.25 | elric | and theres only 1 channel that does that |
| 08:51.27 | elric | australia |
| 08:52.03 | *** join/#asterisk atmel (~vlad@ip68-4-101-199.oc.oc.cox.net) |
| 08:52.34 | elric | so Zap/3 receives the call and dials Zap/1 |
| 08:52.39 | elric | no one answers |
| 08:52.46 | elric | and the caller hangs up |
| 08:52.59 | elric | but Zap/1 doesnt stop ringing |
| 08:53.02 | Zeeek | ZAP/3 is FXO? |
| 08:53.19 | *** join/#asterisk pranav (pranav@202-149-48-220.broadband.isp.exatt.net) |
| 08:53.45 | elric | signalling = fxs_ls |
| 08:53.49 | elric | for zap 3,4 |
| 08:54.16 | Zeeek | Have you scoured (not scanned) the mailing list about au and hangup detection? |
| 08:54.50 | elric | yeah nothing about au |
| 08:54.54 | Zeeek | International issues are the one area where asterisk can become weak |
| 08:54.58 | elric | i have busydetect=yes |
| 08:55.03 | Zeeek | there are posts about au |
| 08:55.31 | elric | hang on |
| 08:55.35 | elric | let me recheck |
| 08:55.48 | pranav | hello Zeeek |
| 08:55.55 | Zeeek | hi pranav - |
| 08:56.24 | Zeeek | elric do a umungous google on asterisk australia |
| 08:56.31 | pranav | as u said i have downloaded the x-lite phone but it is not getting registered with asterisk |
| 08:56.44 | Zeeek | sory to hear that |
| 08:57.01 | elric | Zeeek, yeah i just did that |
| 08:57.02 | elric | :) |
| 08:57.21 | Zeeek | I know there's 10,000 pages of crap, but it's your only chance |
| 08:57.27 | Zeeek | or |
| 08:57.35 | Zeeek | if some Oz brethern show up |
| 08:58.11 | SwedMiro | pranav.what error message do you get? |
| 08:58.52 | pranav | see i get the dial tone but when i try to any number the dial tone goes |
| 08:59.15 | SwedMiro | does it say "logged in" |
| 08:59.24 | pranav | no |
| 08:59.45 | pranav | but the call does not reflect on the asterisk cli screen |
| 08:59.49 | booyeah7 | sweet, i got app_conference working |
| 09:00.00 | booyeah7 | had to hack some c code |
| 09:00.23 | SwedMiro | and have you set a extension in the extensions.conf? |
| 09:00.55 | pranav | no |
| 09:01.08 | Zeeek | pranav the dial tone is generated by the phone |
| 09:01.13 | pranav | ok |
| 09:01.22 | SwedMiro | ahh..you see you have to set up an extension for it in the extension.conf |
| 09:01.28 | booyeah7 | can some people help me test out my conference system |
| 09:01.43 | SwedMiro | with a password and an extension number |
| 09:02.17 | SwedMiro | and then you have to set the phones settings with the right password and extension number |
| 09:02.24 | SwedMiro | so you log into the asterisk |
| 09:02.41 | SwedMiro | after that you can call |
| 09:02.55 | booyeah7 | call 8662291552 |
| 09:03.18 | booyeah7 | thats the test conference |
| 09:05.53 | *** part/#asterisk Mazda-RX8 (~leolarrel@220-130-142-43.HINET-IP.hinet.net) |
| 09:09.03 | Silik0n | hah crack whores on hbo |
| 09:10.01 | *** join/#asterisk feral_kid (~not@209.205.207.130) |
| 09:10.33 | feral_kid | What purpose does sip_nat.conf serve? |
| 09:12.42 | Silik0n | sip_nat.conf? |
| 09:13.14 | *** join/#asterisk mesi (~player@dsl-082-083-135-089.arcor-ip.net) |
| 09:13.22 | *** join/#asterisk christo (~chris@office.enovi.com) |
| 09:13.49 | *** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
| 09:15.39 | christo | I have a number routed to our equpment (E1 termination into *). How can I get * to detect the incoming call? At the moment, I see nothing in the asterisk console and dialling the no, I just get beeps |
| 09:16.49 | christo | my extensions.conf contains a line like exten => 09050090700,1,Macro(extensions,3005,1) and I have reloaded * |
| 09:17.56 | *** join/#asterisk mbranca (~matteo@81.208.92.210) |
| 09:19.52 | Silik0n | turn debugging onto the console and set verbose 9 you should see what the E1 is sending asterisk when a call comes in |
| 09:20.19 | Silik0n | (edit logger.conf to send debug to the console) |
| 09:22.03 | christo | Silik0n - how do I know which channel to enable debugging on? |
| 09:22.10 | christo | Usage: debug channel <channel> |
| 09:22.23 | feral_kid | Silik0n: Yes, sip_nat.conf |
| 09:22.54 | SexyKen | Anyone know much about call parking with Asterisk? |
| 09:22.59 | *** part/#asterisk pranav (pranav@202-149-48-220.broadband.isp.exatt.net) |
| 09:23.02 | *** join/#asterisk pranav (pranav@202-149-48-220.broadband.isp.exatt.net) |
| 09:23.40 | christo | Silik0n - ok logger.conf has debug => debug, console => notice,warning,error and messages => notice,warning,error |
| 09:23.48 | christo | but I see nothing when I dial the number |
| 09:23.56 | feral_kid | Does anyone use shorewall in front of their * server? |
| 09:24.20 | *** join/#asterisk afe ([jrngIMN7N@c-e616e055.123-1-64736c12.cust.bredbandsbolaget.se) |
| 09:24.49 | *** part/#asterisk feral_kid (~not@209.205.207.130) |
| 09:24.54 | *** join/#asterisk feral_kid (~not@209.205.207.130) |
| 09:29.03 | *** join/#asterisk Delvar (~irc@83.146.53.34) |
| 09:30.38 | feral_kid | I am able to call out to FWD through IAX2, but I cannot get any calls in... Anyone want to give some pointers on what could be the way to troubleshoot this? |
| 09:32.45 | *** join/#asterisk Inv_arp (junya@adsl-3-255-228.mia.bellsouth.net) |
| 09:33.45 | Zeeek | are you registered with FWD in IAX? |
| 09:33.51 | SwedMiro | feral... http://www.fwd.pulver.com/advanced/iax |
| 09:34.13 | Zeeek | and NOT registered in SIP? |
| 09:34.18 | booyeah7 | can someone people help me try to break this call conference i set up |
| 09:34.35 | booyeah7 | im testing out some changes to app_conference |
| 09:34.51 | feral_kid | Zeeek: Yes, I am registered and like I said, I am able to make call out on the trunk.... |
| 09:35.03 | booyeah7 | 8662291552 |
| 09:35.14 | *** join/#asterisk RoyK (~roy@80.239.107.80) |
| 09:35.49 | feral_kid | 69.73.19.178:4569 kcc 209.205.207.130:4569 60 Registered |
| 09:35.49 | feral_kid | 69.90.155.70:4569 24915 <Unregistered> 60 Request Sent |
| 09:35.49 | feral_kid | 65.39.205.121:4569 34062 209.205.207.130:4569 60 Registered |
| 09:37.41 | feral_kid | This as been driving me nuts, because I have PSTN calls routed through FWD and it has been nearly a week since I have gotten a single call in! |
| 09:38.06 | Delvar | and its a good morning to you all |
| 09:38.11 | booyeah7 | fwd is unstable |
| 09:38.25 | booyeah7 | you should just have a sip number |
| 09:38.30 | booyeah7 | like your email address |
| 09:39.06 | SwedMiro | feral..have it worked before? |
| 09:39.44 | christo | I'm going insane here.. |
| 09:39.53 | feral_kid | SwedMiro: No, prior to the switch over, I was just doing FWD through stun... |
| 09:40.20 | christo | <PROTECTED> |
| 09:40.27 | christo | Have I totally overlooked something? |
| 09:40.37 | SwedMiro | i dont know much about iax..but do you have to have an extension set up in the conf files for both incomming and outgoing calls? |
| 09:40.38 | feral_kid | The real problem is that I am behind 2 NAT firewalls, with no control over one of them... |
| 09:41.04 | Zeeek | not a lot of hope for much voIP fun in that case :( |
| 09:41.40 | SwedMiro | hmm..feral..have you tried connecting to fwd with an iax2 phone client? |
| 09:41.47 | SwedMiro | and not asterisk? |
| 09:42.54 | feral_kid | SwedMiro: Nope, that was not an idea that came to mind... The setup is a single Sipura 2000, going through *... |
| 09:44.04 | SwedMiro | download a client from http://www.voip-info.org/wiki-Asterisk+IAX+clients and test if you can get it going |
| 09:44.29 | SwedMiro | if you cant get a phone client up..there might be a problem outside your control |
| 09:45.08 | SwedMiro | the iax ping tool here http://www.voip-info.org/wiki-IAX might help you |
| 09:47.58 | christo | correction - the message I get is 'beep beep bepp.. the number you have dialled has not been recognised. Please check and try again'. Is that a problem at my asterisk end? |
| 09:49.29 | antifuchs | hmmm. |
| 09:50.12 | antifuchs | I'm trying to monitor our asterisk server via sipsak. if I enable quality for the monitor user, it complains that it receives OPTIONS before a "200 OK" response |
| 09:50.17 | SwedMiro | christo..you havent changed prefix number have you? |
| 09:50.37 | feral_kid | SwedMiro: Ok, so far so good... I just downloaded firefly and was able to dial to FWD |
| 09:50.54 | SwedMiro | via iax? |
| 09:51.06 | feral_kid | SwedMiro: Yes |
| 09:51.39 | christo | SwedMiro - not sure what you mean - but I don't think I have :) |
| 09:53.03 | SwedMiro | i have to dial 0 before the phone number to get outside my asterisk |
| 09:53.12 | *** join/#asterisk easydone_ (~easydone@ip3e83a471.speed.planet.nl) |
| 09:53.27 | *** part/#asterisk pranav (pranav@202-149-48-220.broadband.isp.exatt.net) |
| 09:53.30 | SwedMiro | if i forget the first number of the phone number is stripped |
| 09:54.04 | *** join/#asterisk pranav (pranav@202.149.48.220) |
| 09:54.17 | SwedMiro | and i end up colling strange places, people and sometimes the number isnt available.. |
| 09:54.30 | SwedMiro | then i get that message |
| 09:54.35 | christo | SwedMiro - but I'm calling into asterisk.. not out |
| 09:54.48 | SwedMiro | then i have no clue |
| 09:54.53 | christo | :) |
| 09:55.55 | *** join/#asterisk pi-cubic (~pi-cubic@62.111.48.31) |
| 09:56.17 | pi-cubic | hi there, can anyone me recommand a kernel-version which works STABLE with chan_capi? |
| 09:59.18 | simonides | anyone know what the error message "Freshmaker failed register test" means? |
| 10:00.22 | *** join/#asterisk implicit (~implicit@ip68-5-148-1.oc.oc.cox.net) |
| 10:05.11 | clive- | pi, I am using 1.0.2 , seems to be pretty good so far |
| 10:09.14 | feral_kid | SwedMiro: Well, thank you very much... You are the first person to point me in the right direction! The problem in the * setup... I shutdown *, then called out with the Firefly client and then used FWDnet call me to call me back... The phone rang immediately... |
| 10:10.57 | SwedMiro | i suck at programming so im good at error-searching :) |
| 10:12.03 | SwedMiro | no1..find out what part of the setup is crappy..no2 cry a bit..no3 fiddle around until you fix it |
| 10:12.04 | SwedMiro | :) |
| 10:14.45 | *** join/#asterisk meppl (~mephisto@p3E9E2159.dip.t-dialin.net) |
| 10:15.24 | *** join/#asterisk easydone_ (~easydone@ip3e83a471.speed.planet.nl) |
| 10:16.40 | SwedMiro | feral..you probably need one extension that handles incomming calls..and another to handle outgoing calls |
| 10:19.55 | *** join/#asterisk jedirl (~fdsafasdf@213.162.200.226) |
| 10:19.59 | jedirl | Hello |
| 10:20.48 | *** join/#asterisk miller7 (~none@adsl49-static-gw1.access.acn.gr) |
| 10:20.50 | *** join/#asterisk fantomax1 (~fanto@81.208.114.250) |
| 10:20.53 | fantomax1 | hi all |
| 10:20.56 | miller7 | anyone from Spain here? |
| 10:21.24 | fantomax1 | is there anyone that commected * to a Voip/TDM GWY ? |
| 10:21.36 | miller7 | fantomax1: what exactly do you need? |
| 10:26.39 | pi-cubic | clive-: you told my your * version, i ment the kernel version...which one are you using? |
| 10:27.43 | justinnnnnn | anyone played with a 7914 ? |
| 10:29.57 | *** join/#asterisk Fabe (~spamhere@pD95B0BF7.dip0.t-ipconnect.de) |
| 10:31.12 | christo | I'm loosing the will to live |
| 10:33.01 | *** join/#asterisk sysdef (~sysdef@sysdef.admin.debiancenter) |
| 10:35.16 | *** join/#asterisk mesi (~player@dsl-082-083-059-037.arcor-ip.net) |
| 10:41.12 | SwedMiro | <---- is out of coffe... GAAAAAAH |
| 10:41.16 | *** join/#asterisk A-Tuin|work (~A-Tuin@nat.office.legend.net.uk) |
| 10:41.23 | SwedMiro | *runs around waving his arms in panic* |
| 10:44.03 | Leland | *perk* did someone mention coffee ?? |
| 10:44.38 | SwedMiro | *runs around waving his arms in panic* |
| 10:45.19 | *** join/#asterisk Jas_Williams (~Jason@host81-155-66-178.range81-155.btcentralplus.com) |
| 10:46.51 | *** join/#asterisk jeffik (~jeffik@node-423a160a.mdw.onnet.us.uu.net) |
| 10:48.28 | *** join/#asterisk MaeWest (~maewest@207.237.5.3) |
| 10:48.38 | MaeWest | Anyone here use asterlink? |
| 10:51.36 | jedirl | fantomax1: I connected asterisk to a VoIP/TDM GWY |
| 10:52.33 | *** part/#asterisk pranav (pranav@202.149.48.220) |
| 10:54.16 | clive- | PI, -its Kernel 2.4.26 |
| 10:54.16 | MaeWest | anyone know where i can get a list of iax-pstn providers? |
| 10:54.33 | *** part/#asterisk sysdef (~sysdef@sysdef.admin.debiancenter) |
| 10:54.48 | *** join/#asterisk pranav (pranav@202.149.48.220) |
| 10:55.06 | clive- | Mae, how many does one need? |
| 10:55.07 | *** join/#asterisk pranav (pranav@202.149.48.220) |
| 10:55.46 | MaeWest | thats not what i asked you. And a few to determine who has the best service |
| 10:56.26 | clive- | Mae, thats a huge debate on here....are you in the usa? |
| 10:56.31 | MaeWest | yes |
| 10:56.42 | MaeWest | and voicepulse is terrible |
| 10:57.11 | clive- | I use nufone, who are fine...some people complain, but I find them good, also livevoip |
| 10:57.57 | MaeWest | I use nufone too |
| 10:58.03 | MaeWest | but no more 800 numbers |
| 10:58.06 | MaeWest | i just tried to order one |
| 10:58.57 | MaeWest | and their site is so shitty |
| 11:01.00 | clive- | Well, I agree, not a fancy website, but I dont belive a phone company should be judged on its website, but on its phone service |
| 11:01.20 | MaeWest | tru |
| 11:01.22 | clive- | A few poeple have siad livevoip is good |
| 11:01.23 | MaeWest | love the service though |
| 11:01.34 | MaeWest | excellent service ;) |
| 11:01.51 | *** join/#asterisk pratik (~root@202.149.48.220) |
| 11:03.40 | *** join/#asterisk [aryeh] (~aryeh@80.179.208.113.cable.012.net.il) |
| 11:03.44 | *** join/#asterisk __Sparks_ (ringding@bb-195-172-50-212.ukonline.co.uk) |
| 11:17.57 | *** join/#asterisk caesar2 (~igerl@ppp-62-245-210-208.mnet-online.de) |
| 11:19.00 | ManxPower | ~docs |
| 11:19.02 | 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 |
| 11:19.02 | ManxPower | ~mailinglist |
| 11:19.03 | jbot | well, mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ |
| 11:19.33 | *** join/#asterisk RoyK (~roy@80.239.107.80) |
| 11:20.08 | *** join/#asterisk jsmith (~jsmith@smithfam.dsl.xmission.com) |
| 11:23.53 | Zeeek | goog morning ManxPower |
| 11:24.00 | pranav | hi zeeek |
| 11:24.25 | pranav | i have started my fwd |
| 11:24.38 | pranav | can u give me your number for a test call |
| 11:24.39 | *** join/#asterisk jeffik (~jeffik@node-423a160a.mdw.onnet.us.uu.net) |
| 11:25.09 | ManxPower | http://ars.userfriendly.org/cartoons/?id=20050317 |
| 11:25.17 | Zeeek | pranav there are test numbers on the fwd site for that purpose. You should go to the FWD forum and talk to people there. Many are knowledgeable about asterisk |
| 11:25.30 | pranav | ya i tried 612 and 613 |
| 11:25.35 | Zeeek | <PROTECTED> |
| 11:25.35 | pranav | both are working |
| 11:25.59 | pranav | i want to call someone who has a fwd number |
| 11:26.05 | Zeeek | go tot hat forum - everyone publishes their numbers and wants to talk - you can check out quality all over the world! |
| 11:26.24 | Zeeek | they even have little ocons saying they are ONLINE! |
| 11:26.29 | pranav | ok fine zeeek |
| 11:26.44 | Zeeek | You'll find dowens of FWD people dying to help |
| 11:26.58 | Zeeek | if only someone would help me type |
| 11:27.05 | pranav | no i could'nt find anyone to help me for a long time |
| 11:27.20 | pranav | finally i got it working |
| 11:27.30 | Zeeek | pranav I for one have discussed your FWD issues at least four times in this room |
| 11:27.46 | Zeeek | and so have many others - Glad you finally got it to work |
| 11:28.02 | Zeeek | the best FWD test is still the ech |
| 11:28.04 | Zeeek | o |
| 11:28.19 | *** join/#asterisk oej (~oej@40.186.204.213.sol.worldonline.se) |
| 11:28.20 | pranav | ya the echo is working fine with 613 |
| 11:28.52 | pranav | i can also get the time from 612 |
| 11:29.01 | Zeeek | great - that's an important test |
| 11:29.07 | pranav | ya |
| 11:29.30 | feral_kid | pranav: Call 55555 on FWD... That is the Welcome Room... There are people there if you want to hear a voice |
| 11:29.48 | pranav | ok i'll try |
| 11:29.49 | Zeeek | apparently he won't hear yours :) |
| 11:30.49 | feral_kid | Zeeek: Well, with the help of SwedMiro, I have come to realize that I wasn't the problem this time! :) |
| 11:31.20 | *** join/#asterisk zotz (~zotz@24.231.32.191) |
| 11:31.45 | Zeeek | what was the trouble? |
| 11:32.07 | feral_kid | Zeeek: I have been sitting here trying to check firewall connections and many other things... Then I shut down * and fire-up firefly, and I am not only send, but receive just fine... |
| 11:32.28 | Zeeek | so what is the issue? |
| 11:32.45 | Zeeek | asterisk config? |
| 11:33.36 | feral_kid | Zeeek: Seemingly there is some quirk on A@H that I need to sort out... For instance, if I do a register 12345:passwd@iax2.fwdnet.net/200, it will fail to register |
| 11:34.01 | feral_kid | But if I remove the extention portion it will register... |
| 11:34.29 | feral_kid | So I will have to bang away at this later... I be tired! :) |
| 11:34.37 | jeffik | All: is someone using livevoip? |
| 11:35.15 | Zeeek | feral_kid I know this will sound geeky and I am not a qualified linux or asterisk person, but various versions that configure or setup asterisk "for you" don't get you where you want to go. |
| 11:35.25 | MaeWest | where can i get a list of iax-to-pstn providers? |
| 11:35.37 | Zeeek | iaxproviders.com? |
| 11:35.41 | Zeeek | or .org |
| 11:35.54 | Jas_Williams | feral_kid: IAX does not have the extension part only sip !!!! |
| 11:36.26 | feral_kid | Jas_Williams: Really... Then I guess that would be a slight problem... :) |
| 11:36.56 | feral_kid | I really soundn't be doing these things at 3:30 in the morning... |
| 11:36.58 | *** join/#asterisk jmav (~jmav@201.243.76.158) |
| 11:37.10 | Jas_Williams | yes IAX uses the extensions.con and contexts to set the extension to call :) |
| 11:37.12 | jmav | hello all |
| 11:38.03 | [aryeh] | when i use asterisk it locks down /dev/dsp and nothing else can use it... any ideas why? |
| 11:38.58 | Jas_Williams | [aryeh]: It uses it for the chan_alsa console phone if you do not use it place a noload= in modules.conf |
| 11:39.27 | [aryeh] | well the question is really, can't it be nice so it will share the device? |
| 11:40.01 | [aryeh] | maybe it's my system settings that screwed up... but if i start asterisk.. nothing else can use the sound device... if i'm using the sound dev, then asterisk can't use the console |
| 11:40.13 | feral_kid | Jas_Williams: Then that may be where the problem is in how A@H trys to lay out the extention.conf and how it tries to do iax.conf... Something to look at when I am functioning again... |
| 11:41.07 | Jas_Williams | feral_kid: Get a few hours sleep and come back later |
| 11:44.21 | feral_kid | Zeeek: I agree wholeheartedly about turnkey operations, but keep in mind that certain things exist for reasons. For instance, I am sitting here in Mexico, I have one line coming in from the state ran telephone company, and a few connections via ipkall and sipphone. I have 2 analog phones connected through my Sipura and my Cisco 7960. |
| 11:45.23 | *** part/#asterisk kensuke (~bryan@rrba-146-118-90.telkomadsl.co.za) |
| 11:45.33 | feral_kid | So why take a lot of time to configure a full blown * deployment when my needs are met with A@H... Now, when I get ready to do this hotel, that will be be a different story... :) |
| 11:46.07 | *** join/#asterisk dreamcode (~je@81.181.199.39) |
| 11:46.30 | *** join/#asterisk zotz (~zotz@24.231.32.191) |
| 11:47.40 | feral_kid | Zeeek: But you are right... When you allow someone to do a turnkey solution, you are almost at the mercy of the one that developed it.... Almost, because with a little time, and some time to do an indepth analysis of what the developer was up to, you can figure out which way they are going... |
| 11:47.48 | *** join/#asterisk montag___ (~montag@lan.desys.it) |
| 11:48.05 | montag___ | anyone has tried Planet IP phone VIP-152 ??? |
| 11:48.29 | SwedMiro | and don't forget the knowledge gap..turnkey lets people with less knowledge handle complex systems |
| 11:48.57 | feral_kid | SwedMiro: Well said... And with that you can grow it into what you need it to be... |
| 11:49.06 | miller7 | montag___: you mean the one with the huge buttons? |
| 11:49.32 | Delvar | anyone else notice that asterisk HEAD the event_log doesnt do anyhting? |
| 11:49.42 | wildgoose | Whats the cheapest decent fxs card for a home voip setup in the uk? It's not well covered in the wiki on fxs cards... FXO seems to be the main thing listed? |
| 11:49.55 | sudhir492 | hi all |
| 11:50.02 | *** join/#asterisk amir (~amir@shield.guindehi.ch) |
| 11:50.14 | *** join/#asterisk mAsH` (~mAsH@ppp-217-133-150-46.cust-adsl.tiscali.it) |
| 11:50.17 | sudhir492 | anyone using te110p card here? |
| 11:50.23 | mAsH` | hi all |
| 11:51.40 | [aryeh] | is there an easy way to get the voicemail sound filename added to the CDR record so it's more easily referenced and connected? |
| 11:54.12 | montag___ | miller7: yes., the one with five huge button on the left side |
| 11:54.35 | miller7 | montag___: I haven't seen it but welltech products are more or less ok |
| 11:54.51 | montag___ | miller7: http://shop.voismart.it/prodimages/IPhone_302_big.jpg |
| 11:55.08 | montag___ | miller7: you think is a good phone ??? |
| 11:55.11 | miller7 | I know it, I just haven't tested it |
| 11:55.22 | miller7 | It depends for what you need it for |
| 11:55.59 | montag___ | miller7: testing purpose and few calls :-) |
| 11:56.21 | miller7 | could be ok then |
| 11:56.25 | miller7 | it's a low cost phone |
| 11:56.30 | miller7 | not the lowest possible but low cost phone |
| 11:56.47 | miller7 | Usually the sound on welltech voip products is good |
| 11:57.02 | miller7 | but the firmware has its flaws :) |
| 11:57.37 | tzafrir | hi |
| 11:57.44 | ManxPower | that phone seems rather expensive |
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| 12:16.22 | justinnnnnn | anyone using a 7914 yet ? |
| 12:18.21 | *** part/#asterisk miller7 (~none@adsl49-static-gw1.access.acn.gr) |
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| 12:27.24 | *** join/#asterisk mrproper_ (~psynode@CPE-203-45-98-127.nsw.bigpond.net.au) |
| 12:27.52 | mrproper_ | anyone know how id fix this: WARNING[25628]: loader.c:301 __load_resource: /usr/lib/asterisk/modules/app_readfile.so: undefined symbol: ast_read_textfile |
| 12:30.55 | *** join/#asterisk afe ([S5+u3WJ78@c-e616e055.123-1-64736c12.cust.bredbandsbolaget.se) |
| 12:33.46 | slePP | mrproper_: recompile |
| 12:33.47 | slePP | probably :> |
| 12:33.56 | slePP | and remove things from /usr/lib/asterisk/modules/ |
| 12:33.58 | slePP | then reinstall it |
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| 12:54.41 | *** join/#asterisk fugitivo (~ajf@201.255.100.115) |
| 12:55.49 | *** join/#asterisk didz_ (~didz_@200.218.192.3) |
| 12:56.11 | fugitivo | anyone using asterisk with OpenBSD? |
| 12:56.50 | elric | fugitivo, i am using it with FreeBSD |
| 12:57.29 | fugitivo | elric: i'm worried about hardware support |
| 12:57.53 | elric | fugitivo, most drivers are in Beta stage |
| 12:58.05 | elric | not sure about OpenBSD |
| 12:58.14 | elric | but yeah there will be some Hardware issues |
| 12:58.22 | elric | try using sangoma cards |
| 12:58.32 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 12:58.34 | elric | they have excellent *BSD driver support |
| 12:58.37 | puzzled | hi |
| 12:59.01 | fugitivo | elric: thanks, i'll keep that in mind |
| 12:59.10 | elric | heck when I emailed them about solaris drivers they said if you buy a few cards they were willing to port drivers for solaris |
| 12:59.29 | elric | i like to hear that from a hardware manufacturer |
| 13:00.03 | fugitivo | heh, i agree |
| 13:02.40 | elric | fugitivo, what kind of hardware are you looking into? |
| 13:02.47 | elric | t1/e1 interfaces? |
| 13:03.20 | fugitivo | elric: e1, we don't have t1 in my country :) |
| 13:03.27 | elric | ah cool |
| 13:03.46 | elric | well sangoma make e1 cards. we have e1s too |
| 13:04.02 | elric | only they arent govt certified in australia yet |
| 13:04.11 | elric | no a-tick certification |
| 13:04.22 | fugitivo | i'm checking the webpage |
| 13:04.27 | Zgarbi | I think this is bag... somewhere want to write -- I register 2 numbers in my asterisk sip.conf, incoming operation is correct but statistics is incorrect |
| 13:07.43 | fugitivo | elric: it seems they care about drivers |
| 13:07.50 | ManxPower | Zgarbi, Please try not to use slang on this channel. The international nature of this channel means that slang may not be understood by the person that can help you. |
| 13:07.50 | fugitivo | elric: they support linux from 1.0 :) |
| 13:08.09 | fugitivo | elric: and *BSD as you said |
| 13:08.32 | ManxPower | (yes, I am guilty of using slang too, but I try not to) |
| 13:09.39 | mrproper_ | when compiling asterisk-oh323 it kills asterisk from working, i get unresolved symbols after installing oh323, any ideas? |
| 13:10.33 | elric | fugitivo they do Indeed |
| 13:11.24 | elric | anyway can anyone tell me what sort of server hardware would I need to support 30 out going channels and 90 internal sip phones plus trunking with other locations? |
| 13:18.08 | *** join/#asterisk Darwin[laptop] (~darwin-la@c-24-3-226-147.client.comcast.net) |
| 13:18.53 | ManxPower | elric, You have been here long enough to know we need to know the codec. |
| 13:19.43 | ManxPower | as well as how many calls will be trunked and the bandwidth between locations |
| 13:19.43 | elric | ManxPower, oops G729 most probably. the boss has agreed to pay the licensing cost incurred |
| 13:19.47 | Darwin[laptop] | good morning |
| 13:19.48 | ariel_ | elric 30 channels are via an E1. If everything is using ulaw for your connections you should be able do this on a P4 with 1 gig ram. Now once you start to add other codec's and meetme's that is a different story. |
| 13:19.49 | Zeeek | I don need no stinking codec! |
| 13:20.07 | ManxPower | elric, G729 is very CPU intensive. |
| 13:20.11 | ManxPower | Get a dual Xeon |
| 13:20.12 | elric | hrm |
| 13:20.16 | ariel_ | Morning all |
| 13:21.45 | elric | i prefer AMDs though how would a 64 bit processor like opteron do? |
| 13:22.02 | *** join/#asterisk gbdrbob (~drbob@alltalk.demon.co.uk) |
| 13:22.08 | *** part/#asterisk gbdrbob (~drbob@alltalk.demon.co.uk) |
| 13:23.01 | [aryeh] | in the extensions config file, it seems like after you Dial(..), and the line is answered the next part is excuted when the line is hung up, is this correct? |
| 13:24.01 | fugitivo | elric: i worked a lot with opterons, but not with asterisk, still, the performance you get with opteron and a 64bit linux, is incredible |
| 13:24.15 | [aryeh] | how do I trigger something to happen as soon as the line is answered not after the line has been answered and hungup? |
| 13:24.24 | fugitivo | anyone with experience in asterisk and opteron hardware? |
| 13:25.42 | ariel_ | [aryeh], what are you trying to do? |
| 13:25.55 | elric | fugitivo, i would be using FreeBSD though |
| 13:26.22 | elric | which runs on AMD64 |
| 13:26.48 | *** join/#asterisk tessier (~treed@210.245.97.197) |
| 13:26.59 | fugitivo | elric: hmm, i didn't try freebsd with amd64 |
| 13:27.27 | tessier | Anyone know anything about this rtp timing bug? I have seen this bug on a couple of different systems. I thought it was because we did not have a source of timing like zap hardware or ztdummy. |
| 13:27.47 | ariel_ | elric, Freebsd does not have full support for all of digiums cards. you need to check with the people that are working no the freebsd for asterisk to find out which ones work. |
| 13:28.02 | fugitivo | elric: i like gentoo and amd64 |
| 13:28.18 | ariel_ | tessier, what timing bug? |
| 13:29.50 | tessier | http://www.voip-info.org/wiki-Asterisk+bounty+rtp+timing |
| 13:29.51 | tessier | That one. |
| 13:30.01 | tessier | I don't understand their proposed solution. Get timing from an internal source? |
| 13:30.07 | tessier | I thought that's what zap hardware/ztdummy did? |
| 13:30.17 | elric | ariel_, I have talked to Konstantin and he has send me beta for te405p drivers for freebsd |
| 13:30.18 | [aryeh] | ariel_< i'd like to execute a script when a call is answered |
| 13:30.49 | elric | ariel_, also if they dont work perfectly I will use a sangoma E1 card. |
| 13:31.26 | *** join/#asterisk Shoragan (~shoragan@d072.apm.etc.tu-bs.de) |
| 13:31.49 | elric | fugitivo, i just prefer BSD and if I can use it anywhere I will, if not gentoo would be my second choice, |
| 13:32.22 | tessier | elriah: RedHat not l337 or exclusive enough for you? :) |
| 13:33.22 | elric | tessier, heh its just uniformity of deployment through out the enterprise, all our internal servers are FreeBSD, |
| 13:33.23 | ariel_ | tessier, ok this is the silence compression problem. Which I don't think there going to be adding this to the sip rtp. |
| 13:34.25 | ariel_ | elric, looks like your covering your self. Good luck let us know how it goes. |
| 13:34.25 | tessier | ariel_: Silence compression? But it should still be transmitting something to sync on. We don't have VAD or anything turned on. |
| 13:34.53 | tessier | And actually, we are not using compression. We are ulaw all the way. |
| 13:35.20 | ariel_ | tessier, not all sip devices are the same. This is a problem with just some of them. I don't really see this problem much so most people don't bother over it. |
| 13:36.22 | ariel_ | tessier, sorry it's silence suppression |
| 13:39.02 | *** join/#asterisk santiago (~santiago@63.245.86.89) |
| 13:39.24 | [aryeh] | ariel, and ideas? |
| 13:41.34 | ariel_ | [aryeh], you should look into shady dial on the wiki |
| 13:42.54 | [aryeh] | thanks! |
| 13:44.16 | ariel_ | [aryeh], also look at the wake up call it's another way of doing it. |
| 13:44.37 | [aryeh] | i'm looking at the shady dial unsure if it is what i want |
| 13:46.01 | [aryeh] | i'm looking for sometype of "on ANSWER" functionality |
| 13:46.06 | *** join/#asterisk isamar (~isamar@p8131-ipadfx21sasajima.aichi.ocn.ne.jp) |
| 13:46.08 | isamar | hi folks |
| 13:46.45 | pascals | The manager API is cool |
| 13:47.09 | pascals | I have full integration with our custom built helpdesk system within a day :) |
| 13:47.25 | [aryeh] | for instance, lets say I want to be able to pause certain things as soon as I pick up the phone (like say my audio player...) so when I answer the phone this shold trigger a program that iwll then go and mute/pause xmms |
| 13:47.38 | isamar | needing to make receive a call from a SIP server in my asterisk without any authentication.. IP to IP only... |
| 13:47.42 | isamar | is that possible? |
| 13:48.37 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
| 13:48.49 | *** join/#asterisk TheEmperor (TheEmperor@j167.dms41.jaring.my) |
| 13:49.32 | *** join/#asterisk sysdef (~sysdef@sysdef.admin.debiancenter) |
| 13:49.58 | isamar | needing to make receive a call from a SIP server in my asterisk without any authentication.. IP to IP only... |
| 13:50.00 | isamar | is that possible? |
| 13:50.14 | ManxPower | isamar, yes, of course. |
| 13:50.21 | ManxPower | see permit/deny in sip.conf.sample |
| 13:50.25 | *** part/#asterisk sysdef (~sysdef@sysdef.admin.debiancenter) |
| 13:51.01 | isamar | Man.. but I need that only for an specific connection.. not for all |
| 13:53.07 | *** join/#asterisk |nix (~inix@202.148.164.35) |
| 13:53.29 | *** join/#asterisk _NaNDao_ (~my@c9066570.virtua.com.br) |
| 13:53.30 | _NaNDao_ | hello |
| 13:53.39 | *** join/#asterisk mAsH` (~mAsH@ppp-217-133-150-46.cust-adsl.tiscali.it) |
| 13:54.08 | isamar | fala Nandao :-) |
| 13:54.28 | |nix | hi, i've got an asterisk server with realtime, pgsql, and asterisk cvs |
| 13:54.41 | _NaNDao_ | Can help me ? What´s the minimum PC hardware to run using a 30 lines ? |
| 13:54.52 | |nix | everything works great, with 1 exception. when i call an internal extension, and after it hits the voicemail, the system load will go sky high |
| 13:54.59 | |nix | can anyone provide assistance with this? |
| 13:55.03 | *** join/#asterisk Frantic (~ab@24-193-46-85.nyc.rr.com) |
