| 00:00.11 | Qwell[] | Morex: How are you putting the call on hold? |
| 00:00.18 | [TK]D-Fender | Morex : With AgentLogin you'd have to transfer the call to a parking-lot. |
| 00:00.19 | Morex | Using parked call at the moment |
| 00:00.26 | Morex | Pressing #700 |
| 00:00.29 | Qwell[] | so use the parking lot to pick it back up |
| 00:00.33 | Morex | Question is, how do I get it back? |
| 00:00.44 | Qwell[] | it should read you a number back when you park it |
| 00:00.44 | Morex | Without hanging up? |
| 00:00.48 | Morex | Yup |
| 00:00.52 | Qwell[] | so dial that |
| 00:00.54 | wunderkin | Qwell[]: sounds like chase bank |
| 00:01.11 | Qwell[] | wunderkin: not gonna say what bank. :) |
| 00:01.14 | wunderkin | heh |
| 00:01.22 | Morex | Trouble is, when you put the call on hold, the agent is now just in the AgentLogin call, and you can't transfer out of that to the parking lot... |
| 00:01.28 | Qwell[] | we're big though...one of the biggest |
| 00:01.31 | *** join/#asterisk Umaro (n=umaro@68.142.142.105) |
| 00:01.35 | websae | i knew there was reason i closed my chase accounts |
| 00:01.50 | Umaro | hi guys.. anyone here hosting their asterisk server at serverbeach? |
| 00:03.36 | DeadZen | dharma is hot |
| 00:04.11 | DeadZen | im out guys |
| 00:04.13 | rayvd | and she has legs'o'doom |
| 00:04.14 | DeadZen | thanks for your help |
| 00:04.20 | DeadZen | stay cool |
| 00:05.14 | *** part/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 00:07.03 | *** part/#asterisk gorauskas (n=gorauska@66-224-20-131.atgi.net) |
| 00:07.24 | *** join/#asterisk Pids (i=pids@38.113.222.9) |
| 00:08.19 | FuriousGeorge | so its probably my client that appends the domain to the cid, since i dont remember seeing that over zap, am i wrong? |
| 00:08.24 | FuriousGeorge | sip clients specifically |
| 00:09.01 | *** join/#asterisk _Soul_ (n=Soul@87-196-2-15.net.novis.pt) |
| 00:10.04 | *** join/#asterisk Eitch (n=hugo@unaffiliated/eitch) |
| 00:10.07 | [av]bani | http://news.yahoo.com/s/afp/20060222/tc_afp/netherlandsnigeriaus |
| 00:10.52 | *** join/#asterisk tainted- (n=somewher@mail.k2usa.com) |
| 00:11.13 | SwK[Work] | anyone using t38modem ? |
| 00:11.25 | Qwell[] | SwK[Work]: Just you |
| 00:11.32 | *** join/#asterisk backblue (n=moo@87.196.0.177) |
| 00:12.03 | SwK[Work] | appearently |
| 00:12.05 | SwK[Work] | whats up Qwell |
| 00:12.30 | file[laptop] | ^_^ |
| 00:12.44 | SwK[Work] | unlink file |
| 00:12.53 | file[laptop] | noes! |
| 00:13.35 | trixter | ha |
| 00:13.57 | file[laptop] | SwK[Work]: how much beer do you owe me? |
| 00:14.08 | SwK[Work] | about a case |
| 00:14.17 | file[laptop] | what's the currency exchange to muffins for that? |
| 00:14.29 | SwK[Work] | 1muffin == 24 beers |
| 00:14.29 | Qwell[] | oz to oz |
| 00:14.33 | file[laptop] | HA |
| 00:17.12 | FuriousGeorge | err, what does a call with unknown callerid look like to asterisk here in eastern US. im trying something, but i dont know how to test |
| 00:17.21 | FuriousGeorge | how can i sumulate unknown cid |
| 00:17.30 | *** join/#asterisk robin_sz (n=robin@adsl.redpoint.org.uk) |
| 00:17.37 | robin_sz | MEEP! |
| 00:17.53 | robin_sz | w00t! .. new GXP2000 software is out at last ;) |
| 00:18.39 | *** part/#asterisk LeonardoCabelo (n=leonardo@unaffiliated/leonardocabelo) |
| 00:20.00 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 00:26.47 | websae | shnew software?? |
| 00:26.57 | FuriousGeorge | now that i look at the cdr, the funny thing is one box dispalys "Maria" <103> (a zap channel> and another does "Asterisk" <0> |
| 00:27.17 | FuriousGeorge | if i could figure out where its getting that couldnt i set it to unknown there vs. putting it in my dialplan? |
| 00:27.26 | *** join/#asterisk |Vulture| (n=Vulture@82.115.205.68.cfl.res.rr.com) |
| 00:27.31 | FuriousGeorge | oh, btw, that happens when i get an unknown cid |
| 00:27.39 | |Vulture| | Anyone know why this AEL statement doesn't work? http://pastebin.ca/42845 |
| 00:27.58 | |Vulture| | are 91XXXXXXXXXX not allowed in case statements? |
| 00:28.20 | trixter | I dont think it pattern matches there |
| 00:28.30 | trixter | at least not externally |
| 00:28.44 | trixter | if it were to I am sure a preface of _ would be in order |
| 00:28.53 | |Vulture| | okay so I would have to move it out |
| 00:29.02 | trixter | ultimately it will do pattern matching (look at what 'default' gets mapped to) ... |
| 00:29.07 | |Vulture| | kk |
| 00:29.22 | trixter | but I am not sure that you can create a pattern, try it with the _ first to be sure though |
| 00:29.24 | FuriousGeorge | grrr |
| 00:29.54 | *** join/#asterisk zaf (n=tfournet@ip68-226-162-240.lf.br.cox.net) |
| 00:39.47 | |Vulture| | trixter: no even with the _ it errors |
| 00:40.13 | |Vulture| | strange thing is it does: Added extension 'sw-1-_9NXXXXXX' priority 1 to outbound then I call and it says "Channel 'SIP/201-fbe9' sent into invalid extension 'sw-1-94038409' in context 'outbound', but no invalid handler" |
| 00:41.04 | trixter | yeah that is why I was saying ultimately it does pattern matching, however I think that the way its inserted you cant specify patterns there |
| 00:41.28 | file[laptop] | |Vulture|: _ goes at the front |
| 00:41.45 | |Vulture| | I have _9. |
| 00:41.52 | |Vulture| | then in my case it is case _9NXXXXXX: |
| 00:42.07 | file[laptop] | _sw-1-9NXXXXXX |
| 00:42.13 | file[laptop] | switches = twitches |
| 00:42.25 | *** join/#asterisk Eitch (n=hugo@unaffiliated/eitch) |
| 00:43.12 | file[laptop] | I'm too tired to care :P |
| 00:43.48 | file[laptop] | drugs are bad mmmk? |
| 00:43.51 | |Vulture| | file[laptop]: hmm not sure I follow |
| 00:44.10 | file[laptop] | |Vulture|: the _ at the front tells Asterisk to use pattern matching, putting _ in farther is useless |
| 00:44.47 | |Vulture| | file[laptop]: I have a _9. so it should start matching there... correct? |
| 00:44.53 | |Vulture| | then it does the cases |
| 00:45.06 | file[laptop] | it'll match the one that is closest... |
| 00:46.05 | |Vulture| | file[laptop]: this is what it looks like right now http://pastebin.ca/42845 |
| 00:46.16 | file[laptop] | ugh AEL |
| 00:46.36 | |Vulture| | lol |
| 00:46.53 | |Vulture| | AEL a waste still? |
| 00:47.20 | Kizmet | |Vulture|, there is AEL2 in beta now -_- |
| 00:47.30 | file[laptop] | well in the end it all gets translated into the same thing in the PBX core |
| 00:47.42 | Qwell[] | file[laptop]: 0's and 1's? |
| 00:47.48 | file[laptop] | Qwell[]: muffins |
| 00:47.58 | Qwell[] | pbx_muffin? |
| 00:48.00 | brookshire | qwell: just 0s |
| 00:48.06 | file[laptop] | brookshire: !!!!!!!!!!!! |
| 00:48.06 | |Vulture| | isn't AEL2 the same as AEL just with some more commands? |
| 00:48.11 | brookshire | file! |
| 00:48.13 | Qwell[] | brookshire: OH NOEZ! |
| 00:48.15 | file[laptop] | |Vulture|: sorta but not |
| 00:48.17 | Qwell[] | I've been brookshire'd |
| 00:48.22 | kFuQ | howto declare a system command as a variable.. i.e. something like SetVar(date=/bin/date %F-%T) |
| 00:48.26 | Kizmet | BBL |
| 00:48.48 | |Vulture| | I guess Ill just refine my huge extensions.conf files |
| 00:49.13 | brookshire | :/ |
| 00:49.27 | JunK-Y | brookshire: ! |
| 00:51.17 | *** join/#asterisk rajiv (n=irc@gentoo/developer/rajiv) |
| 00:53.23 | robin_sz | wowo .... 20 minutes and my GXP2000 is still alive .. :) |
| 00:55.34 | *** join/#asterisk bkw_ (n=bkw_@md80afa48.tmodns.net) |
| 00:55.41 | Qwell[] | bkw_: !! |
| 00:55.54 | Qwell[] | You only get two, because your DNS is ugly |
| 00:56.22 | brookshire | junk-y! |
| 00:56.41 | *** join/#asterisk AlexCTI (n=alex@weston-69.65.87.249.myacc.net) |
| 00:57.12 | *** join/#asterisk peppertrader (n=peppertr@216.134.0.189) |
| 01:00.59 | peppertrader | I'm a newbie and need some help configuring Asterisk |
| 01:01.27 | Qwell[] | ~docs |
| 01:01.30 | jbot | methinks docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 01:03.19 | brookshire | don't forget asterisk-docs.org |
| 01:03.25 | peppertrader | Thanks you guys are really a lot of help |
| 01:03.40 | Qwell[] | I'll assume that wasn't sarcasm. You're welcome. |
| 01:04.53 | peppertrader | I have looked at many documentation sites and forums but I have been unable to find an answer. |
| 01:05.09 | Qwell[] | I've been unable to find a question... |
| 01:05.14 | Qwell[] | so, until one of those things give... |
| 01:05.32 | Qwell[] | ie; ask your question, and you might get an answer |
| 01:07.11 | peppertrader | I just setup Asterisk with a TDM400 with 1 FXO and 1 FXS. I have an analog telephone connected to the FXS and I'm able to dial out the PSTN with no problem. When I try to dial out from my SIP phone, it rings my analog phone. |
| 01:07.38 | Qwell[] | You're using the wrong zap channel in extensions.conf for the SIP phone |
| 01:08.37 | Qwell[] | Zap/1/${EXTEN} vs Zap/2/${EXTEN} |
| 01:10.53 | FuriousGeorge | lol |
| 01:10.58 | peppertrader | I am not familiar with the config files, can this be done from AMP? |
| 01:11.12 | rayvd | :( |
| 01:11.14 | *** join/#asterisk Aum-Aum (i=Aum-Aum@66.226.248.2) |
| 01:11.41 | FuriousGeorge | im gonna sneeze |
| 01:11.43 | FuriousGeorge | ah ah |
| 01:11.44 | FuriousGeorge | ~amp |
| 01:11.46 | jbot | rumour has it, amp is NOT supported here! people using it should join #amportal |
| 01:11.50 | FuriousGeorge | jbot: thanks |
| 01:11.50 | jbot | FuriousGeorge: bitte |
| 01:12.05 | FuriousGeorge | j/k peppertrader |
| 01:12.54 | rayvd | or #ampersand! |
| 01:13.23 | peppertrader | Sorry guys I'm new to this and I intend to learn more of the inner workings of Asterisk but now I'm just trying to get a couple of phones to work. |
| 01:14.47 | *** join/#asterisk redondos (n=redondos@190.48.46.22) |
| 01:17.00 | FuriousGeorge | you'll be fine once you understand the configs and extensions.conf |