| 13:55.21 | _NaNDao_ | Isamar, preciso saber qual o hardware minimo para rodar o asterisk ! |
| 13:55.43 | |nix | verbose and debug turned up nothuing |
| 13:55.48 | |nix | i'm running gentoo |
| 13:55.54 | _NaNDao_ | Isamar, irei utilizar um E1 - 30 linhas |
| 13:55.59 | *** join/#asterisk MikeJ[Laptop] (~icechat5@65.170.43.34) |
| 13:56.31 | *** join/#asterisk cgeek (~cgeek@pl6.lawrence.org.uk) |
| 14:00.27 | tessier | |nix: Was realtime fairly easy to set up? Do you find it easier to maintain the configs than text files? |
| 14:01.01 | TheEmperor | hi, can anyone help with jitterbuffer problems? |
| 14:01.17 | TheEmperor | having trouble between 2 locations/extensions, terrible jitters |
| 14:02.28 | ManxPower | TheEmperor, Fix the network jitter. |
| 14:02.38 | TheEmperor | ManxPower: where can I fix that? |
| 14:02.54 | ManxPower | TheEmperor, On the routers between the two Asterisk servers |
| 14:03.10 | TheEmperor | ManxPower: is it in the conf files? |
| 14:03.20 | ManxPower | You *could* also try the experimental patches for jitterbuffer. |
| 14:03.31 | ManxPower | TheEmperor, I don't know. Do your routers use conf files? |
| 14:03.58 | ManxPower | What brand of routers do you use? |
| 14:04.07 | TheEmperor | ManxPower: the problem seems to be on the other side, I can hear them well, but they can't hear me well |
| 14:04.17 | ManxPower | What brand of routers do you use? |
| 14:04.17 | TheEmperor | We use Linksys and D-Link routers |
| 14:04.26 | ManxPower | You can't fix the problem using those routers. |
| 14:04.36 | TheEmperor | what do you recommend then? |
| 14:04.41 | ManxPower | It sounds like you have a cheap internet connection to a cheap ISP. |
| 14:04.45 | mstocco | I knew he was going to say Linksys :) |
| 14:05.17 | TheEmperor | also..when i try and use livevoip/nufone, the delay is like 4-5 secs! I don't think it's them, perhaps its my connection or the other parties connection |
| 14:05.35 | TheEmperor | in my office, i have a 1 Mbps leased line |
| 14:05.41 | ManxPower | TheEmperor, You cannot FIX jitter unless you control EVERY router between the two ends of the link. |
| 14:06.09 | TheEmperor | ManxPower: I don't think it's the routers... |
| 14:06.15 | ManxPower | The_Ball, So you have a 1Mbps leased line between the two offices? Is it a T-1 leased line or a frame relay leased line or an ATM leased line or a PPoE leased line? |
| 14:06.23 | ManxPower | ..e.r.. TheEmperor |
| 14:06.43 | ManxPower | TheEmperor, do you UNDERSTAND what jitter is? |
| 14:06.46 | TheEmperor | no..it's 1Mbps in my office and the other offices are using broadband |
| 14:07.01 | mstocco | TheEmperor: you could have 100M to the internet but those little soho routers won't handle the throughput |
| 14:07.15 | TheEmperor | ManxPower: oh.. |
| 14:07.22 | ManxPower | TheEmperor, then there is NOTHING you can do to fix the jitter. You can only hide it using Asterisk's terribly bad jitterbuffer (but only if you are using IAX) |
| 14:07.44 | ManxPower | If you control every router between the two asterisk servers, then you can FIX the jitter. |
| 14:08.03 | ManxPower | But since you don't control every router between the servers, you cannot fix jitter, you can only hide it. |
| 14:08.18 | TheEmperor | we are using SIP to call other extensions |
| 14:08.35 | ManxPower | TheEmperor, Then there is nothing you can do. Asterisk does not have a jittebuffer for SIP. |
| 14:08.52 | TheEmperor | ManxPower: I see. So you think it's a bad internet connection? |
| 14:08.54 | ManxPower | Welcome to one of the dirty little secrets of VoIP. |
| 14:09.20 | TheEmperor | ManxPower: would it help if I were to purchase a g729 license? |
| 14:09.26 | ManxPower | TheEmperor, Do this: ping www.yahoo.com |
| 14:09.29 | TheEmperor | instead of using GSM |
| 14:09.35 | ManxPower | See the differences in ping time? |
| 14:09.41 | ManxPower | The DIFFERENCE is jitter. |
| 14:09.51 | TheEmperor | yeah |
| 14:10.52 | ManxPower | Asterisk -> Linksys -> ISP router -> ISP router -> ISP router -> ISP router -> D-Link -> Asterisk |
| 14:10.58 | TheEmperor | ManxPower: so do you think it would help if I get a g729 license and use that instead of gsm? |
| 14:11.15 | ManxPower | Didn't I just say you cann't fix jitter with SIP? |
| 14:11.36 | TheEmperor | yes yes, I got that part :) |
| 14:11.44 | ManxPower | If you had routers at your offices that supported giving voice traffic priority then you might be able to REDUCE jitter. Maybe. |
| 14:11.52 | ManxPower | Changing codes will not help. |
| 14:11.59 | TheEmperor | i see |
| 14:12.04 | TheEmperor | so you think it's a crappy internet connection? |
| 14:12.12 | mstocco | arghhhh |
| 14:12.20 | ManxPower | TheEmperor, No. All internet connections without QoS have jitter. |
| 14:12.38 | ManxPower | TheEmperor, now, diagram what your ACTUAL network is, just like I did with the example above. |
| 14:12.54 | ManxPower | Do you have Asterisk on both sides of the link, or only one side of the link? |
| 14:12.55 | *** join/#asterisk mogorman (~mogorman@dhcp-162.digium.com) |
| 14:13.01 | |nix | tessier: i wasn't the one who set it up |
| 14:13.10 | TheEmperor | asterisk is on one side |
| 14:13.23 | *** join/#asterisk Hmmhesays (negative3k@66.173.103.108) |
| 14:13.35 | |nix | i've got a colleague who did it up, and left it alone when he left the company |
| 14:13.36 | mstocco | TheEmperor: it would help if you didn't use $40 Linksys/D-Link routers |
| 14:13.43 | TheEmperor | user 1 -> isp-> asterisk-> isp-> user 2 |
| 14:14.01 | *** part/#asterisk Grooby (~Grooby@ip24-250-126-171.dc.dc.cox.net) |
| 14:14.12 | TheEmperor | mstocco: but some of my clients already use d link and linksys, what do i do about them? |
| 14:14.32 | zoa | ManxPower: the jitter buffer is no longer bad |
| 14:14.36 | zoa | and it also exists for SIP |
| 14:14.39 | ManxPower | Asterisk is hosted at the ISP? |
| 14:14.48 | zoa | although not in stable or in dev |
| 14:15.12 | TheEmperor | ManxPower: we are hosting the * server |
| 14:15.21 | TheEmperor | ManxPower: I think I drew it wrong... |
| 14:15.23 | ManxPower | TheEmperor, Then your diagram is wrong. |
| 14:15.40 | zoa | this is the difference between no jitter buffer and jitter buffer: |
| 14:15.41 | zoa | http://astertest.com/downloads/ast-without-jb.wav |
| 14:15.45 | zoa | http://astertest.com/downloads/ast-with-jb.wav |
| 14:15.51 | zoa | results for iax2 or sip should be similar |
| 14:15.59 | ManxPower | zoa, if it's not even stable/tested enough to be in CVS I can't really recommend it to some poor n00b that doesn't even know the design of his own network. |
| 14:16.12 | zoa | hehe |
| 14:16.13 | zoa | true |
| 14:16.17 | zoa | i agree with that :) |
| 14:16.17 | *** join/#asterisk wildgoose (~edward@xunil.mail.wildgooses.com) |
| 14:16.27 | ManxPower | TheEmperor, What is the correct diagram? |
| 14:16.43 | TheEmperor | ManxPower: user1 -> *server -> router ->dsl line ->isp ->router -> user 2 |
| 14:16.55 | TheEmperor | Again, i could have drawn it wrong |
| 14:17.05 | TheEmperor | please excuse my ignorance, I am new to this :) |
| 14:17.11 | ManxPower | The_Ball, And user1 hears bad auudio or user2 hears bad audio. |
| 14:17.19 | TheEmperor | user 2 |
| 14:17.23 | TheEmperor | user 1 never has a problem |
| 14:17.28 | ManxPower | TheEmperor, user2 has a SIP phone? |
| 14:17.40 | TheEmperor | ManxPower: user2 using a softphone right now |
| 14:18.15 | ManxPower | TheEmperor, The SIP softphone should have jitterbuffer that you can tune. |
| 14:18.19 | TheEmperor | ManxPower:how come user 1 never has problems (no matter where he connects) |
| 14:18.20 | ManxPower | Remember you can only have a jitterbuffer apply to INCOMING audio. |
| 14:18.35 | ManxPower | TheEmperor, maybe there's a virus on the network user2 is on? |
| 14:19.32 | TheEmperor | ManxPower:I don't think so...also, when I try and call a PSTN or mobile going through my * server, the other party hears me only after a few secs delay? |
| 14:19.39 | mstocco | TheEmperor: user1 -> Asterisk -> overloaded soho router -> saterated upstrem dsl -> internet -> another $40 dsl router -> user2 |
| 14:19.53 | MikeJ[Laptop] | zoa, is the sip stuff seperate or in steves mega patch? |
| 14:20.08 | *** join/#asterisk CosmicRay (~jgoerzen@2002:4463:7269:1:20e:a6ff:fe66:c5a3) |
| 14:20.09 | zoa | its seperate |
| 14:20.12 | TheEmperor | mstocco: i don't think the router is overloaded :) |
| 14:20.23 | zoa | my patches also include steves stuff i think |
| 14:20.33 | mstocco | TheEmperor: how do you know? |
| 14:20.53 | TheEmperor | mstocco: there is only 1 server connected to it right now? |
| 14:21.05 | TheEmperor | mstocco: maximum would be 2 |
| 14:21.28 | MikeJ[Laptop] | well.. hopefully steves stuff will hit cvs today, there are a lot of people wiht it installed now |
| 14:21.51 | mstocco | TheEmperor: how many NAT connections is that router dealing with atm? |
| 14:22.20 | TheEmperor | mstocco: probably just 3 |
| 14:22.56 | mstocco | TheEmperor: probably doesn't cut it, you have to find out. |
| 14:23.38 | TheEmperor | mstocco: well, when i was testing today, it was only 1 nat connection : e.g user 1 to user 2 and no one else.. |
| 14:23.59 | *** join/#asterisk BruceX (~bruce@200.29.14.68) |
| 14:24.46 | BruceX | hi :) |
| 14:24.49 | Hmmhesays | hmmm i'm curious... why is enum not more popular |
| 14:25.05 | Hmmhesays | after reading the documentation on it... |
| 14:25.09 | BruceX | anyone have a minute to give me a hand with the atxfer feature ? |
| 14:25.48 | Hmmhesays | ask a question... you'll get a faster answer |
| 14:25.50 | JunK-Y | how is called the switch with the name similar to quantim or something like that? |
| 14:26.08 | zoa | quantum |
| 14:26.13 | mstocco | TheEmperor: how many windows boxes do you have surfing the internet from the same LAN as the asterisk server? |
| 14:26.14 | zoa | quintum |
| 14:26.15 | zoa | sorry |
| 14:26.19 | BruceX | im using the atxfer feature, and it works for the calling party... but i cant make it work for the called party :( |
| 14:26.35 | JunK-Y | zoa: just got it :) |
| 14:26.40 | Hmmhesays | quintum finally released a firmware that allows for multiple sip registrations |
| 14:26.41 | mutilator | behind nat, a person can make and receive calls, but after the first 1-2 seconds he can't hear the other party talking but they can hear him |
| 14:26.42 | JunK-Y | im like lost with all notes here and there. |
| 14:26.47 | mutilator | what causes that? and is there a fix? |
| 14:26.51 | mutilator | using sip |
| 14:27.08 | mutilator | i have qualify=1000 in their userconf and nat=yes |
| 14:27.43 | SwedMiro | anyone got a tip on a good router with VoIP handling? |
| 14:27.48 | Hmmhesays | get out ethereal monitor the connection.... watch for the rtp stream to stop |
| 14:28.22 | mutilator | this is at a customers house.. |
| 14:28.38 | mutilator | i can't duplicate it here.. |
| 14:28.41 | mutilator | works fine |
| 14:30.00 | *** join/#asterisk NTJOCK (~brian@txshirts.com) |
| 14:30.05 | NTJOCK | good morning |
| 14:30.40 | NTJOCK | What's a good way to disable echo cancellation for fax calls? |
| 14:30.41 | BruceX | im using the atxfer feature, and it works for the calling party... but i cant make it work for the called party :( , any hints on that ? |
| 14:30.56 | SwedMiro | Question.....anyone got a tip on a good router with VoIP handling? |
| 14:30.59 | ManxPower | NTJOCK, Asterisk will disable it the moment it hears a fax tone |
| 14:31.04 | NTJOCK | hmm, ok |
| 14:31.10 | NTJOCK | I keep having incoming faxes hangup . |
| 14:31.31 | ManxPower | NTJOCK, what makes you think it's echocan causing it? |
| 14:31.42 | NTJOCK | I was going to implement Spandsp (and plan to)... but * dies when those modules load and someone this wknd said they saw a bug... |
| 14:31.50 | NTJOCK | no reason... just a wild guess. |
| 14:31.59 | NTJOCK | faxes work fine on Analog lines directly. |
| 14:32.02 | NTJOCK | Only thing in between is the digium card. |
| 14:32.21 | NTJOCK | POTS, FXO port, inbound, goes back out on a FXS port to a modem 3' cable feet away |
| 14:32.29 | NTJOCK | modems are Hayes Accuras |
| 14:32.34 | fantomax1 | is there anyone from Germany here ? |
| 14:32.36 | NTJOCK | I turned off ECM on the modems. |
| 14:32.55 | NTJOCK | I'm basically executing a fax, 1, Dial (zap/6/1) |
| 14:33.20 | TheEmperor | mstocco: none |
| 14:33.26 | NTJOCK | the modem is terminating the call after chirping for 20 seconds |
| 14:33.41 | *** join/#asterisk coppice (~chatzilla@111.196.17.210.dyn.pacific.net.hk) |
| 14:33.45 | NTJOCK | Hi coppice |
| 14:33.50 | coppice | hi |
| 14:33.57 | TheEmperor | mstocco: the asterisk server has it's own 1Mpbs leased line which is not shared with anyone/anything |
| 14:34.07 | NTJOCK | I was just telling manx about some headaches with my inbound faxes.... |
| 14:34.26 | NTJOCK | sure wish I could get spandsp to load... :) |
| 14:35.16 | coppice | hundreds do |
| 14:35.43 | NTJOCK | I have it installed and compiled... * just dies when it hits the TX or RX modules |
| 14:35.43 | coppice | probably thousands by now. hard to tell, really |
| 14:35.46 | NTJOCK | :) |
| 14:36.01 | *** join/#asterisk mentat (~Mentat@pcp01260498pcs.nhaven01.ct.comcast.net) |
| 14:36.06 | NTJOCK | might be millions in a few minutes at that rate. |
| 14:36.07 | NTJOCK | :) |
| 14:36.20 | BruceX | <PROTECTED> |
| 14:36.35 | BruceX | <PROTECTED> |
| 14:37.28 | Hmmhesays | now that's what I call verbose |
| 14:37.40 | BruceX | hhahah |
| 14:37.42 | BruceX | sorry |
| 14:37.43 | BruceX | :) |
| 14:37.59 | NTJOCK | or a cat on the keyboard |
| 14:38.00 | NTJOCK | ;) |
| 14:38.20 | *** join/#asterisk mjgeorge (~george@216.157.203.105) |
| 14:38.42 | mjgeorge | Hello everyone! I'm having trouble detecting incoming faxes on a zap channel. |
| 14:38.56 | NTJOCK | just put a FAX line in your dial plan |
| 14:39.00 | NTJOCK | should pick it right up and shove it over |
| 14:39.03 | NTJOCK | example |
| 14:39.05 | mjgeorge | I have read the mailing list and the wiki, but they just say to have the extension fax in your dialplan. |
| 14:39.11 | NTJOCK | exten => fax,1,Dial(zap/8/1) |
| 14:39.13 | mjgeorge | I have it in there, but it doesn't seem to detect it. |
| 14:39.28 | NTJOCK | is it in the dial plan where your zap channels are? |
| 14:39.34 | mjgeorge | exten => fax,1,Dial(${FaxExt}) |
| 14:39.44 | NTJOCK | hmmm, what is fax ext set to? |
| 14:39.55 | mjgeorge | when I tried to fax to it, I saw it hit the right cnotext... |
| 14:39.59 | mjgeorge | faxext is set to Zap/1 |
| 14:40.18 | NTJOCK | is your fax modem ringing? |
| 14:40.46 | NTJOCK | Manx/Coppice: I just went and switched fax modems and apparently a class2 has headaches and a class1 works. |
| 14:40.52 | mjgeorge | The fax machine called adn I could watch the call go through the incoming dialplan for the afterhours stuff that I included. |
| 14:41.06 | NTJOCK | hmm.... |
| 14:41.15 | mjgeorge | the fax machine was sending the tone, but * never did anything except go on to send the call to voicemail. |
| 14:41.30 | NTJOCK | it works for me, I'm not an expert here.... you want to post your dialplan to pastebin and I'll go peek? |
| 14:41.31 | mjgeorge | Is there something special that I have to set in zapata.conf that enables fax detectino? |
| 14:41.41 | NTJOCK | oh yea.... |
| 14:41.43 | NTJOCK | let me loock |
| 14:41.44 | NTJOCK | look |
| 14:41.45 | NTJOCK | :) |
| 14:42.11 | NTJOCK | try faxdetect= |
| 14:42.23 | NTJOCK | options are inbound, outbound, no (never or something like that), and both |
| 14:42.26 | NTJOCK | I have it set to both :) |
| 14:42.39 | mjgeorge | I see it. Wht's the default? |
| 14:42.50 | NTJOCK | no idea. |
| 14:42.53 | NTJOCK | ;( |
| 14:43.02 | NTJOCK | Try that, and restart and let me know if it helped. |
| 14:43.09 | SwedMiro | Question.....anyone got a tip on a good router under 250$ with VoIP handling? |
| 14:43.24 | mjgeorge | Okay. the commented out line said "incoming", are you sure about "inbound"? I'll try to find it in the wiki. |
| 14:47.45 | zoa | swedmiro: draytec |
| 14:47.47 | zoa | dryatek |
| 14:48.38 | zoa | draytek |
| 14:48.40 | zoa | vigor |
| 14:49.22 | mutilator | ok so my rtp stream is dieing behind the nat |
| 14:49.32 | mutilator | what can i do about that |
| 14:49.45 | *** join/#asterisk mmckernan (mmckernan@c211-28-35-204.sunsh1.vic.optusnet.com.au) |
| 14:49.48 | mmckernan | hey all. |
| 14:49.52 | mentat | hi |
| 14:50.10 | mjgeorge | NTJOCK: I have put faxdetect=incoming into my zapata.conf file for my FXO and restarted |
| 14:50.13 | mmckernan | does anyone know of a bug with parking? in the sense of: 1. Park the call, 2. Let call timeout and come back. 3. Unable to repark the call |
| 14:50.30 | mjgeorge | However, when I "zap show channel 4" it says "Fax Handled: no" |
| 14:50.38 | mjgeorge | should it say "yes"? |
| 14:50.39 | mmckernan | i have just knocked up a quick patch, just wondering if i should raise it in bugtraq |
| 14:50.56 | *** part/#asterisk _NaNDao_ (~my@c9066570.virtua.com.br) |
| 14:51.32 | dreamcode | what is the difference between using ChanIsAvail and just making a looong dial line ? |
| 14:52.14 | *** join/#asterisk tekati (~captain@cpe-66-75-215-63.bak.res.rr.com) |
| 14:52.49 | rious | has anyone ever had trouble with outgoing lines not detecting DTMF ? |
| 14:52.54 | *** join/#asterisk wmoran (~wmoran@pa-plum-cmts1e-68-68-113-64.pittpa.adelphia.net) |
| 14:53.13 | wmoran | Anyone have any suggestions about fixing echo problems ... other than what's on the wiki? |
| 14:53.39 | wmoran | It doesn't seem like "echotraining" is doing anything at all. |
| 14:53.56 | |nix | wmoran |
| 14:54.01 | |nix | you using e1? |
| 14:54.11 | wmoran | Nope, PSTN |
| 14:54.18 | |nix | if you're using the tdm series, then checkout the latest CVS |
| 14:54.19 | |nix | and fxotune |
| 14:54.22 | |nix | quite effective |
| 14:54.31 | wmoran | How stable is the latest CVS? |
| 14:54.44 | wmoran | This is my office system, I can't really risk it being unstable |
| 14:55.43 | *** join/#asterisk _tekati_ (~captain@cpe-66-75-215-63.bak.res.rr.com) |
| 14:55.45 | |nix | best way is to test it out yourself |
| 14:55.50 | wmoran | Heh ... |
| 14:55.55 | wmoran | Not really ;) |
| 14:56.19 | wmoran | I'll have to take time in the evening or on a weekend |
| 14:56.38 | wmoran | Does echotraining actually work for anyone? |
| 14:57.46 | BruceX | im running cvs |
| 14:57.54 | BruceX | heavy load, no core |
| 14:58.00 | wmoran | That's good news |
| 14:58.14 | wmoran | Are there any docs on this new fxotune thing, or will I need to read the source? |
| 14:58.27 | BruceX | but i havent been able to get atxfer working for the called party :( |
| 14:58.39 | BruceX | i think you should read the source |
| 14:58.47 | BruceX | asterisk sucks on documentation :/ |
| 14:59.29 | MikeJ[Laptop] | if you are so unhappy with docs, right somthing down... the wiki needs your help |
| 14:59.41 | MikeJ[Laptop] | right=>write |
| 15:00.00 | MikeJ[Laptop] | only you can fix the documentation problem.. only you. |
| 15:00.19 | BruceX | yeh, im trying |
| 15:00.21 | wmoran | A coworker just wrote up how he got SpanDSP installed ... he's submitting it to the wiki today |
| 15:00.29 | BruceX | but i'm in the work of understanding the source first :( |
| 15:00.39 | mmckernan | I have submitted a new bug with patch to bugid 3777, in regards to the problems of re-parking a call after it timed out in the parking lot. |
| 15:01.13 | coppice | wmoran: why can't people just foloow the documentation on how to install spandsp? |
| 15:01.27 | wmoran | What documentation? |
| 15:01.44 | wmoran | It took us 30 minutes just to figure out that it needed installed separately |
| 15:01.47 | coppice | why don't people bother to look? |
| 15:02.06 | wmoran | Then the install process didn't do an ldconfig, and we had to do it manually ... |
| 15:02.17 | wmoran | So I believe some docs on the wiki would help |
| 15:02.36 | SwedMiro | Question.....anyone got a tip on a good router under 250$ with VoIP handling? |
| 15:02.49 | coppice | the autoconf tools normally handle the ldconfig OK |
| 15:03.37 | coppice | do you have much that is meaningful to add to http://www.soft-switch.org/installing-spandsp.html |
| 15:03.46 | ariel_ | SwedMiro, I uset the linksys WRT54GS then upgrade the firmware to Sveasoft |
| 15:05.15 | ariel_ | coppice, Hello. Great to hear about your project for T.38. I am looking forward to it. |
| 15:05.33 | *** join/#asterisk mhnoyes (~mhnoyes@user-38lc09b.dialup.mindspring.com) |
| 15:06.27 | *** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com) |
| 15:06.28 | coppice | wmorna: twice people have posted workarounds for bugs in spandsp on the wiki without telling me about the problem. I think people just want to get their name in the wiki |
| 15:08.29 | RoyK | :) |
| 15:08.34 | RoyK | Mar 17 16:07:34 NOTICE[4216]: chan_sip.c:7867 handle_request: Unknown SIP command 'PUBLISH' from '80.239.107.69' |
| 15:08.38 | RoyK | wtf does that mean? |
| 15:09.47 | *** join/#asterisk cbachman (~chatzilla@victory.ece.northwestern.edu) |
| 15:09.52 | wmoran | coppice: two things: 1) The problem with ldconfig, 2) We had to fidget around a bit to get the correct libtiff libraries installed |
| 15:10.06 | wmoran | coppice: Admittedly, #2 could be a Fedora only problem. |
| 15:10.07 | tessier | RoyK: It means asterisk desperately needs better error messages. |
| 15:10.45 | *** join/#asterisk Nugget (nugget@dazed.slacker.com) |
| 15:10.51 | wmoran | coppice: I'll have this guy email you his howto before he posts it. |
| 15:11.12 | ariel_ | RoyK, look at all of these http://www.google.com/search?hl=en&lr=&q=Unknown+SIP+command+%27PUBLISH%22&btnG=Search |
| 15:11.17 | *** join/#asterisk eKo1 (~bernd@63.245.57.70) |
| 15:11.33 | wmoran | coppice: the most annoying thing was that we spent a lot of time surfing the wiki before we even realized that SpanDSP was a addon, and that it was the one we wanted. |
| 15:11.38 | coppice | wmoran: the libtiff issue is the first paragraph of the installation notes. the ldconfig issue is something odd you have. it normally works OK. an autoconfig "make install" usually runs ldconfig |
| 15:12.36 | wmoran | I'm not exactly sure how to make the wiki clearer on how SpanDSP works, though ... once you've fought through finding something and now know where it is, it's hard to remember where to go back to to add links |
| 15:13.01 | ariel_ | RoyK, looks like either your missing a context or something like that. |
| 15:13.10 | wmoran | The libtiff advice in your docs wasn't good enough for him ... he had to figure out some stuff |
| 15:13.21 | RoyK | ariel_: er - what? how can this be a context error? |
| 15:13.40 | wmoran | coppice: I'm not saying your docs are bad ... I'm just saying a few minor additions would make it easier for peope to get up and running ... and that's what it's all about, if you ask me. |
| 15:13.40 | ariel_ | wmoran, it just needs a note that says you need to follow these instructions on: put the spandsp installation guide url there. |
| 15:14.26 | ariel_ | RoyK, did you read the pass list on the google search? |
| 15:14.27 | coppice | wmoran: oh, the "your instructions are good enough because I refused to follow them" school of thought :-) |
| 15:14.42 | wmoran | Now, if I could just get these echo problems fixed ... |
| 15:14.46 | wmoran | coppice: ??? |
| 15:15.02 | elriah | Hi all, anyone know of a faq that will show how to push iax or sip through a NAT'd firewall? i.e., what ports to forward to asterisk? |
| 15:15.02 | mmckernan | oej: you in here? |
| 15:15.26 | ariel_ | elriah, it's on the wiki |
| 15:15.28 | RoyK | ariel_: it works with another client |
| 15:15.38 | wmoran | I'm pushing _out_ of a nated firewall with no special config ... can't say anything about pushing in, though |
| 15:15.53 | elriah | tnx |
| 15:17.57 | *** join/#asterisk stevekstevek (~stevekste@slim-eth0.horizonlive.net) |
| 15:19.41 | wmoran | Is it possible to compile asterisk without echotraining support? |
| 15:19.55 | epoch | anything's possible! |
| 15:20.07 | wmoran | Well ... yeah ... |
| 15:20.19 | elriah | Are there any "good" iax2 phones out there? I was about to purchase 10 polycom phones, but this SIP nat issue might get me to switch to an IAX phone if they exist. Any suggestion? |
| 15:20.26 | wmoran | I guess what I mean is: is it likely that I accidentally did that and that's why I'm having echo problems? |
| 15:20.27 | *** join/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
| 15:21.15 | ariel_ | elriah, good ones not yet. But polycom work you just need to make sure of a few settings in the sip.conf like externip= and open your ports. |
| 15:21.28 | *** join/#asterisk trym (trym@linux.debian.us) |
| 15:21.58 | elriah | Ok, I'm ordering the Polycom 500 series phones then. |
| 15:22.15 | ariel_ | elriah, I have in my sip.conf externip = 68.142.XX.XXX |
| 15:22.15 | ariel_ | localnet = 192.168.88.0/255.255.255.0 |
| 15:22.15 | ariel_ | ; |
| 15:22.36 | elriah | tnx |
| 15:22.45 | *** join/#asterisk Cherebrum (~jgarland@216.32.77.10) |
| 15:22.47 | Cherebrum | Morning! |
| 15:22.58 | wmoran | Indeed |
| 15:23.23 | wmoran | coppice: I've been trying to figure out where this guy had problems, and I think I know where. |
| 15:23.36 | ariel_ | Cherebrum, crate... can I have one,,, can I? |
| 15:23.44 | Cherebrum | heh heh |
| 15:23.58 | wmoran | coppice: the install doc on the web site explains the libtiff thing perfectly, but the README not so well ... It may be better to just have the readme point to the web page |
| 15:24.17 | wmoran | I think he tried to work off the README and didn't actually read the install web page. |
| 15:24.42 | *** join/#asterisk Bob_ (~icechat5@196.7.14.183) |
| 15:24.56 | wmoran | Are those Polycom good phones? |
| 15:24.58 | Zeeek | coppice - wouldn't it make sense to centralize all that is know about spandsp in one place, like a spandsp wiki? |
| 15:25.00 | wmoran | We need a better phone |
| 15:25.13 | wmoran | Our Snom phone wore out too quickly |
| 15:25.19 | ariel_ | wmoran, there good |
| 15:25.28 | wmoran | Expensive? |
| 15:25.40 | ariel_ | wmoran, less then the snom |
| 15:25.45 | wmoran | ! |
| 15:26.00 | Bob_ | Hi there. Does anyone out there have experience with using conferencing with SER (SEMS module) and with Asterisk (Meetme module)? |
| 15:26.40 | Bob_ | I am currently using SER and SEMS, but am looking at the possibility of migrating to Asterisk |
| 15:26.43 | *** join/#asterisk _-Jon-_ (~jon@CPE000d8861e8f7-CM00080d290642.cpe.net.cable.rogers.com) |
| 15:26.52 | tessier | Bob_: Why asterisk instead of SEMS? |
| 15:26.53 | Darwin[laptop] | cool xdesk on windows is nice 6 virtual desktops |
| 15:27.04 | ariel_ | Bob_, asterisk works just fine with SER |
| 15:27.15 | tessier | Darwin[laptop]: Wow. Only 15 years behind Linux! :) |
| 15:27.16 | Moc____ | hi everyone |
| 15:27.22 | Darwin[laptop] | hehhe |
| 15:27.31 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.rr.com) |
| 15:27.31 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:27.35 | Bob_ | It seems that Asterisk might be more stable and have a wider user community |
| 15:27.56 | Darwin[laptop] | I use fbsd but needed a windows laptop for program for programing payphones |
| 15:28.01 | ariel_ | Bob_, asterisk does allot. What are your needs? |
| 15:28.17 | _-Jon-_ | Hey, I'm wondering if someone could help me out with a little problem I'm having.. I use the CID information to validate who I am so that I can call out through another line, however my CID info is never correct when I call from my cell phone. Gives me a different number each time. Is there a way I can validate myself with * by entering my actual number? |
| 15:28.30 | coppice | wmoran: fair point. I just removed all the instructions from the README file, and left a pointer to the web page. |
| 15:28.33 | jluk | Has anyone ever played with distinctive ring on a 7960G - I'm just wondering what sort of things are possible ? |
| 15:28.45 | Bob_ | Well, basically softphone conferencing. SER and SEMS is providing what I need, but I am a bit worried that it won't be that stable with many users |
| 15:28.47 | Darwin[laptop] | fixcid in the wiki |
| 15:28.47 | Zeeek | _-Jon-_ see DISA |
| 15:28.57 | Bob_ | I am looking for move into a production environment |
| 15:29.00 | wmoran | coppice: Hah! That means our fumbling around _was_ useful! |
| 15:29.39 | Darwin[laptop] | whenis 1.0.7 going to be released so we can get the port updated |
| 15:30.01 | _-Jon-_ | Zeeek, thanks, I will look into that |
| 15:30.05 | ariel_ | Bob_, I don't know SEMS But I know Asterisk and it's great but it really depends on how many meetme users your going to have at one time. |
| 15:30.11 | Zeeek | it authenticates incoming callers |
| 15:30.39 | Zeeek | unfortunately, the line has to be answered to use it |
| 15:31.02 | Bob_ | What is the capacity capability of conferencing with Asterisk - I would need multiple conferences concurrently - maybe 10 conferences with 10 people each |
| 15:31.16 | Zeeek | _-Jon-_ better solution is to find why CID is wrong in the first place |
| 15:32.07 | ariel_ | Bob_, what codec will you be using? |
| 15:32.46 | *** join/#asterisk srosebush (~steve@initrd.com) |
| 15:32.52 | Bob_ | mmm - not too sure |
| 15:34.01 | Bob_ | ariel - do you know anyone with experience in using SEMS in a live environment - I would rather not move servers if it is not a necessity |
| 15:34.45 | ariel_ | Bob_, I have never heard of SEMS much less know someone with it. |
| 15:35.21 | Bob_ | I don't think that it is widely used |
| 15:35.32 | Bob_ | Have you used conferencing with Asterisk |
| 15:37.22 | ariel_ | Bob_, yes lots of times |
| 15:38.10 | srosebush | hey there -- im trying to get caller id to work on the lastest version of asterisk and zaptel in the netherlands but it isnt working, i have it set to this: usecallerid=yes, cidsignalling=dtmf, cidstart=polarity.. ugh im using a X100P clone... i've read on the net that it should work with these settings but it doesnt? i can confirm i do have caller id on the line as if i hook a phone directly to it, it shows up. |
| 15:38.18 | _-Jon-_ | Zeeek, I think my cell provider has some issues |
| 15:38.51 | _-Jon-_ | Zeeek, As I travel around different areas that are in different area codes it gives me a random number in that area code |
| 15:39.08 | elriah | Can anyone help me out with my extensions.conf? I just setup my first pbx last night, all seems well when connecting with a softphone and going through the 1000 extension and the 500 extension default demos, but I can't dial another softphone extension and I'm sure it's something to do with my extensions.conf that I haven't edited from defaults yet. Any help would be appreciated. |
| 15:40.54 | Bob_ | ariel - Is is difficult to configure - I have installed Asterisk and want to use conferencing with SIP |
| 15:41.08 | ariel_ | elriah, just make sure you setup the extensions in the same context I think in the sample it's extensions but I have not used it in years. |
| 15:41.12 | Zeeek | _-Jon-_ well you'll need DISA then, but the line will have o answer and present an IVR menu (and when you hit * or something is will wait for a PIN) |
| 15:41.23 | Zeeek | lot of shit to enter |
| 15:41.47 | *** join/#asterisk Uther_P (~uther_p@66.180.120.83) |
| 15:42.14 | ariel_ | Bob_, it is really easy once you get it started. But you need to make sure you have a timing device setup first like a zap board or ztdummy |
| 15:42.59 | _-Jon-_ | Zeeek, that's fine. I already have a menu that comes up with some BS. It's just that most recently it saw my number as "asterisk" so I feel I have to open the number up to accept calls from anyone and then verify me once I'm in |
| 15:44.18 | Bob_ | Would I need to install the Zaptel module for ztdummy? |
| 15:44.29 | *** join/#asterisk afe ([gpLJaGHAl@c-e616e055.123-1-64736c12.cust.bredbandsbolaget.se) |
| 15:44.33 | Bob_ | Or is it better to get a Zap board and how much would it cost |
| 15:45.37 | *** part/#asterisk srosebush (~steve@initrd.com) |
| 15:46.29 | Moc____ | is nufone down ? |