| 01:17.23 | FuriousGeorge | ive been at it for a year or so and i only ask qwell for help like twice a week now :) |
| 01:18.51 | Qwell[] | speaking of which |
| 01:18.58 | Qwell[] | when do I get paid? :p |
| 01:19.59 | _0_0_ | How deep can I include contexts in AEL?, If I include a context that includes a context asterisk doesn't recognise the extension, but if I include the context directly it works. |
| 01:20.09 | *** join/#asterisk Supaplex (n=supaplex@shell.aros.net) |
| 01:22.25 | peppertrader | Is this where the change needs to be made? [ext-local] |
| 01:22.26 | peppertrader | include => ext-local-custom |
| 01:22.26 | peppertrader | exten => 100,1,Macro(exten-vm,100,100) |
| 01:22.26 | peppertrader | exten => ${VM_PREFIX}100,1,Macro(vm,100) |
| 01:22.26 | peppertrader | exten => 100,hint,ZAP/1 |
| 01:22.26 | peppertrader | exten => 200,1,Macro(exten-vm,200,200) |
| 01:22.27 | Supaplex | is $25/port for pri pretty much the going rate? ELI quoted me they'd require a minimum of 12 channels. (not in my ballpark). I need like 3 dids shared among 4-5 lines. (for my toll free numbers to point to). Any suggestions? |
| 01:22.28 | peppertrader | exten => ${VM_PREFIX}200,1,Macro(vm,200) |
| 01:22.30 | peppertrader | exten => 200,hint,SIP/200 |
| 01:22.50 | Supaplex | peppertrader: in a word, nopaste |
| 01:22.56 | peppertrader | Sorry |
| 01:29.23 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 01:29.39 | *** join/#asterisk kkun (n=none@c-67-186-207-234.hsd1.ut.comcast.net) |
| 01:29.52 | *** join/#asterisk doughecka (n=Miranda@unaffiliated/doughecka) |
| 01:33.41 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 01:33.45 | *** join/#asterisk NewSole (n=dave@d226-105-29.home.cgocable.net) |
| 01:36.59 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm) |
| 01:37.11 | *** join/#asterisk iq (n=iq@71-214-6-43.omah.qwest.net) |
| 01:39.47 | *** join/#asterisk Micc (n=dotirc@c-24-16-228-130.hsd1.wa.comcast.net) |
| 01:41.24 | *** join/#asterisk livinded (n=livinded@cpe-24-24-190-252.socal.res.rr.com) |
| 01:41.56 | *** join/#asterisk dasuberdavid (n=root@68.62.203.147) |
| 01:42.35 | dasuberdavid | where can I get res_odbc? |
| 01:43.06 | JunK-Y | asterisk-addons? |
| 01:43.10 | dasuberdavid | its not in there |
| 01:43.16 | dasuberdavid | whats up JunK-Y !!!! |
| 01:43.21 | JunK-Y | whats up david! |
| 01:43.37 | JunK-Y | not much, ive my new car! |
| 01:43.39 | dasuberdavid | i diddnt find res_odbc in asterisk-addons |
| 01:44.44 | livinded | when i try modprobing zaptel or ztdummy i get http://pastebin.com/567884 would someone mind telling me hats wrong? |
| 01:44.51 | JunK-Y | wait |
| 01:46.40 | *** join/#asterisk steveaj (n=steve@62.55.147.53) |
| 01:46.43 | dasuberdavid | livinded: what kernel version are you running? |
| 01:46.51 | livinded | 2.6.15.4 |
| 01:47.15 | livinded | is it the slackware problem with udev? |
| 01:47.51 | cpm | So, when you "see dmesg" what do you see? |
| 01:48.29 | livinded | cpm, theres actually nothing in there about zaptel or ztdummy |
| 01:49.31 | cpm | fresh module build? |
| 01:49.35 | livinded | yes |
| 01:49.36 | cpm | I'd build'em again |
| 01:49.39 | livinded | fresh install also |
| 01:49.43 | livinded | ok i'll try that |
| 01:49.47 | pauldy | if he was running another kernel it would have said unknown module, problem is probably related to another module not auto loading and the symbols it reffers to will be found in that module |
| 01:49.53 | cpm | doesn't look right. |
| 01:50.08 | *** join/#asterisk justinu (n=Justin@cpe-72-129-86-208.socal.res.rr.com) |
| 01:50.28 | livinded | i didn't compile this on the original kernel |
| 01:50.33 | livinded | i compiled it on this one |
| 01:51.15 | pauldy | then there are probably kernel modules whose symbols dont match |
| 01:51.32 | *** part/#asterisk dasuberdavid (n=root@68.62.203.147) |
| 01:52.05 | *** join/#asterisk Nand0 (i=nando@unaffiliated/nand0) |
| 01:52.07 | livinded | pauldy, how would i fix that? |
| 01:57.55 | pauldy | well I wouldnt try compiling it against the kernel your running |
| 01:59.55 | *** join/#asterisk jyukes (n=jameshot@pool-71-244-78-223.atc.east.verizon.net) |
| 02:00.05 | justinu | had a nice 6 hour cable modem outage |
| 02:04.12 | livinded | pauldy, whats wrong with 2.6.15.4? |
| 02:06.08 | *** join/#asterisk firestrm (n=no@S01060013105c1c69.gv.shawcable.net) |
| 02:12.40 | *** join/#asterisk litage (n=nick@203.220.55.70) |
| 02:13.35 | litage | how old is Asterisk? |
| 02:15.33 | mzo | young |
| 02:17.10 | trixter | it started in 1999 |
| 02:17.13 | litage | 12 months? 18? |
| 02:17.24 | trixter | and look at it after all this time |
| 02:17.31 | litage | thanks |
| 02:21.37 | firestrm | websae, you around? |
| 02:22.27 | lunaphyte | what's the difference between exten => 7508,s followed by exten => 7508,1 and exten => 7508,1 followed by exten => 7508,2 ? |
| 02:24.26 | lunaphyte | oooh. i think i read that wrong.. s is not a valid priority, is it? |
| 02:26.00 | litage | is it possible to set the packetisation rate (eg: 20ms) in Asterisk, or is that an end-user-device setting/ |
| 02:27.43 | lunaphyte | i see that the s extension is used when there is no known number called in the context - when would that happen? |
| 02:32.09 | *** join/#asterisk jeffik (n=Jeff@CPE0050babf4cd6-CM014350000760.cpe.net.cable.rogers.com) |
| 02:33.32 | *** join/#asterisk rezEdit (n=rezEdit@dhcp-153.omnigroup.com) |
| 02:39.20 | rezEdit | hey... can anyone help with configuring multiple Digium Wildcards? We've had one up and running for some time and just added a new one. Zaptel seems happy, but Asterisk complains about not finding a device for our two new channels |
| 02:39.49 | trixter | those cards are so wild there is no configuring them!! |
| 02:40.00 | rezEdit | LOL |
| 02:40.06 | Nugget | don't forget to plug in the power molex. I'd never be that dumb, but maybe you were. ;) |
| 02:40.20 | mzo | nugget, make my zyxel suck less plzkthx |
| 02:41.37 | rezEdit | http://pastebin.ca/42857 has snippets from zaptel and zapata conf files |
| 02:42.51 | rezEdit | Unfortunately Digium hardware support seems to be closed :-( |
| 02:42.51 | *** join/#asterisk iq|tablet (n=iq@71-214-4-12.omah.qwest.net) |
| 02:42.58 | *** join/#asterisk mogorman (n=mogorman@user-24-236-84-48.knology.net) |
| 02:45.58 | *** join/#asterisk SpaceBass (n=CumSlut2@static-71-251-230-2.rcmdva.fios.verizon.net) |
| 02:46.00 | SpaceBass | hey folks |
| 02:46.18 | SpaceBass | anyone use voxbone? i'm having problems get my DID routed to my * box |
| 02:46.22 | rezEdit | hey Sp |
| 02:46.28 | rezEdit | SpaceBass even |
| 02:47.23 | rezEdit | SpaceBass: nice hostname. |
| 02:48.14 | SpaceBass | hostname? |
| 02:48.25 | SpaceBass | WHAT THE F?!?!? |
| 02:48.27 | SpaceBass | I didn't set that! |
| 02:48.36 | SpaceBass | BRB |
| 02:49.05 | *** join/#asterisk dogtanian (i=dogtania@eye.ham.zee.walroos.com) |
| 02:49.13 | tronix | heh |
| 02:49.25 | *** join/#asterisk SpaceBass (n=SpaceBas@static-71-251-230-2.rcmdva.fios.verizon.net) |
| 02:49.26 | *** join/#asterisk dasuberdavid (n=mike@68.62.203.147) |
| 02:49.32 | SpaceBass | Seriously...I'm going to kill someone |
| 02:49.35 | firestrm | hostname? i missed the joke:( |
| 02:49.37 | rezEdit | LOL |
| 02:49.47 | SpaceBass | I have a buddy who thinks that kind of shit is funny...and its getting old |
| 02:49.56 | lunaphyte | i thought it was funny |
| 02:50.04 | SpaceBass | and people wonder why I lock my keyboard even if I turn around |
| 02:50.11 | dasuberdavid | does anyone know how to get res_dobc to build |
| 02:50.15 | SpaceBass | might be funny if I had thought to do it to someone else |
| 02:50.16 | SpaceBass | :) |
| 02:50.24 | SpaceBass | anyway...voxbone....anyone using them? |
| 02:50.25 | rezEdit | yeah THEN it would be hilarious! |
| 02:50.30 | firestrm | im still missing the joke.. |
| 02:50.45 | rezEdit | firestrm: his hostname included 'CumSlut2' |
| 02:50.48 | SpaceBass | firestrm practical joke on me...my hostname was set to something obscene |
| 02:50.51 | firestrm | lol |
| 02:51.04 | firestrm | i love it.. |
| 02:51.14 | SpaceBass | payback is a bitch :) |
| 02:51.14 | dasuberdavid | can someone tell me how to get res_odbc to build |
| 02:51.43 | firestrm | thats somthing i would do.. lol |
| 02:51.49 | rezEdit | So, anyone here using multiple Digium PCI cards? |
| 02:52.01 | firestrm | no but i want to try it.. |
| 02:52.11 | *** join/#asterisk Utah_Dave (n=boucha@c-67-172-255-244.hsd1.ut.comcast.net) |
| 02:52.35 | firestrm | i imagine it would work is you had enough spare interrupts |
| 02:52.49 | *** part/#asterisk FuriousGeorge (n=Brian@ool-44c5a9b8.dyn.optonline.net) |
| 02:52.51 | dasuberdavid | yes, you can run more than one digium card in a pc |
| 02:53.11 | firestrm | nice nic :) |
| 02:53.18 | rezEdit | We've had one up and running for some time and just added a new one. Zaptel seems happy, but Asterisk complains about not finding a device for our two new channels |
| 02:53.43 | firestrm | sounds like a zapata.conf problem.. |
| 02:53.45 | rezEdit | http://pastebin.ca/42857 has snippets from zaptel and zapata conf files |
| 02:54.07 | rezEdit | going off of stuff seen on http://www.voip-info.org/wiki/index.php?page=Asterisk+config+zaptel.conf |
| 02:54.10 | tronix | rezEdit: you run ztcfg -vvv ? |
| 02:55.37 | rezEdit | tronix: oh yeah, it's only seeing 4 channels of the 6 (the original card) |
| 02:55.39 | rezEdit | duh |
| 02:56.02 | tronix | I'm sure you'll figure it out. |
| 02:56.07 | SpaceBass | ok...so voxbone just started working for me...odd |
| 02:57.18 | rezEdit | tronix: oh, wait, I had commented out the config |
| 02:57.25 | lunaphyte | why would an inbound sip call to an extension ignore it's context and go to the default context? |
| 02:57.25 | justinu | voxbone is pretty touchy |
| 02:57.27 | rezEdit | tronix: it recognizes all 6 |
| 02:57.43 | SpaceBass | justinu begining to notice that |
| 02:58.25 | justinu | one of those too good to be true type things :( |
| 02:59.35 | SpaceBass | seems to be working now...but i'm gettin the wrong digital receptionist for some reason |
| 03:01.12 | *** join/#asterisk trym (n=trym@194.63.254.6) |
| 03:01.38 | litage | is it possible to set the packetisation rate (eg: 20ms) in Asterisk, or is that an end-user-device setting? |
| 03:02.33 | SpaceBass | and....its quit working again |
| 03:03.22 | [av]bani | so how it goes with 1.0.2.13 ? |
| 03:04.13 | rezEdit | Stupid me. Turns out that I just needed to specify the 2 new modules on the as channels 7 & 8 instead of 5 & 6 because those modules are in spots 3 & 4 on the new card, not spots 1 & 2 |
| 03:04.34 | rezEdit | tronix was right. :-) |
| 03:07.10 | xachen | evil iax |
| 03:08.50 | *** join/#asterisk Darwin35 (n=Darwin@c-24-9-75-234.hsd1.co.comcast.net) |
| 03:09.31 | *** join/#asterisk opsys (n=opsys@68-235-141-52.miamfl.adelphia.net) |
| 03:11.37 | iq|tablet | Hi, I've a dumb question. How do we choose FXO vs FXS on T1 lines? Why would we chose FXS if we can not connect any analog phones to it? |
| 03:12.01 | xachen | T1 has nothing to do with FXO or FXS |
| 03:12.24 | xachen | or maybe i'm being stupid myself ;p |
| 03:12.56 | iq|tablet | :) |
| 03:13.14 | xachen | from my understanding the channelbank takes care of that stuff |
| 03:14.05 | rezEdit | xachen: the channelbank can break the digital T1 down into analog lines... |
| 03:14.17 | rezEdit | iq|tablet: what are you trying to do? |
| 03:15.53 | iq|tablet | rezEdit, I want my sip extensions to be able to receive and mail calls. Was wondering if I could connect one T1 so 23 people can use it at a time. |
| 03:16.05 | rezEdit | iq|tablet: you would want FXS interfaces to plug in analog phones, and FXO interfaces if you are trying to bring analog lines into your PBX |
| 03:16.24 | rezEdit | a T1 only has 16 channels... no? |
| 03:16.39 | iq|tablet | rezEdit, T1. no analog |
| 03:17.10 | rezEdit | iq|tablet: so you'd need one of the T1 interface cards... |
| 03:17.11 | iq|tablet | 23 or 24 |
| 03:17.34 | iq|tablet | rezEdit, already got a card |
| 03:17.35 | Abydos313 | 24 |
| 03:17.52 | iq|tablet | Abydos313, ISDN T1 = 23, Rob bit = 24 |
| 03:18.44 | rezEdit | iq|tablet: so can't you configure that card using zaptel to handle incoming and outgoing calls? |
| 03:18.56 | *** join/#asterisk kuku5 (i=kuku@c-67-175-218-223.hsd1.il.comcast.net) |
| 03:19.06 | kuku5 | How do I add a host from which anything can be sent, in exim |
| 03:19.17 | iq|tablet | rezEdit, my question was. Why do we have concept of fxo/fxs in T1 |
| 03:19.38 | justinu | it's signalling |
| 03:20.02 | justinu | it's how you tell a channel bank to replicate the signalling on an analog line |
| 03:20.23 | iq|tablet | justinu, no channel bank involved. T1 -> Asterisk -> SIP Clients |
| 03:20.34 | rezEdit | iq|tablet: :-) |
| 03:20.38 | justinu | t1 from what? |
| 03:20.45 | *** join/#asterisk ziggurat (n=ziggurat@dsl081-114-028.dfw1.dsl.speakeasy.net) |
| 03:21.07 | justinu | if your telco uses fxo/fxs to signal you over the T1, that sucks |
| 03:21.09 | iq|tablet | justinu, to give SIP clients access to outbound and inbound calls. So they can send/receive calls |
| 03:21.12 | justinu | very legacy |
| 03:21.20 | justinu | but it works |
| 03:24.35 | *** join/#asterisk L|NUX (n=linux@202.5.145.56) |
| 03:24.43 | iq|tablet | still confused :( |
| 03:25.44 | *** join/#asterisk |omni| (n=cathode@net130.limelyte.net) |
| 03:26.00 | *** part/#asterisk mogorman (n=mogorman@user-24-236-84-48.knology.net) |
| 03:27.15 | *** join/#asterisk annonimous (n=annonimo@201.152.130.49) |
| 03:27.25 | annonimous | hello |
| 03:28.20 | annonimous | anybody here uses the gateway audiocodes? (mp104 fxs) |
| 03:30.26 | *** part/#asterisk |omni| (n=cathode@net130.limelyte.net) |
| 03:34.06 | *** join/#asterisk |omni| (n=cathode@net130.limelyte.net) |
| 03:35.53 | sevard | Alright, I'm totally stuck on this Asterisk + VPN. I need help. |
| 03:36.20 | annonimous | asterisk +vpn? |
| 03:36.34 | sevard | An nmap shows port 5060 being open yet the phone still can't register. It works fine with they're on the same LAN. |
| 03:37.26 | sevard | Is * set to only listen on one IP? I'm pretty sure that I have it set to 0.0.0.0 which should technically be "listen on all interfaces" but maybe I'm wrong. Anyone? |
| 03:40.11 | iq|tablet | sevard, VPN or NAT? |
| 03:40.46 | sevard | VPN. I'm using Hamachi |
| 03:41.12 | iq|tablet | sevard, this machine has a public IP ? |
| 03:41.42 | sevard | I can connect to services just fine on the machine over the VPN. SSH, HTTP, telnet, whatever, I've nmaped port 5060/udp on the machine, it says it's open, but the sip phone won't register with * |
| 03:41.46 | sevard | awesome. |
| 03:43.21 | *** join/#asterisk _sciallo_ (n=rick@adsl-218-239-186.jax.bellsouth.net) |
| 03:43.48 | _sciallo_ | I get a bunch of "asterisk[2858]: Failed attempt from 127.0.0.1" in the event log... Anyone knows why? |
| 03:44.39 | *** join/#asterisk iq (n=iq@71-214-4-12.omah.qwest.net) |
| 03:44.54 | annonimous | i have a Gw audiocode (fxs) and i want to use it for 4 phones but i cannto setit up cause asterisk drop it any idea of how to do it? (i have put the security on invite cause its invited but i cant connect |
| 03:44.56 | *** join/#asterisk jorgito (n=jorge@office.inway.cz) |
| 03:44.58 | jorgito | hi |
| 03:45.11 | annonimous | hi |
| 03:45.59 | jorgito | just compiled asterisk against uclibc , with about 6 megas it is very big package, what can i remove, i think some modules wont be needed. What are the core modules wihtout which will be asterisk unusable? |
| 03:49.15 | *** join/#asterisk bmg505 (n=leon@dsl-146-41-176.telkomadsl.co.za) |
| 04:01.00 | *** join/#asterisk Kernel_core (i=Kernel_C@217.218.80.242) |
| 04:03.14 | _0_0_ | could somebody please send me the rxfax and txfax modules from spandsp, I'd like to let my package manager handle asterisk, so I dont really want to build/install it by hand. |
| 04:03.32 | *** join/#asterisk dijit0 (n=dijit0@adsl-68-127-136-253.dsl.pltn13.pacbell.net) |
| 04:04.38 | Kernel_core | anybody used asterisk-h323 ? |
| 04:04.52 | startkeylogger | never, not even the developers! |
| 04:05.24 | Kernel_core | startkeylogger: so H323 in asterisk is for FUN ?! |
| 04:05.49 | startkeylogger | I'm kidding. cut to the chase and ask your real question |
| 04:05.59 | startkeylogger | eg, start with "How ..." |
| 04:06.21 | *** join/#asterisk juice (n=juice@209.33.110.137) |
| 04:06.51 | *** join/#asterisk Defraz (n=t0tal@24-116-219-55.cpe.cableone.net) |
| 04:07.00 | dijit0 | are there any other voip providers that offer pay-as-you-go like Teliax that include voicemail that anyone aware of?? |
| 04:07.19 | Kernel_core | startkeylogger: I am running Debian , I use Asterisk 1.2.4 , but I can't install the Native H323 in source directory of Asterisk... nor binary package doesn't work(asterisk-h323 , depends on libopenh323-1.18.0 which doesn't exist!!!) |
| 04:08.06 | startkeylogger | what tree are you tracking? |
| 04:08.24 | Kernel_core | I want to use H323 in Asterisk |
| 04:09.10 | Kernel_core | I can't install the module , when I want to compile the source , it needs PWLib , when I compile PWLib I get too many errors ... |
| 04:09.42 | Kernel_core | I tried to install debian package... |
| 04:09.53 | [av]bani | Kernel_core: use oh323 instead |
| 04:10.14 | Kernel_core | [av]bani: I heard it isnot stable and doesn't function properly |
| 04:10.19 | _sciallo_ | Has anyone ever gotten "asterisk[2858]: Failed attempt from 127.0.0.1" in the event log? |
| 04:10.23 | [av]bani | Kernel_core: it works for me, nothing else does |
| 04:10.48 | [av]bani | Linux asterisk 2.6.12-10-amd64-k8 #1 Mon Jan 16 17:23:13 UTC 2006 x86_64 GNU/Linux |
| 04:10.57 | *** join/#asterisk gongoputch (n=gongoput@c-68-82-194-31.hsd1.de.comcast.net) |
| 04:11.05 | [av]bani | asterisk*CLI> oh323 show conf |
| 04:11.11 | [av]bani | <PROTECTED> |
| 04:11.14 | [av]bani | Version: 0.7.3 |
| 04:11.34 | Kernel_core | [av]bani: I am running XEON 2.8 without transcoding does it support 400 simultaneously channels ? |
| 04:11.39 | *** join/#asterisk Nukemizer (n=Nuke@160.7.249.15) |
| 04:11.40 | startkeylogger | very hot salsa on a grilled burger with seasoning salt = YUMM |
| 04:11.49 | _sciallo_ | Nothing on google either... I guess I'll figure out what's trying to connect from localhost |
| 04:11.51 | [av]bani | Kernel_core: no idea, try it |
| 04:12.19 | [av]bani | ; Set the maximum number of inbound/outbound/simultaneous |
| 04:12.19 | [av]bani | ; H.323 connections. |
| 04:12.30 | [av]bani | you can set it to whatever you like i guess, it defaults to 100 |
| 04:13.27 | Kernel_core | [av]bani:after I installed oh-h323 on Asterisk1.2.4 asterisk crashes ! |
| 04:14.00 | justinu | oh noes |
| 04:17.39 | *** join/#asterisk tengulre (n=tengulre@221.11.5.180) |
| 04:18.14 | [av]bani | Kernel_core: after i installed asterisk-h323 on asterisk1.2.4 asterisk crashes ! |
| 04:18.44 | [av]bani | we can play this all night :) |
| 04:19.09 | *** join/#asterisk NovceGuru (n=asdf@oh-71-53-48-11.dhcp.sprint-hsd.net) |
| 04:19.40 | jorgito | just compiled asterisk against uclibc , with about 6 megas it is very big package, what can i remove, i think some modules wont be needed. What are the core modules wihtout which will be asterisk unusable? |
| 04:20.18 | [av]bani | 6 megs o_o |
| 04:20.20 | Kernel_core | [av]bani: so... ? |
| 04:21.23 | justinu | hah |
| 04:22.16 | [av]bani | Kernel_core: only oh323 ever worked for me. i don't have any other solution for you. |