| 15:49.37 | Darwin[laptop] | the whole world is down it seems |
| 15:49.46 | Darwin[laptop] | its got to be a backbone issue |
| 15:49.57 | CosmicRay | wow, I have heard a lot of complaints about nufone in the last couple of days |
| 15:49.58 | Moc____ | 85% of my outgoing call drop |
| 15:50.10 | Moc____ | or get congestioned |
| 15:50.15 | Darwin[laptop] | Nufone/BV/VP |
| 15:50.17 | ariel_ | Bob_, look at asterisk@home it has eveything you need there including a really nice setup fo a newer meetme 2. |
| 15:50.26 | Darwin[laptop] | they are all having issues |
| 15:50.44 | ariel_ | It comes preconfigured with ztdummy or you can get an x100p board for you system. |
| 15:51.29 | SwedMiro | oh...is ztdummy preinstalled in *@home |
| 15:51.34 | SwedMiro | :) |
| 15:51.39 | CosmicRay | it is trivial to install elsewhere though |
| 15:53.11 | Bob_ | ariel - thanks, appreciate it |
| 15:53.16 | SwedMiro | anoyone know of a tool to check out how a router works..ehh..how well it works as a router for * traffic |
| 15:54.48 | Bob_ | ariel - sorry are you refering to the asterisk website - www.asterisk.org? |
| 15:55.16 | Darwin[laptop] | morning Kram |
| 15:55.21 | *** part/#asterisk Moc____ (~mochouina@h66-201-214-109.gtconnect.net) |
| 15:55.24 | kram | morning |
| 15:55.42 | Darwin[laptop] | how goes the struggle |
| 15:55.49 | ariel_ | Bob_, look at http://asteriskathome.sourceforge.net/ |
| 15:56.07 | ariel_ | morning kram hope everything is going well. |
| 15:56.24 | Bob_ | ariel - thanks. |
| 15:56.55 | Katty | ariel_: way to set off my hilight (= |
| 15:57.27 | ariel_ | Katty, morning hope everything is well. Are you over your cold? |
| 15:57.51 | eKo1 | Wow, now I have about 70 ghost channels. |
| 15:58.00 | Katty | ariel_: yub yub (= |
| 15:58.08 | Katty | ariel_: how is you and family? |
| 15:58.11 | *** join/#asterisk alcaselzer (ALCASELZER@mirko.hta-bi.bfh.ch) |
| 15:58.16 | eKo1 | Say, what's the limit on the number of active channels on *? |
| 15:58.17 | alcaselzer | Hi |
| 15:58.32 | *** join/#asterisk bannerman (~bannerman@209.216.176.42) |
| 15:58.40 | ariel_ | Katty, just fine everyone here is over the cold too. Thanks |
| 15:58.43 | jluk | eKo1: that depends on your hardware and the types of channels |
| 15:58.57 | bannerman | Hi. I've got a bit of an emergency.. I was happily working away on my server and dialplan, and don't have my new phones in yet, and everyone decided to move to the new office today. |
| 15:59.08 | alcaselzer | I need a S-bus card for my Asterisk gateway.. someone that know? |
| 15:59.14 | eKo1 | jluk: eh, I'm talking about the soft limit, not what my hardware can handle. |
| 15:59.17 | bannerman | So, I need to learn in about 10 minutes how the receptionist can transfer a call to a cell phone |
| 15:59.20 | ariel_ | eKo1, I don't think there is a real limit but it depends on the use coding and allot of factors. |
| 15:59.33 | bannerman | How would I setup an extension that would dial a particular cell number? I use voipjet |
| 15:59.42 | *** join/#asterisk caesar2 (~igerl@ppp-62-245-163-62.mnet-online.de) |
| 15:59.57 | bannerman | Normally I'd hit the wiki and figure it out.. |
| 16:00.15 | sivana | bannerman: you would just exten => s,1,Dial(voipjet/cellnumber) |
| 16:00.22 | bannerman | sivana, I love you. |
| 16:00.33 | sivana | fill in the blanks with your particular info |
| 16:00.38 | wmoran | Is there an equivalent to rxgain and txgain for SIP phones? |
| 16:00.50 | eKo1 | If the number of channels is stored as an usigned int, then the theoretical max. is 2^32 channels. |
| 16:01.20 | jluk | eKo1: there's a lot more to it than that. |
| 16:01.46 | *** join/#asterisk heison (~heison@ns.somanetworks.com) |
| 16:01.48 | alcaselzer | I need a dual port S-bus (BRI) card for my Asterisk gateway.. |
| 16:01.50 | heison | ~seen JerJer |
| 16:01.54 | jbot | jerjer <~JerJer@dsl-106-170.che.centurytel.net> was last seen on IRC in channel #asterisk, 17d 23h 42m 59s ago, saying: 'mrgoby: sure'. |
| 16:01.54 | *** join/#asterisk Skysky (~Miranda@host6614613596.biz.tor.fcibroadband.com) |
| 16:02.00 | sivana | ~seen heison |
| 16:02.01 | jbot | heison is currently on #asterisk (15s). Has said a total of 1 messages. Is idling for 11s |
| 16:02.01 | eKo1 | jluk: What else? If there are more channels than that, there will be an integer overflow and that is a no-no. |
| 16:02.11 | caesar2 | bad thing... i use 4 channels rigth no so i have almost no chans left :-) |
| 16:02.16 | sivana | hey heison :) |
| 16:02.34 | jluk | eKo1: you need to take into account file descirptors and stuff like that I'd guess. |
| 16:02.35 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
| 16:02.39 | wmoran | Is there an equivalent to rxgain and txgain for SIP phones? |
| 16:03.00 | sivana | ~seen eric_ |
| 16:03.01 | jbot | eric_ is currently on #asterisk (5d 7h 59m 19s). Has said a total of 41 messages. Is idling for 10h 47m 51s |
| 16:03.07 | sivana | hehe |
| 16:03.19 | coppice | wmoran: most phones have some form of volume control |
| 16:03.47 | jluk | eKo1: theoretical max based on an unsigned integer is pointless - no hardware that * runs on can handle that kind of load. |
| 16:03.49 | eKo1 | Man, I have so many ghost channels and * hasn't crashed. |
| 16:03.53 | bannerman | Can I do "reload" when I have calls up, or will that drop thme? |
| 16:03.58 | alcaselzer | ceasar.. you are using 4 channels.. i want to buy a 2 BRI port card for my PC.. wich card are you using? |
| 16:04.09 | sivana | bannerman: reload will not drop channels, only restarts |
| 16:04.16 | *** join/#asterisk KennethPorter (~chatzilla@segw.microprecisionautomation.com) |
| 16:04.42 | jluk | eKo1: pse explain what is you mean by a ghost channel |
| 16:05.03 | bannerman | sivana: thanks again |
| 16:05.38 | eKo1 | A ghost channel is a channel that exists when you do a 'show channels' on the cli but you know for a fact that the channel is already dead. |
| 16:06.40 | *** join/#asterisk Moc____ (~mochouina@h66-201-214-109.gtconnect.net) |
| 16:07.33 | jluk | eKo1: don't know where things top out in that case. A ghost channel would/should have released most of it's resources and be just sat there waiting for final destruction. |
| 16:08.09 | Koshatul | anyone know of a way to call an extension, tell them the calling number and ask for a "press 1 to receive call, press 2 to push to voicemail" ? my idea is i have a remote call forwarding system, so when we are all out of the office we cna setup remotely the call forward, but i want it to ask us before we get the call, not just ring and drop on us ? |
| 16:08.59 | Koshatul | i've been googling for about 5 minutes on it, i just can't think of some good keywords for it |
| 16:09.14 | Koshatul | i was thinking something to do with parking ? |
| 16:09.15 | eKo1 | jluk: Yeah, I usually kill them using 'soft hangup'. |
| 16:09.40 | KennethPorter | At http://www.asterisk.org/index.php?menu=support#irc would be nice to have irc://irc.freenode.net/asterisk link |
| 16:09.54 | eKo1 | I wish my mrtg memory usage graph was working. Then I could check how much memory these ghost channels are eating. |
| 16:10.47 | KennethPorter | On same page, 2nd draft of Handbook is advertised, but file says last edit was 3/2003; is this no longer maintained? |
| 16:12.06 | *** join/#asterisk jeffik (~jeffik@208-41-192-106.client.dsl.net) |
| 16:13.14 | sivana | eKo1: restart now :) |
| 16:15.44 | Essobi | anyone had a case where rtp.c is generating bad checksums? |
| 16:17.11 | eKo1 | sivana: No. Not yet. |
| 16:17.22 | sivana | :) |
| 16:17.40 | *** join/#asterisk Frantic (~ab@TechnologicPartners35.dsl.concentric.net) |
| 16:17.49 | eKo1 | I'll just wait until * gets into a completely useless state. |
| 16:17.53 | tzanger | sivana: put down the keyboard and step away from the computer. Who said you could provide help? :-p |
| 16:18.36 | Katty | tzanger (= |
| 16:18.43 | sivana | hehe |
| 16:19.02 | Katty | gosh |
| 16:19.15 | Katty | tzanger )= |
| 16:19.54 | sivana | hey hey |
| 16:19.54 | tzanger | eek |
| 16:20.01 | sivana | don't be bashing ASP.. hehe |
| 16:20.09 | Katty | sivana: bash bash bash bash bash |
| 16:20.11 | tzanger | ASP!! My only weakness... how did she know... someone must have told her... SIVANA?!?! |
| 16:20.12 | Katty | sivana: BASH |
| 16:20.24 | eKo1 | don't bash bash |
| 16:20.33 | Katty | sivana: i can bash all i want. i have 22 pages at font size 4 right here. |
| 16:20.40 | Katty | sivana: and it should only be 4, not 22 |
| 16:20.43 | sivana | dammit I'm telling Billie |
| 16:20.59 | BruceX | im using the atxfer feature, and it works for the calling party... but i cant make it work for the called party :( , any hints on that ? |
| 16:21.32 | sivana | Katty: are you working with ASP? |
| 16:21.35 | Moc____ | hi sivana |
| 16:21.38 | sivana | hey dude |
| 16:21.41 | sivana | I got inventory!! |
| 16:21.42 | Katty | sivana: no, i'm just sitting here staring at it |
| 16:21.53 | Katty | sivana: how else would i know that it should only be 4 pages instead of 22? |
| 16:22.05 | Katty | gosh, i'm grumpy today. |
| 16:22.07 | Jas_Williams | BruceX: include a T after the dial |
| 16:22.17 | sivana | heh |
| 16:22.19 | BruceX | i've tried T , t and tT |
| 16:22.24 | Essobi | sivana You poor poor man. |
| 16:22.24 | Katty | bye now. |
| 16:22.30 | sivana | it's fun :) |
| 16:22.47 | Essobi | Someone wrote an * .net mgmt tool.. beta on voip-info |
| 16:22.57 | Jas_Williams | tT should allow both parties to transfer :( |
| 16:23.02 | sivana | Essobi: where? |
| 16:23.03 | CosmicRay | sivana: does it work with mono? |
| 16:23.07 | Essobi | sivana main page |
| 16:23.08 | BruceX | Dial(IAX2/ast/1234,100,Ttr) |
| 16:23.15 | Essobi | It BEATS your management port to death. |
| 16:23.17 | sivana | CosmicRay: no, don't think so |
| 16:23.26 | BruceX | with t and T individually too.. but dont work for the called party :( |
| 16:23.40 | sivana | I think AgiNamu was working on .NET with mono |
| 16:23.42 | Essobi | Anyone know where UDP checksum generation actually occurs? In * or Linux? |
| 16:24.05 | *** join/#asterisk emacsen (hidden-use@gw.coderyte.net) |
| 16:24.08 | Essobi | don't have a very high debug level if you're going to run that .net program |
| 16:24.22 | Essobi | I'm taking a different approach with a perl daemon |
| 16:24.25 | DyOS | Hey does anyone know where I can get an example of a dialplan that uses different messages for different hours? I have mine set up like that but in my default incoming dial plan i have my closed message after my open 8-5 message...so when my open message gets done it always goes to the next priority and plays the closed one |
| 16:24.25 | emacsen | Is there a way to do GW<->GW encryption with AIX or SIP? |
| 16:24.29 | Essobi | (to * management) |
| 16:24.30 | *** join/#asterisk slePP (~slepp@S01060040f48412ad.ed.shawcable.net) |
| 16:24.53 | jontow | AIX? ew |
| 16:25.12 | *** join/#asterisk boch (~as24@200.59.172.98) |
| 16:25.21 | emacsen | jontow: I meant IAX, sorry |
| 16:25.26 | eKo1 | didn't sgi abandon aix in favor of linux |
| 16:25.28 | afe | DyOS: There's a working example on www.voip-info.org (in the asterisk tips and tricks section) |
| 16:25.29 | jontow | ;) |
| 16:25.49 | eKo1 | oh wait, that's irix |
| 16:26.00 | eKo1 | aix is ibm right? |
| 16:26.18 | Darwin[laptop] | yes |
| 16:26.32 | Darwin[laptop] | AIX = IBM-Unix |
| 16:26.46 | DyOS | thanks afe |
| 16:26.59 | eKo1 | rephrasing: didn't ibm abandon aix in favor of linux |
| 16:27.04 | CosmicRay | hell no |
| 16:27.16 | CosmicRay | ibm will still sell aix to you |
| 16:27.21 | CosmicRay | despite the fact that it sucks |
| 16:27.24 | riksta | haha |
| 16:27.30 | riksta | doesn't it come with CDE |
| 16:27.33 | riksta | the most ugly thing ever |
| 16:27.35 | BruceX | they will still sell you OS/2 xD |
| 16:27.59 | CosmicRay | riksta: yeah but I have yet to figure out why. nobody buys aix for a workstation. |
| 16:27.59 | eKo1 | os/2?! NOO! |
| 16:28.00 | sivana | Essobi: I don't see it on wiki |
| 16:28.06 | riksta | CosmicRay: yeah |
| 16:28.19 | CosmicRay | riksta: it took me about 2 days to bootstrap a working C compiler in aix |
| 16:28.23 | Essobi | sivana opening page |
| 16:28.27 | eKo1 | what os do ibm's eServers use? |
| 16:28.29 | Darwin[laptop] | os/2 was a good os |
| 16:28.35 | Essobi | Called IPSwitchboard |
| 16:28.35 | BruceX | is nice |
| 16:28.38 | sivana | Essobi:ya |
| 16:28.39 | Darwin[laptop] | till os2-merlin |
| 16:28.41 | sivana | not the same thing I'm doing |
| 16:28.42 | BruceX | i got a copy some dayg ago :) |
| 16:28.43 | CosmicRay | eKo1: those are PCs, right? so it would be whatever pc os you like |
| 16:28.44 | Darwin[laptop] | then it died |
| 16:28.44 | riksta | it's nice for mark to have got a modest hot tub :) |
| 16:28.52 | Essobi | Ahh. That one is purty nice |
| 16:28.58 | sivana | I'm doing a complete customer/system mgmt |
| 16:29.01 | CosmicRay | or is that the xseries |
| 16:29.01 | CosmicRay | I forget |
| 16:29.04 | Essobi | Does UDP checksum generation occur in * or Linux kernel? Anyone know? |
| 16:29.06 | eKo1 | ok, what about high end clusters, like the parallel sysplex. |
| 16:29.31 | eKo1 | Essobi: kernel. |
| 16:29.32 | sivana | Essobi: what's the issue with that one? |
| 16:29.36 | lqdengr | i'm having a problem with my inbound iax connection to voicepulse timing out, is there a debug command that will show me if the connection is "active"? |
| 16:29.38 | CosmicRay | Essobi: udp checksum would always be in the kernel |
| 16:29.48 | Essobi | sivana You poor bastard.. I think you'll find that the management port isn't what you want to do that with.. |
| 16:29.51 | CosmicRay | Essobi: whether or not there is an additional checksum in the protocol that uses udp is a different question. |
| 16:29.56 | CosmicRay | Essobi: asterisk may do something too. |
| 16:30.04 | *** join/#asterisk habakuk (~chatzilla@24-117-8-113.cpe.cableone.net) |
| 16:30.06 | sivana | Essobi: I'm not talking about interfacing with * in that manner |
| 16:30.08 | Essobi | eKo1 Well i got an RTP checksum that's failing.. any ideas where to start? |
| 16:30.13 | BruceX | aaah |
| 16:30.17 | coppice | asterisk's reports about bad checksums are bogus |
| 16:30.24 | Essobi | Umm. |
| 16:30.28 | sivana | my system will generate configs and "push" it up to * then use the management to simply reload |
| 16:30.31 | Essobi | It's asterisk generating the bad checksum. |
| 16:30.46 | eKo1 | the checksums are due to network fucks ups i think. |
| 16:31.01 | eKo1 | Somewhere in the RTP stream, a bit got flipped. |
| 16:31.03 | BruceX | ny1 has any idea why i cant get atxfer works for the called party ?? (i tweak the code, and i can see the channel_bridge fucntion copying only frames from caller->called, not viceversa) .... |
| 16:31.05 | Essobi | console doesn't report it.. I have a one way DTMF problem, and that's why.. tethereal is showing me a packet that had a bad checksum so the opther end of my h323 is dropping it. |
| 16:31.28 | coppice | Essobi: * doesn't generate checksums. the system does. Asterisk often reports bad checksums, but it lies |
| 16:31.36 | Essobi | It's not reporting them. |
| 16:31.58 | Essobi | My tethereal sniffer says */linux is generating them. |
| 16:32.01 | CosmicRay | eKo1: not possible if it is udp |
| 16:32.05 | eKo1 | coppice: How do you know it is lying? |
| 16:32.54 | Beirdo | note to self: don't forget the admin password to your fucking ATA boxes! |
| 16:32.59 | coppice | if * gets an EAGAIN, and reports bad checksum. there are a number of reasons for getting EAGAIN. bad checksums are just one |
| 16:33.05 | Essobi | I see some options to disable RTP checksum generation.. MM. |
| 16:33.11 | BruceX | Beirdo , important note.. |
| 16:33.33 | eKo1 | Essobi: on *? |
| 16:33.37 | Essobi | Maybe my network driver is the culprit? What's the checksum based on? |
| 16:33.41 | Beirdo | I had to factory reset my SPA-2100 last night because, like a moron, I forgot the password |
| 16:33.41 | Essobi | eKo1 in rtp.c |
| 16:33.57 | Essobi | you can disable checksums for UDP from rtp.c |
| 16:34.08 | AgiNamu | Hey, in which file is the CDR uniqueid generated? I looked in cdr.c; doesn't seem to be there. |
| 16:34.33 | eKo1 | AgiNamu: Look at the app_dial.c. |
| 16:34.35 | sivana | AgiNamu: you still working on .NET/mono thingy |
| 16:34.50 | Essobi | AgiNamu grep -i uniqueid *.c | more |
| 16:35.07 | sivana | heh |
| 16:35.12 | Essobi | AgiNamu I see it in a few places.. it's scoped to the channels.. |
| 16:35.33 | DyOS | Can someone tell me what to search for on Wiki I want to be able to have my customers dial one ext and it will ring multiple cell phones and then the first one that answers will get that call is that possible? |
| 16:35.42 | Essobi | looks like pbx.c is generating the id |
| 16:35.49 | sudhir492 | anyone using TE110p card here? |
| 16:35.57 | Essobi | DyOs |
| 16:35.58 | AgiNamu | Oh yea, I see it now. |
| 16:36.01 | Essobi | DyOs app dial |
| 16:36.05 | *** join/#asterisk DevilFish (~me@staff211.qtm.net) |
| 16:36.13 | AgiNamu | It just copies the channel's uniqueid, which AFAIK, isn't globally unique. |
| 16:36.24 | AgiNamu | Is there any reason to copy the channel's ID? |
| 16:36.30 | AgiNamu | I'd like to fix it to use a GUID |
| 16:36.32 | sivana | you should modify the CDR writer to use an autoincrement or something |
| 16:36.37 | sivana | or that |
| 16:36.43 | Essobi | AgiNamu HUH? |
| 16:36.44 | sivana | do it on the DB end |
| 16:36.46 | AgiNamu | no, use a GUID |
| 16:36.47 | eKo1 | What's the point of having a uniqueid anyways? |
| 16:36.54 | DevilFish | I'm having trouble getting Asterisk to autologoff my queue members...anyone have any experience with this... my members are dynamically added if that helps. |
| 16:36.54 | Essobi | Uniqueip is a global ID. |
| 16:37.06 | AgiNamu | the point is to make sure we don't import the same record twice. |
| 16:37.14 | AgiNamu | for instance, in my system, CDRs are written realtime via SOAP |
| 16:37.24 | AgiNamu | BUT, if they fail |
| 16:37.27 | *** part/#asterisk emacsen (hidden-use@gw.coderyte.net) |
| 16:37.31 | AgiNamu | then it instead writes to XML |
| 16:37.34 | habakuk | Hi I have a business interested in some asterisk consulting. Can anyone give me some pointers on what the going rate is these days? |
| 16:37.36 | AgiNamu | on the local disk |
| 16:37.38 | sivana | AgiNamu: manage the unique id on the db side, which will help with scaling |
| 16:37.39 | Essobi | Dude.. You need to share that soap interface. ;) |
| 16:37.41 | AgiNamu | and my system will pick it up later. |
| 16:37.49 | AgiNamu | sivana, but the DB isn't always accessibile. |
| 16:37.54 | sivana | oh I see |
| 16:38.02 | AgiNamu | essobi: it's specific to my app, so I doubt it'd help anyone |
| 16:38.10 | Essobi | Why not generate them on insert time.. |
| 16:38.18 | sivana | couldn't two * boxes possibly generate the same GUID? |
| 16:38.20 | AgiNamu | because there is no insert if the SOAP calls fail |
| 16:38.31 | AgiNamu | not if linux's GUID library is worth shit :) |
| 16:38.35 | sivana | hehe |
| 16:38.53 | habakuk | AgiNamu, what soap libarary are you using? |
| 16:39.17 | eKo1 | AgiNamu: Maybe it would be better to ensure the availability of your SOAP writes. |
| 16:40.07 | AgiNamu | ASP.NET and gSOAP |
| 16:40.15 | AgiNamu | eKo1, yea, it'd be nice if I could count that nothing would ever fail |
| 16:40.17 | *** join/#asterisk tvmkid (~chatzilla@202.88.232.132) |
| 16:40.20 | AgiNamu | but that's not very robust |
| 16:41.11 | BruceX | 2 |
| 16:43.03 | habakuk | AgiNamu, so how is the gsoap library? Does it seem to work pretty well? I was thinking of using it for other purposes. i.e. send commands to the manger interface via soap |
| 16:43.20 | Damin | Hmm... |
| 16:43.21 | AgiNamu | gSOAP is an order of magnitude better than anything I thought could possibly exist for linux. |
| 16:43.27 | AgiNamu | It's still not .NET |
| 16:43.34 | AgiNamu | but gettign close to that (as far as client-ease) |
| 16:43.40 | habakuk | cool |
| 16:43.43 | AgiNamu | www.atrevido.net <-- I just wrote about the perf I got out of it |
| 16:43.45 | AgiNamu | gSOAP + ASP.NET |
| 16:43.58 | AgiNamu | 470rps with average response time of under 0.01 secs |
| 16:44.05 | AgiNamu | *off of a win XP development machine* |
| 16:44.10 | habakuk | cool! I'll check that out thanks |
| 16:44.15 | AgiNamu | and a cheap linux box, running X at the same time :P |
| 16:44.50 | heison | any dialplan experts here? |
| 16:44.59 | bjohnson | no |
| 16:45.01 | sivana | no |
| 16:45.09 | Nugget | do you really need an expert, or just someone who knows the answer to your question? |
| 16:45.10 | bjohnson | heison: just you |
| 16:45.29 | heison | if i have exten=>s,1,Wait(1) |
| 16:45.32 | eKo1 | You say you need a uniqueid so that you don't have import the same record twice. Why would it do that? |
| 16:45.45 | heison | exten=>s,2,AGI(callerID.php) |
| 16:45.50 | heison | exten=>s,3,Answer |
| 16:45.55 | sivana | AgiNamu: if the db is down, it writes it to an xml? |
| 16:46.04 | Damin | Alright.. |
| 16:46.10 | Damin | Question for the developers.. |
| 16:46.25 | heison | Asterisk doesn't answer the line until callerID.php finish executing |
| 16:46.30 | eKo1 | Damin: go to #asterisk-dev |
| 16:46.32 | heison | is there a way to do this in the background? |
| 16:46.46 | heison | would system() do any different? |
| 16:46.52 | AgiNamu | sivana, yea, if the SOAP call fails for any reason, then it writes to xml |
| 16:46.53 | heison | s/do/be/ |
| 16:47.00 | AgiNamu | and my system will swing by and grab that XML every 5 minutes |
| 16:47.06 | lqdengr | do iax peers always show up in the peers list, or only when a cal is active? |
| 16:47.11 | heison | bjohnson, sivana? |
| 16:47.13 | sivana | i c |
| 16:47.23 | sivana | heison: I don't know the answer, never used AGI |
| 16:47.36 | *** join/#asterisk gmcinnes (~gmcinnes@67.71.63.9) |
| 16:47.47 | sivana | heison: does it help by putting it in a macro? |
| 16:48.06 | julianjm | heison: what does that agi script do? |
| 16:48.21 | eKo1 | heison: you can't execute two commands concurrently. |
| 16:48.26 | Nugget | the proper approach would be to decouple the dialplan from whatever activity callerid is doing that takes time. |
| 16:48.32 | heison | julianjm: connect to jabber and display callerID |
| 16:48.42 | Nugget | what's callerid.php do? |
| 16:48.54 | DevilFish | heison, couldn't you just use System() to call a script the breaks off in the background on its own |
| 16:49.07 | julianjm | heison: and does it take that long? |
| 16:49.14 | heison | Nugget: it connects to jabber and display callerID |
| 16:49.15 | DevilFish | like a perl script that breaks off and then executes the php for you |
| 16:49.32 | heison | julianjm: yes, it took almost one extra ring |
| 16:49.52 | heison | callerid.php takes >2 seconds |
| 16:49.59 | Nugget | that sounds excessive |
| 16:50.15 | Nugget | I also suggest using system with creative use of "&" somewhere |
| 16:50.25 | Chuji | Are their other causes of jitter than just latency? |
| 16:50.26 | Nugget | but there's absolutely no reason why a jabber message should take 2 seconds |
| 16:50.34 | AgiNamu | HAHA! And if we use UUID/GUIDs instead of this Asteirsk-specific id, then we save 16bytes of memory on each CDR!~ |
| 16:50.39 | Chuji | I have pretty bad one way jitter on a couple of our sipuras |
| 16:50.49 | heison | Nugget: does that work? system(callerID.php&)?? |
| 16:50.50 | *** join/#asterisk brc-tux (~brc-tux@pD9E9A25F.dip0.t-ipconnect.de) |
| 16:50.56 | Nugget | I have no idea |
| 16:51.03 | Nugget | try it and let us know |
| 16:51.11 | antifuchs | AgiNamu: funny, I thought you were talking about lisp's CONS cells (-: |
| 16:51.15 | Nugget | at a minimum you can system(somescript) where somescript does the fork |
| 16:51.20 | gmcinnes | hi all. can someone explain how to concatenate multiple voice files? |
| 16:51.44 | Nugget | gmcinnes: first explain why you would want to |
| 16:51.49 | heison | Nugget: but there is always a potential for any other apps to take more time than a few hundred milliseconds and affecting the diaplan |
| 16:51.50 | AgiNamu | antufuchs ? |
| 16:51.57 | AgiNamu | LOL, read the uuid_generate docs |
| 16:51.59 | gmcinnes | I want to play many sequentially together and get a single string of digits representing the users response |
| 16:52.10 | Nugget | heison: yes, which is why my original suggestion is to decouple the process. |
| 16:52.21 | AgiNamu | some genius dev decided that uuid_generate should just return a random number intead of a GUID, because GUIDs can be traced. dumbass. |
| 16:52.33 | Nugget | but -- that said -- if your jabber thing is taking >2 seconds, there's a problem somewhere. that's insane |
| 16:52.45 | antifuchs | AgiNamu: you were talking about CDRs... memory efficiency of cons cells was once a great concern for lisp implementations (-: |
| 16:52.51 | antifuchs | (now, it's mostly a solved problem) |
| 16:53.00 | gmcinnes | Nugget: something like "Your test result from"+"January"+"20th"+"2004"+"is available. Push 1 to hear it" |
| 16:53.02 | antifuchs | AgiNamu: bwaha |
| 16:53.10 | heison | Nugget: that's because i'm not connecting to a local jabber server |
| 16:53.11 | gmcinnes | Nugget: Make sense? |
| 16:53.11 | Nugget | gmcinnes: you do not want to concatenate the files. |
| 16:53.14 | Nugget | play them in series |
| 16:53.24 | Nugget | exactly like the sample dialplan demonstrates |
| 16:53.24 | heison | Nugget: the roundtrip time is over 300ms |
| 16:53.31 | bjohnson | heison: hell if I know .. you're the expert |
| 16:53.31 | Nugget | heison: nutty |
| 16:53.42 | *** part/#asterisk brc-tux (~brc-tux@pD9E9A25F.dip0.t-ipconnect.de) |
| 16:54.49 | julianjm | Chuji: could it be related to Silence Suppresion? |
| 16:54.49 | julianjm | gmcinnes, what kind of files? if they are gsm... cat file1.gsm file2.gsm > filemix.gsm |
| 16:54.50 | jontow | so use Background() to play each individually? |
| 16:54.50 | `Sauron | Nugglet |
| 16:54.50 | Beirdo | hmmm |
| 16:55.06 | Beirdo | too bad I won't be able to get the employer to send me to VON Canada :) |
| 16:55.15 | Beirdo | I think I'll go for exhibits though |
| 16:55.28 | gmcinnes | Nugget: Did I miss that? My problem is the respose could be multi-digit. Each get_data call could get one of the digits. so if the user wants to enter "123" each get_data call could get part of the entry, no? |
| 16:55.44 | Nugget | you've lost me |
| 16:56.00 | gmcinnes | julianjm: have you tried that? will it work? is there any header info on gsm files that might trip it up? |
| 16:56.01 | Nugget | in any event, it's not customary to cat files for playback. play them in series. |
| 16:56.37 | Nugget | I don't know what get_data does in the context of what you've told us |
| 16:56.44 | Juggie | Beirdo, my employer is sending me, where do u work |
| 16:56.55 | Beirdo | in Toronto :) |
| 16:56.56 | Beirdo | heh |
| 16:57.10 | gmcinnes | Nugget: Ok. Thanks for your help. I hope this is a theoretical problem and not a real one. |
| 16:57.15 | Juggie | duh, what company or is it gov? |
| 16:57.24 | Beirdo | I'm a UNIX Sysadmin, so they won't pay for me to go when other people are doing the VoIP stuff |
| 16:57.34 | Beirdo | MCI Canada |
| 16:57.35 | Nugget | I strenously suggest that you stop pursuing a solution which involves modifying files. you'll regret it later |
| 16:57.51 | Beirdo | hey WTF? |
| 16:57.56 | Beirdo | we have a booth?! |
| 16:58.25 | bjohnson | Beirdo: hide in the display crates .. you'll get there (eventually) |
| 16:58.30 | Beirdo | hehe |
| 16:58.49 | Beirdo | well, that won't get me into the conference part though |
| 16:58.55 | heison | Beirdo: on Adelaide? |
| 16:59.01 | Beirdo | yes |
| 16:59.03 | gmcinnes | Nugget: :) Ok. I'm not going to modify files. |
| 16:59.29 | heison | Beirdo: same here |
| 16:59.30 | Nugget | yay :) |
| 16:59.49 | DevilFish | I'm having trouble getting Asterisk to autologoff my queue members...anyone have any experience with this... my members are dynamically added if that helps. |
| 17:00.15 | Beirdo | I should hunt down the sales dweebs, see if they can at least get me free admission to the exhibits :) |
| 17:02.12 | BruceX | ny1 has any idea why i cant get atxfer works for the called party ?? (i tweak the code, and i can see the channel_bridge fucntion copying only frames from caller->called, not viceversa) .... |
| 17:03.35 | Alexis | bye |
| 17:03.37 | *** part/#asterisk Alexis (~alexis@www.trim.it) |
| 17:03.58 | *** join/#asterisk husher (~andrew@68.143.92.130.nw.nuvox.net) |
| 17:04.24 | *** join/#asterisk A-Tuin|work (~A-Tuin@nat.office.legend.net.uk) |
| 17:04.25 | *** join/#asterisk outtolunc (~me@adsl-66-218-53-170.dslextreme.com) |
| 17:07.46 | hardwire | ugh |
| 17:07.53 | AgiNamu | uh, what's hte way to format a char* as hex? |
| 17:08.02 | AgiNamu | supposing i have a char[16] of random data |
| 17:08.04 | *** join/#asterisk Jer13261 (~Jer@rdu57-251-152.nc.rr.com) |
| 17:10.41 | *** join/#asterisk _THEEND_ (~DrEaM@80.18.184.226) |
| 17:11.14 | _THEEND_ | someone uses voicemail on cisco phones? |
| 17:11.36 | Nugget | sure |
| 17:11.37 | husher | _THEEND_: yes |
| 17:12.14 | _THEEND_ | i have a problem |
| 17:12.19 | Beirdo | AgiNamu: printf( "%02X", blah[0] ) |
| 17:12.21 | _THEEND_ | when i call |
| 17:12.22 | husher | ok |
| 17:12.25 | *** join/#asterisk w0p (~DJ-Pyro@firewall.brevient.net) |
| 17:12.26 | _THEEND_ | the voicemail |
| 17:12.32 | Beirdo | for each char (in a loop would be neatest) |
| 17:12.38 | Nugget | you also seem to have a problem with your enter key. |
| 17:12.39 | _THEEND_ | for look at new messages |
| 17:13.05 | Jer13261 | anyone know why * could core dump when it fails to neg a codec with a sip device? |
| 17:13.08 | _THEEND_ | it does't recognize number pressed on the phone |
| 17:13.11 | Jer13261 | would* |
| 17:13.26 | _THEEND_ | someone knows why? |
| 17:13.42 | Jer13261 | lol..... |
| 17:14.57 | _THEEND_ | .... |
| 17:15.39 | *** part/#asterisk j_berry (~jberry@acs-24-154-150-174.zoominternet.net) |
| 17:15.51 | husher | _THEEND_: it's probably in your dtmf settings; what are your phones set to? |
| 17:16.31 | *** join/#asterisk planet_guru (~chris@office.enovi.com) |
| 17:17.19 | _THEEND_ | dtmfmode=inband |
| 17:17.22 | _THEEND_ | in sip.conf |
| 17:18.21 | husher | try dtmfmode=rfc2833 in sip.conf |
| 17:18.29 | bkw_ | also put an answer in there |
| 17:18.32 | bkw_ | and a wait 1 |
| 17:18.34 | planet_guru | I can't detect an inboud call on an E1 here - the caller gets a 'number not recognised' message and nothing shows in the asterisk console when I dial the number. I have a line like extensions.conf:exten => 0800#######,1,Goto(extensions,3005,1) in my extensions.conf. What else might be missing? |
| 17:18.36 | bkw_ | before voicemail just in case |
| 17:18.56 | husher | and dtmf_inband: 0 and dtmf_outofband: avt_always in SIPDefault.cfg |
| 17:19.42 | *** join/#asterisk Grooby (~Grooby@66.160.105.186) |
| 17:19.57 | _THEEND_ | ok |
| 17:19.58 | _THEEND_ | :D |
| 17:20.00 | _THEEND_ | now it works |
| 17:20.19 | husher | planet_guru: i think you need an 's' in front of the number in extensions.conf |
| 17:21.25 | *** join/#asterisk jdb1968 (~jdb1968@S010600045af29653.cg.shawcable.net) |
| 17:21.28 | jontow | try _0800XXXXXXX,1,Goto.... |
| 17:21.37 | Nugget | yeah, |
| 17:21.41 | planet_guru | husher - like exten => s0800#######,1,Goto ... bla bla ? |
| 17:21.43 | Nugget | "#" isn't doing what you think it is. |
| 17:21.45 | planet_guru | aaaah |
| 17:21.49 | tzanger | ?? whta's s0800?? |
| 17:22.06 | jdb1968 | hi all, what happened to chanspy? wiki says it was pulled 19 oct 2004 with no explanation |
| 17:22.22 | husher | planet_guru: i think, but try a _ like jontow said too |
| 17:22.27 | jdb1968 | link to bug 2379 on wiki page says bug doesn't exist (2379) |
| 17:24.36 | planet_guru | husher,jontow - I tried exten => _08001425bla,1,Goto(extensions,3005,1) but still nothing at all when the number is dialled |
| 17:25.36 | *** join/#asterisk salviadud (~dude@201.129.86.185) |