| 04:22.27 | *** join/#asterisk xtrvd (n=j@d209-121-36-44.bchsia.telus.net) |
| 04:23.51 | Kernel_core | [av]bani: is it stable for you ? |
| 04:25.20 | [av]bani | i dont use it much, but its more stable than all the others for me |
| 04:25.30 | [av]bani | asterisk_h323 and ooh323 didnt work at all, or crashed |
| 04:26.29 | Kernel_core | [av]bani: thank you :) |
| 04:31.19 | *** part/#asterisk Utah_Dave (n=boucha@c-67-172-255-244.hsd1.ut.comcast.net) |
| 04:31.57 | *** join/#asterisk fafnir (i=hahaha@unaffiliated/fafnir) |
| 04:43.15 | Abydos313 | anyone can help for a min? looking for advise on hardware setup |
| 04:43.59 | mzo | bleh, fwd still doesn't work for me ;P |
| 04:45.57 | *** join/#asterisk annonimous (n=annonimo@201.152.130.49) |
| 04:46.00 | annonimous | hello |
| 04:47.33 | annonimous | i have settuped my ivr but iwant to know how to create a little app who asks to the caller his name to show to the other side the name of the people who is calling, i get busted cause i dont know how to do it |
| 04:54.39 | *** join/#asterisk diclophis (n=diclophi@65.203.37.58) |
| 04:54.43 | diclophis | howdy all |
| 04:54.59 | diclophis | does anyone happen to know of a big list of cellphone "email2sms" gateway domains |
| 04:55.01 | diclophis | and formats? |
| 04:55.25 | mzo | if you have one for finland, please hollar. thx |
| 04:56.19 | *** join/#asterisk salviadud (n=mofo@201.133.209.101) |
| 04:57.56 | pauldy | startkeylogger |
| 04:59.01 | *** join/#asterisk salviadud (n=mofo@201.133.209.101) |
| 04:59.12 | startkeylogger | yes? |
| 04:59.18 | salviadud | Feb 22 22:56:34 NOTICE[22110]: chan_iax2.c:7398 socket_read: Registration of '651692' rejected: 'Registration Refused' from: '192.246.69.186' |
| 04:59.25 | salviadud | its been 3 days |
| 04:59.33 | salviadud | will this ever work? |
| 05:07.58 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 05:08.19 | dijit0 | anyone know any good pay-as-you-go voip providers that have voicemail and allow BYOD? |
| 05:08.38 | diclophis | broadvoice? |
| 05:09.04 | Qwell | dijit0: Why need voicemail? |
| 05:09.10 | Qwell | dijit0: You'll be using asterisk, won't you? |
| 05:09.28 | dijit0 | well, originally i was, but i got lazy |
| 05:09.34 | dijit0 | lol |
| 05:10.13 | dijit0 | i got DSL, cant really run it here |
| 05:11.09 | dijit0 | the internet connection keeps getting lost and registration to nufone times out and i gotta reboot for it to work again |
| 05:12.03 | livinded | can someone help me with zaptel issues, i got asterisk installed but when i when i installed zaptel it worked fine bu when i went to modprobe it i got this error http://pastebin.com/567884 i then fixed it so that it would load the zaptel modules by enabling a library in my kernel but i still can't get ztdummy to load |
| 05:12.29 | dijit0 | its not reliable... i mean if i had a remote shell i could host it on... i aint paying a million bucks for that though |
| 05:13.46 | dijit0 | is broadvoice as bad as all these reviews i am reading say it is??? |
| 05:13.50 | jorgito | just compiled asterisk against uclibc , with about 6 megas it is very big package, what can i remove, i think some modules wont be needed. What are the core modules wihtout which will be asterisk unusable? |
| 05:14.06 | jorgito | sorry for repeating |
| 05:18.49 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 05:22.50 | livinded | when i compile zaptel i get this this warning http://pastebin.com/568105 the make and make install work fine but when i go to modprobe ztdummy i get this error http://pastebin.com/568107 could someone help me fix this? |
| 05:26.38 | *** join/#asterisk Psykick (n=anon@203.167.226.250) |
| 05:26.41 | Psykick | hi guys |
| 05:27.11 | Psykick | what could cause conversations to cut in out? |
| 05:27.51 | livinded | not enough bandwidth |
| 05:27.56 | RV-Dioxide | Psykick: the QoS on your packets |
| 05:28.01 | RV-Dioxide | and bandwidth |
| 05:28.31 | Psykick | we've got probably less than 30 people making calls maybe all at the same time |
| 05:28.35 | RV-Dioxide | I think I'm finally understanding how the extensions file works |
| 05:28.44 | Psykick | and we're using 2 100Mbit Managed Switches |
| 05:29.12 | RV-Dioxide | hrmm |
| 05:29.25 | RV-Dioxide | who's your provider? ISP and VoIP |
| 05:29.28 | Psykick | I can provide QoS on each port on the switch |
| 05:29.48 | Psykick | we're using an E1 connection to our telco |
| 05:30.04 | RV-Dioxide | hrmm |
| 05:30.09 | RV-Dioxide | what's an IP you have? |
| 05:30.18 | RV-Dioxide | err |
| 05:30.21 | RV-Dioxide | wait, that won't matter |
| 05:30.50 | RV-Dioxide | Psykick: how does your connection work? |
| 05:30.51 | Psykick | we are using the IAX protocol if that makes any difference |
| 05:31.31 | Psykick | basically in our setup we have AAH running on a dual xeon 3ghz server with 1gb ram |
| 05:31.34 | RV-Dioxide | you have an E1 connection to your telco, does that mean the VoIP goes straight through them, or do you mean you have an E1 and you are using somebody else like nufone for VoIP? |
| 05:31.54 | Psykick | we have an E1 PRI |
| 05:32.01 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 05:32.09 | justinu | Psykick: dude, where's your buddy, out to lunch? |
| 05:32.37 | RV-Dioxide | me? |
| 05:32.54 | RV-Dioxide | I'm just guessing here, I'm still mostly learning |
| 05:32.55 | Psykick | each of our reps use a softphone to connect to AAH which are on the same LAN |
| 05:33.06 | jorgito | Psykick, if you have 30 calls then that is your e1 full LOL |
| 05:33.12 | RV-Dioxide | yeah |
| 05:33.14 | Abydos313 | livinded did you uncomment the # in front of zaptel in Makefile? |
| 05:33.18 | RV-Dioxide | I was just looking that up |
| 05:33.32 | justinu | voice cutting out with softphones? |
| 05:33.33 | RV-Dioxide | jorgito: well, it's 32 |
| 05:33.38 | livinded | Abydos313, what # in the makefile? |
| 05:33.42 | Psykick | justinu: yip |
| 05:33.48 | justinu | e1 pri to telco? |
| 05:33.52 | Psykick | justinu: yip |
| 05:33.55 | jorgito | RV-Dioxide, yes but the 16th is for signaling |
| 05:33.56 | justinu | which side voice cuts out? |
| 05:34.15 | RV-Dioxide | oh |
| 05:34.16 | Psykick | lemme just talk to the rep |
| 05:34.17 | Abydos313 | if you want the ztdummy compiled you take the # in front of ztdummy. it's in the howto |
| 05:34.17 | Psykick | bbiab |
| 05:34.41 | jorgito | well did you anybody tried to compile asterisk for embedded dev? |
| 05:34.49 | Psykick | ok ... |
| 05:34.49 | RV-Dioxide | Psykick: I would do something to check your connection to your telco |
| 05:34.54 | livinded | i really need to get the asterisk book back from my friend |
| 05:35.02 | *** join/#asterisk Evanrude (n=david@ip68-107-162-212.lu.dl.cox.net) |
| 05:35.12 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
| 05:35.13 | justinu | ~thebook |
| 05:35.15 | jbot | hmm... thebook is Asterisk: The Future of Telephony, released under a Creative Commons license and available at http://www.asteriskdocs.org << Read the book online! |
| 05:35.18 | justinu | you can read it online, yo |
| 05:35.18 | Psykick | apparently it would go dead then come back ... then go dead and come back ... etc etc |
| 05:35.35 | jorgito | how much does DIDs cost ? |
| 05:35.37 | justinu | psykick go completely dead? in both directions? |
| 05:35.48 | RV-Dioxide | wait, is this soft phone to soft phone, inside the company? or going through the E1? |
| 05:35.51 | justinu | Psykick: think of the call as two seperate calls in each direction. which is affected? both? rx? tx? |
| 05:35.52 | RV-Dioxide | jorgito: depends on who |
| 05:36.03 | livinded | Abydos313, where it says MODULES:=zaptel tor2 torisa wcusb wcfxo wctdm wctdm24xxp \ |
| 05:36.03 | livinded | <PROTECTED> |
| 05:36.03 | livinded | <PROTECTED> |
| 05:36.15 | RV-Dioxide | jorgito: I get my 1-866 through NuFone for free |
| 05:36.22 | Psykick | justinu: both |
| 05:36.24 | jorgito | RV-Dioxide, where ? |
| 05:36.27 | RV-Dioxide | some companies charge 10/mo |
| 05:36.32 | jorgito | RV-Dioxide, lol |
| 05:36.32 | RV-Dioxide | nufone.net |
| 05:36.37 | jorgito | i see |
| 05:36.40 | RV-Dioxide | what? |
| 05:36.41 | justinu | psykick iax softphone? |
| 05:36.45 | justinu | which one? |
| 05:36.54 | Psykick | justinu: yip ... firefly |
| 05:36.57 | justinu | psykick: this is what I would do, pay attention ok? |
| 05:37.03 | jorgito | RV-Dioxide, do you know what are the prices in Europe? |
| 05:37.08 | RV-Dioxide | no idea |
| 05:37.12 | Psykick | justinu: pm me |
| 05:37.13 | jorgito | RV-Dioxide, want to sell |
| 05:37.17 | justinu | on the affected machines, run ethereal in capture mode. capture all network traffic on the interface |
| 05:37.26 | justinu | i want to say it in channel for the loggers |
| 05:37.29 | *** part/#asterisk diclophis (n=diclophi@65.203.37.58) |
| 05:37.57 | RV-Dioxide | ah |
| 05:38.04 | justinu | use ethereal to analyze the packet flow between ast and the softphone |
| 05:38.11 | Psykick | justinu: would it make a difference if there are timing issues on the card |
| 05:38.12 | justinu | you can do all sorts of neat stuff with ethereal |
| 05:38.12 | RV-Dioxide | no way in hell I'm buying, I've got a company that's full of them :) |
| 05:38.23 | jorgito | like tcpdump more |
| 05:38.24 | justinu | psykick: i'm thinking your issue is on the iax/softphone side, not the pRI |
| 05:38.34 | RV-Dioxide | as long as ethereal is run securly, with all of it's holes |
| 05:38.36 | justinu | tcpdump sucks, its only good for writing pcap files |
| 05:38.49 | RV-Dioxide | tcpdump is good if you know how to parse it |
| 05:38.50 | jorgito | Psykick, you can trace the calls with you telco operator |
| 05:38.56 | justinu | i use tcpdump on my server, and scp the file back to my workstation |
| 05:39.08 | jorgito | RV-Dioxide, yes i know little ... LOL |
| 05:39.28 | RV-Dioxide | I'm tired |
| 05:39.34 | jorgito | me to |
| 05:39.35 | jorgito | me too |
| 05:39.50 | Psykick | jorgito: err ... how? |
| 05:40.23 | jorgito | Psykick, just call them. send them email, they will trace the calls for you in the soft switch or other pstn device you connect to |
| 05:40.30 | justinu | psykick: also, iax does qos exhcange info at the end of the call. not sure if/where this is logged, but you can see it when you turn on iax2 debug |
| 05:40.44 | *** part/#asterisk _0_0_ (n=000000@81-178-251-94.dsl.pipex.com) |
| 05:40.47 | justinu | pstn is rarely help |
| 05:41.09 | Psykick | ok let me do the ethereal thing |
| 05:41.21 | Psykick | is there a win32 ethereal available? |
| 05:41.24 | justinu | yep |
| 05:41.44 | jorgito | justinu, why ? |
| 05:41.48 | justinu | look from the server side too |
| 05:41.57 | justinu | why aren't they helpful? i don't know |
| 05:42.25 | justinu | they just aren't knowledgeable enough in my experience |
| 05:42.52 | jorgito | justinu, well i am working in telco and we trace callswithout problems. bleee |
| 05:43.04 | justinu | cool, which one? |
| 05:43.13 | Psykick | thing with this issue is it happens on different machines and at different times |
| 05:43.43 | jorgito | justinu, czech republic |
| 05:43.58 | jorgito | justinu, this is the country not the telco LOL |
| 05:44.44 | jorgito | Psykick, you should realy ask your telco if they dont see any problems in l1 |
| 05:45.21 | jorgito | Psykick, well i am not good in SS1 or ISUP .. |
| 05:45.42 | justinu | jorgito: cool, do you know benjk? |
| 05:45.52 | jorgito | who is benjk? |
| 05:45.59 | justinu | he is czech, i believe |
| 05:46.09 | justinu | but now in japan |
| 05:46.12 | jorgito | justinu REALY ? |
| 05:46.22 | justinu | yeah - i met him recently here in la |
| 05:46.30 | Qwell | justinu: eww, why? |
| 05:46.30 | jorgito | justinu, what does he do ? |
| 05:46.39 | RV-Dioxide | wait, where did czech come in? |
| 05:46.43 | justinu | telco |
| 05:46.52 | justinu | well, pbx stuff |
| 05:46.54 | RV-Dioxide | ah |
| 05:47.02 | jorgito | Psykick, really contact them maybe some sinchronization problems with your e1 *timing* |
| 05:47.11 | RV-Dioxide | I was thinking political stuff coming into play, I should stop reading out of context :/ |
| 05:47.15 | livinded | Abydos313, i fixed it, just needed to build in rtc support into my kernel |
| 05:47.18 | justinu | political? |
| 05:47.32 | jorgito | livinded, ntpd ? |
| 05:47.53 | livinded | ? |
| 05:48.05 | livinded | ztdummy wasn't loading becuase rtc wasn't defined |
| 05:49.40 | jorgito | livinded, well rtc is real time clock not ? |
| 05:49.48 | livinded | yes |
| 05:50.02 | jorgito | livinded, if you need to synchronize time with ntp network time protocol |
| 05:50.03 | livinded | i didn't have it built in before |
| 05:50.17 | jorgito | livinded, yes i know, i made same mistake also ... LOL |
| 05:50.30 | RV-Dioxide | night guys |
| 05:50.35 | jorgito | by |
| 05:50.35 | Psykick | night |
| 05:50.40 | livinded | i'm fairly new to linux and asterisk so i've been learning a lot through error |
| 05:51.01 | jorgito | livinded, yes... me too. |
| 05:51.15 | livinded | i'm just happy i'm learning, its just a little frustrating when you can't figure something out and theres nobody there to help you with it |
| 05:51.21 | Psykick | i'm new to asterisk .. not to linux |
| 05:51.56 | livinded | i think my next step is to get the mysql support working with asterisk |
| 05:52.32 | jorgito | livinded, yestrday i was getting to run festival in czech ... uf |
| 05:52.41 | livinded | lol |
| 05:53.08 | livinded | my lug is actually working on a way to implement fesitval and sphinx together and possibly integrate to asterisk |
| 05:53.19 | livinded | i'm hoping to help work on that project |
| 05:53.38 | jorgito | livinded, i was looking at sphinx but found it realy extremely hard..to understand |
| 05:58.03 | *** join/#asterisk jaike (n=a@203.131.137.76) |
| 05:59.49 | *** part/#asterisk jeffik (n=Jeff@CPE0050babf4cd6-CM014350000760.cpe.net.cable.rogers.com) |
| 06:03.12 | *** join/#asterisk ebryn (i=ebryn@eins.itlx.com) |
| 06:05.17 | jorgito | for what is mkdep ? |
| 06:07.57 | jorgito | what is doing trap ? |
| 06:28.58 | Psykick | justinu: you still here? |
| 06:31.51 | Psykick | jorgito: do you know what this means: IAX Packet 386972 from circuit ids 945->87conflicts with earlier call with circuit ids 480->87 |
| 06:32.29 | Psykick | does anyone know? |
| 06:33.23 | Psykick | hello? |
| 06:36.05 | jorgito | Psykick, what version of asterisk ? |
| 06:36.11 | *** join/#asterisk L|NUX (n=linux@202.5.145.56) |
| 06:36.20 | Psykick | 1.2.0 |
| 06:36.31 | Psykick | I'm upgrading to AAH 2.5 this sunday |
| 06:36.46 | Psykick | which I think uses 1.2.4 |
| 06:38.30 | jorgito | Psykick, sorry cant find anything |
| 06:38.38 | Psykick | yeah me neither |
| 06:40.19 | Psykick | doing a packet capture with ethereal |
| 06:42.29 | *** join/#asterisk jeebusroxors (n=jeebus@29palms-cuda1-68-170-36-65.losaca.adelphia.net) |
| 06:44.12 | jorgito | just compiled asterisk against uclibc , with about 6 megas it is very big package, what can i remove, i think some modules wont be needed. What are the core modules wihtout which will be asterisk unusable? |
| 06:44.14 | *** join/#asterisk |ryan| (n=foo@c-67-174-157-188.hsd1.ca.comcast.net) |
| 06:44.38 | |ryan| | will a 266MHz 586 box run asterisk ok? |
| 06:45.34 | sl16 | NOTICE[3952]: chan_sip.c:9521 handle_response_invite: Failed to authenticate on INVITE to '"11" <sip:11@x.x.x.x>;tag=as622eebfc' |
| 06:45.40 | sl16 | what is the mean of this ? |
| 06:46.24 | sl16 | i cannot make outbound calls :( i think i've set evrything correct in extensions.conf |
| 06:50.40 | jaike | sl16: making outbound calls on your sip phone via your asterisk server? |
| 06:52.32 | *** join/#asterisk clive- (n=pirch@dsl-145-23-22.telkomadsl.co.za) |
| 06:53.08 | clive- | anyone here using iax2 trunking ? |
| 06:53.13 | sl16 | jaiger: yes |
| 06:56.16 | jaike | sl16: check your sip.conf, if the username= and secret= values of you sip phone config is the same as the ones you configured on the phone |
| 06:58.07 | sl16 | i think they are not, should they be ? |
| 06:59.46 | jaike | yes. your sip phone has to authenticate to your * server |
| 07:00.06 | *** join/#asterisk CrashHD (n=crashhd@c-24-7-168-46.hsd1.ca.comcast.net) |
| 07:00.08 | CrashHD | hello |
| 07:00.15 | CrashHD | is BYEXTENSION depreciated? |
| 07:00.21 | CrashHD | would ${EXTEN} work the same? |
| 07:00.29 | CrashHD | what is the difference? |
| 07:00.41 | CrashHD | *crickets* |
| 07:00.41 | CrashHD | heh |
| 07:00.45 | CrashHD | that's wednesday night for ya |
| 07:01.24 | Qwell | never heard of it |
| 07:01.32 | sl16 | jaiger: the sip phone is already authenticated to * with some username and secret and can make calls in the segment |
| 07:01.43 | *** join/#asterisk CaRb0n^ (n=Genocide@203.81.212.87) |
| 07:01.53 | sl16 | and i am using exten => _9.,1,Dial(SIP/${EXTEN:1}@sip.provider.com,30,rT) |
| 07:02.07 | CaRb0n^ | any one knowz about chan spy |
| 07:02.08 | sl16 | to outbound dialing, and i am getting this notice |
| 07:02.16 | CaRb0n^ | how to install it on asterisk@home 1.5 |
| 07:02.22 | jaike | oh ok. your * server isnt authenticating properly with your sip provider |
| 07:02.40 | sl16 | it says that it is registered |
| 07:02.47 | sl16 | is this enought ? |
| 07:02.49 | jaike | no |
| 07:02.55 | jaike | registered is for inbound calls |
| 07:03.11 | jaike | for outbound..your username/password is usually included in the Dial statement |
| 07:03.49 | jaike | unless your provider does authentication by ip address |
| 07:04.05 | sl16 | so if i want to make outbound calls from any of the internal sip phones thru * i have to make username and secret match those in my provider-account |
| 07:05.18 | Psykick | clive: I'm using IAX2 trunks |
| 07:06.31 | *** join/#asterisk stoffell (n=stoffell@fw.catsanddogs.com) |
| 07:09.10 | WAudette | Is anyone available to help me troubleshoot a problem I am having with one of my SIP extentions loosing outgoing voice after activating a second or third line? I am a novice and I am using AMP for almost everything. I am running A@H 2.5 on CentOS with all the yum updates. |
| 07:09.20 | CrashHD | do patterns match before actuall numbers, for instance: will Dial digits "911" in a 2 line context match _X.,1 if it is first, first? or will it match 911,1,.....? |
| 07:09.35 | *** join/#asterisk ThaZZa_Work (n=me@203.80.44.200) |
| 07:09.56 | CrashHD | the specifics I found on sorting really only reference includes and non includes |
| 07:09.57 | jaike | sl16: yes |
| 07:11.38 | *** join/#asterisk af_ (n=af@ip-165-17.sn2.eutelia.it) |
| 07:12.08 | WAudette | Otherwise, everthing else seems to work fine. I am using a Polycom 500 on x201. If I restart asterisk it starts working again. It only occurs after running multiple lines and only after the first time. It's almost like it is keeping a channel open somewhere and not fully hanging up. |
| 07:12.28 | sl16 | jaiger: ok, i'll try, thank you |
| 07:13.03 | WAudette | The ZAP trunk still passes voice while this is happening BTW, via an analog x200 on Channel 1 |
| 07:13.14 | jaike | sl16: on your Dial statement it should be something like Dial(SIP/username:secret@sip.provider.com/${EXTEN:1},30,rT) |
| 07:13.32 | jaike | try changing your Dial statement 1st |
| 07:13.40 | sl16 | ok |
| 07:13.51 | WAudette | So, I think it has to do with the SIP extention itself. If I keep it down to one call at a time it seems to work fine forever. |
| 07:14.07 | *** join/#asterisk oej (n=oej@swissco012231-3-3.clients.easynet.fr) |
| 07:14.37 | *** join/#asterisk Eggplant (n=none@dsl-352.cascadeaccess.com) |