| 17:26.03 | Nugget | put in a logging line for the incoming context you're using for the e1 |
| 17:26.11 | Nugget | see what ${EXTEN} is being sent |
| 17:26.11 | heison | ~seen JerJer |
| 17:26.13 | jbot | jerjer <~JerJer@dsl-106-170.che.centurytel.net> was last seen on IRC in channel #asterisk, 18d 1h 7m 18s ago, saying: 'mrgoby: sure'. |
| 17:26.18 | salviadud | does anyone have a sipura 3000? |
| 17:26.18 | *** join/#asterisk Shoragan (~shoragan@d072.apm.etc.tu-bs.de) |
| 17:26.20 | w0p | when I do a "zap show channel 1" it shows echo cancellation off, can I turn that on? |
| 17:26.30 | w0p | this is for a T110P card |
| 17:26.40 | Nugget | [e1context] |
| 17:26.45 | w0p | there seems to be a little echo on the t1 channels coming in |
| 17:26.58 | Nugget | exten => _.,1,NoOp(Incoming call on E1 from ${CALLERID} for ${EXTEN} |
| 17:27.04 | Nugget | ^ something like that, for debugging |
| 17:27.24 | w0p | I have echocancelwhenbridged turned on |
| 17:27.27 | Nugget | whatever you see for ${EXTEN} will show you what the exten needs to resemble |
| 17:27.28 | planet_guru | Nugget - let me try that |
| 17:27.48 | husher | w0p: put echocancel=yes in your zapata.conf |
| 17:28.07 | w0p | also have that |
| 17:28.26 | w0p | echocancel=yes |
| 17:28.26 | w0p | echocancelwhenbridged=yes |
| 17:28.42 | BruceX | ny1 has any idea why i cant get atxfer works for the called party ?? (i tweak the code, and i can see the channel_bridge fucntion copying only frames from caller->called, not viceversa) .... |
| 17:28.48 | *** join/#asterisk t3t (~t3t@cust018.mke.attron.net) |
| 17:30.36 | planet_guru | Nugget I have added: exten => s,2,NoOp,${CALLERID} to extensions.conf, run asterisk -rvvvvvvv, reload, dial number.. and still nothing at all |
| 17:30.44 | Nugget | that's not what I suggested. |
| 17:30.52 | Nugget | so I'm not surprised it didn't work. :) |
| 17:31.16 | husher | w0p: try checking the channel when it's in use |
| 17:31.26 | planet_guru | Nugget - ummkay - I'm not understanding I guess |
| 17:31.36 | Nugget | well, you changed several things from what I typed. |
| 17:31.37 | Nugget | why? |
| 17:31.55 | Nugget | you changed it from _. to "s" and you put it at priority 2 instead of 1. |
| 17:32.46 | planet_guru | Nugget - sorry - that was a typo I see what you mean.. yes I do get a message. |
| 17:33.31 | Nugget | so what's the message? |
| 17:33.45 | *** join/#asterisk Tall-guy (tall-guy@hssxrg207-195-103-110.sasknet.sk.ca) |
| 17:33.47 | Zeeek | 42 |
| 17:33.47 | planet_guru | well... |
| 17:34.51 | planet_guru | Nugget - sry, that message I guess was nothing to do with my call - because when I re-try I get nothing. I guess it was just a conicidence that it popped up at the same time |
| 17:35.23 | Jer13261 | this is strange the second i put allow=ulaw in my sip.conf it crashes * when i call it |
| 17:35.33 | AgiNamu | I've got a char[16]. I want ot turn it into a char[32] hex string |
| 17:35.35 | planet_guru | iax2_poke_noanswer: Peer 'magrathea' is now UNREACHABLE! - altho I'm not doing anything which uses magrathea |
| 17:35.43 | planet_guru | hmm |
| 17:35.59 | Nugget | use pastebin.ca to paste the whole context. |
| 17:36.12 | *** join/#asterisk Gh0sty (~Ghosty@ip-81-11-230-198.dsl.scarlet.be) |
| 17:36.14 | Jer13261 | me? |
| 17:36.19 | Nugget | no, planet_guru. |
| 17:36.29 | planet_guru | Nugget.. hmmmm |
| 17:36.39 | Nugget | there's no way for anyone here to help you unless you can be accurate in telling us what is in your dialplan |
| 17:36.55 | Nugget | also, you do understand the role of contexts in this, right? |
| 17:36.58 | Nugget | that's not the problem? |
| 17:37.56 | planet_guru | Nugget - okay, well I have a line in my [default] context, which reads exten => _,1,NoOp,${CALLERID} as you suggested |
| 17:38.03 | Nugget | no. |
| 17:38.06 | Nugget | not your default context. |
| 17:38.12 | Nugget | the context for your e1 |
| 17:38.23 | Nugget | and that ISN'T what I suggested. |
| 17:38.28 | Nugget | you changed it from "_." to "_" |
| 17:38.32 | tvmkid | hi, i am looking for a simple example sip.conf and extension.conf connecting to FWD for sending and receiving calls to a asterisk extension, for some reason, the samples at voip-info is not working (just a newbie here, trying to learn how * works) |
| 17:38.36 | Nugget | and you removed the ${EXTEN} part. |
| 17:38.46 | Nugget | do you realize how difficult you are making it to help you? |
| 17:39.15 | mikegrb | Nugget: <3 |
| 17:39.18 | planet_guru | Nugget - sorry. I've extremely tired. I guess I should come back tomorrow when I've slept and eaten. :/ |
| 17:39.23 | Nugget | yes, I suggest that. |
| 17:39.24 | mikegrb | Nugget: he is telling you he doesn't need help |
| 17:40.09 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 17:40.10 | *** join/#asterisk jaiger (~jaiger@fire.innovationsw.com) |
| 17:40.29 | *** join/#asterisk ManxPower (~eric@ip-209-16-83-10.i-55.com) |
| 17:41.05 | *** join/#asterisk Los415 (~los415@c-24-126-63-233.we.client2.attbi.com) |
| 17:41.18 | jaiger | hey all |
| 17:41.19 | *** part/#asterisk Darwin[laptop] (~darwin-la@c-24-3-226-147.client.comcast.net) |
| 17:41.27 | jaiger | has anyone used the cuphone with asterisk? |
| 17:41.27 | Los415 | hey guys has anyone had any issues running * on rh enterprise 4 |
| 17:41.42 | Nugget | I know that zaptel had troubles building in FC3. |
| 17:41.47 | Jer13261 | darwin yo there? |
| 17:41.57 | Jer13261 | duh |
| 17:42.15 | Nugget | as far as asterisk is concerned, though, unix is unix is unix. |
| 17:42.20 | Nugget | even linux. |
| 17:42.27 | Bentley | hey all, what happened to chan_spy? I was just about to implement it, and now i see it's been pulled |
| 17:42.32 | Los415 | so zaptel is the one to be worried about |
| 17:42.39 | Los415 | how about in rh enterprise 3 |
| 17:42.46 | Nugget | zaptel is just picky about kernel sources on the machine matching the running kernel. |
| 17:42.58 | Los415 | ahh oj |
| 17:42.59 | Los415 | err ok |
| 17:43.01 | Nugget | and there were some redhatisms in FC3 that made the kernel different enough to confuse zaptel |
| 17:43.56 | bannerman | hm.. I've got a Sipura SPA-1001 thingy, and when I call that extension, Asterisk seems to think it's ringing.. but it doesn't make my analog phone ring |
| 17:44.10 | sivana | too bad zaptel is weak |
| 17:44.20 | sivana | wish Digium would make some real robust drivers |
| 17:49.32 | Nugget | now that the sipura doesn't have that weird double-dialing artifact I'd do an spa-3000 instead. |
| 17:49.37 | Nugget | going sipura lets you avoid zaptel and also means you're not stuck with linux. it's a double win |
| 17:49.43 | t3t | bannerman: I assume that you are using a TDMxx card; do you have the power connector attached to the card? |
| 17:49.53 | Nugget | platform independence and more reliable |
| 17:50.33 | bannerman | t3t: no idea what a TDMxx card is, actually |
| 17:50.47 | bannerman | t3t: I've just got a Sipura SPA-1001 connecting my analog phone to my network |
| 17:50.52 | bannerman | I can dial out fine with taht phone |
| 17:51.10 | bannerman | but when I dial that extension, Asterisk thinks its ringing, but my phone doesn't get a ring signal |
| 17:51.16 | t3t | bannerman: sorry, I combined 2 unrealted conversations :) |
| 17:51.28 | bannerman | t3t: I'll forgive you if you can help me! :-P |
| 17:51.42 | t3t | bannerman: is the 1001 behind a firewall/NAT? |
| 17:51.55 | bannerman | no |
| 17:52.05 | bannerman | it is, but it's on the same subnet as my asterisk box |
| 17:52.14 | t3t | it's on the same ethernet segment as the * box? |
| 17:52.26 | bannerman | yeah |
| 17:52.56 | bannerman | as far as I can tell everything looks normal on my debug log and things, when it's ringing |
| 17:52.58 | bannerman | but no ring tone |
| 17:53.18 | bannerman | and it doesn't pick up the call, when I do pick up the handset |
| 17:53.29 | *** join/#asterisk ManxPower (~eric@ip-209-16-83-10.i-55.com) |
| 17:53.46 | t3t | If the 1001 is accepting the call, maybe the ringer is off on the phone or the 1001 is set to not ring the phone |
| 17:53.59 | *** join/#asterisk Dandan (dandan@234.88.149.195.in-addr.arpa.virt-ix.net) |
| 17:54.02 | Dandan | re |
| 17:54.05 | *** join/#asterisk marc324 (~marc32344@69-90-241-15.dsl.teksavvy.com) |
| 17:54.14 | t3t | ok... doesn't pick up the call when you pick up the handset means that it's not a no-ringing issue... |
| 17:54.14 | bannerman | the 1001 I'm guessing is not set up to ring the phone |
| 17:54.22 | bannerman | or connect the call at all :-P |
| 17:54.26 | Dandan | Mar 17 12:49:13 WARNING[1706]: app_queue.c:375 changethread: Can't change device '**Unknown**' with no technology! |
| 17:54.26 | Dandan | Mar 17 12:49:13 WARNING[1707]: app_queue.c:375 changethread: Can't change device '**Unknown**' with no technology! |
| 17:54.34 | Dandan | what is that when leaving a voicemail? |
| 17:54.38 | Dandan | anyone can help me? |
| 17:54.45 | hardwire | I am sure anyone could. |
| 17:54.53 | Dandan | can anyone help me pls? |
| 17:55.19 | hardwire | thats one freakish looking error. |
| 17:55.34 | Dandan | yeah :/ |
| 17:56.17 | hardwire | no idea what it means |
| 17:56.32 | hardwire | but usually technology refers to the codec right? |
| 17:56.40 | hardwire | or maybe you have no voicemail format set up |
| 17:56.46 | hardwire | like wav49|gsm|blah |
| 17:56.48 | AgiNamu | Wow, easiest patch I've ever done :) |
| 17:56.56 | hardwire | AgiNamu: whats it dooooo?! |
| 17:57.04 | mstocco | Dandon: my guess is you have a caller in a queue trying to transfer to a bad exten |
| 17:57.07 | AgiNamu | Adds a GUID to the CDRs. |
| 17:57.12 | AgiNamu | instead of a "unique" id. |
| 17:57.24 | hardwire | oooooh |
| 17:57.32 | hardwire | Group ID I take it? |
| 17:57.33 | t3t | bannerman: sorry, i haven't used the 1001. I'm not sure what could be causing your issues.. anyone else? |
| 17:57.39 | AgiNamu | groupd ID? |
| 17:57.46 | AgiNamu | no, Globally Unique ID |
| 17:57.48 | hardwire | GUID? |
| 17:57.53 | hardwire | HAHHALLEOOGHOOOOOH |
| 17:57.54 | hardwire | !! |
| 17:58.00 | hardwire | sorry.. having issues |
| 17:58.12 | hardwire | my girlfriend wants me to be worried about things that worry her. |
| 17:58.17 | hardwire | so I am worried about her. |
| 17:58.27 | mstocco | Dandan: technology being SIP Zap IAX; device being the extension |
| 17:58.28 | hardwire | but she isn't worried about herself.. so that doesn't work out to my benifet. |
| 17:59.13 | *** part/#asterisk tzanger (~tzanger@165.154.13.35) |
| 17:59.16 | salviadud | whats the difference between FXO and FXS? |
| 17:59.18 | *** join/#asterisk tzanger (~tzanger@165.154.13.35) |
| 17:59.30 | Dandan | mstocco: hmmm |
| 17:59.39 | t3t | ~fxo |
| 17:59.40 | 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 |
| 17:59.43 | Dandan | anyway I can fix that? |
| 17:59.43 | t3t | ~fxs |
| 17:59.44 | jbot | extra, extra, read all about it, fxs is foreign exchange system - or the type of port you need to connect a analog device (phone, fax machine) to a pbx |
| 17:59.54 | hardwire | ~lart |
| 18:00.07 | hardwire | hehe |
| 18:00.09 | hardwire | ~lart hardwire |
| 18:00.12 | Dandan | ~stfu |
| 18:00.14 | jbot | hmm... stfu is Shut the F*** Up!, or http://www.linuks.mine.nu/stfu-noob.jpg |
| 18:00.14 | Dandan | :) |
| 18:00.17 | *** join/#asterisk Darwin[laptop] (~darwin-la@c-24-3-226-147.client.comcast.net) |
| 18:00.20 | salviadud | yeah, i am a newbie... |
| 18:00.22 | hardwire | heh |
| 18:00.26 | hardwire | salviadud: voip-info.org |
| 18:00.36 | salviadud | hey, but im serious! |
| 18:00.45 | hardwire | HAH |
| 18:00.46 | salviadud | i got this really big dilemma for hardware here |
| 18:00.47 | hardwire | ok. |
| 18:00.48 | mstocco | Dandan: you could disallow the transfer sure |
| 18:00.59 | salviadud | should i buy a sipura 3000 |
| 18:01.01 | hardwire | salviadud: didn't your question get answered? |
| 18:01.05 | hardwire | salviadud: yes.. |
| 18:01.08 | hardwire | I will even sell them to you :) |
| 18:01.11 | t3t | salviadud: nothing wrong with being a newbie... as long as you are willing to learn on your own before you ask for help |
| 18:01.40 | hardwire | salviadud: but be afraid.. be very afraid.. configuring an SPA-xx00 is an amazing adventure. |
| 18:01.56 | t3t | salviadud: fxo/fxs is a good question. Google is often the simplest way to answer these types of definition questions though |
| 18:02.06 | hardwire | its worthile.. and you feel like the grand poobah of VoIP after correctly configuring one. |
| 18:02.09 | salviadud | im the adventurous type |
| 18:02.26 | salviadud | i've never used linux, and the first distro i've tried. slackware |
| 18:02.29 | hardwire | I have had to resist the urge to flood this channel with questions about the ones I have configured |
| 18:02.38 | hardwire | thankfully.. the wiki is my saviour |
| 18:02.38 | mstocco | Dandan: what does your exten for that queue look like? exten => x,x,Queue(queue_name|options) |
| 18:02.40 | hardwire | and I bow.. |
| 18:02.42 | Dandan | slav: the best one |
| 18:02.43 | hardwire | *bow* |
| 18:02.45 | hardwire | all of you!!! |
| 18:02.51 | Dandan | salviadud: the best distro :) |
| 18:03.15 | Dandan | mstocco: i have no queues |
| 18:03.20 | Dandan | it is 2 extensions |
| 18:04.01 | Dandan | exten => 311,1,SetCallerId,"Daniel Nowacki" |
| 18:04.01 | Dandan | exten => 311,2,Dial(SIP/311,20) |
| 18:04.02 | Dandan | exten => 311,3,Voicemail(u311) |
| 18:04.02 | Dandan | exten => 311,103,Voicemail(b311) |
| 18:04.02 | Dandan | exten => 311,104,Hangup |
| 18:04.08 | Sedorox | ~pastebin |
| 18:04.09 | jbot | rumour has it, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 18:04.24 | mstocco | Dandan: Mar 17 12:49:13 WARNING[1706]: app_queue.c:375 changethread: Can't change device '**Unknown**' with no technology! |
| 18:04.42 | Dandan | Sedorox: 5 lines :) |
| 18:04.44 | mstocco | Dandan: we all see app_queue in your error msg |
| 18:04.53 | Dandan | mstocco: hold on... |
| 18:05.01 | Dandan | will pastebin the whole thing... |
| 18:05.27 | salviadud | so here's my question in a nutshell, im a first time user of asterisk, and the hardware i would need for this new adventure should it be? a sipura 3000, or just your IAXY thingy? |
| 18:06.40 | Nugget | those two devices do different things. |
| 18:06.44 | Nugget | the iaxy has no fxo |
| 18:07.06 | rvhi | is there a way to comment a block of code in extension.conf? |
| 18:07.09 | salviadud | damn, the answer is simple... i want to get a good deal on a sipura 3000 |
| 18:07.12 | rvhi | like /* */ in c? |
| 18:09.05 | AgiNamu | sheesh, 3 lines of code is all it was |
| 18:09.13 | Dandan | mstocco: http://pastebin.ca/7651. |
| 18:09.13 | AgiNamu | Was it that hard to use a UUID from the start? :) |
| 18:09.15 | Dandan | mstocco: http://pastebin.ca/7651 |
| 18:09.58 | *** join/#asterisk Qwell (~north@70-32-102-18.ontrca.adelphia.net) |
| 18:10.57 | *** join/#asterisk postel (~canonical@postel.user) |
| 18:11.33 | Hmmhesays | heh ser is a handful when you first look at it |
| 18:13.32 | AgiNamu | does anyone have a test server to try some load testing? |
| 18:13.43 | AgiNamu | I want to see how asterisk reacts to having 1000s of IAX2 registrations |
| 18:13.51 | AgiNamu | I have 2 machines, but one only has a 512k link... |
| 18:14.13 | *** join/#asterisk [cc]smart (~smart@gw.ptr-62-65-149-158.customer.ch.netstream.com) |
| 18:14.44 | jontow | i've got a bunch of test servers, but the bandwidth here is .. ass |
| 18:15.15 | AgiNamu | jontow, but between them? |
| 18:15.26 | AgiNamu | I can just send you the .conf files |
| 18:15.28 | anthm | rvhi. ;-- ... --; <- brought to you by cluecon |
| 18:15.30 | AgiNamu | and you could run it and see |
| 18:16.17 | AgiNamu | oh, you just need to patch the iax loading code, since it's shit |
| 18:16.18 | jontow | heheh |
| 18:16.38 | AgiNamu | just remove 8 lines of code, and all of a sudden it goes from N*N to N :) |
| 18:16.43 | AgiNamu | 100,000 friends load in 4 seconds |
| 18:17.03 | jontow | hah |
| 18:17.22 | *** join/#asterisk afe ([buQmIFRk+@c-e616e055.123-1-64736c12.cust.bredbandsbolaget.se) |
| 18:17.38 | AgiNamu | assuming you dont use multiple peers/users with the same [category] |
| 18:19.04 | jaiger | tzanger, hows the echo canceler working? |
| 18:19.14 | tzanger | jaiger: the software echo can just started working one day |
| 18:19.17 | tzanger | haven't tried the hardware one yet |
| 18:19.43 | jaiger | I was wondering about the hardware |
| 18:19.47 | tzanger | I know :-) |
| 18:20.38 | jaiger | has anyone used any of the tjnet USB devices with *? |
| 18:20.47 | jontow | aginamu; tar those configs up and put 'em on the web :) ill grab 'em and give it a shot |
| 18:21.08 | jontow | i've got a 1ghz gentoo machine as a client and a 2.8ghz gentoo machine as a server.. on a 10/100 LAN |
| 18:21.40 | jaiger | jontow, there's your problem - not enough horsepower |
| 18:21.43 | *** join/#asterisk Dr-Linux (Dreamer@202.147.168.142) |
| 18:21.48 | jontow | huh? :P |
| 18:21.50 | AgiNamu | jon, ok |
| 18:21.53 | jontow | the 2.8ghz xeon is fine |
| 18:21.55 | jontow | the 1ghz is just a crashbox |
| 18:22.06 | jontow | im not having any problems :) |
| 18:22.09 | jaiger | jontow, just kidding - I've got * running on a 200Mhz pentium |
| 18:22.23 | jontow | hehe |
| 18:22.41 | AgiNamu | yea, well, with 10,000 registrations, CPU is definately the bottleeck ;P |
| 18:22.48 | AgiNamu | ok, one sec. |
| 18:22.52 | *** join/#asterisk MattH (~matth@noc-wireless.chilitech.net) |
| 18:23.23 | MattH | Hi.. I have 2 sipura phones running on asterisk... they both succesfully GET message waiting notifications and turn on the MWI and stutter-tone... but they aren't clearing after the messages are deleted... what am I missing? |
| 18:23.27 | Dr-Linux | does asterisk support Dialogic card ? |
| 18:24.00 | Mike | anyone here working with chan_h323? |
| 18:24.08 | *** join/#asterisk Qwell (~north@70-32-102-18.ontrca.adelphia.net) |
| 18:25.09 | wmoran | Heh ... finally got my echo problems fixed ... |
| 18:26.17 | AgiNamu | www.telefinity.com/test_iax.zip 25,000 IAX friends + 25,000 IAX register=> lines |
| 18:26.21 | AgiNamu | #include em :) |
| 18:26.30 | AgiNamu | and oh, lemme get you a patch so it doesnt take a minute to load them |
| 18:26.49 | Qwell | It'll take longer to patch, compile, and start, then it will to load them :p |
| 18:27.00 | AgiNamu | Oh, yea. |
| 18:27.08 | AgiNamu | unless you reload once :P |
| 18:27.25 | AgiNamu | Oh, and a suggestion, in the text_iax_reg.conf, goto line 1000 and put [ignore] , so you only test 1000 |
| 18:27.30 | AgiNamu | then move it up to see where the limit it |
| 18:29.32 | AgiNamu | My *guess* is that the perf sucks because the access check is inside a mutex AND it uses a linked list. |
| 18:29.32 | *** join/#asterisk Juxt (~Juxt@adsl-068-213-216-087.sip.bct.bellsouth.net) |
| 18:29.38 | Juxt | good afternoon |
| 18:30.06 | AgiNamu | no wait, hm |
| 18:30.15 | tclark | jaiger: the tjnet usb devices id the digium s100u :) |
| 18:30.27 | bannerman | hm my X-lite is echoing to the remote caller |
| 18:30.46 | wmoran | Anyone having echo problems, read and follow this: http://lists.digium.com/pipermail/asterisk-users/2004-November/071301.html |
| 18:31.03 | wmoran | If your rxgain and txgain aren't correct, you'll never get rid of your echo! |
| 18:31.24 | AgiNamu | yea, i think im correct |
| 18:31.31 | AgiNamu | it traverses a linked list EACH damn time |
| 18:31.47 | AgiNamu | so with 25,000 peers.... all of a sudden, that's a lot of work |
| 18:32.08 | AgiNamu | so essentially, handling registrations is an O(N*N) problem too |
| 18:32.31 | AgiNamu | well, R*N where R=registering pers and N=total peers |
| 18:32.42 | Juxt | i have the following question |
| 18:32.52 | AgiNamu | fire |
| 18:32.58 | Juxt | i'd like to have my * dial 2 numbers and have them talk to each other |
| 18:33.04 | Juxt | would that require a conference? |
| 18:33.08 | Juxt | e.g. meet-me? |
| 18:33.17 | *** join/#asterisk leandro_pt (~leandro@82.155.117.148) |
| 18:34.49 | AgiNamu | just 2 people? |
| 18:34.52 | Juxt | yes |
| 18:34.53 | AgiNamu | No. |
| 18:34.59 | AgiNamu | that's just a call :P |
| 18:35.02 | AgiNamu | use a .call file |
| 18:35.17 | Juxt | ok that's what i thought |
| 18:35.25 | AgiNamu | http://www.voip-info.org/wiki-Asterisk+auto-dial+out |
| 18:35.41 | AgiNamu | so long it's just 2 channels, there's no problem. |
| 18:36.07 | AgiNamu | bbl |
| 18:36.16 | Juxt | thank you! |
| 18:36.28 | Juxt | next question |
| 18:36.35 | Juxt | i am using firefly (iax2) |
| 18:36.40 | Juxt | it doesn't have any transfer buttons |
| 18:36.48 | Juxt | how would i transfer calls to other extensions |
| 18:36.53 | *** join/#asterisk drachenfels (~ba@82.133.34.211) |
| 18:37.01 | Qwell | hookswitch? |
| 18:37.22 | Juxt | huh |
| 18:37.27 | Juxt | (forgive the newbie) |
| 18:37.41 | *** join/#asterisk mbranca_home (~matteo@host-84-222-9-45.cust-adsl.tiscali.it) |
| 18:37.54 | jontow | hmm |
| 18:38.09 | jontow | so add [ignore] after registration #1000 to ignore the rest of the file.. |
| 18:38.11 | drachenfels | anyone know much about faxing? |
| 18:39.10 | drachenfels | I've spent the past 48hrs trying to get the damn thing working, but I'm still having problems.... |
| 18:39.30 | Juxt | 48 hrs is just a start LOL |
| 18:39.53 | drachenfels | I haven't been to bed for 48hrs..! |
| 18:39.55 | husher | drachenfels: maybe you should consider a nap :) |
| 18:39.56 | drachenfels | solid! |
| 18:40.15 | drachenfels | I was hoping someone might point me in the right direction.... |
| 18:40.20 | Juxt | oh man so you're sponsored by Folgers or some other coffee co? |
| 18:40.22 | *** join/#asterisk Unrea1 (~nschmidt@65.244.121.132) |
| 18:40.45 | Juxt | i am on my 4th espresso and i am already thinking out loud |
| 18:41.03 | Unrea1 | Quick question. Im trying to setup prompt recording from a headset. I have it so when you dial x205 it beeps and starts recording. How do I get it to stop recording after 2 seconds of silence |
| 18:41.06 | jaiger | drachenfels, what do you need to know? |
| 18:41.21 | drachenfels | I've got a TDM400P digium card, with 4 fxo cards on it |
| 18:41.26 | Unrea1 | exten => 205,2,Record(/usr/src/temp:gsm) |
| 18:41.29 | Unrea1 | is the line right now |
| 18:41.31 | drachenfels | there the red ones right? |
| 18:41.32 | jontow | aginamu; whats with all the ^M's ? :) |
| 18:41.55 | *** join/#asterisk Rival (~rival@66.177.249.219) |
| 18:41.57 | jaiger | drachenfels, don't recall the color but they should be marked |
| 18:42.34 | drachenfels | First of all I had the problem where I couldn't get any negotiation going, asterisk recognised that it was a fax coming in, passes it over to spandsp |
| 18:42.39 | jontow | aginamu; and we're up to 3mins of cpu time :) |
| 18:42.40 | drachenfels | and nothing. |
| 18:42.51 | Juxt | what do you use to actually receive the fax |
| 18:42.53 | Juxt | hylafax? |
| 18:43.03 | bannerman | that's frustrating, even if I mute my mic, they have echo, unless I turn down my system sound.. which means X-lite is using my sound card's output instead of my microphone channel |
| 18:43.06 | drachenfels | I eventually realised this was down to playing with my rxgain and txgain settings |
| 18:43.20 | jaiger | I use hylafax and/or regular fax machine on a fxs channel to receive faxes |
| 18:43.34 | drachenfels | All on the PBX |
| 18:43.43 | drachenfels | I want it to just email me the faxes |
| 18:43.50 | Unrea1 | the line is Record(filename.format[|silence][|maxduration][|option]) but I cant get it to stop after 2 seconds of silence |
| 18:43.51 | jaiger | my hylafax can do that |
| 18:44.29 | drachenfels | however, I used the ztmonitor program while setting my levels, |
| 18:44.55 | drachenfels | as soon as it passes control over to app_rxfax the receive level drops loads.... |
| 18:45.07 | Juxt | uugh |
| 18:45.09 | drachenfels | I've had to set my rxgain at 18.0 |
| 18:45.33 | drachenfels | Is that quite high, it seems high to me! |
| 18:45.38 | jaiger | drachenfels, unfortunately I don't know anything about spandsp |
| 18:46.05 | drachenfels | I was wondering if it had anything to do with bug 2023 |
| 18:46.08 | jaiger | my gains are 0 but then I haven't tried to tweak them |
| 18:46.43 | drachenfels | My point is though, my rx level drops considerably when it passes it out to the program |
| 18:46.45 | jaiger | my hardware echo canceler has auto gain control which seems to work well |
| 18:46.54 | *** join/#asterisk afe ([BpVDOJx0t@c-e616e055.123-1-64736c12.cust.bredbandsbolaget.se) |
| 18:47.00 | jontow | aginamu; takes all sorts of time |
| 18:47.52 | jontow | but admittedly.. not all of the registered |
| 18:48.39 | drachenfels | What hardware do you need to run hylafax? |
| 18:48.56 | Dandan | drachenfels: hardware modem |
| 18:48.59 | Dandan | and hylafax |
| 18:49.00 | drachenfels | Will it run with my digium card, or do you need a separate modem in there? |
| 18:49.09 | drachenfels | k |
| 18:49.52 | bannerman | Nobody knows the answer to Unrea1's question? |
| 18:49.54 | jontow | good news is.. they register in batches and it DOES seem to work.. |
| 18:50.06 | bannerman | I'd love to know how to do that as well. |
| 18:50.43 | jaiger | drachenfels, you need a modem that appears as a serial modem with AT command set for hylafax |
| 18:50.44 | *** join/#asterisk PMantis (~Miranda@66.251.89.34) |
| 18:50.54 | jaiger | dunno if the digium cards give you that access yet |
| 18:50.57 | jontow | Record(mysound:gsm|2) |
| 18:50.58 | jontow | <PROTECTED> |
| 18:51.10 | jaiger | drachenfels, and you |
| 18:51.24 | drachenfels | hmmm, I think I've got an old sportster 56k knocking about somewhere |
| 18:51.26 | jaiger | drachenfels, and you'll need some sort of fxs channel in the pbx |
| 18:51.37 | mikegrb | . |
| 18:51.45 | mikegrb | /////////////////////////////////////////////////////////////////////////////////////////// |
| 18:51.51 | jontow | unrea1; the impression one gets from "show application record" on the asterisk CLI is that the paremeter directly after 'format' is a number in seconds which to allow for silence before cutting the person off |
| 18:51.56 | PMantis | Anyone have experience with WaitForSilence or MachineDetect? |
| 18:52.34 | Rival | ey im stupid but what would the function be called to setup a extention to transfer to the phone port on a digium x100p card? |
| 18:53.23 | mogorman | ? |
| 18:53.34 | drachenfels | errr. something like exten => s,1,Dial(ZAP1/1) |
| 18:53.55 | Rival | thanks |
| 18:54.04 | Rival | i just need a hint where to look or read about it |
| 18:54.04 | jontow | aginamu; it doesn't work well.. ;) |
| 18:54.29 | drachenfels | look at the asterisk wiki |
| 18:54.43 | drachenfels | http://www.voip-info.org/ |
| 18:56.14 | *** join/#asterisk chevyman440 (~matt@65.122.76.58) |
| 18:58.27 | PMantis | Can anyone comment on the effectiveness of WaitForSilence or MachineDetect? |
| 18:59.32 | elriah | Hi all - anyone know of some free music on hold sites? |
| 19:01.14 | *** join/#asterisk elvisthedj (~kris@lnhe6-49.2wcm.comporium.net) |
| 19:02.27 | *** join/#asterisk xpasha (~pavel@217.30.252.68) |
| 19:02.44 | elvisthedj | hello all, I was just looking to see if anyone was aware of an agi that made use of allison's weather recordings? I see the .gsm's, but the only agi's i see are for festival |
| 19:02.47 | Rival | dranchenfels i have a hard time finding stuff on that site quickly |
| 19:03.01 | Rival | specially when i dunno what im looking for exactly |
| 19:03.12 | elvisthedj | just wondering if there is a script to parse weather from somewhere and use playbacks to make allison give the weather |
| 19:03.30 | Juxt | what is allison? |
| 19:03.45 | bjohnson | elvisthedj: must be .. a lot of weather sound bites in asterisk-sounds. but I haven't seen such an app |
| 19:03.59 | Qwell | Juxt: a person |
| 19:03.59 | bjohnson | Juxt: the woman whose voice is used |
| 19:04.19 | elvisthedj | bjohnson: you'd think. i promise i've spent lots of time on google before coming here to ask such a flameworthy question :D |
| 19:04.26 | Juxt | oh sheesh i thought there was a voice synth on the block |
| 19:04.28 | Zeeek | elvisthedj ManxPower had a page on this at some point |
| 19:04.45 | bjohnson | ~alison |
| 19:04.54 | bjohnson | ~allison |
| 19:05.01 | elvisthedj | Zeeek: thank you Zeeek |
| 19:05.05 | bjohnson | nothing |
| 19:05.27 | Zeeek | you only need to type asterisk weather to get several pages about this |
| 19:05.34 | Zeeek | in google |
| 19:05.43 | bjohnson | doh .. busted |
| 19:05.52 | Zeeek | ~got milk |
| 19:05.53 | jbot | ACTION chugs down a carton |
| 19:05.58 | elvisthedj | i do voiceovers for a living and have recorded most prompts for asterisk (all for the default sound) |
| 19:05.59 | *** join/#asterisk kpfleming (~chatzilla@ip68-3-230-141.ph.ph.cox.net) |
| 19:06.27 | bjohnson | any funny ones? |
| 19:06.28 | Zeeek | what's new bjohnson |
| 19:06.29 | elvisthedj | Zeeek: that was my first search. But like i said, i only find festival scripts. e.g., pull the weather, text2wav and (playback) |
| 19:06.46 | Zeeek | you want a way to pull weather? |
| 19:07.03 | bjohnson | Zeeek: struggling with a wrt54g to get it to a) do pppoe b) do wifi c) do asterisk |
| 19:07.13 | Zeeek | or, au contraire, you anted to know why allison was talking about high winds? |
| 19:07.25 | Zeeek | I have one of those bjohnson |
| 19:07.39 | Zeeek | in fact I'm talkin at ya right now "over it" |
| 19:07.56 | elvisthedj | Zeeek: no, there are lots of sources for the forecast, i just wondered if someone had written a script to parse the forecast so i could use playback(allisonsstuff) instead of using festival to read the forecast |
| 19:08.06 | bjohnson | v2.2 seems to make it a little too leading edge .. only works with experimental release of openwrt |
| 19:08.48 | *** join/#asterisk steve123 (~stevetdi@pD9E2F2BA.dip.t-dialin.net) |
| 19:09.06 | steve123 | hello everybody! |
| 19:09.09 | Zeeek | I have it connected to a DSL modem in bridge mode |
| 19:09.44 | Zeeek | there is one thing I can't understand: why don't wifi modems have a simple SWITCH to turn off the radio signal? |
| 19:09.56 | Zeeek | I keep it off when not in use for obvious reasons |
| 19:10.10 | AgiNamu | Zeeek, it's better to get a consistent amount of radiation. |
| 19:10.15 | steve123 | is anybody german speaking here? |
| 19:10.28 | elvisthedj | I don't understand why wifi doesn't use DSS |
| 19:10.37 | mmlj4 | nein |
| 19:10.46 | CosmicRay | mmlj4: I was going to say that :-) |
| 19:10.48 | bjohnson | Zeeek: like a hardware switch? I don't know |
| 19:10.52 | mmlj4 | :-) |
| 19:10.57 | steve123 | mmlj4 ? |
| 19:11.05 | mmlj4 | steve123: i was joking |
| 19:11.16 | bjohnson | Zeeek: I want it to connect to my dsl modem but the modem is not a router .. so the wrt must use pppoe through it |
| 19:11.24 | mmlj4 | probably someone does speak german here |
| 19:11.42 | bjohnson | elvisthedj: I thought wifi WAS DSS |
| 19:11.45 | Zeeek | I am using ppoA |
| 19:11.46 | PMantis | Can anyone comment on the effectiveness of WaitForSilence or MachineDetect? |
| 19:12.05 | CosmicRay | steve123: do you not speak english? |
| 19:12.23 | steve123 | a little.... |
| 19:12.25 | mmlj4 | don't harass him, dude |
| 19:12.34 | bjohnson | Zeeek: I guess they've changed stuff around in experimental and pppoe wasn't really tested. Next version of experimental is supposed to work |
| 19:12.34 | hardwire | steve123: what do you speak? |
| 19:12.39 | hardwire | just wondering |
| 19:12.51 | steve123 | german.. i am from germany |
| 19:12.55 | Zeeek | bjohnson you're using the open source version, is that it? |
| 19:12.58 | hardwire | well.. happy german day! |
| 19:13.12 | elvisthedj | bjohnson: if it was, routers and nics would have to come in sets :) |
| 19:13.18 | bjohnson | Zeeek: openwrt .. the sveasoft doesn't support the v2.2 yet either |
| 19:13.23 | *** join/#asterisk Weezey (~Weezey@206.210.109.226) |
| 19:13.29 | mmlj4 | irish, gr... |