| 07:14.58 | jaike | waudette: you have enough bandwidth? one ulaw call uses up 84kpbs up/down |
| 07:15.17 | CrashHD | nevermind I figured out my question |
| 07:15.40 | CrashHD | (Asterisk only sorts patterns, extensions are left in the order you put them in) |
| 07:15.49 | WAudette | Plenty of bandwidth as the SIP extention is local and the traffic outbound it via an POTs line on ZAP trunk 2 |
| 07:15.55 | jaike | oh ok |
| 07:16.10 | WAudette | jaike: Thanks though, good thinking. |
| 07:16.11 | bsdfreak | uh |
| 07:16.13 | bsdfreak | wtf bbq |
| 07:16.50 | WAudette | jaike: Any other ideas? |
| 07:16.58 | *** join/#asterisk sternn (n=sternn@user-0c938ku.cable.mindspring.com) |
| 07:17.41 | jaike | not running any firewalls? udp ports 10000-20000 should be open for rtp |
| 07:17.43 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 07:17.50 | *** join/#asterisk flashnet (i=flashnet@Darkstar.AceShells.com) |
| 07:17.57 | jaike | cant think of any other cause |
| 07:18.30 | WAudette | jaike: It's all open, without a firewall between the Polycom and the Asterisk Server, but again, a very good question. |
| 07:18.47 | WAudette | jaike: I should have thought to put that in my background information. |
| 07:18.59 | WAudette | jaike: Sorry... |
| 07:20.08 | WAudette | I can replicate the issue pretty easily. I am going to try seeing if just restarting sip will clear the issue in an attempt to narrow down the problem. |
| 07:21.12 | WAudette | At first I thought I had to resart the whole server to clear the problem, then I narrowed it down to restart gentle (I think that was the command, must look at notes) |
| 07:21.36 | WAudette | Now I am going to force it to happen again and try just restarting sip. |
| 07:21.55 | sl16 | jaike: No such host: sip.provider.com/082824259 .... i have route to this gate, ping everythiing!??! |
| 07:22.32 | *** join/#asterisk welles (n=welles@222.90.91.20) |
| 07:22.32 | jaike | sl16: replace sip.provider.com with your provider :P |
| 07:22.59 | sl16 | jaiger: i am changing it only in here butin my conf is correct ... |
| 07:23.04 | luke-jr_ | Is there a way to dump the extentions.conf equivalent of AEL? |
| 07:23.18 | jaike | waudette: try reload chan_sip.so |
| 07:23.31 | welles | hi all |
| 07:24.26 | CrashHD | any way to get console colors when connecting with -r? |
| 07:25.03 | jaike | -rc? |
| 07:25.33 | CrashHD | nope |
| 07:25.37 | CrashHD | still b&w |
| 07:25.41 | CrashHD | I tried that a bit earlier |
| 07:25.44 | CrashHD | I've searched around |
| 07:25.47 | jaike | think youll have to restart * with -vvvvc |
| 07:25.48 | CrashHD | can't seem to find an answer |
| 07:25.59 | CrashHD | feature request! |
| 07:26.00 | CrashHD | lol |
| 07:26.50 | WAudette | jaike: I'll try reload chan_sip.so as soon as I can get it to repeat the problem... LOL I thought it was easier to reproduce than this! <grin> |
| 07:27.24 | luke-jr_ | I usually do about -vvvvvvvvvvvvvvvvvvvvv |
| 07:27.30 | luke-jr_ | :) |
| 07:27.33 | CrashHD | lol |
| 07:27.39 | CrashHD | never hurt to have a few v's :) |
| 07:28.00 | luke-jr_ | "Verbosity was 20 and is now 22" |
| 07:28.41 | sternn | I use set verbose 90. You think that might be overkill? |
| 07:29.13 | welles | i meet a strang problem.i have two iax softphone behind the same nat.one use port 18110(B) and the other use 18720(C). when asterisk A try to transfer between softphone B and another asterisk D .i found asterisk D send TXCNT to port 18720(C) while not 18110(B). so they always TXREJ. what's wrong? |
| 07:29.19 | luke-jr_ | sternn: ;) |
| 07:30.16 | luke-jr_ | BTW, 'switch...pattern' should be documented |
| 07:31.34 | CrashHD | do included contexts always get last priority? |
| 07:31.39 | CrashHD | I want to create a 911 context |
| 07:31.41 | CrashHD | (not a macro) |
| 07:31.53 | CrashHD | but want it to always take priority |
| 07:32.17 | welles | any ideas? |
| 07:33.01 | CrashHD | nm I got it |
| 07:33.31 | WAudette | jaike: I give up, I can't get it to reproduce the error. I guess I'll resume in the morning. It seems to happen over night anyway. I'll just have to call myself first before I place a call to a customer in the morning. It's embarasing when I can hear them but they can't hear me. |
| 07:34.13 | WAudette | I just assumed it happened because of all the testing I did the night before while running multiple lines at once. |
| 07:35.36 | WAudette | BTW, who here has had good experience with the Polycom phone line? I have to decide between Polycoms and Snom for a 10 station phone roll out. I am partial to the Polycom hardware, it just feels nice. |
| 07:35.50 | welles | sometimes cmd meetme list <confno> can not show correct channel |
| 07:36.16 | DarthClue | i've had great results with the ip501s |
| 07:36.44 | jaike | waudette: weve 70 polycoms...tough phone |
| 07:37.06 | WAudette | DarthClue: I havent had any problems upgrading the BootROMs and SIPs on the 500 either. It'll be a mix between 501 and 601s. |
| 07:37.27 | WAudette | jaike: Wow, I just happened to hit on the right bunch of people! |
| 07:37.39 | *** join/#asterisk lorinc (n=ang@caracas-2297.adsl.interware.hu) |
| 07:37.59 | luke-jr_ | CrashHD: I think they take the priority based on where you place them |
| 07:38.12 | WAudette | Others were telling me to go with the Grandstreams or Snoms. Has anyone had a Snom and a Poly and would like to compare the two? |
| 07:39.31 | DarthClue | i've used the grandstreams as well, the poly is better imho. haven't ever touched a snom, but the poly still 'looks' better than the snoms i've seen. |
| 07:39.41 | jaike | cons about polycoms, they take a lot of time to boot up after reconfiguration |
| 07:40.04 | jaike | but once set, theyre pretty stabl |
| 07:40.47 | WAudette | Jaike: BTW, what does the Split function do? I haven't found that in the manual. Seems to come up while in confrencing? I have an Avaya Definity G3 background and am finding the transition in terminology to be a bit difficult at times. |
| 07:41.46 | jaike | waudete: no idea bout that |
| 07:41.46 | WAudette | Jaike: Are you using FTP to program your phones also? |
| 07:41.48 | jaike | sorry |
| 07:41.58 | WAudette | jaike: NP |
| 07:41.59 | jaike | yes, ftp |
| 07:42.21 | jaike | set ftp server on phone..copy config of other phone, change a few lines, its finished |
| 07:42.28 | WAudette | It works pretty slick. Any other tips you can think of? |
| 07:42.56 | jaike | none i can think of right now |
| 07:43.02 | *** join/#asterisk jeebusroxors (n=jeebusro@29palms-cuda1-68-170-36-65.losaca.adelphia.net) |
| 07:43.41 | WAudette | Message button. I haven't gotten it to work yet, or directory, Services, Call Lists. |
| 07:43.55 | WAudette | Not sure what they should do anyway. |
| 07:44.14 | WAudette | Well I guess Call lists does work. |
| 07:45.00 | WAudette | jaike: Geesh! Good luck on the backups. |
| 07:47.08 | *** join/#asterisk jeebusroxors (n=jeebusro@29palms-cuda1-68-170-36-65.losaca.adelphia.net) |
| 07:47.55 | *** join/#asterisk wundaboy (n=asdf@c-24-21-100-201.hsd1.or.comcast.net) |
| 07:48.00 | *** join/#asterisk vgster (n=vg@host217-45-221-53.in-addr.btopenworld.com) |
| 07:51.02 | *** join/#asterisk ToTo (n=ToTo@host20-162.pool870.interbusiness.it) |
| 08:00.24 | *** join/#asterisk xorsysd (n=variant@cpe-24-175-60-183.houston.res.rr.com) |
| 08:01.58 | xorsysd | Everytime I receive or try to make a call I'm getting WARNING[8759]: chan_sip.c:1210 retrans_pkt: Maximum retries exceeded on transmission 2ff02f2-9c@147.135.28.128 for seqno 1 (Critical Response), can anyone offer advice? |
| 08:02.31 | *** join/#asterisk akrall (i=user@201.152.155.229) |
| 08:03.18 | akrall | Guys.. how can I make a queue member always be registered on the queue without the need to login or do agentcallbacklogin? |
| 08:05.42 | evilbuny | xorsysd: looks like a firewall issue |
| 08:05.46 | evilbuny | firewall/nat |
| 08:06.03 | jaike | xorsysd: used to get that error when my sip provider was sending my * server a call but the DID wasnt in extensions.conf |
| 08:06.04 | xorsysd | I thought I had fixed that |
| 08:06.38 | jaike | adding the DID fixed it..outbound? no idea |
| 08:06.56 | *** join/#asterisk KaBewM (n=DA-MAN@66-215-7-106.dhcp.psdn.ca.charter.com) |
| 08:08.55 | trym | Asterisk has just borked, and is trying to register like a manic.. but I get timed out errors.. even though the sip server is fine and the internet connection is up and running.. any idea? |
| 08:09.34 | jaike | borked? |
| 08:09.44 | xorsysd | borked?? |
| 08:10.02 | shido6 | heh |
| 08:10.23 | vgster | akrall: dont you just add member=whatever in the config for the queue |
| 08:10.24 | trym | borked.. as in nothing works ;) |
| 08:10.31 | trym | that is.. its not registering to the sip server |
| 08:13.06 | *** join/#asterisk Camonz (i=CamonZ@200.8.20.210) |
| 08:13.07 | Camonz | hi |
| 08:13.43 | trym | it happened after a power failure |
| 08:15.25 | vgster | hi |
| 08:17.16 | *** join/#asterisk ToTo (n=ToTo@host20-162.pool870.interbusiness.it) |
| 08:17.41 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 08:19.02 | akrall | vgster: but if you add a member in agents.conf, they need to then dial something and enter their password to log into the acd dont they? |
| 08:19.35 | Camonz | anybody has any idea how hosted pbx systems work?, i was just cheking a website for a company that says the only thing they install is a voip phone |
| 08:20.08 | Camonz | how do they make the routing from the pstn to their own network so companies keep their phone number |
| 08:20.55 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-225-183.claranet.co.uk) |
| 08:21.22 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-225-183.claranet.co.uk) |