| 19:13.29 | *** part/#asterisk Juxt (~Juxt@adsl-068-213-216-087.sip.bct.bellsouth.net) |
| 19:13.37 | Zeeek | I'm using stock firmware and it performs great |
| 19:13.40 | *** join/#asterisk fugitivo (~ajf@201.255.100.115) |
| 19:13.49 | elvisthedj | bjohnson: i mean spread spectrum ala cordless phones.. jumping frequencies every 1/10th of a sec.. but the router, card would have to have the same codes.. in sync w/ each other |
| 19:13.50 | bjohnson | Zeeek: with asterisk running on it? |
| 19:14.00 | bjohnson | Zeeek: sorry .. I guess you didn't say that |
| 19:14.04 | Zeeek | the WAG54g at the office reboots after any change in cofig - that's irritating |
| 19:14.16 | *** join/#asterisk implicit (~implicit@dhcp-251254.mobile.uci.edu) |
| 19:14.22 | Zeeek | no I have no interest in running asterisk on the router |
| 19:14.37 | bjohnson | Zeeek: jerjer says he does it with up to 4 concurrent calls |
| 19:14.40 | Nugget | 6 hours, 43 minutes, and 54 seconds until epoch 1111111111 |
| 19:14.58 | Zeeek | My interest would be on the contrary to have an asterisk box with no disk |
| 19:15.12 | *** join/#asterisk ckruetze (ckruetze@cpc1-cmbg7-5-0-cust31.cmbg.cable.ntl.com) |
| 19:15.14 | bjohnson | Zeeek: how is that different? |
| 19:15.23 | CosmicRay | steve123: Don't worry about your english skills here; most people can understand what you mean in many cases |
| 19:15.34 | CosmicRay | steve123: your english is probably far better than my german |
| 19:15.35 | Zeeek | bjohnson my last nufone call must have been one of those 4 :) |
| 19:15.40 | steve123 | ok... |
| 19:15.56 | hardwire | steve123: I bet you can spell better in english than I can :).. and I am American :) |
| 19:15.57 | steve123 | i have a problem with the extensions.conf |
| 19:16.11 | bkw_ | if you are on the dev call go to #devcall |
| 19:16.22 | steve123 | i use chan_capi.so |
| 19:16.33 | darius_ | bkw_: can I go if I'm not on the dev call? |
| 19:16.45 | bkw_ | ya |
| 19:16.48 | darius_ | :) |
| 19:17.00 | *** join/#asterisk bile_one (~bile_one@pcp03281999pcs.gillst01.ar.comcast.net) |
| 19:17.19 | bkw_ | of if you can't get into the call join #devcall too |
| 19:17.21 | CosmicRay | steve123: I understand you just fine, but I don't use isdn, so probably can't help |
| 19:18.25 | *** part/#asterisk kpfleming (~chatzilla@ip68-3-230-141.ph.ph.cox.net) |
| 19:18.36 | Zeeek | steve123 maybe it would be easier for you to just use pastebin.ca to show you error messages and the part of extensions you need to check? |
| 19:18.48 | Zeeek | ~pastebin |
| 19:18.49 | jbot | hmm... pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 19:18.53 | Katty | hi bkw_ (= |
| 19:19.43 | Katty | i've just read 3 chapters of a+ certification, and i haven't learned a single thing yet :< |
| 19:20.20 | Zeeek | why in heavens name do you want to certify a pluses? |
| 19:20.28 | Katty | because teh boss said so |
| 19:20.28 | *** join/#asterisk brc_ (~brian@brc.base.supporter.pdpc) |
| 19:20.29 | Zeeek | don't they have enough troubles? |
| 19:20.30 | Katty | dun dun dun |
| 19:20.44 | epoch | nothing wrong with A+ |
| 19:20.46 | Katty | he says it'll look pretty on a resume too |
| 19:20.55 | Zeeek | ya, Michael Jackson has one |
| 19:20.57 | epoch | it doesn't make you dumber, so it's gotta be a good thing :) |
| 19:21.05 | Katty | i think my portfolio is lots prettier than a silly certification on a bullet line |
| 19:21.10 | Zeeek | which is not true of a decent beer |
| 19:21.15 | *** join/#asterisk crich1999 (~crich@pD9E77819.dip.t-dialin.net) |
| 19:21.45 | Zeeek | epoch ^^^ |
| 19:21.46 | epoch | I dunno man, beer makes me smart |
| 19:21.51 | Katty | i'm just not so sure i want to memorize I/O and irqs and stuff |
| 19:21.52 | epoch | ;) |
| 19:21.57 | CosmicRay | does an include bring over the s extension? |
| 19:21.59 | Zeeek | ya, then the smart guy wants one, eh? |
| 19:22.01 | CosmicRay | in extensions.conf? |
| 19:22.12 | *** join/#asterisk pi-cubic (~pi-cubic@62.111.48.31) |
| 19:22.48 | Katty | pi-cubic: 31.0062767 |
| 19:23.05 | *** join/#asterisk yertle (yertle@ip68-6-98-122.sb.sd.cox.net) |
| 19:23.05 | PMantis | Can anyone comment on the effectiveness of WaitForSilence or MachineDetect? |
| 19:23.27 | wildgoose | what is a cheap fxs card for use in a home voip setup? I have x100p card just arrived for incoming, need to wire up the old analogue phones now |
| 19:23.45 | CosmicRay | wildgoose: sip converter boxes are cheaper |
| 19:23.46 | CosmicRay | about $60 |
| 19:23.55 | CosmicRay | wildgoose: they'll have one ethernet and one analog port |
| 19:23.58 | jontow | wildgoose; i just bought a Sipura SPA-2000 .. $88 shipped, dual-FXS |
| 19:23.58 | pi-cubic | hi there, can anyone recommend me a kernel version running stable with chan_capi? |
| 19:24.02 | wildgoose | any brand names? |
| 19:24.04 | pi-cubic | Katty: right ;> |
| 19:24.08 | jontow | and Sipura makes a 1port model too |
| 19:24.10 | wildgoose | the wiki doesn't seem to cover this? |
| 19:24.13 | jontow | im very impressed with it so far.. |
| 19:24.14 | jontow | yes it does |
| 19:24.19 | CosmicRay | wildgoose: Sipura SPA-1001, very nice, about $65 IIRC |
| 19:24.36 | wildgoose | aha, would this have been better than my x100p then...? |
| 19:24.45 | jontow | no, the X100P is an FXO device |
| 19:24.52 | jontow | thats your PSTN connection |
| 19:24.58 | hardwire | POTS |
| 19:25.01 | jontow | vs. the FXS which are "ATA"'s (Analog Telephone Adapters) |
| 19:25.27 | Dandan | ~pstn |
| 19:25.30 | jbot | hmm... pstn is Public Switched Telephone Network |
| 19:25.35 | Dandan | ~pots |
| 19:25.36 | jbot | it has been said that pots is Plain Old Telephone Service as in "Old Analogue Crap" |
| 19:25.39 | wildgoose | yeah. sorry was confused, thought the sipura was one of those gadgets with both inside |
| 19:25.52 | Zeeek | there is one with both |
| 19:25.57 | PMantis | ~iax |
| 19:25.58 | jbot | iax is, like, 4569 and 5036, or pronounces "Eeks" |
| 19:26.11 | hardwire | wildcard0: the 3000 series has both |
| 19:26.12 | jontow | http://www.voip-info.org/wiki-ATA |
| 19:26.15 | jontow | go there, wildgoose :) |
| 19:26.37 | wildcard0 | ? |
| 19:26.41 | Zeeek | how does a dual-FXS SIP box work? It uses 2 ip addresses? |
| 19:26.42 | *** join/#asterisk FarrisG (~jrush@h-68-164-19-170.dllatx37.covad.net) |
| 19:26.55 | jontow | no, 1 IP.. 2 registrations, iirc |
| 19:26.57 | Zeeek | no, it must address the ports |
| 19:26.59 | yertle | 2 regs |
| 19:27.03 | wildcard0 | oh. borked nick completion. i see :) |
| 19:27.03 | FarrisG | Can anyone recommend a good SIP softphone for linux? |
| 19:27.17 | Nugget | there's no such thing as a good softphone. |
| 19:27.21 | yertle | gnophone? |
| 19:27.24 | Zeeek | here we go |
| 19:27.24 | wildgoose | Aha. Interestingly the "FXS" page on the wiki doesn't seem to lead me there, hence my confusion... Thanks |
| 19:27.32 | yertle | fxo page on wiki is the one |
| 19:28.22 | bjohnson | jontow: ATA also come with fxo ports |
| 19:28.24 | PMantis | FarrisG: IAXComm worked great for me. Kinda ugly, and hung once in a while upon closing it, but worked beautifully otherwise. |
| 19:28.29 | bjohnson | jontow: some ATAs also come with fxo ports |
| 19:28.35 | *** join/#asterisk w0w0 (~w0w0@80.26.165.152) |
| 19:28.49 | PMantis | gnophone only supports the original IAX protocol AFAIK |
| 19:29.35 | *** join/#asterisk Qwell (~north@70-32-102-18.ontrca.adelphia.net) |
| 19:29.50 | *** join/#asterisk nani707 (~nene@adsl-69-109-117-188.dsl.pltn13.pacbell.net) |
| 19:29.54 | nani707 | hi |
| 19:29.57 | CosmicRay | gnophone *sucks* |
| 19:30.00 | hardwire | does the whole world go green on st patricks day? |
| 19:30.05 | CosmicRay | it was dropped from debian for being unmaintained upstream, btw |
| 19:30.12 | hardwire | for instance.. do africans wear green tunics today? |
| 19:30.17 | Rival | can a x100p be using to output calls to a analogue phone also? |
| 19:30.31 | hardwire | Rival: no. |
| 19:30.33 | CosmicRay | Rival: to an analog pstn line, yes. to an analog phone, no. |
| 19:30.40 | Darwin[laptop] | only if you have a hard line plugged into it |
| 19:30.52 | hardwire | Rival: unless you have two lines in your house.. and you use it to call the other one :) |
| 19:31.00 | *** join/#asterisk eKo1 (~bernd@63.245.57.70) |
| 19:31.02 | hardwire | via the phone company. |
| 19:31.10 | Rival | well could i use my cellphone |
| 19:31.12 | Rival | call the box |
| 19:31.23 | hardwire | it woul dbe just like calling your house |
| 19:31.23 | Rival | and have it output to a line as a extention? |
| 19:31.32 | Rival | i cant get it to work |
| 19:31.42 | nani707 | Hi, I have two SIP outbound providers, how to load balance them, or use round robin? |
| 19:31.44 | hardwire | Rival: then you just aren't trying hard enough :) |
| 19:32.01 | hardwire | nani707: put them both in a call agent group? |
| 19:32.08 | hardwire | somehow. |
| 19:32.13 | Rival | exten => 111,1,Dial(ZAP1/1,20) |
| 19:32.15 | hardwire | ?? |
| 19:32.23 | PMantis | nani707: Write a macro that flip-flops a variable, from 1 or 0... then use one or the other based upon the variable contents. |
| 19:32.25 | hardwire | Rival: that will dial out |
| 19:32.30 | nani707 | call agent group ( new term) let me lookup |
| 19:32.33 | *** join/#asterisk unixgeek (~unixgeek@12.45.238.189) |
| 19:32.36 | hardwire | ooh.. I like what PMantis suggested |
| 19:32.47 | hardwire | nani707: I wouldn't bother.. its a whim thought from a crazy freak. |
| 19:32.49 | *** part/#asterisk darius_ (darius@c-67-171-44-101.client.comcast.net) |
| 19:33.00 | nani707 | ok |
| 19:33.04 | Rival | it should dial out to the phone hooked into the x100p right? |
| 19:33.07 | bjohnson | Rival: that won't actually dial anything .. just pick u the line |
| 19:33.11 | bjohnson | Rival: that won't actually dial anything .. just pick up the line |
| 19:33.13 | hardwire | Rival: no |
| 19:33.17 | hardwire | it will dial to the phone company |
| 19:33.19 | rvhi | try to comment out a block of code in extensions.conf |
| 19:33.21 | hardwire | given you have it plugged into the wall |
| 19:33.27 | Rival | it is |
| 19:33.30 | rvhi | anthm told me to use ;-- ... --; |
| 19:33.35 | rvhi | not sure what that means |
| 19:33.47 | hardwire | Rival: that extension should give you dialtone to the phone company then |
| 19:33.57 | hardwire | sorta |
| 19:34.01 | hardwire | that extension is wrong |
| 19:34.02 | hardwire | but still |
| 19:34.05 | Rival | heh k |
| 19:34.17 | hardwire | more like Dial(Zap/1) |
| 19:34.24 | Rival | i tried it that way too |
| 19:34.29 | hardwire | or Zap/g1 and put the fxs in its own little group in zapata.conf |
| 19:34.29 | *** join/#asterisk jhiver (~jhiver@AStDenis-103-1-5-132.w80-13.abo.wanadoo.fr) |
| 19:34.35 | jhiver | ~seen p0lar |
| 19:34.36 | jbot | p0lar <~p0lar@64.254.225.62> was last seen on IRC in channel #asterisk, 6d 2h 5m 21s ago, saying: 'closer'. |
| 19:34.42 | anthm | serious ? |
| 19:34.44 | hardwire | Rival: digium provides newbie support don't they? |
| 19:34.48 | anthm | ;-- = start comment |
| 19:34.54 | anthm | --; = end |
| 19:35.10 | *** join/#asterisk r0d3nt|m (anonymous@soveliss.luniac.com) |
| 19:35.37 | Rival | hardwire if ya could just point me to documentation that kinda explains the functions id be more then happy to waste my time and not everyone elses |
| 19:35.50 | Rival | http://www.voip-info.org/wiki-Asterisk+config+extensions.conf |
| 19:36.22 | *** join/#asterisk ManxPower (~eric@ip-209-16-83-10.i-55.com) |
| 19:36.30 | Rival | is what ive been looking at but what i dont know is the command to pass the incoming call to ring the phone hooked into the phone port of the x100p |
| 19:36.31 | hardwire | Rival: sure |
| 19:36.34 | hardwire | www.voip-info.org |
| 19:36.40 | Rival | hehe |
| 19:36.45 | Rival | i hate that site =( |
| 19:36.50 | Zeeek | Starter tutorial: |
| 19:36.50 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html |
| 19:36.50 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html |
| 19:36.50 | Zeeek | http://www.automated.it/guidetoasterisk.htm |
| 19:36.50 | Zeeek | THE reference of the moment: |
| 19:36.50 | Zeeek | http://www.asteriskdocs.org |
| 19:36.51 | hardwire | learn to love it |
| 19:36.54 | Zeeek | what about one of those? |
| 19:36.57 | Rival | so confusing to me |
| 19:37.07 | hardwire | spend more time on it |
| 19:37.08 | Zeeek | every one of the above has the answers |
| 19:37.08 | Rival | ill check those i havent looked at any of them yet zeeek |
| 19:37.10 | hardwire | start at the main menu :) |
| 19:37.10 | Rival | thanks |
| 19:37.15 | bjohnson | Rival: look at all the examples listed there |
| 19:37.17 | Zeeek | every one |
| 19:37.19 | *** part/#asterisk yertle (yertle@ip68-6-98-122.sb.sd.cox.net) |
| 19:37.33 | CosmicRay | you have voip-info? |
| 19:37.38 | CosmicRay | it's the best wiki I've ever seen |
| 19:37.49 | Zeeek | like democracy is the best form of gov't |
| 19:37.57 | afe | Rival: Asterisk can't call the phone port on a x100p |
| 19:37.58 | hardwire | http://www.voip-info.org/wiki-Asterisk+config+zaptel.conf |
| 19:38.05 | bjohnson | many, many, answers on the wiki |
| 19:38.10 | Rival | k |
| 19:38.11 | hardwire | http://www.voip-info.org/wiki-Asterisk+Zaptel+Installation |
| 19:38.13 | Zeeek | many wrongs ones too |
| 19:38.32 | rvhi | anthm, i use stable code, is it supported? |
| 19:38.36 | Rival | i thought there might be a way to use the x100p as a ata device also |
| 19:38.39 | rvhi | i tried exactly the way you told me |
| 19:38.47 | afe | Rival: there isn't |
| 19:38.48 | hardwire | Rival: http://www.abptech.com/mainpages/products/asteriskbook.html |
| 19:38.50 | Zeeek | Rival no, do some reading of the basics |
| 19:38.52 | hardwire | super awesome newb book |
| 19:39.11 | rvhi | it says --; has error |
| 19:39.11 | ManxPower | I don't suppose anyone has seen situation where you CANNOT get a Digium T-1 card to get sync from the telco? It ALWAYS says internally clocked. |
| 19:39.22 | anthm | bah |
| 19:39.45 | hardwire | ManxPower: no :( |
| 19:39.47 | anthm | stable shnable |
| 19:40.18 | Rival | i just needed to know it wasnt possible |
| 19:40.25 | rvhi | ? |
| 19:41.27 | anthm | I can't help you if you use stable |
| 19:41.33 | anthm | its against my contract |
| 19:41.37 | *** join/#asterisk SpaceBass (~sp@24.125.33.214) |
| 19:41.39 | SpaceBass | hey folks |
| 19:41.51 | SpaceBass | not sure if this is networking or asterisk related |
| 19:42.17 | SpaceBass | I am going to add some wifi sip phones that do not support WPA, so I have to set up a wifi network that has WEP |
| 19:42.26 | SpaceBass | so I set that up on a 2nd (protected) subnet |
| 19:42.27 | *** join/#asterisk KalD|WORK (~KalD@proxy.corp.telesym.com) |
| 19:42.27 | wildgoose | ok, any tips on buying the sipura 1001 in the uk? |
| 19:42.43 | SpaceBass | I am passing port 5060 back to the sip server, but still unable to connect |
| 19:42.56 | SpaceBass | anyone know of a reason why I cannot be on a different subnet than my asterisk box? |
| 19:43.19 | jaiger | SpaceBass, firewall/router issues? |
| 19:43.28 | SpaceBass | wildgoose any luck on ebay? |
| 19:43.38 | SpaceBass | jaiger well, other traffic passes correctly |
| 19:43.41 | jaiger | SpaceBass, 5060 isn't the only port involved in sip |
| 19:43.50 | SpaceBass | jaiger what else will I need to open? |
| 19:44.08 | wildgoose | nothing which doesn't have a high shipping cost... |
| 19:44.17 | Zeeek | SpaceBass how does the client talk to the server? What route? |
| 19:44.19 | wildgoose | ends up cheaper to buy from a mainstream dealer here |
| 19:44.20 | jaiger | SpaceBass, not sure off the top of my head but I know that SIP uses more than one port - similar to ftp |
| 19:44.29 | SpaceBass | Zeeek what do you mean? |
| 19:44.38 | SpaceBass | jaiger dynamically or certian ports? |
| 19:44.40 | Zeeek | wildgoose voiptalk.org |
| 19:44.52 | wildgoose | spacebass: check your rtp.conf file |
| 19:45.02 | wildgoose | if you are using sip then those ports need to be mapped |
| 19:45.08 | Zeeek | SpaceBass client:192.168.1.50 server:192.168.2.1 ? |
| 19:45.14 | wildgoose | I shrunk the range quite a bit |
| 19:45.30 | bjohnson | SpaceBass: and nat=yes in sip.conf |
| 19:45.40 | bjohnson | SpaceBass: and it still might not work |
| 19:45.43 | SpaceBass | Zeeek client 10.1.1.250 server: 10.1.0.30 |
| 19:45.50 | Dandan | intercom on snom phones - is it doable? the hack on v-info is old and useless |
| 19:45.51 | Zeeek | and the netmùask is ? |
| 19:45.56 | Zeeek | netmask? |
| 19:46.04 | SpaceBass | 24 for both |
| 19:46.07 | bjohnson | SpaceBass: search for asterisk nat |
| 19:46.14 | *** join/#asterisk husher (~andrew@68.143.92.130.nw.nuvox.net) |
| 19:46.19 | wildgoose | is the ata-486 considered to be as good as the sipura 1001? |
| 19:46.25 | Zeeek | there is no NAT between the two, is there? |
| 19:46.38 | SpaceBass | zeek no nat |
| 19:46.39 | wildgoose | By the way I have asterisk working on a nat setup - works quite well |
| 19:46.45 | _-Jon-_ | wildgoose, don't get the HT |
| 19:46.53 | SpaceBass | bjohnson there is no nat b/t the two |
| 19:46.56 | ManxPower | ixx, on the other hand, cannot help anyone that uses CVS-HEAD |
| 19:46.56 | wildgoose | whats an HT? |
| 19:47.01 | Zeeek | SpaceBass so the clients can ping the * server? |
| 19:47.06 | _-Jon-_ | Handytone ATA-486, right? |
| 19:47.10 | wildgoose | aha |
| 19:47.20 | wildgoose | why is it so bad? |
| 19:47.24 | SpaceBass | Zeeek there is a firewall b/t the two, I can only open select ports |
| 19:47.29 | bjohnson | wildgoose: no |
| 19:47.39 | _-Jon-_ | I have the ATA-286 and I've been less than impressed. it stops working at random |
| 19:47.43 | bjohnson | wildcard0: not bad .. just less features |
| 19:47.46 | Zeeek | you will have to open a shitload of ports |
| 19:48.10 | SpaceBass | Zeeek well my choices are open a shitton of ports or allow a accesspoint with only wep on my network |
| 19:48.16 | wildgoose | Zeeek: just change rtp.conf to open perhaps 20 ports, then also map 5060 and you are done |
| 19:48.22 | wildgoose | This was all I needed to do |
| 19:48.27 | SpaceBass | apparently the ZyXel 2000w phhone doesnt support wpa |
| 19:48.34 | Zeeek | can't you open all connections from the other subnet (not knowing who's on it... I dunno) |
| 19:48.56 | SpaceBass | Zeeek thats what I want to avoid... opening everything |
| 19:49.13 | Zeeek | but it's ytour subnet - but maybe the public is on it? |
| 19:49.29 | SpaceBass | I guess the other option is to throw another nic in the asterisk box and put it on that subnet too |
| 19:49.44 | SpaceBass | Zeeek there are at least 10 other wireless networks in my neighborhood alone... I trust no one |
| 19:49.55 | SpaceBass | all it takes to hack wep is some scriptkiddie with google |
| 19:50.16 | Zeeek | yeah |
| 19:50.43 | Zeeek | except they don't need google |
| 19:50.45 | Grooby | bjohnson, you listening to the dev conf? |
| 19:50.58 | *** join/#asterisk clive- (~pirch@rrba-146-73-222.telkomadsl.co.za) |
| 19:51.26 | bjohnson | yes |
| 19:51.27 | Zeeek | SpaceBass that is exactly what I did |
| 19:51.33 | unixgeek | Does anyone have any experience with placing a SIP call through Lingo? |
| 19:51.37 | Grooby | ok |
| 19:51.38 | Grooby | hehe |
| 19:51.45 | Zeeek | then the script kiddies only have access to your astezrisk machine |
| 19:51.51 | bjohnson | Grooby: #devcall |
| 19:51.53 | Grooby | they mentioned your name |
| 19:51.59 | bjohnson | wooohoooo !!! |
| 19:52.02 | bjohnson | I'm famous |
| 19:52.14 | Grooby | lol |
| 19:52.38 | SpaceBass | Zeeek I had wanted to avoid doing that, but I'm not 100% opposed |
| 19:52.41 | SpaceBass | if that is what i have to do |
| 19:52.50 | SpaceBass | I had thought sip just needed 5060 |
| 19:53.27 | jhiver | SpaceBass what's your problem? |
| 19:53.29 | Zeeek | for that you'll need to read up on SIP a little |
| 19:53.38 | jhiver | sorry I'm late in the conversation :) |
| 19:53.44 | jhiver | Hi Zeeek |
| 19:53.54 | *** part/#asterisk PMantis (~Miranda@66.251.89.34) |
| 19:53.56 | Zeeek | maybe you'll bring a fresh tropical breeze of knowledge in |
| 19:54.04 | SpaceBass | jhiver - adding a wifi sip phone, only supports wep, don't want to put a wep AP on my network, so i have it on its own subnet with a firewall b/t the two |
| 19:54.10 | Zeeek | it's about talking to other subnets |
| 19:54.18 | SpaceBass | it should be clear that I'm a complete asterisk newbie |
| 19:54.40 | SpaceBass | jhiver and I can only open ports b/t the two subnets |
| 19:55.10 | *** join/#asterisk buddah (~hnic@208.179.86.5) |
| 19:55.10 | Zeeek | as bjohnson pointed out, look at the sip.conf nat=yes |
| 19:55.10 | *** join/#asterisk Juxt (~Juxt@adsl-068-213-216-087.sip.bct.bellsouth.net) |
| 19:55.20 | Juxt | i can't unsubscribe from the asterisk list |
| 19:55.24 | Juxt | the mailman isn't responding |
| 19:55.25 | Zeeek | and forward ports 1000-10100 to the asterisk box |
| 19:55.34 | Zeeek | and see what happens |
| 19:55.43 | jhiver | SpaceBass, is the SIP phone firewall a linux box? |
| 19:55.45 | SpaceBass | just changed the nat line |
| 19:55.54 | Zeeek | sip reload |
| 19:56.05 | SpaceBass | jhiver yeah, of some sort, its called ipcop www.ipcop.org |
| 19:56.08 | SpaceBass | similar to smoothwall |
| 19:56.15 | AgiNamu | Hey, anyone know of a site that has "stanard" numbers for things like *69, *67, etc? |
| 19:56.24 | afe | SpaceBass: check which ports the WiFi phone uses for RTP, and just open those and 5060 in the firewall |
| 19:56.32 | jhiver | Then you should prolly combine OpenVPN + ethernet bridging |
| 19:56.51 | SpaceBass | wow... the nat=yes did it |
| 19:56.55 | jhiver | Everything will go thru one port, no nat issues, and encryption as a bonus... |
| 19:57.06 | SpaceBass | well... maybe it didnt |
| 19:57.12 | SpaceBass | was able to log in, but not connecting |
| 19:57.29 | SpaceBass | jhiver cannot VPN from the phone |
| 19:57.42 | SpaceBass | and the router is the same box |
| 19:57.45 | jhiver | No but the phone connects to the firewall and the firewall can VPN can't it? |
| 19:57.46 | SpaceBass | bot both subnets |
| 19:57.56 | jhiver | that's why u have to be using ethernet bridging |
| 19:58.07 | jhiver | so basically the phone's subnet is 1 ethernet |
| 19:58.16 | jhiver | the vpn is a 'virtual' ethernet |
| 19:58.19 | jhiver | and you bridge the 2 |
| 19:58.26 | jaiger | AgiNamu, I think those are called Vertical Service Codes - use google |
| 19:58.32 | SpaceBass | doesnt that basically open that phone subnet wide open then? |
| 19:58.38 | jaiger | ~vsc |
| 19:58.49 | *** join/#asterisk NexterB (~be_nexter@lns-vlq-23-lyo-82-255-97-194.adsl.proxad.net) |
| 19:58.58 | *** join/#asterisk husher (~andrew@68.143.92.130.nw.nuvox.net) |
| 19:59.01 | AgiNamu | http://www.nanpa.com/number_resource_info/vsc_assignments.html |
| 19:59.02 | AgiNamu | :) |
| 19:59.03 | AgiNamu | thanks |
| 19:59.11 | jhiver | I guess... |
| 19:59.45 | SpaceBass | ok... what if I just had the sip phone and was going to connect to something like vonage... how do they do it with only one port or behind nat? |
| 19:59.59 | NexterB | Hello. |
| 20:00.12 | AgiNamu | What's the recommended way for speed dialing then? |
| 20:00.13 | jhiver | but you could use something like FireHol to allow only for SIP on that subnet I suppose... |
| 20:00.24 | AgiNamu | *xx isn't ok, since there's codes. #xx ? |
| 20:00.27 | SpaceBass | might have to look into that |
| 20:00.44 | SpaceBass | again, I can just plug that wep basestation into a 2nd nic in my asterisk box |
| 20:00.47 | SpaceBass | that will work too |
| 20:00.48 | afe | SpaceBass: you just open the ports the wifi phone uses - not all those used by asterisk |
| 20:01.18 | SpaceBass | afe, i guess thats what I'm after- what ports are needed for sip b/t the phone and asterisk |
| 20:01.53 | afe | SpaceBass: you might be able to config it on the phone - you need ports for SIP and RTP |
| 20:02.14 | afe | SpaceBass: and if its not configurable, it should be documented somewhere |
| 20:02.23 | SpaceBass | trying with just a softphone right now- dont have the actual phone in question yet |
| 20:02.35 | Rival | i gotta question why would you use ata device when the device costs more then most ip phones |
| 20:02.51 | SpaceBass | Rival fax machins, TiVos, etc |
| 20:02.52 | SpaceBass | ? |
| 20:03.07 | afe | SpaceBass: which softphone? |
| 20:03.12 | Zeeek | and analogue phones some people prefer |
| 20:03.36 | SpaceBass | afe xlite |
| 20:03.46 | bjohnson | Rival: plus they don't cost more than ip phones |
| 20:03.56 | Zeeek | SpaceBass are these phones for an office or public use (if I may) |
| 20:03.59 | bjohnson | Rival: plus cordless phone usage |
| 20:03.59 | Rival | whats a cheap ata device then? |
| 20:04.03 | jaiger | AgiNamu, I think usually xx# but that's just a guess |
| 20:04.10 | bjohnson | Rival: linksys pap2-na |
| 20:04.14 | bjohnson | sipura spa 2000 |
| 20:04.33 | afe | SpaceBass: I think x-lite uses port 8000-8007 (or maybe less) for RTP and 5060 for SIP |
| 20:04.37 | bjohnson | $30 to $35 USD per port |
| 20:04.47 | Jer13261 | linksys pap2-na ....nice device |
| 20:04.59 | jhiver | what does it do? |
| 20:04.59 | jaiger | Rival, define cheap |
| 20:05.19 | bjohnson | Jer13261: I want to get one (given to me) .. they look slick |
| 20:05.23 | steve123 | ok peoples... i need to go now.... |
| 20:05.24 | jaiger | I like the sipura 2000 ata |
| 20:05.26 | steve123 | cu later! |
| 20:05.28 | NTJOCK | hey gang |
| 20:05.32 | Jer13261 | they are nice ... |
| 20:05.36 | Jer13261 | i have 5 in use atm |
| 20:05.38 | SpaceBass | been looking at the linksys myself does it have one or two fxs? |
| 20:05.42 | Jer13261 | 2 |
| 20:05.43 | Zeeek | SpaceBass for info, I have set all my SIP devices to begin RTP ports at 10000 |
| 20:05.43 | Katty | hmm |
| 20:05.45 | jhiver | what 'nice'? |
| 20:05.49 | jhiver | they look nice? |
| 20:05.52 | NTJOCK | I have this problem with faxes coming in across my Zap interfaces. (FXS to FXO). I only get part of them. |
| 20:05.56 | jhiver | they have fancy colors? |
| 20:05.56 | SpaceBass | afe thans, I'll try those ports |
| 20:06.00 | afe | SpaceBass: so try opening those UDP ports both ways |
| 20:06.02 | jhiver | they do QOS? VPN. |
| 20:06.04 | Zeeek | NTJOCK common |
| 20:06.05 | jhiver | ? |
| 20:06.06 | Jer13261 | they looks nice ......yes BUT only 1 call w/ g729 |
| 20:06.11 | NTJOCK | anything I can do with it? |
| 20:06.13 | NTJOCK | or to fix it? |
| 20:06.21 | NTJOCK | like slow down the fax? |
| 20:06.25 | Zeeek | other than purchasing a $35 fax machine... |
| 20:06.28 | NTJOCK | Does spandsp work better? |
| 20:06.28 | bjohnson | jhiver: they look nice (AND have fancy colors) .. and are small |
| 20:06.38 | NTJOCK | hmm, I have a nice Brother fax machine.... |
| 20:06.38 | Jer13261 | yea its blue :) |
| 20:06.40 | Jer13261 | hehee |
| 20:06.41 | bjohnson | jhiver: they are SPA 2000 in new packaging |
| 20:06.46 | jhiver | duh... |
| 20:06.47 | Jer13261 | not an MFC i hope lol |
| 20:06.53 | mmlj4 | or use hylafax, it's free and great |
| 20:06.56 | jaiger | NTJOCK, are the fxo/fxs ports on the same physical * box? |
| 20:06.58 | Zeeek | NTJOCK I misread your post, sorry |
| 20:06.59 | NTJOCK | yes |
| 20:07.01 | NTJOCK | no probelm. |
| 20:07.05 | NTJOCK | possibly same card |
| 20:07.06 | SpaceBass | last question (for now) anyone use the ipicasso phones? |
| 20:07.15 | NTJOCK | we have 2 zap cards, 5 fxo, 3fxs |
| 20:07.17 | jaiger | I do that with no problem |
| 20:07.26 | jhiver | What I'd really like to see would be a TDM4xxP soekris-like system with a debian-based linux distro on a CF card |
| 20:07.33 | NTJOCK | I think echo cancel is on.... |
| 20:07.40 | jhiver | would make an awesome PBX system |
| 20:07.48 | jaiger | echo cancel should auto-disable when it hears a fax tone |
| 20:07.50 | NTJOCK | it's really wired... with ECM disabled it's crap... with ECM I get the first 1/3rd of a page. |
| 20:07.59 | NTJOCK | I tried installing Spandsp. |
| 20:08.01 | NTJOCK | it wasn't pretty |
| 20:08.07 | Zeeek | jhiver throw in world peace and an end to hunger |
| 20:08.16 | NTJOCK | and it still won't load even though it compiled, and I completed the treasure hunt. |
| 20:08.25 | jhiver | well potentially you can do it yourself... |
| 20:08.25 | BruceX | ny1 has any idea why i cant get atxfer works for the called party ?? (i tweak the code, and i can see the channel_bridge fucntion copying only frames from caller->called, not viceversa) .... |
| 20:08.25 | Zeeek | that's why I misread your post - I thought you WERE using spandsp |
| 20:08.26 | NTJOCK | when * gets to the TX or RX module it just dies.... |
| 20:08.30 | jontow | grr.. can't get this *one* iax registration to work |
| 20:08.33 | jhiver | I'll have to take the time to do that one day |
| 20:08.38 | jontow | out of 8 this is the only one that doesn't fly.. go figure :) |
| 20:08.47 | bjohnson | Zeeek: and a nude girl parachuting out of the sky to land in his lap |
| 20:08.49 | jhiver | might be a bit expensive but so nice... |
| 20:09.10 | jhiver | what, you don't have nude girl parachuting out of the sky landing on yours? |
| 20:09.16 | jaiger | NTJOCK, mine works fine on the same wct100p card to a brother fax machine |
| 20:09.18 | jhiver | what's wrong with you man? :) |
| 20:09.31 | bjohnson | jhiver: I don't llive in your world |
| 20:09.36 | NTJOCK | hmm, i have two fax modems... class 2 is no game... class 1 works. both Hayes. |
| 20:09.42 | NTJOCK | I have a brother fax machine let me route them there |
| 20:09.48 | Zeeek | bjohnson well if we're gonna start there.... |
| 20:09.48 | NTJOCK | we would really like fax to email. |
| 20:09.56 | jhiver | hey that's too bad.. lots of pretty girls on my island :) |
| 20:10.07 | Zeeek | NTJOCK you too could wish for world peace with that |
| 20:10.24 | Zeeek | jhiverwhat are you doing up so late anyway? |
| 20:10.36 | jhiver | but i do admit they don't do parachute |
| 20:10.38 | buddah | i concur |
| 20:10.42 | NTJOCK | well we actually want to eliminate a Windows App |
| 20:10.44 | NTJOCK | :) |
| 20:10.46 | NTJOCK | yeha |
| 20:10.49 | NTJOCK | die microsoft! |
| 20:10.58 | Zeeek | nakedness would lead to world peace. Wanna know why? Nowhere to hang the canteen |
| 20:11.00 | *** join/#asterisk husher (~andrew@68.143.92.130.nw.nuvox.net) |
| 20:11.06 | jhiver | yeah it should be covered by national insurance (lap dancing, not m$) |
| 20:11.08 | jaiger | lapdances on the house! |
| 20:11.45 | SpaceBass | i was just about to ask for some serious newibe help in dsigning my system |
| 20:11.51 | SpaceBass | but I like the lapdanc topic better |
| 20:11.55 | jhiver | lol |
| 20:12.03 | Zeeek | ~serious |
| 20:12.14 | jhiver | in fact forget about the dance and just keep the lapping bit... errr what am i talking about |
| 20:12.29 | Zeeek | joli jeu de mots! |
| 20:12.31 | *** join/#asterisk zooko (~user@blk-222-185-250.eastlink.ca) |
| 20:12.32 | jhiver | darn it's late, i'll just go fetch a beer |
| 20:12.39 | Zeeek | you don't need one! |
| 20:12.45 | SpaceBass | jhiver where are you ? |
| 20:12.46 | jontow | wtf.. it worked when i just set 'em both to dynamic and used register -> .. |
| 20:12.53 | jhiver | reunion island, gmt+4 |
| 20:13.01 | Zeeek | what phone jontow? |
| 20:13.03 | bjohnson | jhiver: don't forget to get one for your parachutist |
| 20:13.12 | SpaceBass | back to being serious |
| 20:13.37 | SpaceBass | im going to convert my house to voip with asterisk- b/c I have nothing better to do |
| 20:13.45 | NTJOCK | does hylafax work better then spandsp? for receiving? |
| 20:13.57 | SpaceBass | and I'm trying to plan my attack here |
| 20:14.13 | jhiver | my parachutist is asleep |
| 20:14.32 | SpaceBass | will asterisk support distinctive ring? |
| 20:14.42 | jhiver | she's pissed off because I've had to do some cool shit with x less server + wine + nullsoft nsis |
| 20:14.48 | *** part/#asterisk zooko (~user@blk-222-185-250.eastlink.ca) |
| 20:15.04 | jhiver | so i'll drink both beers :) |
| 20:15.23 | afe | SpaceBass: you're planning to use the wifi phone just at home? |
| 20:15.25 | Zeeek | jhiver this is what you need for the island: http://www.zapatatelephony.org/app_rpt.html |
| 20:15.47 | SpaceBass | afe yep |
| 20:15.54 | SpaceBass | afe is that a bad idea for some reason? |
| 20:16.12 | Zeeek | why a wifi phone over an ATA + cordless phone? |
| 20:16.20 | Zeeek | aprat from the geek edge |
| 20:16.23 | afe | SpaceBass: well, afaik wifi phones have lower range and louse battery performance |
| 20:16.28 | SpaceBass | totally about the geek factor |
| 20:16.30 | Zeeek | which is profusely bleeding atm |
| 20:16.40 | Zeeek | the edge I mean |
| 20:16.42 | SpaceBass | can't be worse than my current analogue phone's battery |
| 20:16.50 | afe | SpaceBass: ATA + a good DECT phone would probably be a lot better |
| 20:16.50 | *** join/#asterisk jas_williams (~Jason@host81-155-66-178.range81-155.btcentralplus.com) |
| 20:17.05 | SpaceBass | dect? |
| 20:17.10 | Zeeek | I'm convinced (having read reviews of geeks) that those phones aren't that great |
| 20:17.14 | jhiver | Zeek i'll take a look? in a nutshell, what is it all about? |
| 20:17.24 | afe | SpaceBass: cordless (DECT is a digital cordless standard) |
| 20:17.25 | SexyKen | Hey guys doesnt anyone know how to get the Polycom XML screen to show cool info like queue info |
| 20:17.30 | Zeeek | Ham radio repeater + asterisk |
| 20:17.36 | SpaceBass | Zeeek currenlty have one for a decent price on e-bay... thought i could at least play around wiht it |
| 20:17.50 | Zeeek | SexyKen there's some stuff on the mailing list about the phones |
| 20:17.58 | SpaceBass | going to put a wired phone in each room (IP phone) and have one wifi phone |
| 20:18.05 | Zeeek | SpaceBass may I ask... what price? |
| 20:18.20 | SpaceBass | my wifi coverage in my house is much better than my analog base station coverage |
| 20:18.24 | SpaceBass | Zeeek $100 usd |
| 20:18.35 | Zeeek | ya, for that I'll let you do it |
| 20:18.50 | SpaceBass | what would be too much (so I know my bidding limit ) :) |
| 20:18.54 | Zeeek | with links to pages about what a piece of shit they are |
| 20:18.57 | SpaceBass | of course the WAF on this project is very low |
| 20:19.03 | Zeeek | withholding the links |
| 20:19.06 | afe | SpaceBass: hehe, ok - you won't be able to get a nice cordless and an ATA that cheap :) |
| 20:19.25 | husher | SpaceBass: stay the hell away from the ZyXel WIFi phones |
| 20:19.26 | SpaceBass | afe was about to say that too |
| 20:19.34 | Zeeek | My one experience with eBay sucked |
| 20:19.39 | SpaceBass | ata seems to be about $100 |
| 20:19.44 | SpaceBass | and I will eventually need one |
| 20:19.45 | mgth | husher why? |
| 20:19.47 | Zeeek | there are some for less |
| 20:19.48 | jhiver | Zeeek: same here |
| 20:20.11 | Zeeek | eBay France. I got my money back and all, but what a f$cking waste of time |
| 20:20.22 | SpaceBass | i haven't bought a lot |
| 20:20.26 | Zeeek | I do,'t trust electrocis to eBay deals |
| 20:20.28 | SpaceBass | about 5 or 6 things... |
| 20:20.34 | husher | i have one, and it sucks |
| 20:20.37 | SpaceBass | cico switch for 15$ (guy didnt know what he had) |
| 20:20.48 | Zeeek | hey there are a lot of people that love it and some even are addicted the act of selling |
| 20:20.55 | infernall-work | Hey guys, would any be able to recommend a configuration and approx price for an asterix setup if i provide details? |
| 20:20.56 | SexyKen | Anyone know why this wont go to voicemail when I call in: |
| 20:20.56 | SexyKen | exten => 212203xxxx,1,DIAL(SIP/1.200,20,tr) |
| 20:20.56 | SexyKen | exten => 212203xxxx,104,GoTo(voicemail,s,1) |
| 20:21.06 | SexyKen | Says no rule for t, timeout....and gives a fast busy signal |
| 20:21.19 | Zeeek | yeah yiou're missing the wildcard ____ |
| 20:21.23 | *** join/#asterisk Remowylliams (~Mare@168.215.138.106) |
| 20:21.29 | SpaceBass | ebay is a good source for stuff like $100 PCs to run asterisk on |
| 20:21.32 | SexyKen | What do you mean? |
| 20:21.47 | Zeeek | is this one number or several with XXX on the end? |
| 20:21.49 | Remowylliams | Hello all. |
| 20:22.03 | SexyKen | Zeeek -- it's one actually number just didn't want the entire world to see it. |
| 20:22.03 | Beirdo | SexyKen: shouldn't it be 102, not 104? |
| 20:22.07 | jas_williams | SexyKen: change exten => 212203xxxx,104,GoTo(voicemail,s,1) to exten => 212203xxxx,102,GoTo(voicemail,s,1) |
| 20:22.09 | Zeeek | ok |
| 20:22.09 | SexyKen | Second or 3rd place I've asked ;-) |
| 20:22.19 | Beirdo | aka 1 + 101 |
| 20:22.19 | infernall-work | SexyKen- Detroit area? |
| 20:22.21 | Zeeek | 102 |
| 20:22.30 | Zeeek | right |
| 20:22.30 | infernall-work | er NY |
| 20:22.39 | jas_williams | SexyKen: do you have a context called voicemail |
| 20:23.15 | Remowylliams | I'm curious. Why is the Wildcard X100P listed as discontinued? Is there a replacement model in the works? |
| 20:23.35 | Zeeek | replaced by the TDM FXO modules |
| 20:23.47 | SexyKen | jas_williams I have a 'macro-voicemail' context |
| 20:24.08 | Zeeek | not the same thing |
| 20:24.17 | jas_williams | SexyKen: then you need to goto(macro-voicemail,s,1) |
| 20:24.37 | SpaceBass | Remowylliams you can get a x100p on ebay for $7 |
| 20:24.42 | SpaceBass | fyi |
| 20:24.42 | Zeeek | SexyKen what were the first two places you asked in? |
| 20:24.53 | SpaceBass | still curious if the i.picasso phone will work with asterisk |
| 20:24.56 | SpaceBass | cannot find any info on it |
| 20:25.15 | Remowylliams | So those that get the Wildcard X100P cards while they last are the lucky ones. |
| 20:25.16 | SexyKen | exten => s,1,SetVar(VMBOX=${MACRO_EXTEN}) |
| 20:25.16 | SexyKen | exten => s,2,GoTo(voicemail,s,1) |
| 20:25.20 | Beirdo | SpaceBass: + $15 shipping or so |
| 20:25.22 | SexyKen | That's my macro thing. |
| 20:25.41 | SexyKen | What should my incoming exten => look like now |
| 20:25.53 | *** join/#asterisk harryvv (~leonardo@S010600055d210201.vs.shawcable.net) |
| 20:25.58 | jas_williams | Macro wont work like that either |
| 20:26.09 | SpaceBass | Beirdo yeah... good point... i just got 2 for $25 total |
| 20:26.14 | bjohnson | SexyKen: use macro() instead of goto |
| 20:26.27 | afe | SexyKen: if you want to go to a macro you should not goto but do a Macro(voicemail) |
| 20:26.48 | bjohnson | SexyKen: otherwise it isn't a macro .. just another context |
| 20:27.48 | SexyKen | Still get this: Mar 17 14:33:48 WARNING[30522]: pbx.c:2016 ast_pbx_run: Timeout, but no rule 't' in context 'incoming' |
| 20:27.48 | SexyKen | <PROTECTED> |
| 20:27.49 | *** join/#asterisk JerJer[mobile] (~jj@feth100-fw.fament.net) |
| 20:27.56 | Remowylliams | It's cool the TDM400p will be availible I don't however see them on Ebay or a price for them. |
| 20:28.01 | SexyKen | Not sure what that means. |
| 20:28.19 | *** join/#asterisk Evanrude|w (~david@ama-dslst.69616131.amaonline.com) |
| 20:28.26 | afe | SexyKen: do you want the macro to kick in when busy, or when the extension is not answered? |
| 20:28.37 | jas_williams | SexyKen: why are you using a macro at all ? whynot just exten => 212203xxxx,102,Voicemail(u${EXTEN}) |
| 20:28.41 | Zeeek | SpaceBass that looks interesting but they never mention SIP anywhere - only than any existing installation can be UPGRADED to talk to the phone |
| 20:28.43 | Evanrude|w | hey |
| 20:28.54 | SpaceBass | Zeeek where are you seeing that? |
| 20:28.55 | Zeeek | it looks like ti might work |
| 20:29.01 | Jer13261 | hey can someone help me with FWD? :/ |
| 20:29.05 | Zeeek | on one of the first three lines of a simple google |
| 20:29.08 | SexyKen | •afe• When the ext is not answered. |
| 20:29.13 | Zeeek | * |
| 20:29.15 | SpaceBass | Zeeek what did you google? |
| 20:29.16 | Zeeek | <PROTECTED> |
| 20:29.22 | Evanrude|w | i have a question for anyone who has a Poloycom IP500 phone... |
| 20:29.23 | SexyKen | jas_williams I'll use whatever is best. |
| 20:29.38 | afe | then you need to use priority 2 instead of 104 |
| 20:29.40 | bjohnson | SexyKen: you have a GoTo(voicemail,s,1) INSIDE macro-voicemail? I think you're creating a loop |
| 20:29.48 | SpaceBass | Zeeek thanks! |
| 20:29.49 | jas_williams | SexyKen: do you want to go to voicemail on busy or no reply or both ? |
| 20:29.59 | harryvv | nice looking phone. |
| 20:30.00 | Zeeek | SpaceBass let us know, looks geeky |
| 20:30.04 | SexyKen | jas_williams - both busy and no reply |
| 20:30.05 | harryvv | zeek, have one of those? |
| 20:30.12 | Rival | are some of the linksys pap2 devices locked to certain providers? |
| 20:30.17 | bjohnson | SexyKen: why don't you just use voicemail() anyway |
| 20:30.27 | afe | SpaceBass: I would suggest you take a look at some of the example dial plans at voip-info.org |
| 20:30.53 | SpaceBass | afe, i will thanks! been reading digium's handbook... trying to understand the terms and how dialplans work |
| 20:30.55 | Evanrude|w | has anyone encountered a problem when dialing into systems with IVR (menus when you type an extension, password, etc.) and not being able to enter anything |
| 20:31.01 | Evanrude|w | I am having this problem on my IP500 |
| 20:31.06 | SpaceBass | Rival some are, some arent |
| 20:31.09 | afe | SpaceBass: wasn't really intended for you ;) |
| 20:31.15 | Chuji | Hmm, anyone know why I would still be having signifigant jitter on my Sipura 2000 even though ping times look great? |
| 20:31.16 | afe | I meant that for SexyKen |
| 20:31.17 | harryvv | hi bjohnson |
| 20:31.30 | Chuji | VPN wouldn't cause jitter would it? |
| 20:31.42 | harryvv | ch, is that outgoing into a voip carrier? |
| 20:31.43 | SexyKen | jas_williams, What do you think it should be if I want to go to voicemail on busy & no reply |
| 20:31.45 | bjohnson | SexyKen: here's my plug .. use the superdial macron on the wiki. It'll make your life easier |
| 20:32.08 | bjohnson | harryvv: hi |
| 20:32.11 | Evanrude|w | if anyone has any ideas please p/m me, I am going to be afk for a while |
| 20:32.28 | Remowylliams | Holy Buttfloss! $85.00 for an FXO module. |
| 20:32.31 | SexyKen | Can anyone suggest a Gruopware to use for projectmanagement? |
| 20:32.33 | bjohnson | SexyKen: I accept thanks in the form of nude, parachuting women |
| 20:32.51 | tzanger | damn |
| 20:32.56 | tzanger | I need to start accepting that kind of currency |
| 20:33.15 | Zeeek | why the need for a parachute? |
| 20:33.23 | Chuji | tzanger : you working on the jitter buf? |
| 20:33.26 | Zeeek | place them gently on the doorstep |
| 20:33.28 | Juxt | achievo <- nice groupware |
| 20:33.32 | bjohnson | Zeeek: it's the whole image thing |
| 20:33.33 | Juxt | achievo.org |
| 20:33.34 | Beirdo | so it doesn't hurt when they land on you, Zeeek |
| 20:34.06 | bjohnson | Zeeek: parachute settling over the 2 of us .. how 007ish |
| 20:34.35 | tzanger | Chuji: sort of |
| 20:34.38 | Moonwick | wee, looks like nufone's fucked up again |
| 20:34.43 | bjohnson | SexyKen: doublechocolatelatte is a web based project management app |
| 20:34.44 | tzanger | Juxt: I prefer exchange4linux myself |
| 20:34.45 | jas_williams | SexyKen: look at this macro it will do what you want http://pastebin.ca/7663 |
| 20:34.49 | tzanger | dcl I never liked that |
| 20:34.55 | bjohnson | me either |
| 20:34.57 | bjohnson | but other do |
| 20:35.05 | Moonwick | just got cut off in the middle of a call, and my redials net a "no one is available to answer at this time" |
| 20:35.13 | bjohnson | I use phpgroupware .. but it isn't really project management |
| 20:36.55 | clive- | is anyone here using ASTC? |
| 20:36.57 | Beirdo | phpgroupwhore? |
| 20:37.07 | clive- | ASTCC I mean |
| 20:37.25 | jas_williams | SexyKen: to call an extension exten = exten => 212203xxxx,1,Macro(stdexten,${EXTEN},${EXTEN}) |
| 20:37.37 | BruceX | ny1 has any idea why i cant get atxfer works for the called party ?? (i tweak the code, and i can see the channel_bridge fucntion copying only frames from caller->called, not viceversa) .... |
| 20:38.20 | tzanger | phpgroupware is *NASTY* |
| 20:38.46 | SpaceBass | it'll do ANYTHING |
| 20:38.53 | tzanger | SpaceBass: hahaha |
| 20:39.04 | SpaceBass | tons of backdoors |
| 20:39.22 | Zeeek | heh |
| 20:39.28 | tzanger | man I don't see the desire to plug a chick in the ass |
| 20:39.31 | tzanger | I just don't see it |
| 20:39.33 | Chuji | tzanger : Since you are familiar with jitter, mind helping try and trouble shoot why I'm seeing so much? |
| 20:39.40 | tzanger | Chuji: sure I can give it a shot |
| 20:39.59 | Beirdo | tzanger: different strokes for different folks, etc |
| 20:40.02 | SpaceBass | what is the syntax to check incoming caller id from PSTN? |
| 20:40.15 | SpaceBass | likeif I want calls from my cell phone to go stright to the voicemail system |
| 20:40.25 | Jer13261 | ext/num |
| 20:40.29 | SpaceBass | thanks |
| 20:40.30 | CosmicRay | Over at http://www.voip-info.org/wiki-Asterisk+tips+911, I'm having trouble understanding how these lines work: |
| 20:40.33 | *** join/#asterisk yertle (yertle@ip68-6-98-122.sb.sd.cox.net) |
| 20:40.36 | jhiver | tzanger -> it's more degrading for women => more fun! (ok that was bad...) |
| 20:40.41 | CosmicRay | exten => s,1,SetVar(SET_EMERG_FLAG=0) |
| 20:40.41 | CosmicRay | exten => s,n(checkavail),ChanIsAvail(${EMERGENCY_TRUNK}) |
| 20:40.41 | CosmicRay | exten => s,n,SetGlobalVar(EMERGENCY=1) |
| 20:40.42 | CosmicRay | exten => s,n,SetVar(SET_EMERG_FLAG=1) |
| 20:40.43 | CosmicRay | exten => s,n(dial),Dial(${EMERGENCY_TRUNK}/${EMERGENCY_NUM}) |
| 20:40.45 | CosmicRay | exten => s,s+2(trunkbusy),GotoIf($[${EMERGENCY} = 1]?inprogress) |
| 20:40.47 | CosmicRay | how does the "n" priority work? |
| 20:40.47 | *** part/#asterisk yertle (yertle@ip68-6-98-122.sb.sd.cox.net) |
| 20:40.51 | *** join/#asterisk donis (donis@office.unique.lt) |
| 20:40.55 | CosmicRay | and what is the deal with "s+2(trunkbusy)"? |
| 20:40.55 | Chuji | tzanger : <asterisk> -->Firewall VPN-> Internet <- Firewall VPN <-Cablemodem <- <Sipura 2000> |
| 20:41.10 | elvisthedj | man.. ulaw is a hog |
| 20:41.20 | Chuji | tzanger : That is my setup. I get jitter on the sipura tx'ing back to asterisk. Both over 711 and 729 |
| 20:41.23 | tzanger | elvisthedj: no way, try slinear :-) |
| 20:41.34 | Chuji | tzanger : But only one way jitter. Is that odd? |
| 20:41.39 | bjohnson | tzanger: I like it a lot |
| 20:41.42 | elvisthedj | i thought it was 64kbs... that must be w/ tos=mincost cuz i'm on reliability and it's costing 12.5KB :/ |
| 20:41.46 | tzanger | Chuji: what network |
| 20:41.55 | harryvv | Cosmic is that for 911 |
| 20:41.59 | Jer13261 | is there anyway i can TEST g729? besides buying it lol |
| 20:42.06 | elvisthedj | tzanger: I'm using sip w/ fwd ... i think ulaw is all they do |
| 20:42.06 | CosmicRay | harryvv: yeah, it's from a 911 example |
| 20:42.08 | Zeeek | I don't suppose anyone here has .NET 2.0 ? |
| 20:42.15 | Chuji | tzanger : Qwest on my side, and RoadRunner on the other |
| 20:42.22 | tzanger | elvisthedj: I'm just saying if you think ulaw's bad try slinear |
| 20:42.27 | *** join/#asterisk _Raptor_ (RaptorX@dsl-082-083-169-005.arcor-ip.net) |
| 20:42.32 | _Raptor_ | hi |
| 20:42.41 | tzanger | Chuji: well the SIP jitter buffer's not implemented unless you're using zoa's patches |
| 20:42.42 | Remowylliams | Raptor: Hi |
| 20:42.45 | donis | can i use my 56k modem with asterisk to connect to outside phone line? |
| 20:42.53 | Chuji | tzanger : Smokeping shows about 80ms average ping and no packet loss |
| 20:42.53 | bjohnson | Jer13261: get your sugar daddy to buy it for you |
| 20:42.54 | jas_williams | donis: no |
| 20:43.04 | harryvv | Cosmic if thats also for a larger network oh phones I highly recomend you also put in one analog phone in for purpouses like that and make it only a emergency phone. this in the case the pbx fails in the worst of times. |
| 20:43.10 | Moonwick | maybe what the voip industry needs is some good ol' fashioned regulation. |
| 20:43.19 | Chuji | Zeeek : I'm sure we have it on our MSDN disks |
| 20:43.28 | elvisthedj | tzanger: Oh :D I'm gonna change my tos and see if i can get it closer to 8KB so I can squeeze another call in. I can't believe ipkall/fwd allows 5 connects... crazy (you know.. since it's free) |
| 20:43.29 | Chuji | Zeeek : Is it not publicly available as a beta? |
| 20:43.34 | Zeeek | come to thinkof it I didn't notice it on mine |
| 20:43.37 | Zeeek | sure |
| 20:43.46 | bjohnson | Moonwick: yeah .. things always get better when politicians get involved :P |
| 20:43.53 | harryvv | outa here |
| 20:43.54 | tzanger | Jer13261: for $10 just try it, jesus christ I bet you'll spend more than that for a one night stand |
| 20:43.58 | Zeeek | I haven't tried it but since there was an intersting asterisk monitor app posted I was interested |
| 20:44.01 | Moonwick | bjohnson: my fucking POTS line doesn't drop out in the middle of a call. |
| 20:44.13 | tzanger | elvisthedj: uh your TOS won't change the bitrate |
| 20:44.19 | Jer13261 | well i want to test it between 8 places |
| 20:44.19 | Jer13261 | lol |
| 20:44.23 | Moonwick | or give me problems where I can hear the person on the other end, but they can't hear me. |
| 20:44.24 | Zeeek | Chuji fact is, I hate updating .NET and I hate .NET :) |
| 20:44.26 | Jer13261 | its per connection right? |
| 20:44.29 | elvisthedj | tzanger: i thought that set the quality.. |
| 20:44.31 | tzanger | Jer13261: for a test test it one place ot the other ONE at a time |
| 20:44.32 | Jer13261 | or is it per NIC? |
| 20:44.35 | Chuji | tzanger : Yeah, I'm not real excited about trying the patches, I'm just curious why I'm seeing jitter on my call. Doesn't seem like I should given the ping times |
| 20:44.38 | tzanger | elvisthedj: no it sets the type of service field for the packet |
| 20:44.52 | tzanger | Jer13261: you said you wanted to test it... I assumed that meant call quality |
| 20:45.01 | Jer13261 | right... |
| 20:45.01 | tzanger | Chuji: ping has *nothing* to do with jitter |
| 20:45.04 | tzanger | at all |
| 20:45.08 | *** join/#asterisk bah (048830696@ACAB3DD9.ipt.aol.com) |
| 20:45.19 | bjohnson | tzanger: he's joking |
| 20:45.31 | Jer13261 | hmmmmmm n/m i know how i can test it ....duh studpid me |
| 20:45.35 | elvisthedj | tzanger: is there a way to adjust the quality, or is ulaw just ulaw.. if it is, i don't understand why i'm using 12.5K upload per connect when it should be 8K |
| 20:45.42 | tzanger | bjohnson: who |
| 20:45.46 | tzanger | ulaw is ulaw |
| 20:45.54 | Chuji | tzanger : What's a good way for me to troubleshoot it then? |
| 20:45.56 | tzanger | now jitter, packet loss, delay... they all play in to call quality |
| 20:46.08 | _Raptor_ | i have a problem concering oh323 conferencing: in my extentions.conf is [voip-h323] and one line below exten => conf,1,meetme(100) and the context in oh323.conf is set to voip-h323 but when calling asterisk it is falling back to s and at least to default where it plays me the demo |
| 20:46.10 | Juggie | elvisthedj, its called network overhead |
| 20:46.13 | bjohnson | tzanger: Jer13261 |
| 20:46.13 | tzanger | and different codecs can handle those network issues differently |
| 20:46.15 | jas_williams | ulaw is about 80k bidirectional |
| 20:46.17 | Juggie | every packet needs a rtp and ethernet header |
| 20:46.20 | tzanger | iLBC, for example, has insane MOS for packet loss |
| 20:46.24 | jas_williams | with headers |
| 20:46.25 | Essobi | Mar 17 15:37:02 DEBUG[511]: channel.c:498 ast_channel_walk_locked: Avoiding initial deadlock for 'H323/ <-- anyone offer any insight to that before I start digging in channel.c? |
| 20:46.27 | tzanger | jas_williams: exactly |
| 20:46.39 | Juggie | ulaw is actually 81.7kbps i believe :) |
| 20:46.50 | jas_williams | :P |
| 20:46.59 | BruceX | 222 |
| 20:47.00 | BruceX | ny1 has any idea why i cant get atxfer works for the called party ?? (i tweak the code, and i can see the channel_bridge fucntion copying only frames from caller->called, not viceversa) .... |
| 20:47.34 | Jer13261 | BruceX what extactly do you mean? |
| 20:47.49 | BruceX | i want the called party to make atxfer |
| 20:47.52 | *** part/#asterisk chevyman440 (~matt@65.122.76.58) |
| 20:47.54 | *** part/#asterisk xilch (~xilch@66.239.17.226.ptr.us.xo.net) |
| 20:47.55 | BruceX | as when you call to the operator |
| 20:48.03 | elvisthedj | Juggie: ty. I'm aware of overhead, 50% just seemed excessive. But I gotcha |
| 20:48.16 | tzanger | elvisthedj: it's just the nature of the beast |
| 20:48.21 | Jer13261 | make sure you have T or t dependning on which way your dialplan is configured |
| 20:48.22 | _Raptor_ | i don't know whether it is a problem with oh323 or with meetme |
| 20:48.26 | BruceX | doesnt work |
| 20:48.28 | tzanger | UDP has specific packet overheads no matter what the data size is |
| 20:48.34 | BruceX | i've tried both, just t, and just T |
| 20:48.37 | *** join/#asterisk r3d5un (~r3d5un@83-64-19-110.dynamic.adsl-line.inode.at) |
| 20:48.44 | Jer13261 | what ver of * you running |
| 20:48.53 | BruceX | latest cvs |
| 20:49.00 | bjohnson | parachutes |
| 20:49.23 | BruceX | i've been playing with the code, and noticed that it is only checking frames on the originating peer |
| 20:49.47 | Remowylliams | Errr Ok going to have to stock up on OEM X100P FXO PCI cards if they turn out to work decent. As the world is runningout and it's going to get pricey from here on out once the cards are gone. |
| 20:50.30 | Zeeek | doesn't the N under NAT in sip show peers mean "no"? If this is wrong, what does it mean? Naked? |
| 20:50.39 | _Raptor_ | == Starting OH323/R28812 at voip-h323,s,1 still failed so falling back to context 'default' << can anyone tell me what could be the reason for this |
| 20:50.41 | bjohnson | parachutes |
| 20:50.42 | Zeeek | because my phone says N |
| 20:50.51 | Zeeek | but the peer entry has nat=yes |
| 20:51.03 | Zeeek | I wonder where this is coming from? |
| 20:51.13 | Essobi | _Raptor_ chan_h323 or oh323? |
| 20:51.22 | _Raptor_ | Essobi: oh323 |
| 20:51.32 | Zeeek | I wonder where I'm coming from? |
| 20:51.47 | Essobi | No idea. |
| 20:51.56 | _Raptor_ | :-( |
| 20:52.02 | ACiDV | N = NAT ... not NO |
| 20:52.09 | Zeeek | ah the answer! |
| 20:52.12 | Zeeek | thank you |
| 20:52.20 | Remowylliams | No nat nat no N nat nat Yes No. |
| 20:52.23 | Essobi | oh323 doesn't support transfers properly.. I don't use it. |
| 20:52.24 | Remowylliams | OOPS sorry |
| 20:52.28 | bjohnson | where have all the parachutes gone? |
| 20:52.52 | Zeeek | in fact, NOW it's obvious - a programmer would want to save writes tot he output buffer so instead of Y/N it's N |
| 20:53.49 | _Raptor_ | Essobi: but the h323 delivered with debian sarge didnt send any data :( |
| 20:53.53 | *** join/#asterisk DannyF (~wizardone@h49n5c1o848.bredband.skanova.com) |
| 20:54.10 | bjohnson | my cat says meow |
| 20:54.57 | *** join/#asterisk NK123 (~p645@cpe-024-163-078-012.nc.rr.com) |
| 20:55.33 | *** join/#asterisk mozilla (sabre@69.149.209.83) |
| 20:56.15 | NK123 | does the digium/govarion TE410P support MFC R2? |
| 20:56.41 | Zeeek | This is interesting reading if you don't mind opening a Word document: http://www.telradconnegy.com/tips.htm |
| 20:56.59 | Zeeek | Details of bandwidth calculations for the voice streams in a VoIP network |
| 20:57.16 | Katty | you know what's also interesting reading? |
| 20:57.26 | Katty | how to make stirfry. |
| 20:57.30 | Katty | now that's /hot/ |
| 20:57.36 | Zeeek | no, caramel is better reading |
| 20:57.49 | Zeeek | making caramel is my specialty |
| 20:57.59 | Katty | post gifs. |
| 20:58.04 | donis | hm i also can't use my modem to accept incoming calls right? |
| 20:58.11 | *** join/#asterisk bannerman (~bannerman@209.216.176.42) |
| 20:58.18 | Katty | donis: what do you mean? |
| 20:58.20 | Zeeek | sur, wint WIndows |
| 20:58.30 | Katty | donis: Accept incoming calls to do /what/? |
| 20:58.31 | ariel_ | Katty, stirfry is great. I love the Mongolian barbieque. |
| 20:58.42 | Katty | ariel_: that sounds distinctly non-vegan :< |
| 20:58.49 | jontow | ok |
| 20:58.49 | bannerman | Well, I have an X-lite softphone working-- dialing and receiving calls is fine on this softphone and on IAX phones, but my Sipura SPA-1001 just doesn't do any ringing :( |
| 20:58.57 | donis | am. to act like answering machine |
| 20:59.01 | donis | or redirect calls |
| 20:59.06 | *** part/#asterisk NK123 (~p645@cpe-024-163-078-012.nc.rr.com) |
| 20:59.12 | donis | to local or outer phones |
| 20:59.18 | *** join/#asterisk NK123 (~p645@cpe-024-163-078-012.nc.rr.com) |
| 20:59.28 | Zeeek | donis you'll need a real FXO card for that |
| 20:59.53 | bannerman | Katty: Mongolian Barbecue is actually somewhat hospitible for vegans, I guess.. tofu n'stuff.. lots of veggies! |
| 21:00.00 | SexyKen | jas_william,s you there? |
| 21:00.02 | bannerman | at least, my local one is |
| 21:00.14 | *** part/#asterisk NK123 (~p645@cpe-024-163-078-012.nc.rr.com) |
| 21:00.55 | *** join/#asterisk kpfleming (~chatzilla@ip68-3-230-141.ph.ph.cox.net) |
| 21:00.56 | donis | right. |
| 21:00.59 | bjohnson | Zeeek: maple syrup |
| 21:01.03 | donis | thanks a lot |
| 21:01.27 | *** part/#asterisk kpfleming (~chatzilla@ip68-3-230-141.ph.ph.cox.net) |
| 21:01.37 | Zeeek | are we imagining what we'll be pouring on the parachute girls, bjohnson? Have you read any Tom Robbins? |
| 21:02.12 | Zeeek | donis - check out ebay (which I hate) you'll find cheap cards to play around with before you buy the real Digium ones |
| 21:02.13 | bjohnson | stream of sub-conscious thought |
| 21:02.26 | Beirdo | tofu.... shudder |
| 21:02.30 | Zeeek | no it's more hip narrative with some heavy insights |
| 21:02.34 | jhiver | barbecue with... what? |
| 21:02.52 | SexyKen | Who would figure it's this hard to get a voicemail to pickup if there's no answer or it's busy |
| 21:02.59 | Zeeek | In fact a lot of Tom Robbins could make really good sigs |
| 21:03.03 | jhiver | vegetables? |
| 21:03.04 | bjohnson | SexyKen: nobody would .. it's easy |
| 21:03.08 | jhiver | nyeeeeew |
| 21:03.14 | Zeeek | SexyKen why is it so hard? |
| 21:03.18 | bannerman | jhiver: ever been to a mongolian bbq? |
| 21:03.30 | bjohnson | never been .. heard it's good |
| 21:03.33 | SexyKen | Becuase I did what jas_willias told me and it doesn't work. |
| 21:03.34 | jhiver | nope I admit... |
| 21:03.41 | bjohnson | SexyKen: no you didn't |
| 21:03.51 | SexyKen | bjohnson, yes, I did. |
| 21:04.00 | Zeeek | did not! |
| 21:04.16 | bannerman | I don't think I'd enjoy it without using disproportional amounts of carnivore food, but they do cater towards vegetarians as well. |
| 21:04.25 | Silik0n | ok stoopid questions... is there a SPA-841 w/ a builtin 2port switch? |
| 21:04.37 | bjohnson | here's a test .. make a 1000 vm in voicemail.conf, then make an extension like exten => 111,1,voicemail(u1000) .. test it |
| 21:04.58 | bjohnson | Silik0n: no. other brands have phones with switches |
| 21:04.58 | SexyKen | johnson |
| 21:05.13 | SexyKen | I want it to go to voicemail AFTER the phone rings for 20 seconds OR if the phone is busy |
| 21:05.14 | bjohnson | you can just call me big |
| 21:05.16 | Katty | bannerman: oooh... |
| 21:05.20 | Katty | bannerman: post recipe :> |
| 21:05.25 | Zeeek | notch |
| 21:05.40 | Katty | bannerman: also, make sure my /nick is in the hilight so i see it when i come back (((= |
| 21:05.43 | bjohnson | SexyKen: yes I know .. doing my test will show you/us that voicemail is working |
| 21:05.43 | Katty | kthxbi |
| 21:05.59 | jhiver | but you now... /me psyche => barbecue => meat meat meat meat meat meat meat meat meat meat!!!! |
| 21:06.09 | Zeeek | SexyKen there is an example from the 1940's that's in almost every dilpaln on earth, the oneline macro. Haven't you seen that? |
| 21:06.16 | SexyKen | bjohnson - uh voicemail works for other extensions. |
| 21:06.57 | bjohnson | macro-std-exten I believe on the wiki |
| 21:07.07 | Zeeek | that's 1990's version! |
| 21:07.35 | Zeeek | what is the extension-specific problem? Or to say it another way, what is the significant diff between what works and what doesn't? |
| 21:07.43 | bannerman | Katty: Afraid I can't help you, I'm a big fan of canivorous habits and I've never dabbled with the tofu stuff. I do have a friend who is vegan, who loves mongolian- she does crazy tofu stuff that actually looks appetizing. Me, I enjoy my simple recipe: 3x the meat, and lots of spices! |
| 21:08.23 | Remowylliams | Does anyone have experience with the IAXy adaptors. How are they for quality and all that? |
| 21:08.26 | SexyKen | •Zeeek• Whoar e you talking to? |
| 21:08.26 | Zeeek | spice is the key to tofu and that other rubbery stuff with a suspicious foreign name |
| 21:08.46 | SexyKen | Zeeek-- I think I have something called the oneline macro |
| 21:08.49 | Zeeek | SexyKen to you - what exactly is the difference between the ones that do and do not work. Therein lies the 42 |
| 21:08.52 | jas_williams | SexyKen: whynot post the full output from the cli in pastebin.ca so we can see what is happening |
| 21:09.20 | Zeeek | SexyKen that was when I thought no vmail was working |
| 21:09.24 | afe | SexyKen: this does what you want for extension 600: http://pastebin.ca/7664 |
| 21:10.25 | *** part/#asterisk Grooby (~Grooby@66.160.105.186) |
| 21:10.28 | *** join/#asterisk Koshatul (~evangelio@CPE-138-217-191-19.qld.bigpond.net.au) |
| 21:10.46 | rvhi | is dialplan case insensitive? just want to make sure |
| 21:10.54 | Zeeek | jas_williams you might be interested in this if you haven't seen it: |
| 21:11.06 | Zeeek | <PROTECTED> |
| 21:11.08 | jas_williams | afe: no it does not, it plays the unavailiable message at all times |
| 21:11.54 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
| 21:12.16 | SexyKen | Okay I got it to workwith the help of Jer13261 |
| 21:12.20 | afe | jas_williams: well, unless you want different messages for unavailable and busy it works |
| 21:13.55 | bannerman | Question-- do most of you guys just edit your iax.conf, sip.conf, extensions.conf by hand? |
| 21:13.57 | Zeeek | could someone sum up so I can go to bed? |
| 21:14.15 | *** join/#asterisk neopher (~crazy@mail.techhelpresources.com) |
| 21:14.22 | Zeeek | I have my secretary do it |
| 21:14.34 | Zeeek | with notepad |
| 21:14.39 | bannerman | Zeeek -- the answer to rubbery stuff with a suspicious foreign name is "no" imo |
| 21:14.42 | *** part/#asterisk Juxt (~Juxt@adsl-068-213-216-087.sip.bct.bellsouth.net) |
| 21:15.03 | Zeeek | bannerman if you ever come to Paris, we go to a veg place that has a pine nut roast to die for! |
| 21:15.05 | hardwire | anybody using teliax for sip oubound w/o register? |
| 21:15.14 | hardwire | I keep getting a 404 from teliax |
| 21:15.17 | Jer13261 | you will |
| 21:15.18 | neopher | hey guys, is there something special i have to do to get speex to work with asterisk? i compiled and installed speex but it doesn't show up in the translations |
| 21:15.24 | Jer13261 | 404 is not found |
| 21:15.40 | Jer13261 | what does console say |
| 21:15.42 | bannerman | Zeeek: do they have steak, in Paris? |
| 21:15.46 | hardwire | it says 404 not found :) |
| 21:15.53 | Zeeek | steak is better in the USofA |
| 21:15.55 | hardwire | in the sip debug |
| 21:16.01 | *** join/#asterisk Derkommissar (~derk@66.64.215.7.nw.nuvox.net) |
| 21:16.14 | hardwire | Zeeek: what about kobe? |
| 21:16.15 | Derkommissar | I need some help with a TE110P |
| 21:16.23 | Zeeek | I have had it and it is good |
| 21:16.29 | hardwire | I haven't |
| 21:16.32 | Derkommissar | for some reason it keeps saying ZT_CHANCONFIG failed on channel 25: No such device or address (6) |
| 21:16.32 | hardwire | it would be nice |
| 21:16.35 | Zeeek | had it in Japan |
| 21:16.42 | hardwire | good stuff eh? |
| 21:17.00 | Zeeek | Like eating steak-flavored butter |
| 21:17.00 | Derkommissar | :-/ trying to configure it in r2 signaling |
| 21:17.00 | hardwire | Zeeek: :) |
| 21:17.18 | Zeeek | Good thing we don't taste that good or we'd have eaten each other off the planet |
| 21:17.52 | CosmicRay | Zeeek: you're saying steak in japan is bad? |
| 21:18.03 | johnnyb | Is there HEAD on digium CVS the same as the stable head? |
| 21:18.04 | Zeeek | no I'm saying it was too good |