| 08:21.57 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-225-183.claranet.co.uk) |
| 08:22.18 | MGSsancho | read the asterisk manual :) |
| 08:22.30 | Camonz | is there a manual? :-> |
| 08:24.03 | trixter | ~docs |
| 08:24.08 | jbot | somebody said docs was probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 08:24.08 | Camonz | :-> |
| 08:24.36 | Camonz | thks |
| 08:25.05 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-225-183.claranet.co.uk) |
| 08:26.41 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-225-183.claranet.co.uk) |
| 08:26.55 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-225-183.claranet.co.uk) |
| 08:28.39 | *** join/#asterisk dragonheart (n=dragonhe@gentoo/developer/dragonheart) |
| 08:37.39 | akrall | Guys.. how can I make a queue member always be registered on the queue without the need to login or do agentcallbacklogin? |
| 08:37.47 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 08:38.12 | luke-jr_ | any simple way to get a reason Asterisk is rejecting a SIP registration? |
| 08:38.40 | diLLec | <PROTECTED> |
| 08:38.40 | diLLec | <PROTECTED> |
| 08:38.40 | diLLec | <PROTECTED> |
| 08:38.48 | luke-jr_ | that doesn't seem to give any reasons |
| 08:39.16 | trixter | no it requires just a little bit of knowledge to read the headers and see what error is being returned |
| 08:39.20 | trixter | but not a lot of knowledge |
| 08:39.57 | luke-jr_ | I can see the headers fine, but they're too vague |
| 08:40.15 | trixter | what specifically is vague? you should see a 4xx or 5xx error code returned |
| 08:40.28 | mzo | heh, what's error 29? :P |
| 08:40.28 | trixter | and that error message tells you exactly what isnt working the way you think it should |
| 08:40.52 | luke-jr_ | 401 Unauthorized |
| 08:40.58 | luke-jr_ | that tells me nothing except that it isn't working |
| 08:41.36 | dragonheart | mzo: the error after 28 |
| 08:41.43 | my007mssrv | hello all |
| 08:41.48 | my007mssrv | pleas i need help |
| 08:41.58 | luke-jr_ | http://pastebin.ca/42874 |
| 08:42.18 | trixter | it tells you to verify both the user name and password |
| 08:42.21 | my007mssrv | i have install new card tdm2424E |
| 08:42.30 | luke-jr_ | trixter: I'm not even getting prompted for my password |
| 08:42.34 | luke-jr_ | and username is fine |
| 08:42.50 | trixter | if you have a limit on domains set in sip.conf it also tells you to check the sip domain |
| 08:43.01 | trixter | what specifically is not prompting you for a password? |
| 08:43.09 | my007mssrv | but i see that /etc/zapte.conf have all like this |
| 08:43.25 | my007mssrv | # Span 1: WCTDM/0 "Wildcard TDM2400P Prototype Board 1" |
| 08:43.25 | my007mssrv | # channel 1, WCTDM, inactive. |
| 08:43.32 | my007mssrv | in all 24 chanel |
| 08:43.41 | my007mssrv | what is wrong with my cad |
| 08:43.45 | my007mssrv | ?? |
| 08:44.50 | luke-jr_ | trixter: KPhone, which prompts me on the old Asterisk setup |
| 08:45.17 | luke-jr_ | trixter: I'm not aware of any domain restrictions, so unless one is hiding from the examples I modified... |
| 08:45.56 | trixter | well that is the problem with kphone, it should read the 401 and prompt you for your username and password or let you enter it somewhere else |
| 08:46.02 | my007mssrv | any one use TDM4009 b4 |
| 08:46.30 | luke-jr_ | trixter: so why does it work w/ the old config? |
| 08:46.55 | trixter | the domain part is allowexternaldomains=yes and then you have to do domain=blah.com or whatever for each domain that you want to allow |
| 08:47.02 | trixter | you can also bind a domain to a context if you really wanted to |
| 08:47.08 | luke-jr_ | hrm |
| 08:47.16 | trixter | so if you didnt adjust that then you should be ok |
| 08:47.22 | luke-jr_ | ok... more info: with old config, it works via '127.0.0.1', but not 'localhost' |
| 08:47.42 | trixter | becuase I think the default is to allow anything so 'it just works' - like how microsoft allows everything so people dont have to configure as much to get stuff to work |
| 08:48.38 | trym | what can cause asterisk to say a sip server is unreachable when its not ? |
| 08:48.51 | Winkie | trym: do you have a qualify statement? |
| 08:49.25 | luke-jr_ | Feb 23 09:00:19 NOTICE[20652]: pbx.c:1729 pbx_extension_helper: Cannot find extension context 'home' |
| 08:49.25 | luke-jr_ | :( |
| 08:50.46 | trym | winkie: yes |
| 08:51.01 | trym | winkie: well it is actually unreachable to asterisk.. but in reality it shouldnt be |
| 08:51.05 | trym | well |
| 08:51.10 | trym | asterisk communicates with the sip server |
| 08:51.17 | trym | it tries register (without password I think) and get a 401 |
| 08:51.47 | my007mssrv | is there is any how to know if i have pc 2.2 or not ?? |
| 08:51.53 | my007mssrv | is there is any how to know if i have pci 2.2 or not ?? |
| 08:52.05 | vgster | you coudl check the mobo manual |
| 08:53.04 | Winkie | trym: i'm honestly not sure what a 401 is :) |
| 08:53.06 | my007mssrv | i think it's error that many face it i see that all chanel have something like this # channel 1, WCTDM, inactive. |
| 08:53.14 | my007mssrv | what this mean |
| 08:53.18 | Winkie | just a qualify timeout was the first thing i thought of |
| 08:54.09 | my007mssrv | did dtm4000 need something strange i install many card but no one make this error |
| 08:56.00 | my007mssrv | genzaptelconf know it's have 24 chanel but is not make config it |
| 08:57.33 | *** join/#asterisk masked (n=masked@static-203-87-16-192.vic.chariot.net.au) |
| 08:57.38 | masked | is anyone using a standalone fxo device in australia like the linksys pap2 or wrtp54g? |
| 08:57.49 | masked | or wrt54gp2 for that matter. |
| 08:58.22 | *** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk) |
| 09:00.12 | *** join/#asterisk Eggplant (n=none@dsl-352.cascadeaccess.com) |
| 09:02.06 | vgster | has anyone tried any of the options available in zconfig.h in zaptel? |
| 09:02.59 | masked | yep |
| 09:03.01 | masked | i had to |
| 09:03.08 | masked | for .au phone lines |
| 09:04.11 | vgster | did youy use CONFIG_CALC_XLAW at all? |
| 09:04.25 | vgster | im interested to know what small number of channels or small L2 cache means by small |
| 09:04.47 | vgster | and would CONFIG_ZAPTEL_MMX make a great deal of difference on intel machines |
| 09:04.52 | masked | i guess small l2 cache means if you have a celeron or semperon or alike |
| 09:05.04 | masked | well it couldn't hurt |
| 09:05.17 | vgster | ok |
| 09:05.48 | vgster | ive enabled them but couldnt find much info about them apart from what it says in the .h file |
| 09:06.01 | masked | tried voip-info.org? |
| 09:06.40 | *** part/#asterisk akrall (i=user@201.152.155.229) |
| 09:08.10 | vgster | yes but all i could find was stiuff i already had |
| 09:08.28 | vgster | but i dont think it will be a negative to enable them |
| 09:12.30 | *** join/#asterisk welles (n=welles@222.90.91.20) |
| 09:18.42 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 09:20.18 | *** join/#asterisk WasPhantom (n=neil@203-86-197-11-lightning.thepacific.net) |
| 09:21.44 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
| 09:23.24 | trym | hmm |
| 09:23.27 | trym | this is very strange |
| 09:23.27 | trym | lord |
| 09:26.10 | luke-jr_ | http://pastebin.ca/42875 -- any idea what, if anything, is wrong with this? :\ |
| 09:26.25 | luke-jr_ | I get stuff like: Feb 23 09:36:00 WARNING[22811]: ast_expr2.fl:187 ast_yyerror: If you have questions, please refer to doc/README.variables in the asterisk source. |
| 09:26.34 | luke-jr_ | followed by: Feb 23 09:36:00 ERROR[22811]: app_cut.c:391 acf_cut_exec: CUT() requires an argument |
| 09:30.55 | trym | http://tryms.no/dump.txt <-- Why doesnt asterisk send a password with the register? |
| 09:32.05 | trixter | are you looking for a password in plaintext? |
| 09:32.13 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
| 09:32.51 | trym | well Im looking for a reason why asterisk gets 401 forbidden from the sip server |
| 09:33.50 | trym | trixter: any ideas? |
| 09:35.30 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 09:37.26 | Ikarus | hmmmm, wonder how I can make a web-dial-form with Asterisk (so my users submit a phone number and the name of a local SIP phone and Asterisk first rings the SIP phone and then makes the call out |
| 09:37.51 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 09:38.23 | jaike | Ikarus: manager api |
| 09:38.43 | *** join/#asterisk d^k (i=foobar@0x50a6992a.bynxx15.adsl-dhcp.tele.dk) |
| 09:41.21 | Ikarus | jaike: documentation on that is where on the wiki ? |
| 09:41.58 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 09:43.50 | jaike | were implemented that using PHP |
| 09:43.53 | jaike | weve |
| 09:43.56 | d^k | can anyone advise decent books on *? I've never touched neither * nor voip before |
| 09:44.57 | jaike | d^k: asterisk by oreilly. excellent book |
| 09:45.08 | brookshire | d^k: the o'reilly book is pretty good |
| 09:45.19 | brookshire | it's available in pdf too |
| 09:45.27 | jaike | brookshire: :P |
| 09:45.41 | welles | trym, do u have any ideas? |
| 09:45.46 | Ikarus | jaike: yes, but I need the documentation |
| 09:45.51 | welles | and trixter ？ |
| 09:45.55 | brookshire | http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 09:45.57 | d^k | and anything about voip in general but not too general? |
| 09:46.00 | trym | welles: not really ? |
| 09:46.06 | trym | how do I specify which source port asterisk should use |
| 09:46.12 | welles | yes |
| 09:46.17 | welles | trym, yes |
| 09:46.32 | welles | i can see from iax2 debug |
| 09:47.00 | trym | hmm ? |
| 09:47.46 | brookshire | wow.. i should goto bed now |
| 09:48.08 | jaike | d^k: the oreilly book includes topics on pstn, voip, voip protocols, etc |