| 21:18.08 | CosmicRay | ah |
| 21:18.11 | bannerman | Zeeek: how can you be sure we don't taste that good? |
| 21:18.18 | Zeeek | because I... never mind |
| 21:18.28 | *** join/#asterisk ManxPower (~eric@ip-209-16-83-10.i-55.com) |
| 21:18.37 | afe | johnnyb: no such animal as stable head |
| 21:18.37 | Zeeek | well, she was taking suppos^H^H^H |
| 21:18.47 | *** join/#asterisk brc_ (~brian@brc.base.supporter.pdpc) |
| 21:19.07 | ManxPower | ~docs |
| 21:19.09 | 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 |
| 21:19.09 | Zeeek | stable head? Yeah, military intelligence |
| 21:19.12 | ManxPower | ~mailinglist |
| 21:19.13 | jbot | from memory, mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ |
| 21:19.40 | *** part/#asterisk sezuan (sezuan@port-212-202-55-249.dynamic.qsc.de) |
| 21:19.54 | Remowylliams | New tact.. my hardware = 300 Mhz PII celeron with 160 megs of ram. Given that I've not been able to find a minimum hardware spec. Do you think this will serve me well or am I on the end of what can be done? The machine works though it takes many seconds if I use the Festival function. |
| 21:20.03 | *** join/#asterisk tessier (~treed@210.245.96.118) |
| 21:20.29 | afe | Remowylliams: what do you want do do with that machine? |
| 21:20.39 | Zeeek | run a call center with 12 lines |
| 21:21.00 | tzanger | JerJer[mobile]: you around? |
| 21:21.22 | *** join/#asterisk JohnWayne (jeff@doublewide.hitechredneck.net) |
| 21:21.27 | johnnyb | afe: The digium support people are asking me to run Asterisk CVS Stable/HEAD and Zaptel CVS Stable/HEAD |
| 21:21.38 | tzanger | johnwayne... wow there's a name I haven't seen in a while |
| 21:21.43 | JohnWayne | tzafrir: howdy |
| 21:21.47 | Zeeek | cough cough |
| 21:21.48 | JohnWayne | err, tzanger |
| 21:21.51 | tzanger | heh |
| 21:21.57 | JohnWayne | damn nick complete |
| 21:21.59 | tzanger | yeah there are a dozen "tz..." names on here |
| 21:22.05 | tzanger | first channel ever for that to happen to me |
| 21:22.08 | JohnWayne | hehe |
| 21:22.15 | afe | johnnyb: they probably want you to test with the stable branch and also with the head branch |
| 21:22.29 | johnnyb | afe: which branch is stable? |
| 21:22.31 | neopher | after installing speex, do i need to recompile asterisk? |
| 21:22.59 | afe | johnnyb: cvs checkout -r v1-0 zaptel libpri asterisk |
| 21:23.14 | tzanger | neopher: yes, I do not have the speex codec in my builds since i don't have ogg libraries |
| 21:23.26 | afe | johnnyb: if you skip the -r v1-0 you'll get HEAD |
| 21:23.26 | JohnWayne | I am javing a problem with IAX hanging and not completing calls, I restart the asterisk process every hour but it doesn't help all the time, it's done this ever since 1.0.0 or so and I am on 1.0.5 now. Anyone else ever run across a similar problem? |
| 21:23.45 | neopher | tzanger: do i need ogg lib's too? |
| 21:23.49 | tzanger | JohnWayne: I think 1.0.5 has serious bugs |
| 21:23.52 | tzanger | try 1.0.6 or 1.0.7 |
| 21:23.53 | sivana | tzanger: |
| 21:23.55 | Remowylliams | afe: I want to accept calls (have a x100p card showing up soon), I want to route calls, handle conferences of 3-12 people, buckets of voice mail boxes and maybe faxing |
| 21:24.07 | tzanger | don't leave me hanging, sivana... |
| 21:24.07 | JohnWayne | tzanger: it did it on 1.0.0, 1.0.1, 1.0.2, etc |
| 21:24.19 | tzanger | it hung on 1.0.1 and 1.0.2 too? |
| 21:24.21 | tzanger | that's odd |
| 21:24.28 | JohnWayne | yes, everything since 1.0.0 |
| 21:24.30 | afe | Remowylliams: might be a bit low on memory |
| 21:24.45 | tzanger | JohnWayne: what's happenning network-side? do you get an Iax2 stream? |
| 21:25.11 | *** join/#asterisk nicknick (~nicholas@81-86-107-241.dsl.pipex.com) |
| 21:25.11 | JohnWayne | tzanger: it stops at this line: -- Called xxxxxxxx@voicepulse/18007425877 |
| 21:25.20 | sivana | heh |
| 21:25.21 | JohnWayne | and just hangs there |
| 21:25.29 | tzanger | JohnWayne: ahh .. voicepulse... try another provider, i don't trust those fuckers |
| 21:25.34 | tzanger | they're the broadvoice of the IAX2 world IMO |
| 21:25.42 | JohnWayne | tzanger: I've tried at least six providers, it does it with all of them |
| 21:25.52 | tzanger | JohnWayne: you can also try calling IAX2/echo@165.154.13.13 (that's my box's echo test) |
| 21:25.57 | Zeeek | tzafrir with all due respect, VP has done well for me outgoing |
| 21:26.15 | Zeeek | but others need to be tried, that's sure |
| 21:26.29 | JohnWayne | I've had problems with everyone except voicepulse |
| 21:26.42 | tzanger | odd |
| 21:26.51 | JohnWayne | some people's sound quality is garbage, some people don't let me send my own outgoing caller id, etc |
| 21:26.54 | Zeeek | voIP mileage is very variable |
| 21:27.02 | johnnyb | Has anyone here had problems with their zap lines not picking up, or them just making a loop back to themselves instead of ringing? |
| 21:27.08 | roamer323 | does VP force compression codec on incoming? |
| 21:27.26 | tzanger | JohnWayne: odd, I use nufone almost exclusively and have no issues... did you try the echo test on my system? |
| 21:27.36 | JohnWayne | tzanger: no, hold on |
| 21:29.00 | *** join/#asterisk WilliamK (~wkeller@c-24-0-130-177.client.comcast.net) |
| 21:29.01 | Zeeek | nite folks |
| 21:29.06 | jhiver | cya |
| 21:29.10 | JohnWayne | ok, it's now hanging at: -- Called echo@165.154.13.13 |
| 21:29.12 | *** part/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
| 21:29.28 | JohnWayne | my phone will timeout eventually |
| 21:30.13 | *** join/#asterisk Gershwin (~fake@63.250.233.162) |
| 21:30.26 | Remowylliams | afe I can beef up the memory no problem but yu think there's enough cpu to go the distance? |
| 21:31.08 | afe | Remowylliams: the CPU might get a lot of work if you do a lot of MeetMe (conferencing) |
| 21:31.35 | Remowylliams | In reality I suspect it won't get used all that much. |
| 21:32.17 | afe | Remowylliams: I suppose you'll have to try, and if it doesn't work get a better box :) |
| 21:32.21 | NTJOCK | Hi tzanger |
| 21:32.53 | *** join/#asterisk admin0 (~admin0@202.161.146.209) |
| 21:33.59 | tzanger | hello |
| 21:34.04 | tzanger | JohnWayne: I didn't see it |
| 21:34.20 | tzanger | JohnWayne: it isn't even leaving your network from what I can tell |
| 21:34.22 | buddah | whats a good place to buy snom phones? and do they get any cheaper than 225ish for the 190s? |
| 21:34.33 | JohnWayne | tzanger: yeah, that's what it looks like |
| 21:34.48 | Remowylliams | afe: Piffle .. ok new tact .. what/where may one find diagnostic tools to monitor and check where problems may lay in a phone system? Such as drops, and jitter and all that. |
| 21:35.05 | JohnWayne | tzanger: think it could be a problem with my firewall? I have the asterisk box DMZ'd, and remote users work most of the tiem |
| 21:35.11 | DannyF | howdy folks, hope everyone is having and excellent day ;) |
| 21:35.14 | tzanger | JohnWayne: pastebin ONLY the [genral] section of your iax.conf, maybe it's something stupid up there, I'm assuming you just said exten => 555,1,Dial(IAX2/echo@165.154.13.13) |
| 21:35.20 | tzanger | in your dialplan (and thus no user or peer entry) |
| 21:35.28 | Remowylliams | HI DannyF |
| 21:35.31 | JohnWayne | tzanger: yeah that's the line I used to dial you |
| 21:35.59 | *** join/#asterisk kahuna_ (~sootroom@rtl-2.i2k.com) |
| 21:36.10 | tzanger | JohnWayne: ok use pastebin.ca and paste just the [general] section of iax.conf |
| 21:36.16 | Hmmhesays | argh "why doesn't <insert generic voip equipment here> work with my nat and <insert crappy service provider here> |
| 21:36.30 | afe | Remowylliams: In my experience trial and error is the best method - if you set it up on a local network it won't cost you anything but time ;) |
| 21:36.31 | Hmmhesays | i'm so sick of that question |
| 21:37.02 | JohnWayne | tzanger:http://pastebin.ca/7667 |
| 21:38.15 | afe | Remowylliams: if you need * for serious use *any* new computer will be able to handle your requirements |
| 21:38.23 | clive- | hmmm, anyone expereince dialstatus saying Cancel, but the call seems to be answered...making billing a headache for me... |
| 21:38.26 | Unrea1 | Does anyone have any advice on SIP or IAX2 Transit providers. My company is looking to move to an asterisk server with VoIP calling. Most likely 5 - 20 calls will be being made at a time. |
| 21:38.36 | tzanger | JohnWayne: uh |
| 21:38.39 | tzanger | comment out that port |
| 21:38.44 | Remowylliams | afe: Arrrg.. I say.. Arrrg .. to cope with insufficient BW, HZ when you don't really even know how good it's suppose to work. Truely this will take time |
| 21:38.48 | JohnWayne | tzanger: ok |
| 21:38.51 | jhiver | hmmhesays what can you do? NAT *is* a real drag... |
| 21:38.57 | tzanger | oh and KILL KILL DIE ERASE ZAP that fucking delayreject |
| 21:38.58 | tzanger | it doesn't work |
| 21:38.58 | *** part/#asterisk admin0 (~admin0@202.161.146.209) |
| 21:39.02 | tzanger | at least it never did when I used it |
| 21:39.09 | heison | ~seen JerJer[mobile] |
| 21:39.16 | jbot | jerjer[mobile] is currently on #asterisk (1h 11m 27s) |
| 21:39.16 | JohnWayne | tzanger: ok |
| 21:39.24 | Unrea1 | anyone? |
| 21:39.28 | tzanger | one of those two should fix it |
| 21:39.36 | afe | Remowylliams: for a SoHo environment your 300Mhz will do fine, if you put some more RAM into it |
| 21:39.37 | tzanger | Unrea1: I highly recommend nufone |
| 21:39.43 | JohnWayne | tzanger: should I set the bindaddr? |
| 21:39.56 | tzanger | Unrea1: BUT they are not known for customer service... but then again I do NOT ever have a problem with them... and I've used them for over a year |
| 21:40.07 | tzanger | Unrea1: about 5000 minutes a month, multiple simultaneous calls, etc. |
| 21:40.13 | tzanger | JohnWayne: not unless you have reason to |
| 21:40.14 | Remowylliams | afe top says I've got about 34MB just hanging in free but thanks. |
| 21:40.24 | JohnWayne | tzanger: hehe, I dunno if I have a reason to |
| 21:40.25 | Unrea1 | Thank You |
| 21:40.29 | Unrea1 | I will look into them |
| 21:40.30 | tzanger | JohnWayne: then you don't :0p |
| 21:40.38 | JohnWayne | tzanger: ok, let me try it now |
| 21:41.05 | tzanger | JohnWayne: I don't know if you can just reload for those changes or if you'll have to reload chan_iax2.so |
| 21:41.23 | JohnWayne | tzanger: it works now, but it could just be because I restarted it. restarting it 2352 times usually cures it temporarily |
| 21:41.35 | JohnWayne | I just restarted it |
| 21:41.40 | tzanger | JohnWayne: odd... you didn't call echo@me... but give that a shot |
| 21:41.50 | tzanger | let me know if it stops working, we'll do some packet tracing |
| 21:42.04 | JohnWayne | I just called you |
| 21:42.47 | Remowylliams | afe: Thanks so much for the information. |
| 21:43.04 | tzanger | JohnWayne: yes I see that |
| 21:43.26 | afe | Remowylliams: I'm glad if I could be of any help :) |
| 21:43.35 | Remowylliams | EEEK I'm using cvs 103 |
| 21:43.48 | tzanger | 103? |
| 21:44.00 | Remowylliams | Asterisk CVS-v1-0-03/08/05-12:44:47 |
| 21:44.08 | tzanger | yikes |
| 21:44.16 | JohnWayne | tzanger: thanks for you help. I commented the restart command out of cron so I will see how long it works |
| 21:44.18 | Remowylliams | It's what came with Asterisk@home |
| 21:44.24 | tzanger | JohnWayne: ok :-) |
| 21:44.47 | Nugget | that's cvs built march 8th, not cvs 103. |
| 21:45.00 | Nugget | it's only nine days old |
| 21:45.19 | *** join/#asterisk mitcheloc (~mitchel@69-169-28-46.anhmca.adelphia.net) |
| 21:45.27 | Remowylliams | Oh |
| 21:46.40 | tzanger | whee |
| 21:46.49 | tzanger | the packet loss concealing jitter buffer just got committed to CVS HEAD |
| 21:47.00 | Nugget | yay |
| 21:47.20 | *** join/#asterisk r0d3nt|m (anonymous@soveliss.luniac.com) |
| 21:47.31 | jhiver | cool, that means it's gonna work better over the crappy connections I have over here :) |
| 21:47.47 | Remowylliams | Kinda makes it hard to figure out what version you've got this way. hmmm |
| 21:49.02 | *** join/#asterisk jontow (jontow@ws.woflsys.net) |
| 21:49.57 | buddah | antone know of a good place to buy snom phones? and do they get any cheaper than 225ish for the 190s? |
| 21:51.51 | *** join/#asterisk _THEEND_ (~DrEaM@host37-42.pool8248.interbusiness.it) |
| 21:52.29 | *** part/#asterisk nicknick (~nicholas@81-86-107-241.dsl.pipex.com) |
| 21:52.29 | _THEEND_ | who uses cisco phone and musiconhold? |
| 21:52.29 | eKo1 | Is there an easy way to determine the number the user dialed when receiving a call from your provider? |
| 21:52.43 | Jer13261 | DNIS |
| 21:52.57 | bannerman | I have 10 people using my asterisk box here locally, using IAX2 through nufone and voipjet for all our calls. I have a full T1 .. am I going ot have to do some QOS in order to not have weirdness and dropped calls? |
| 21:53.04 | eKo1 | what now? |
| 21:53.20 | eKo1 | ~DNIS |
| 21:53.21 | jbot | extra, extra, read all about it, dnis is A telephone service that identifies the number that the caller dialed for the receiver of the call. DNIS is a common feature of 800 and 900 services, and can identify the number originally dialed when multiple 800 or 900 numbers terminate on the same destination trunks. DNIS works by passing the dialed number to the ... |
| 21:53.34 | _THEEND_ | noone uses cisco phone?! |
| 21:54.06 | jas_williams | _THEEND_: I do works fine what version of * are you running ? |
| 21:54.11 | johnnyb | In CVS HEAD, it seems the plc.h is missing. Anyone know anything about this? |
| 21:54.19 | tzanger | haha |
| 21:54.19 | eKo1 | So I'm supposed to get DNIS from my provider right? |
| 21:55.39 | jas_williams | eKo1: yes how are you connected to your provider? |
| 21:56.22 | eKo1 | SIP |
| 21:56.53 | jas_williams | ah! |
| 21:57.39 | *** part/#asterisk Remowylliams (~Mare@168.215.138.106) |
| 21:57.54 | jas_williams | eKo1: when you register send each number to its own extension of the dialled number then route accordingly |
| 21:58.13 | JohnWayne | tzanger: well that didn't last long, calls are hanging again |
| 21:58.24 | tzanger | urgh |
| 21:58.28 | tzanger | call my echo test again |
| 21:58.33 | jas_williams | JohnWayne: are you behind nat ? |
| 21:58.33 | *** join/#asterisk kuj (~kuj@c-67-165-241-16.client.comcast.net) |
| 21:58.49 | kuj | howdy |
| 21:58.53 | JohnWayne | jas_williams: yes, but the asterisk box is in the dmz |
| 21:59.03 | JohnWayne | tzanger: calling, but it's hanging |
| 21:59.03 | eKo1 | jas_williams: How about receiving a call, then using a context to do the trunking? |
| 21:59.07 | tzanger | yeah I see nothing |
| 21:59.32 | jas_williams | JohnWayne: do you have a qualify= in the iax peer ? |
| 22:00.14 | JohnWayne | jas_williams: negative |
| 22:00.25 | eKo1 | But anyways, say I have no clue what number was used. How do I figure that out from my sip provider? |
| 22:00.42 | *** join/#asterisk HoopyCat (user@nocrtucker.netaccnt.net) |
| 22:00.46 | jas_williams | JohnWayne: add qualify=20 and try again. |
| 22:01.03 | tzanger | 20?? |
| 22:01.06 | tzanger | are you mad? |
| 22:01.10 | tzanger | or just spend a fortune on bandwidth |
| 22:01.11 | JohnWayne | 20 seems a little low |
| 22:01.23 | *** join/#asterisk yashax (~yasha_x@69.15.218.218) |
| 22:01.26 | tzanger | I use qualify=750 on my peers (nufone and the like) |
| 22:02.01 | JerJer[mobile] | i use qualify=no |
| 22:02.13 | Beirdo | heya, JerJer[mobile] :) |
| 22:02.15 | JerJer[mobile] | or simply don't put anything |
| 22:02.17 | jas_williams | JohnWayne: it depends how long your nat router keeps the rtp ports open |
| 22:02.19 | tzanger | JerJer[mobile]: that's because you don't give a shit about anyone else :-) |
| 22:02.32 | JerJer[mobile] | yeah, this is true |
| 22:02.34 | JohnWayne | jas_williams: I dunno, it's a linksys |
| 22:02.37 | tzanger | jas_williams: qualify=### does not change the rate of pong messages |
| 22:02.51 | tzanger | it just sets the maximum return time before declaring them unreachable |
| 22:03.01 | jas_williams | sure ? |
| 22:03.07 | tzanger | my * server is behind a WRT54G, just have UDP/4569 pointing to it |
| 22:03.10 | tzanger | jas_williams: yes |
| 22:03.18 | jas_williams | ok hmm... |
| 22:03.31 | *** join/#asterisk Darwin[laptop] (~darwin-la@c-24-3-226-147.client.comcast.net) |
| 22:03.32 | JohnWayne | my asterisk box is dmz'd though |
| 22:03.32 | julianjm | qualify=xxx|no|yes |
| 22:03.32 | julianjm | where XXX is the number of milliseconds used. If yes the default timeout is used, 2 seconds. |
| 22:03.52 | Juggie | you need to set your externip |
| 22:03.54 | Juggie | in sip.conf |
| 22:04.00 | Juggie | so * puts the right ip in sip headers |
| 22:04.04 | tzanger | JohnWayne: that doesn't mean much... you can't have a DMZ with full port forwarding unless you have mulitple IPs |
| 22:04.10 | JohnWayne | Juggie: this is a different problem....remote sip users work fine |
| 22:04.14 | jas_williams | Juggie: He is using iax |
| 22:04.18 | Juggie | whats the problem |
| 22:04.42 | JohnWayne | Juggie: outgoing IAX calls hang randomly, only cure is to restart asterisk a bunch |
| 22:05.15 | mitcheloc | using dyndns is a good way to ensure you kill that internal/external ip problem |
| 22:05.19 | JerJer[mobile] | run cvs -head |
| 22:05.44 | JohnWayne | mitcheloc: I have a static ip |
| 22:05.46 | Juggie | heh... i think someone just broke iax2 in cvs head |
| 22:05.50 | mitcheloc | problem solved |
| 22:06.12 | JohnWayne | I am beginning to think that the problem is the router |
| 22:06.24 | elriah | Anyone here built an asterisk pbx with a via mainboard? I'm using the 1GHz MS10000 with 512MB of ram and a 40GB hard drive. How do you guys think this will perform? |
| 22:06.41 | Juggie | what the hell is a MS10000 |
| 22:07.37 | elriah | An embedded board, Mini-ITX. |
| 22:07.51 | elriah | (motherboard/mainboard) |
| 22:08.09 | *** join/#asterisk Derkommissar (~derko@66.64.215.7.nw.nuvox.net) |
| 22:08.11 | Derkommissar | :-/ |
| 22:08.35 | *** join/#asterisk rephorm (~rephorm@ip67-95-13-60.z13-95-67.customer.algx.net) |
| 22:09.39 | Skysky | elriah: r u using a mini-itx board? |
| 22:09.47 | *** part/#asterisk donis (donis@office.unique.lt) |
| 22:09.51 | elriah | If I wanted to put a couple SIP phones at my house, how many calls could I get over a 3mb upstream/512k downstream DSL connection? No QoS, unfortunately. |
| 22:09.54 | JohnWayne | tzanger: got any more ideas? I guess I could try replacing the router |
| 22:09.56 | elriah | Skysky: Yep. |
| 22:10.23 | JerJer[mobile] | implement QoS then |
| 22:10.26 | JerJer[mobile] | it is trivial |
| 22:10.40 | Skysky | elriah: Im looking for a good looking one for me to use at home.. u have any recommendation of where i can get them? |
| 22:11.22 | johnnyb | elriah: maybe 4, as long as you're not being throttled by your provider. |
| 22:11.28 | bjohnson | elriah: about 10 with gsm |
| 22:11.36 | elriah | Thanks. |
| 22:11.45 | elriah | Skysky: a good looking mainboard? or mini-itx case? |
| 22:12.10 | elriah | The cool thing about the Via Epia MS10000 is it's fanless. |
| 22:13.00 | Skysky | wow.. 3mb upstream can only take 4 calls at once? |
| 22:13.22 | elriah | 3mb downstream, 512k upstream, had it backwards. |
| 22:13.25 | Skysky | elriah: well. both.. |
| 22:13.58 | Skysky | oic.. so. approx 10 kbyte/s/per? |
| 22:14.05 | Skysky | per call? |
| 22:14.22 | bjohnson | for ulaw |
| 22:14.40 | Skysky | wow.. ulaw takes that much bandwidth? |
| 22:14.49 | Juggie | 512k upstreem is only going to do 512 / ~80 |
| 22:14.56 | eKo1 | register => 2345:password@mysipprovider.com/1234 <---- According to the wiki, I should receive calls to the ext. 1234. But what happens when I have multiple 'register => ...' statements with different /XXXX ? |
| 22:15.03 | bannerman | I didn't realize ulaw took that much |
| 22:15.04 | elriah | All the mini-itx cases I've seen are just ugly and look cheap. The only good one I've found is a rackmount unit that holds two mini-itx boards. Here's the link: http://www.casetronic.com/Products/Products_Main.asp Look for the C146. |
| 22:15.05 | JunK-Y | ulaw is 64k+overhead. |
| 22:15.29 | *** join/#asterisk calvinhp (~calvinhp@rrcs-24-123-25-236.central.biz.rr.com) |
| 22:15.44 | elriah | Or a hush system, but they are pricey... by far the best quality/looks. |
| 22:15.59 | Juggie | say with * behind a nat, when i push the rtp ports from the nat device to the * system, do i need to allow that traffic in both directions, or just nat->dmz |
| 22:16.02 | elriah | www.hushtechnologies.net |
| 22:16.08 | Juggie | er, sorry.. nat => asterisk |
| 22:16.26 | bannerman | Is there a decent tool for measuring how much bandwidth I'm currently using? |
| 22:16.37 | Juggie | for linux? |
| 22:16.39 | bannerman | yeah |
| 22:16.42 | Juggie | iptraf |
| 22:16.46 | DannyF | eKo1, i use a genereic incomming context instead |
| 22:16.47 | bannerman | alright, thanks Juggie |
| 22:16.52 | Juggie | if your distro is recent, its probally installed |
| 22:16.54 | Katty | hmm |
| 22:17.19 | eKo1 | DannyF: I have but it doesn't match any of the extensions I have in there.ç |
| 22:17.22 | DannyF | then match on incomming ${EXTEN} works like a charm |
| 22:17.42 | eKo1 | except for s of course. |
| 22:17.47 | *** join/#asterisk mgth (~chatzilla@dsl093-001-038.det1.dsl.speakeasy.net) |
| 22:18.03 | mgth | is a Signalling Type of FXS Kewlstart for a line going to the phone company |
| 22:18.04 | DannyF | you have a external provider? |
| 22:18.09 | *** join/#asterisk MasterYoda (~mnicholso@dhcp-155.digium.com) |
| 22:18.10 | eKo1 | yes |
| 22:18.24 | Skysky | elriah: i think this one is small like mac-mini http://www.mini-itx.com/news/images/story0387-01L.jpg |
| 22:19.14 | DannyF | k when the call comes it in should exten should contain the DID # |
| 22:19.14 | DannyF | extenb should* |
| 22:19.15 | DannyF | sorry it's late here ;) |
| 22:19.15 | elriah | Slick. No PCI slots, though. I need one for my FXO board. |
| 22:19.38 | eKo1 | DannyF: It should, but apparently it doesn't. |
| 22:20.04 | DannyF | make a s,1,NoOp(${EXTEN}) |
| 22:20.06 | eKo1 | Is there a way to find out WTFN is? |
| 22:20.09 | DannyF | and see what comes in |
| 22:20.13 | PTG123 | hey whats the cheapest sip phone that supports g729 you guys would recommend? |
| 22:20.15 | eKo1 | I did that, all I get is s |
| 22:20.22 | DannyF | urk |
| 22:20.34 | DannyF | if u skip the /1234 part too? |
| 22:20.46 | eKo1 | without the /1234 part |
| 22:20.49 | JerJer[mobile] | PTG123: 7960 or nothing |
| 22:21.02 | DannyF | odd |
| 22:21.10 | eKo1 | PTG123: I have a contact a T-Comm that sells some for $55 bucks. |
| 22:21.35 | PTG123 | jejer: i got those for myself, this is for a company who wants to buy a bunch and is too cheap to spring for cisco |
| 22:21.38 | CosmicRay | eKo1: what model is that |
| 22:21.43 | PTG123 | eko1: sells which model? |
| 22:22.12 | eKo1 | CosmicRay: Generic brand. They make a lot of voip shit for a lot of people. |
| 22:22.23 | PTG123 | got a url? |
| 22:22.29 | jas_williams | eKo1: you try looking in this variable and see if it is populated ${SIPCALLID} |
| 22:22.52 | eKo1 | PTG123: Check the T-Comm website. I'm not sure. |
| 22:23.17 | PTG123 | do the sipuras support g729 |
| 22:23.30 | CosmicRay | PTG123: couldn't you check their site? |
| 22:23.42 | PTG123 | doesnt say |
| 22:23.43 | eKo1 | jas_williams: I checked that. Got a long string of nonsense. |
| 22:23.58 | CosmicRay | PTG123: they have datasheets that list all the protocols they speak |
| 22:24.09 | PTG123 | awesome quality, and at 8-10kps a stream,their is no other choice :) |
| 22:24.35 | CosmicRay | I'll give you a hint: http://www.sipura.com/products/spa841.htm |
| 22:24.37 | *** part/#asterisk MasterYoda (~mnicholso@dhcp-155.digium.com) |
| 22:24.37 | johnnyb | PTG123: how does it compare to iLBC -- isn't that just 15kbps/stream? |
| 22:24.40 | CosmicRay | then click on data sheet on the left |
| 22:24.44 | CosmicRay | page 2 :-) |
| 22:24.48 | *** join/#asterisk YoYo (YoYo@dilbert.psknet.com) |
| 22:24.51 | jas_williams | Ah wrong one then |
| 22:24.55 | PTG123 | johnny: uses half the bandwidth, and i like the quality much better |
| 22:24.56 | jas_williams | ekol |
| 22:25.09 | PTG123 | data sheet won't load |
| 22:25.12 | johnnyb | PTG123: Unfortunately, it's patented :( |
| 22:25.15 | PTG123 | pdf error |
| 22:25.16 | CosmicRay | worked fine for me |
| 22:25.21 | jas_williams | eKo1: can you post a sip debug see if I can help you identify the number from that |
| 22:25.23 | eKo1 | jas_williams: I've check almost every variable. It's nowhere to be found. |
| 22:25.24 | PTG123 | johnny: so what $10 a device is no big deal |
| 22:25.30 | CosmicRay | PTG123: it does list g.729a |
| 22:25.38 | PTG123 | CosmicRay: cool |
| 22:25.38 | johnnyb | PTG123: some of us are purists :) |
| 22:25.42 | PTG123 | so would you guys recommend the sipura? |
| 22:25.45 | CosmicRay | PTG123: now go fix your xpdf |
| 22:25.54 | PTG123 | johnnyb: what do they say, you get what you pay for? :) |
| 22:26.01 | PTG123 | CosmicRay: running windows baby, with firefox :) |
| 22:26.08 | PTG123 | CosmicRay: and adob v7 |
| 22:26.13 | johnnyb | PTG123: It's not about the money, actually. It's the FREEDOM, BABY! |
| 22:26.23 | PTG123 | why expect someone to work for free |
| 22:26.27 | eKo1 | jas_williams: I can tell you that the number is in the To: part of the SIP header. |
| 22:26.27 | PTG123 | how does that guy pay his bills? |
| 22:26.31 | CosmicRay | PTG123: "you get what you pay for" hasn't been accurate for any microsoft operating system in the last 2 decades |
| 22:26.45 | johnnyb | PTG123: again, IT'S NOT ABOUT THE MONEY. I would be fine paying more per phone. |
| 22:26.49 | CosmicRay | PTG123: you are equating "no cost" with "freedom" |
| 22:26.51 | PTG123 | CosmicRay: windows was always the cheapest OS :) thats what all the antirtrust stuff is about |
| 22:27.26 | jontow | nice.. my sipura SPA-2000 works great :) |
| 22:27.51 | *** join/#asterisk tuxinator_linuxM (~tuxinator@m410e36d0.tmodns.net) |
| 22:28.09 | eKo1 | The SPAs are pretty reliable since they're made by the same people that made the ATA-186. |
| 22:28.11 | CosmicRay | PTG123: http://www.gnu.org/philosophy/free-sw.html |
| 22:28.26 | pigpen | Question: How might I setup a company wide page using * and polycom phones? Like "John pickup Line 1" |
| 22:28.27 | CosmicRay | PTG123: there are operating systems that are geniunely free, in all senses of the word |
| 22:28.32 | CosmicRay | PTG123: windows is not free in any sense of the word |
| 22:28.44 | Beirdo | nor cheap |
| 22:28.59 | bannerman | it's cheap in some ways |
| 22:29.01 | CosmicRay | PTG123: and no, antitrust is about using a monopoly in one sector to unfairly influence another sector, or forming alliances that defeat the free market competition |
| 22:29.32 | KalD|WORK | will * compile w/ cygwin? |
| 22:29.44 | CosmicRay | pigpen: there is some stuff on the wiki about that, loko for paging, or auto-answer, or stuff like that |
| 22:29.53 | PTG123 | Cosmic: your giving me a headache |
| 22:30.00 | pigpen | k..I have a bit...but nothing definiative... |
| 22:30.21 | johnnyb | PTG123: We're just trying to brainwash you into our cult. A headache is a normal part of the process... |
| 22:30.21 | *** join/#asterisk jeffik (~jeffik@208-41-192-106.client.dsl.net) |
| 22:30.35 | CosmicRay | PTG123: the reason some of us object to software patents is that it takes away from the ability of intelligent programmers to write intelligent programs without having to pay a bunch of lawyers first. |
| 22:31.06 | CosmicRay | PTG123: the more practical reason is that the USPTO is so stupid and most of the world's legal systems so fscked up that software patents are used by large corporations to muscle smaller competitors away |
| 22:31.14 | jas_williams | pigpen: look at http://lists.digium.com/pipermail/asterisk-users/2004-March/040186.html |
| 22:31.20 | johnnyb | pigpen: can you map the "line" buttons to 701, 702, 703, etc? |
| 22:32.17 | caesar2 | has someone managed to run * on embedded systems ? like http://www.soekris.com/Pictures/net4801_T1_Open_m.jpg ?? my * runs at net4501 -> 133 mhz 80486 but its to slow for isdn or compressed codecs |
| 22:32.18 | jas_williams | eKo1: do you have a separate register line for each sip number ? |
| 22:32.23 | jeffik | All: have a quesion about x-lite behind firewall/router |
| 22:32.24 | *** join/#asterisk Beirdo (~gjhurlbu@beirdo.user) |
| 22:32.26 | CosmicRay | PTG123: windows would not be "Free as in Freedom" even if microsoft charged nothing for it; the lack of source code means that I do not have the freedom to use my own computer in any way I wish |
| 22:32.29 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@l02m-4-165.d4.club-internet.fr) |
| 22:32.34 | pigpen | johnnyb: I guess I am missing your suggestion... |
| 22:33.10 | CosmicRay | PTG123: Asterisk, and Linux, on the other hand, are both Free as in Freedom, although you may pay money to get them. Once you have them, you have the source code, and thus ultimate control over how you use them. And you have the fundamental right to share your knowledge with others in the form of exchanging source code. |
| 22:33.27 | eKo1 | jas_williams: Nope. |
| 22:33.28 | PTG123 | Ok linux is not free :) |
| 22:33.33 | Jer13261 | lol |
| 22:33.36 | PTG123 | If i go to use linux code in something else i get in trouble |
| 22:33.41 | PTG123 | FreeBSD is the only free one |
| 22:33.45 | PTG123 | so why do you use linux? |
| 22:33.45 | johnnyb | pigpen: I may have been misunderstanding what you were saying. I know nothing of Polycom phones, but if your goal is to say, "pick up line one", you can use Asterisk call parking (transfer to 700), which will then park the call on 701, 702, 703, etc. |
| 22:34.15 | CosmicRay | PTG123: Linux *is* free. the Gnu GPL, under which the kernel is licensed, simply says that if you give the source code to anyone else, you must give them the same freedoms you enjoy. |
| 22:34.16 | pigpen | johnnyb: ah..yes...thank...sorry..you had the other half of the equation...thanks. |
| 22:34.39 | PTG123 | Cosmic: no there is alot more involved in the gpl license then that, notice we are not allowed to use gpl code in asterisk? |
| 22:34.39 | johnnyb | PTG123: It's not freedom if you can take it away from someone else. |
| 22:34.59 | PTG123 | your in a channel filled with people who all support a project, that is restricted from using any GPL code |
| 22:35.02 | johnnyb | PTG123: Not really. It's just a bunch of legal language to make sure that happens. |
| 22:35.05 | CosmicRay | PTG123: BSD license does not do that, true. BSD people regard it as greater freedom, centering on an individualistic interpretation. I, and other people, view it as lesser freedom, centering on a more expansive perspective; people's net freedom is decreased if someone has the ability to take your freedom away. |
| 22:35.43 | jas_williams | eKo1: not sure how I can help you then :-( |
| 22:35.48 | CosmicRay | PTG123: no, you are perfectly fine using GPL code in asterisk. asterisk itself is GPL. |
| 22:35.51 | eKo1 | jas_williams: Thanks anyways. |
| 22:36.16 | CosmicRay | PTG123: Digium has made an internal decision that they do not want third-party GPL code in asterisk because then they cannot make asterisk proprietary (non-free) |
| 22:36.25 | eKo1 | If you don't have the freedom to take someones freedom away, then that's not really freedom now is it. |
| 22:36.34 | CosmicRay | PTG123: as holders of the copyright of asterisk, they can also license it to people under licenses other than the GPL |
| 22:36.42 | PTG123 | Cosmic: no you are not |
| 22:36.51 | CosmicRay | I am not what? |
| 22:36.58 | PTG123 | Cosmic: try again, i suggst you talk to the developers to understand.. YOU CAN NOT USE GPOL CODE IN ASTERISK |
| 22:37.03 | PTG123 | LGPL maybe |
| 22:37.06 | CosmicRay | PTG123: you are flat wrong. |
| 22:37.14 | CosmicRay | PTG123: asterisk is licensed under the GPL. |
| 22:37.16 | johnnyb | PTG123: Yes, you can. They just won't accept it into CVS. |
| 22:37.19 | PTG123 | no its not |
| 22:37.24 | PTG123 | base your rants on facts :) |
| 22:37.52 | johnnyb | From asterisk.c: |
| 22:37.54 | johnnyb | <PROTECTED> |
| 22:37.54 | johnnyb | <PROTECTED> |
| 22:37.55 | johnnyb | <PROTECTED> |
| 22:37.55 | johnnyb | <PROTECTED> |
| 22:37.55 | CosmicRay | PTG123: idiot. from README in the asterisk 1.0.6 distribution: |
| 22:37.55 | johnnyb | <PROTECTED> |
| 22:37.57 | johnnyb | <PROTECTED> |
| 22:38.01 | johnnyb | <PROTECTED> |
| 22:38.03 | CosmicRay | * LICENSING |
| 22:38.03 | johnnyb | <PROTECTED> |
| 22:38.03 | CosmicRay | <PROTECTED> |
| 22:38.04 | PTG123 | talk to the developers |
| 22:38.05 | johnnyb | <PROTECTED> |
| 22:38.07 | johnnyb | <PROTECTED> |
| 22:38.08 | PTG123 | its not gpl |
| 22:38.09 | johnnyb | <PROTECTED> |
| 22:38.15 | PTG123 | i don't care what that says |
| 22:38.16 | CosmicRay | oh, and: |
| 22:38.17 | JerJer[mobile] | !kick |
| 22:38.19 | PTG123 | its np gpl |
| 22:38.21 | CosmicRay | The GPL also |
| 22:38.21 | CosmicRay | must apply to all loadable modules as well, except as defined below. |
| 22:38.28 | PTG123 | to contribute i got to sign some stupid disclosures for mark |
| 22:38.32 | PTG123 | giving away all my rights |
| 22:38.37 | tzanger | PTG123: that's not GPL |
| 22:38.39 | tzanger | that's for Asterisk |
| 22:38.42 | CosmicRay | PTG123: you are confusing two very different issues. |
| 22:38.51 | tzanger | because they want to be able to sell closed source licenses as needed |
| 22:38.54 | tzanger | it has NOTHING to do with GPL |
| 22:38.57 | CosmicRay | PTG123: You have the source code to Asterisk, which Digium has given to you under the GNU GPL license. |
| 22:39.15 | PTG123 | I said |
| 22:39.17 | CosmicRay | PTG123: With this source code, you have every right to add other GPL'd projects to Asterisk and distribute the resulting work under the terms of the GPL |
| 22:39.22 | PTG123 | we can not use GPL software in asterisk |
| 22:39.23 | PTG123 | only LGPL |
| 22:39.28 | JerJer[mobile] | wrong |
| 22:39.28 | CosmicRay | PTG123: you are incorrect |
| 22:39.46 | elriah | Is GPL even enforceable in US courts? |
| 22:39.49 | JerJer[mobile] | Digium cannot accept non-disclaimed GPL code into the asterisk source tree |
| 22:39.49 | CosmicRay | PTG123: now, Digium wants to make asterisk available to people under terms other than the GPL also |
| 22:39.59 | PTG123 | JerJer: exactly |
| 22:40.02 | CosmicRay | elriah: how coudl it not be? it is all that gives people the right to copy at all |
| 22:40.19 | bannerman | It's not that I'm lazy, or I dislike my job, or working.. I just really, really, really like doing nothing. |
| 22:40.20 | PTG123 | Nah i doubt they will be able to enforce GPL |
| 22:40.26 | PTG123 | GPL is gonna lose those current lawsuits |
| 22:40.27 | elriah | Well, I have asterisk for sale for $10,000 if anyone wants it. ;) |
| 22:40.37 | CosmicRay | PTG123: So, if digium wants to make asterisk available under terms other than the GPL, they must either 1) own all copyright to asterisk code themselves, or 2) have a license to all code under a license that permits them to do this |
| 22:40.42 | johnnyb | elriah: that's actually legit under the GPL. |
| 22:40.52 | elriah | No shit. ebay, here I come. |
| 22:40.53 | CosmicRay | PTG123: If you send them GPL code that you hold the copyright to, they cannot distribute it under non-GPL terms. |
| 22:41.12 | PTG123 | ok well i really don't want this big of a headache :) |
| 22:41.18 | PTG123 | i don't care, argue amongst yourselves |
| 22:41.22 | CosmicRay | PTG123: how is it going to lose a lawsuit? By default, one has no right at all to reproduce someone else's work |
| 22:41.22 | JerJer[mobile] | it is not copyright |
| 22:41.23 | PTG123 | i'll pay for my g729 and enjoy it |
| 22:41.45 | CosmicRay | PTG123: someone that reproduces it must first stipulate to the validity of the GPL, since it is the only thing that gives them the right to copy it in the first place! |
| 22:41.49 | NTJOCK | He who has the most money usually wins in court. |
| 22:42.01 | NTJOCK | see OJ simpson, Ms. Stewart, and etc. |
| 22:42.02 | elriah | NTJOCK: (sigh) the american way. |
| 22:42.04 | CosmicRay | NTJOCK: he who is right usually wins out of court. |
| 22:42.05 | JerJer[mobile] | you still own rights to the code |
| 22:42.10 | NTJOCK | Not usually |
| 22:42.11 | NTJOCK | sometimes |
| 22:42.12 | CosmicRay | NTJOCK: as the FSF has done on numerous occasions |
| 22:42.18 | NTJOCK | He with the best lawyers usually wins. |
| 22:42.26 | NTJOCK | I'm not saying it's right, just how it is |
| 22:42.29 | eKo1 | He who uses OJ Simpsons lawyer wins. |
| 22:42.49 | NTJOCK | Has anyone used Spandsp and Halyfax (or whatever the other one is called)? |
| 22:42.51 | Nugget | lionel hutz? |
| 22:42.51 | bannerman | NTJOCK: I thought martha stewart did time |
| 22:42.51 | johnnyb | The reason why the GPL hasn't gone to court yet is that it's a pretty big slam dunk against the violators. |
| 22:42.59 | NTJOCK | I'm having hell with spandsp. |
| 22:43.10 | johnnyb | Either (a) the company had license to use it, or (b) they did not. |
| 22:43.10 | NTJOCK | Oh you could call her KY vacation home prison |
| 22:43.13 | NTJOCK | the gov't does. |
| 22:43.18 | caesar2 | would it be ok to use * for military purpose ? |
| 22:43.26 | NTJOCK | but roaming the meadows and microwave gourmet isn't exactly prison. |
| 22:43.27 | denon | caesar2: sure |
| 22:43.36 | johnnyb | If the GPL is invalid, then there is nothing that would give the company license to use it, since copyright law defaults to not allowing this. |
| 22:43.51 | johnnyb | Therefore, whether the GPL is valid or invalid, the authors win in court. |
| 22:44.01 | Nugget | it doesn't actually work that way. |
| 22:44.01 | CosmicRay | johnnyb: exactly |
| 22:44.04 | bannerman | NTJOCK: you're dumb. |
| 22:44.19 | CosmicRay | Nugget: lawrence lessig says it does. |
| 22:44.25 | NTJOCK | bannerman: nah... Martha's time was a joke. |
| 22:44.39 | Nugget | lessig isn't exactly impartial. |
| 22:44.44 | johnnyb | Most EULA's, REMOVE rights, The GPL ADDS rights. |
| 22:44.45 | NTJOCK | bannerman: Lea Fastow is doing time... no sunlight, no visits, lockdown at nite. |
| 22:44.48 | Pinhole | If the GPL is termed invalid, the courts could also rule that the code is public domain. |
| 22:44.48 | JerJer[mobile] | no the lawyers always win |
| 22:44.58 | CosmicRay | Nugget: he doesn't have to be. he has won with that argument countless times. |
| 22:44.58 | johnnyb | Pinhole: on what basis? |
| 22:44.59 | Nugget | Pinhole is correct |
| 22:45.02 | caesar2 | but digium could forbit it if it concure with there understanding in using an pbx system... in my case it would be used to coordinate missle attacks.... |
| 22:45.19 | caesar2 | 0 |
| 22:45.20 | Pinhole | Any modification to public domain is now copyrighted by the person making the change. Thus big company owns code. |
| 22:45.27 | CosmicRay | Nugget: no he's not, the GPL is a license; the copyright owner still holds copyright. |
| 22:45.28 | johnnyb | Pinhole: The courts don't have the authority to proclaim copyrighted code public domain. |
| 22:45.39 | Nugget | that's correct but not relevant. |
| 22:45.45 | *** join/#asterisk tuxinator_linux (~tuxinator@m410e36d0.tmodns.net) |
| 22:45.54 | PTG123 | anyone here use the SPA-841 |
| 22:45.56 | NTJOCK | johnnyb: The gov't taketh what it wanteth. |
| 22:46.01 | PTG123 | before i go and buy 12 of them? |
| 22:46.02 | NTJOCK | and when it wants.... |
| 22:46.17 | johnnyb | Pinhole: has anyone here every heard of the courts taking copyrighted material and proclaiming it public domain? |
| 22:46.34 | NTJOCK | it automatically becomes public domain after X number of year.s |
| 22:46.49 | NTJOCK | Disney spent Millions of $$$ getting the law changed to keep mickey copyrighted. |
| 22:46.53 | johnnyb | NTJOCK: Yes, but that's about 100. |
| 22:46.56 | Pinhole | It's the same as when a person owns a junk of land and the comunity uses it for some purpose. If the owner does not protest for a given time, the community owns the land. |
| 22:47.07 | bannerman | NTJOCK: there was also a little more involved in Lea Fastow's case... |
| 22:47.11 | johnnyb | Pinhole: but there are specific laws governing that. |
| 22:47.19 | NTJOCK | Pinhole: actually it can be condemned and take for public interest. |
| 22:47.24 | NTJOCK | anyhow..... |
| 22:47.29 | Nugget | that's a different matter, NTJOCK. |
| 22:47.43 | NTJOCK | spandsp vs. the other fax halyfax. ( or something like that). |
| 22:47.45 | eKo1 | Hmm...all of a sudden #asterisk has turned into #law. |
| 22:47.50 | tzanger | heh |
| 22:47.52 | NTJOCK | does hylafax actualy work? |
| 22:47.52 | Mavvie | #ianal |
| 22:47.55 | johnnyb | Anyway, by these lines of arguments, there's no such thing as a valid license, anyway. |
| 22:47.55 | tuxinator_linux | NTJOCK: eminate domain |
| 22:48.01 | tzanger | I sure hope stevekstevek gets more than a couple lousy karma points for 2532 |
| 22:48.02 | NTJOCK | tuxinator_linux: exactly. |
| 22:48.16 | Pinhole | A valid license is one with a history of working in court. |
| 22:48.16 | tzanger | ~karma stevek |
| 22:48.16 | jbot | stevek has karma of 3 |
| 22:48.29 | NTJOCK | tuxinator_linux: in the middle east it's emirate domain.... :) |
| 22:48.35 | tzanger | hahahaha |
| 22:48.40 | KalD|WORK | who runs/owns the voip-info wiki? |
| 22:48.41 | tuxinator_linux | he he |
| 22:48.43 | Nugget | hylafax works, but it's a bloated and overcomplicated mess. |
| 22:48.46 | johnnyb | Pinhole: actually, I would say it is one that is so obvious that everyone settles out-of-court |
| 22:48.47 | eKo1 | I wonder what the world would be like without copyrights and licenses... |
| 22:48.49 | NTJOCK | and in new york it's just Irate Domain! |
| 22:48.50 | NTJOCK | ;) |
| 22:48.56 | eKo1 | and lawyers... |
| 22:49.05 | Nugget | eKo1: it would be a world of contracts and hardware dongles. |
| 22:49.09 | NTJOCK | Nugget: damn... I just need a simple way to let faxes into the company. |
| 22:49.27 | NTJOCK | I'm having all sorts of issues routing a call across a FXS and FXO port to a fax modem. |
| 22:49.33 | NTJOCK | at best I get 1/3rd of a page. |
| 22:49.41 | NTJOCK | at worst it looks like the terminator got a hold of the page. |
| 22:49.41 | Nugget | it doesn't take any imagination to dream of what the world would be like with weak or nonexistent intelectual property law. That's what the software industry was like in the early '80s |
| 22:49.45 | Nugget | it sucked. |
| 22:49.50 | Nugget | intellectual, even. |
| 22:49.54 | NTJOCK | Nugget: it's what china is like now! |
| 22:50.04 | johnnyb | eKo1: actually, we'd be pretty screwed without lawyers -- we'd either be governed by a mob or people could use tricky legal tricks to get their way. With lawyers, we at least have a fighting chance back. |
| 22:50.05 | KalD|WORK | ~karma kram |
| 22:50.05 | jbot | kram has karma of 3 |
| 22:50.06 | tzanger | Nugget: true but it's assinine now |
| 22:50.14 | tzanger | ~karma chameleon |
| 22:50.14 | jbot | chameleon has karma of 1 |
| 22:50.20 | johnnyb | eKo1: what we really need to do is start teaching ethics again :) |
| 22:50.29 | tuxinator_linux | IMHO, Copyrighing software to point is a good idea |
| 22:50.31 | Nugget | the current police state of dmca and consumer-unfriendly legistlation is the fault of the pirates, not the fault of industry. |
| 22:50.34 | NTJOCK | ethics, like dinosaurs are extinct. |
| 22:50.41 | tuxinator_linux | but I run mostly OSS |
| 22:50.42 | tzanger | Nugget: I don't buy that for a second |
| 22:50.44 | NTJOCK | at least in politics and law. |
| 22:50.55 | CosmicRay | NTJOCK: how did bush get elected then? |
| 22:50.58 | eKo1 | I guess the bottom line is, people are greedy bastards and we need laws to keep them in check. |
| 22:51.04 | CosmicRay | NTJOCK: and why were people so oppsed to him? |
| 22:51.04 | Nugget | all these horrible laws are a reaction to the widespread ip theft that's taking place right now |
| 22:51.07 | tzanger | DMCA is there because it's easier to legislate a fix than put proper engineering into protecting your propertly |
| 22:51.17 | NTJOCK | don't get me started on Bush. |
| 22:51.23 | NTJOCK | Texas has a no return policy. |
| 22:51.25 | johnnyb | Nugget: it's the fault of the industry. |
| 22:51.28 | Pinhole | except in actually disputes. |
| 22:51.31 | NTJOCK | We don't want him back. |
| 22:51.31 | tuxinator_linux | I think Pirates is a cop out answer for a lack of creatvity |
| 22:51.34 | Nugget | I couldn't disagree more, johnnyb. |
| 22:51.35 | CosmicRay | NTJOCK: well, you made a blanket statement and I gave you a glaring counterexample |
| 22:51.41 | *** part/#asterisk PTG123 (~PTG123@ip68-106-24-139.ph.ph.cox.net) |
| 22:51.47 | NTJOCK | Bush did *NOT* get elected by ethics. |
| 22:52.00 | CosmicRay | NTJOCK: of course he did. Didn't you read the exit polls? |
| 22:52.04 | NTJOCK | Frankly it wasn't like it was good vs. evil in the last election. |
| 22:52.05 | johnnyb | Nugget: can you name any piece of widely pirated software whose company went bankrupt from lack of sales? |
| 22:52.11 | CosmicRay | NTJOCK: now, I disagree with the ethics that did that, but c'mon |
| 22:52.11 | Nugget | if tens of millions of people weren't stealing music and movies as fast as their poor cablemodems allowed, the industry would never have pushed for legislation like dmca. |
| 22:52.14 | NTJOCK | it was more like the lesser of the evils. |
| 22:52.33 | Nugget | the pirate2pirate networks are the loss of control that the dmca seeks to remedy |
| 22:52.33 | tzanger | riiiiiiiiight |
| 22:52.34 | CosmicRay | NTJOCK: ah, still you cannot avoid bringing ethics into it :-) |
| 22:52.36 | johnnyb | Nugget: for authors of software and books alike, obscurity is much more the enemy than pirates. |
| 22:52.42 | Pinhole | elections are *ALWAYS* lesser of evils. |
| 22:53.04 | file[laptop] | I just read that as erections |
| 22:53.06 | johnnyb | Nugget: And yet the music and movie industries are having their best years ever. Doesn't sound like such a tragic situation to me. |
| 22:53.06 | tzanger | Nugget: if the industry put as much effort into engineering proper fixes than feeding their lawyers we'd have TRULY secure networks |
| 22:53.10 | tzanger | truly secure devices |
| 22:53.10 | NTJOCK | P2P is the result of price disparity between what people will pay and what the companies demand. |
| 22:53.16 | CosmicRay | Nugget: if the music and movie industry weren't paragons of monopolistic greed, fearful of change and trying to stifle every innovation, they wouldn't be in this situation either |
| 22:53.20 | tuxinator_linux | How do you copyrigh something that is broadcast to the public? Once it's out there, you can't stop it. |
| 22:53.35 | Nugget | I think you are misunderstanding what "monopoly" means. |
| 22:53.39 | CosmicRay | tuxinator_linux: try telling that to the broadcast networks (see broadcast flag) |
| 22:53.40 | Pinhole | How many soft-phones does it take to change a lightbulb? |
| 22:53.44 | Nugget | please point to the company which is a monopoly in music or film |
| 22:53.59 | Nugget | theft is not "innovation", in any event. |
| 22:54.11 | CosmicRay | Nugget: I suggest that the interests represented by RIAA together form a monopoly |
| 22:54.14 | tzanger | Nugget: well it's impossible to get your song or movie into the mainstream without sucking up to the MPAA/RIAA |
| 22:54.18 | CosmicRay | Nugget: note that I did not say illegal monopoly |
| 22:54.23 | Pinhole | Why did the soft-phone cross the road? |
| 22:54.31 | tzanger | you can always go it alone but all of the mainstream distribution channels are locked down |
| 22:54.34 | CosmicRay | Pinhole: to get to the other sip? |
| 22:54.36 | tzanger | so yes, they are a monopoly |
| 22:54.36 | Nugget | CosmicRay: it's still wrong, even without the word "illegal" |
| 22:54.42 | dersteer | epic was VERY much against FM radio... they banned anyone from playing their music over FM.... look at the industry now... now the pay people... and beg them to play their stuff |
| 22:55.06 | tuxinator_linux | Pinhole: To commit suicide (by getting ran over by a car) |
| 22:55.08 | Nugget | you might be able to make the argument that the RIAA or MPAA is an example of industry collusion, but monopoly isn't even close to the right word. |
| 22:55.25 | Pinhole | How does the US Navy connect VoIP calls? |
| 22:55.30 | dersteer | or how about beta.... or cassette tapes... they were against both of those technologies |
| 22:55.33 | *** join/#asterisk jedaustin (~chatzilla@austin-j.its.dist.maricopa.edu) |
| 22:55.34 | Pinhole | They use SIPernet. |
| 22:55.52 | jedaustin | Arg! Anyone here use Broadvoice and have it working? |
| 22:56.02 | dersteer | broadvoice works here :) |
| 22:56.18 | jedaustin | dersteer: I can't for the life of me get it to register |
| 22:56.39 | dersteer | in your config somewhere... its up and running :) |
| 22:56.52 | jedaustin | ? |
| 22:56.53 | dersteer | actually I've never seen it go down |
| 22:57.16 | eKo1 | All that legalality BS made me hungry. |
| 22:57.29 | *** join/#asterisk jarrod (jarrod@dipole.informationwave.net) |
| 22:57.49 | jarrod | anyone know of a frontend like AMP that supports virtual pbx systems for different functions? |
| 22:58.09 | Evanrude|w | does anyone have a polycom IP500 running the 1.4.1 firmware? |
| 22:58.20 | tuxinator_linux | eKo1: I'm in the car, eating beef jerkey |
| 22:58.21 | tuxinator_linux | eKo1: On my way to Cali (well, I am in Cali now) |
| 22:58.31 | tzanger | yeah I'm starved |
| 22:58.37 | tzanger | my kids proably are too |
| 22:58.40 | tzanger | er |
| 22:58.41 | tzanger | probably |
| 22:59.53 | *** join/#asterisk calvinhp (~calvinhp@cpe-65-29-88-222.indy.res.rr.com) |
| 23:01.09 | jedaustin | Any asterisk users using Broadvoice that have it working? |
| 23:02.11 | RickTick | jedaustin: Yeah |
| 23:04.10 | NexterB | anybody have cisco 7940 good experience ? |
| 23:04.38 | jas_williams | NexterB: I have 7960's running using SIP |
| 23:04.54 | calvinhp | I'm running SIP on some 7940's |
| 23:05.02 | bannerman | I really like these ariaVoice phones |
| 23:05.03 | *** join/#asterisk tuxinator_linux (~tuxinator@m110e36d0.tmodns.net) |
| 23:05.16 | bannerman | cheap, functional, and IAX2 :) |
| 23:05.23 | Nugget | I love my 7960s |
| 23:05.25 | NexterB | what are the things to be carreful about when setting up ? |
| 23:06.04 | jas_williams | NexterB: Ensure you have the sip firmware before you start and then follow wiki |
| 23:06.04 | NexterB | I'm just starting in asterisk conf. Spend the last 2 days (well, night) in voip-info.org and newsgroup :) |
| 23:06.41 | NexterB | well, I don't have it. I understand that tonight ... |
| 23:06.44 | tuxinator_linux | Hard to keep my wireless connection up on the road here |
| 23:07.16 | *** join/#asterisk mrguymontag (~montag@81-174-23-160.f5.ngi.it) |
| 23:07.18 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 23:07.22 | NexterB | my cco login is just un gues account, I could'n get firmware. Reading the specs befors baught them, I thaught SIP was build in |
| 23:07.51 | NexterB | so I'm trying with soft phone for now |
| 23:11.56 | *** join/#asterisk mgth (~chatzilla@dsl093-001-038.det1.dsl.speakeasy.net) |
| 23:14.48 | caesar2 | some one from germany here who knows how eg. sipgate or freenet get there calls to fixed network ? the technical side.. just courios... |
| 23:14.51 | *** join/#asterisk wolfson (~hehe@bcp-68-187-180-085.man.nc.charter.com) |
| 23:16.10 | Nugget | NexterB: I'd also suggest that you never fall pray to the tempting thought "I don't *really* need a tftp server and all that mess, i can just do it from the phone itself". |
| 23:16.50 | NexterB | no, i've run out of this stage (but i maybe think that a day) |
| 23:16.53 | Nugget | cisco phones are designed to be centrally managed with dhcp and tftp. They really only pay lip-service to any other approaach and you'll be swimming upstream with no company if you try. |
| 23:17.16 | jarrod | tftp seems so insecure |
| 23:17.18 | tzanger | dhcp+tftp really is a nice way to centrally manage things though |
| 23:17.20 | Nugget | it is. |
| 23:17.20 | NexterB | i'm testing this at home but it's for a small company installation |
| 23:17.56 | johnnyb | tzanger: dhcp+http is even easier, I think. |
| 23:18.03 | tzanger | johnnyb: it's a LOT more complex |
| 23:18.14 | johnnyb | tzanger: in what way? |
| 23:18.22 | tzanger | johnnyb: write an HTTP client vs a TFTP client |
| 23:18.29 | NexterB | i love this approach : every thing on the server, juste reboot to get new conf/firmware. Find that great for large site |
| 23:18.38 | johnnyb | tzanger: well, thankfully, we can just have the phone manufacturers do that for us :) |
| 23:18.41 | tzanger | :-) |
| 23:19.06 | johnnyb | tzanger: grandstream already does |
| 23:20.51 | tzanger | good for them :-) |
| 23:21.41 | Pinhole | a smart tftp server will only give files to address that the dhcp server knows about. |
| 23:22.09 | NexterB | what is the best tftp for linux ? |
| 23:22.16 | NexterB | (tftp server) |
| 23:22.29 | tzanger | NexterB: tftp-hpa |
| 23:22.31 | tzanger | bar none |
| 23:22.36 | *** join/#asterisk bah (048830696@AC8BCFF5.ipt.aol.com) |
| 23:23.59 | NexterB | thanks. I use tftpd maintained by H. Peter Anvin available for fedora in rpm but it lack in config & logging |
| 23:24.52 | tzanger | hpa = H. Peter Anvin |
| 23:26.19 | NexterB | oh ... |
| 23:26.29 | NexterB | so it's the same ;) |
| 23:27.27 | NexterB | didn't find a way to log file request and errros |
| 23:27.42 | tzanger | ? |
| 23:27.44 | tzanger | works for me |
| 23:27.57 | NexterB | where does it log tranfers ? |
| 23:28.21 | NexterB | found nothing in manpage and command line usage |
| 23:28.24 | tzanger | syslog |
| 23:28.28 | tzanger | I forget what facility though |
| 23:28.38 | tzanger | I just add a *.* -/var/log/all and restart syslogd |
| 23:28.53 | NexterB | ok |
| 23:29.31 | *** join/#asterisk bbgun (~zizi@ppp48-63.lns1.adl2.internode.on.net) |
| 23:31.00 | *** join/#asterisk GMsoft (~r0_ot@gmsoft.developer.gentoo) |
| 23:31.04 | GMsoft | hi |
| 23:31.18 | tzanger | NexterB: remember the man page also shows -v |
| 23:31.59 | *** join/#asterisk Guyo|work (~franz@62.211.184.78) |
| 23:32.01 | *** join/#asterisk adker (~adker@67-51-237-86.dsl1.glv.ny.frontiernet.net) |
| 23:32.11 | GMsoft | I got a tdm400 with fxs module but when there is a call coming in and the third party hangup before asterisk answer, asterisk doesn't detect that |
| 23:32.20 | GMsoft | any pointer to a fix ? |
| 23:32.24 | Guyo|work | hi all, is there a specific channel for spandsp related issues ? |
| 23:32.51 | NexterB | without -v I ve just xinet log on new connction |
| 23:34.06 | Guyo|work | does anyone knows is there is a correlation between spandsp problems and bristuff ? I lost 2 working days looking for a fix to my issues and i'm starting to lose hope |
| 23:34.18 | *** join/#asterisk Rett (~rett@c-67-171-236-169.client.comcast.net) |
| 23:36.33 | *** join/#asterisk Grooby (~Grooby@ip24-250-126-171.dc.dc.cox.net) |
| 23:36.39 | bile_one | What security is available for the data calls incoming and outgoing with asterisk? |
| 23:36.41 | Evanrude|w | does anyone know where to go about changing the permissions for a voicemail file? |
| 23:36.46 | Evanrude|w | does that need to be done in the source? |
| 23:37.12 | bile_one | Evanrude Waht do you mean? |
| 23:37.15 | Mavvie | Evanrude|w: maybe it honours the umask of when asterisk was started. |
| 23:37.29 | NexterB | tzanger: ok with -v & xinet restart it log file request. thanks. |
| 23:37.40 | tzanger | :-) |
| 23:37.43 | Evanrude|w | yeah, like right now all my voicemails are written with asterisk:root 0700 permissions |
| 23:37.53 | NexterB | now I can see it requesting the sip firmware I don't have :) |
| 23:38.00 | AgiNamu | Hmm.... my phone doesn't transmit a # after I dial. |
| 23:38.10 | AgiNamu | Like dialing 123# just sends 123 |
| 23:38.13 | AgiNamu | is that a common function? |
| 23:38.44 | bbgun | im running two hopsip 1003 phones, i can call between them locally but the LCD only displays the persons name who is being called, not the Caller ID |
| 23:38.53 | outtolunc | <PROTECTED> |
| 23:38.57 | bbgun | can anyone direct me on how to get this fixed if its possible |
| 23:38.58 | bbgun | ? |
| 23:39.00 | outtolunc | you are blind bkw |
| 23:39.05 | Guyo|work | AgiNamu, i think i read on voip-info wiki that # is reserved |
| 23:39.22 | AgiNamu | So, how do I go about doing speed dialing? |
| 23:39.31 | AgiNamu | I can't use *, because those are reserved service codes. |
| 23:39.40 | bile_one | Evanrude where are yours being stored? |
| 23:39.52 | Guyo|work | check on the wiki, i don't remember exactly |
| 23:40.21 | Evanrude|w | /var/spool/asterisk/voicemail/ |
| 23:40.22 | Mavvie | AgiNamu: which phone? |
| 23:40.39 | AgiNamu | guyo, any idea under what kind of articles? "#" isn't exactly a good keyword :P |
| 23:40.42 | AgiNamu | mavvie, pa168 |
| 23:40.51 | Mavvie | it's setable in the grandstream BT100s: use # for speeddial. |
| 23:40.58 | Mavvie | don't know about the pa168 |
| 23:41.05 | AgiNamu | yea, hosted speedial is what im doing |
| 23:41.16 | Guyo|work | AgiNamu: try "pound sign" or "hash sign" |
| 23:41.16 | AgiNamu | Basically, im trying to decide how I should allow speeddial for my customers. |
| 23:41.20 | AgiNamu | guyo duh |
| 23:41.21 | AgiNamu | :) |
| 23:41.34 | doughecka | is there an easy way to split a voice T1 into 2 so server 1 gets the first 5 channels and server 2g ets the rest? |
| 23:42.05 | tzanger | doughecka: yes, it's called add-drop |
| 23:42.13 | doughecka | oh? |
| 23:42.25 | doughecka | so I could have 2 servers plugged into a single T1? |
| 23:42.25 | Guyo|work | AgiNamu: it was related to the dialplan i thing |
| 23:42.45 | bile_one | Evanrude are you running Asterisk as a non root user? |
| 23:42.59 | caesar2 | dougheacka: use a nife or scissors for line cut :-) |
| 23:43.07 | doughecka | :P |
| 23:43.30 | Evanrude|w | yeah it's running as 'asterisk' |
| 23:43.46 | AgiNamu | guyo, yea, # isn't treated nicely |
| 23:43.56 | AgiNamu | Anyone know how tradional COs implement this? |
| 23:44.04 | AgiNamu | where you dial *70 or somethign to add speed dieal |
| 23:44.05 | bile_one | hummm? Have you set permissons on the folder for root, or asterisk? |
| 23:44.17 | Guyo|work | Note: To have an extension that is triggered by dialing the # symbol, you must use an extension pattern (see below). Asterisk does not recognize # as an ordinary 'digit', even though it appears on all DTMF telephones. |
| 23:44.36 | *** join/#asterisk jeffik (~jeffik@208-41-192-106.client.dsl.net) |
| 23:44.38 | AgiNamu | Yea, and I read that some phones require # to signal end dialing. |
| 23:44.42 | AgiNamu | So I can't do 123# |
| 23:44.45 | AgiNamu | hmm. |
| 23:44.46 | Guyo|work | can't help you more, sorry |
| 23:44.49 | caesar2 | there is a nice set pem. file in amp to set asterisk permissions right... |
| 23:44.57 | Guyo|work | i'm leaving |
| 23:45.10 | caesar2 | and a short howto |
| 23:45.12 | Guyo|work | anyone has a valid fax tiff file for me to test spandsp ? |
| 23:45.21 | bile_one | Evanrude are the folders set for the owner to be asterisk? |
| 23:45.33 | Evanrude|w | bile_one, yes, they are |
| 23:45.33 | AgiNamu | ok thanks |
| 23:45.34 | Evanrude|w | but |
| 23:45.36 | AgiNamu | time to dine |
| 23:45.37 | AgiNamu | late |
| 23:45.49 | Guyo|work | AgiNamu: u welcome |
| 23:45.59 | NexterB | lot of home isp in france provide voip service now. The first one was free.fr with the 3-in-1 freebox. Does anybody know if we can forward voip call from asterisk to those access without using the fxo port of their box ? |
| 23:46.01 | Evanrude|w | bile_one, when the new voicemail file gets written it writes it with 0700 permissions |
| 23:46.43 | bile_one | Evanrude, so does mine |
| 23:46.45 | Evanrude|w | and i can't find anywhere to change that particular setting |
| 23:47.10 | bile_one | Evanrude but the owner and group are asterisk |
| 23:47.20 | Evanrude|w | weird |
| 23:47.37 | bile_one | And mine works |
| 23:47.54 | Evanrude|w | i am trying to check mine with vmail.cgi |
| 23:48.02 | Evanrude|w | and the web server won't open them yet |
| 23:48.06 | bile_one | Anyone in here using Swift for text to speech? |
| 23:48.14 | caesar2 | yes |
| 23:48.28 | rvhi | can macro be nested? call another macro inside one macro? |
| 23:48.44 | bile_one | Evanrude, what are you truying to open them with? Firefox won't do it right. |
| 23:48.51 | Rett | Anyone here run asterisk within multiple instances of Xen? |
| 23:49.05 | rvhi | the problem seems to be ${MACRO_CONTEXT} returns the result of the calling macro, not the original context |
| 23:49.17 | Evanrude|w | yeah.. firefox |
| 23:49.23 | Evanrude|w | but i am getting an error from my webserver |
| 23:49.26 | bile_one | caesar2 can I get an example of your swift.agi |
| 23:49.32 | Evanrude|w | when the permissions are right firefox will open them too |
| 23:50.02 | bile_one | Evanrude I have never gotten Firefox to open them. Try Mozilla or IE and see if they'll open |
| 23:50.24 | bile_one | Evanrude even opera works |
| 23:50.53 | caesar2 | bile_one: not in my office with * right now... so you can't I'm sorry |
| 23:50.54 | bile_one | Evanrude Firefox is missing the plug in and it never installs the correct one. |
| 23:51.16 | bile_one | caesar2 will you email it to me? |
| 23:51.21 | Evanrude|w | bile_one, i think this is a file permission problem -- my firefox just prompts me to open them in <default meida player> |
| 23:51.24 | Evanrude|w | i think i have found it |
| 23:51.33 | Evanrude|w | i can use -G asterisk when i start asterisk |
| 23:51.37 | Guyo|work | anyone has a valid fax tiff file for me to test spandsp ? |
| 23:51.39 | Evanrude|w | and the group won't default to root |
| 23:51.41 | Evanrude|w | let me try that real quick |
| 23:51.50 | Evanrude|w | but, that doesn't fix the 0700 permissions deal |
| 23:51.53 | bile_one | Evanrude okay |
| 23:52.23 | bile_one | Evanrude like I said mine are 0700 too! |
| 23:52.48 | Evanrude|w | but what does your web server run as? |
| 23:52.52 | Evanrude|w | mine runs as www-data |
| 23:52.58 | caesar2 | Guyo: http://www.verypdf.com/pdf2tif/pdf-to-fax.html -> found on google so you can make your tiff |
| 23:53.02 | Evanrude|w | i think that's the problem .. my web server can't access the files |
| 23:53.08 | bile_one | Evanrude hol don. |
| 23:53.32 | Guyo|work | caesar2: thanks, sorry, i'm a little bit overwhelmed with googling :-| sorry for bothering |
| 23:53.39 | bile_one | Evanrude my webserver runs as Asterisk |
| 23:53.39 | Pinhole | bile_one, the cvs version of phpagi has swift support |
| 23:53.47 | Evanrude|w | ahhh |
| 23:53.48 | Evanrude|w | that is why :) |
| 23:54.12 | Evanrude|w | i may resort to that if i can't find another way to make it work |
| 23:54.15 | Guyo|work | BUT... it runs on windows, which i do not have |
| 23:54.23 | Guyo|work | thanks anyway :) |
| 23:54.37 | Guyo|work | going to sleep, thanks and bye all |
| 23:54.39 | bile_one | Pinhole thanks |
| 23:55.52 | *** part/#asterisk Rett (~rett@c-67-171-236-169.client.comcast.net) |
| 23:56.11 | bile_one | Pinhole you got a web link or googleit? |
| 23:56.40 | bile_one | PinHole nm |
| 23:57.28 | *** join/#asterisk epoch (epoch@octane.breakbeats.org) |
| 23:57.34 | simonides | I've got a firefly softphone trying to connect, but * claims that there's no registration info for peer 'simon', when I've got a heading filled out for [simon] in iax.conf |
| 23:57.52 | simonides | am I putting the un/pw in the wrong part of the iax.conf? |
| 23:58.14 | *** join/#asterisk soundguy (~soundguy@zeus.blendtek.com.au) |
| 23:58.44 | simonides | ah crap, why'd it work that time. |
| 23:59.43 | simonides | I guess a better question is, how do I make sure my iax.conf has been reloaded without restarting *? |
| 23:59.48 | bbgun | <PROTECTED> |
| 23:59.53 | bbgun | can anyone direct me on howto display a caller ID name instead of number, if its possible ? |