| 09:48.15 | welles | from database .i saw two iax softphone use the same ipaddr but different port |
| 09:48.44 | Ikarus | brookshire: who is smoking so much crack he packaged a PDF in a zip file, especially as they both use the same compression (zlib) |
| 09:48.49 | trym | ah ok |
| 09:49.08 | brookshire | that would be lief ;) |
| 09:49.21 | brookshire | aka blitzrage |
| 09:49.40 | welles | trym, btw, one asterisk version is 1.2.1 and the other is 1.2.4. is it something with this？ |
| 09:52.32 | welles | trym, also.sometimes i use cmd meetme list <confno>. the channel info in the list sometimes show wrong.it sometimes show ipaddr sometimes show username. when show username sometimes it will show another softphone's username |
| 09:53.06 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 09:57.27 | *** join/#asterisk P0L0 (n=n0n3@62-43-65-175.user.ono.com) |
| 09:59.21 | *** join/#asterisk QuAtRo[NL] (n=dennis@dsl-083-247-051-039.solcon.nl) |
| 09:59.52 | QuAtRo[NL] | Does enoyone have any experience with IVR that does not work? |
| 10:00.28 | QuAtRo[NL] | The playback works, but when pressing the 1 for helpdesk, Asterisk does not respond |
| 10:00.36 | Ikarus | Bleah, annoying, Asterisk has some of the same stupid limitations of other PBX/IVRs |
| 10:01.30 | trym | insmod: error inserting '/lib/modules/2.6.12/misc/ztdummy.ko': -1 Unknown symbol in module |
| 10:01.40 | ChrisUK | its only as good as the person configuring it |
| 10:01.45 | *** join/#asterisk saftsack (n=saftsack@p54A7DAAD.dip.t-dialin.net) |
| 10:02.25 | *** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net) |
| 10:02.40 | jaike | Quatro[NL]: either its a dtmf problem with your phone, or you dont have a exten => 1,1 |
| 10:03.06 | trym | Unable to open IAX timing interface: No such device or address |
| 10:03.51 | Ikarus | Hmmmm, with manager API dial-out stuff, can you make the SIP phone you are using display a special extension or something |
| 10:03.58 | jaike | Quatro[NL]: did you use Playback or Background? |
| 10:05.07 | QuAtRo[NL] | Background, because Playback does not 'see' what the caller does |
| 10:05.16 | QuAtRo[NL] | I doe have: exten => 1,1,Goto(helpdesk,s,1) |
| 10:05.38 | jaike | CLI messages should show the problem |
| 10:06.36 | *** join/#asterisk Bambr (n=Bambr@213-35-238-17-dsl.end.estpak.ee) |
| 10:08.52 | QuAtRo[NL] | Hold on, currently trying dtmf... |
| 10:15.46 | QuAtRo[NL] | Ok, dmtf was the problem |
| 10:15.49 | QuAtRo[NL] | Thanks a lot |
| 10:16.01 | jaike | no problem |
| 10:16.30 | *** join/#asterisk Guest^DJ (i=me@211.24.146.12) |
| 10:19.31 | *** join/#asterisk Kernel_core (i=Kernel_C@217.218.80.232) |
| 10:20.36 | luke-jr_ | blar |
| 10:20.45 | luke-jr_ | AEL has some... issues :\ |
| 10:21.30 | jaike | luke-jr_: new features are often riddled with bugs |
| 10:28.02 | luke-jr_ | !!! |
| 10:28.21 | saftsack | asterisk isnt unstable imho |
| 10:28.26 | saftsack | i mean 1.2.4 |
| 10:28.33 | luke-jr_ | weird, the new asterisk setup I have isn't supposed to be able to interact with the phones yet |
| 10:29.19 | jaike | lol |
| 10:30.39 | *** join/#asterisk fulgas (n=fulgas@82.102.2.254) |
| 10:31.02 | saftsack | pri is dammit expensive. the telekom (telephone provider in germany) takes 280per month for one E1 pri line. + the costs for each call. :( |
| 10:31.24 | RoyK | saftsack: there are issues with all available versions of asterisk, issues that are bad in a large production environment |
| 10:31.38 | RoyK | saftsack: heh |
| 10:31.45 | RoyK | saftsack: you can prolly get it cheaper |
| 10:31.55 | RoyK | saftsack: we don't pay that much up here in .no |
| 10:32.02 | saftsack | .no = norway? |
| 10:32.09 | RoyK | aye |
| 10:32.18 | saftsack | hmm i hate the telekom |
| 10:32.32 | jaike | Royk: id have to agree. asterisk isnt ready for mission-critical, 99.9999% uptime setups |
| 10:32.40 | saftsack | RoyK, but i heard often that asterisk was done for this big environments |
| 10:32.42 | jaike | or even 99% |
| 10:33.00 | saftsack | why? there is imho one factor |
| 10:33.09 | RoyK | saftsack: we're building a 20-pri setup these days and there _are_ issues |
| 10:33.11 | saftsack | the hardware, or are there other issues that prevent mission critcal? |
| 10:33.24 | RoyK | the software |
| 10:33.25 | Kernel_core | jaike: http://www.asterisk.org/developers/astridevconeurope2006 end of 2006 asterisk 1.6 will be released :) |
| 10:33.35 | saftsack | RoyK, do you mean asterisk itself with zaptel? |
| 10:33.43 | RoyK | yes |
| 10:33.58 | RoyK | zaptel is quite stable imo |
| 10:33.59 | saftsack | really? because i heard other things |
| 10:34.07 | saftsack | yes and libpri too |
| 10:34.26 | RoyK | I've also heard other things, like people should be nice to oneanother, never hit, never swear etc |
| 10:34.40 | saftsack | :) |
| 10:34.40 | RoyK | but I beleive that's not the way the world is right now |
| 10:34.56 | *** join/#asterisk Dabba (n=d@ipv6.mfnx.ip6net.net) |
| 10:34.57 | RoyK | libpri also seems stable |
| 10:35.00 | saftsack | there must be a price for the cheapness of a * server |
| 10:35.16 | RoyK | there certainly is |
| 10:36.14 | saftsack | RoyK, so do you mean that there are logical bugs in the dialplan interpreter? |
| 10:36.27 | RoyK | that as well |
| 10:36.33 | *** join/#asterisk cogineo (n=cogineo@81.215.105.114) |
| 10:36.35 | RoyK | i've reported a bug about that |
| 10:36.58 | saftsack | but is there also a problem when it runs once? |
| 10:37.15 | saftsack | because if it runs one day and the drivers are ok it should run ever |
| 10:37.16 | Dabba | has anyone got a pointer to a url for sending email notifcation of missed calls? |
| 10:37.18 | RoyK | as in 'if _0047X. is specified as a pattern and _00479X. is specified as another, the latter isn't necessarily being used' |
| 10:37.21 | jaike | Kernel_core: thats good news..but i only need mixmonitor to run properly now |
| 10:37.29 | RoyK | it's not stable |
| 10:37.32 | RoyK | that's the problem |
| 10:37.49 | saftsack | asterisk itself? do you mean segfaults and so on? |
| 10:38.03 | RoyK | we had one box hanging 3-4 times a day at worst, running 1.2.4, and that didn't have much traffic either |
| 10:38.24 | RoyK | this wasn't segfault, but deadlock afaics |
| 10:38.35 | jaike | royk: if its one box..might be hardware issues |
| 10:38.36 | saftsack | what is deadlock? total system freeze? |
| 10:38.46 | saftsack | i have 2 1.2.4 boxes |
| 10:39.02 | saftsack | my box at home runs stable since 4 days (i set it up 4 days ago) |
| 10:39.07 | jaike | i got fequent segfaults with 1.2.4 when recording meetme conferences..had to disable recording |
| 10:39.08 | saftsack | for 4 days i mean |
| 10:39.40 | saftsack | and another box in the buero runs since yesterday and is frequently used w/o problems yet |
| 10:41.02 | RoyK | jaike: no it might not be that imho. no zaptel. no nothing. |
| 10:41.06 | RoyK | jaike: http://bugs.digium.com/view.php?id=6445 |
| 10:41.22 | saftsack | there are several bugs thats right but if you dont use the feautres which are having such bugs asterisk runs 100% stable |
| 10:42.11 | saftsack | Please try latest SVN trunk or SVN asterisk 1.2, as i've just fixed a bunch of seg faults. |
| 10:42.14 | saftsack | lol a bunch ^^ |
| 10:42.43 | *** join/#asterisk ToTo (n=ToTo@host20-162.pool870.interbusiness.it) |
| 10:42.49 | saftsack | all developers want to force the production people to go to the newest version ^^ because they dont want fix the old ones ^^ |
| 10:43.22 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 10:43.23 | RoyK | saftsack: i have tried latest whatever but the problem is that trunk also has problems (meetme doesn't parse config correctly) |
| 10:43.43 | RoyK | perhaps someone should start an 'asterisk stabilisation project' |
| 10:43.45 | RoyK | hm |
| 10:43.49 | RoyK | btw that's already there |
| 10:43.52 | RoyK | it's called openpbx :P |
| 10:44.45 | saftsack | lol |
| 10:44.56 | saftsack | is openpbx good? |
| 10:45.17 | luke-jr_ | hrm |
| 10:45.24 | luke-jr_ | the asterisk voicemail beep changed |
| 10:46.30 | saftsack | RoyK, is openpbx more stable than asterisk? |
| 10:48.19 | *** join/#asterisk welles (n=welles@219.144.238.87) |
| 10:48.32 | RoyK | I beleive so, yse |
| 10:48.33 | RoyK | yes |
| 10:48.49 | saftsack | so why are u using asterisk and not openpbx? |
| 10:49.03 | saftsack | dont answer with good question please ;) |
| 10:49.12 | *** join/#asterisk my007mssrv (n=my@213.158.171.162) |
| 10:49.17 | my007mssrv | hello all |
| 10:50.06 | Ikarus | Hmmmmm, there is no usable Asterisk Manager API module for Perl (there is a PoCo WIP) |
| 10:50.08 | my007mssrv | i have new DTM2400 card but i think i can not used it coz my mothere board not see it right |
| 10:50.33 | my007mssrv | i have http://www.gigabyte.com.tw/Products/motherboard/Products_Overview.aspx?ClassValue=motherboard&ProductID=1909&ProductName=GA-8I945G and this need to install driver for linux |
| 10:50.42 | my007mssrv | to see pci 2.2 |
| 10:51.44 | my007mssrv | did anyone face like this problem b4 |
| 10:57.21 | Dabba | ANyone do missed call alerts via email ? |
| 10:58.49 | *** join/#asterisk fourcheeze (n=rich@westbury.doilywood.org.uk) |
| 10:58.57 | *** join/#asterisk Eggplant (n=none@dsl-352.cascadeaccess.com) |
| 11:00.35 | trym | Im having trouble getting ztdummy to work.. modprobe always states it does not exist |
| 11:01.25 | fourcheeze | does it exist? |
| 11:03.05 | luke-jr_ | Any idea what would happen if I configured two asterisk systems to register on the same DID? |
| 11:05.02 | darkskiez | luke-jr_: depends on the connection technology and the provider |
| 11:05.11 | jaike | luke-jr_: happened to me once, forgot to shutdown test server that was registering to the same provider, was wondering why i was getting half the number of calls |
| 11:05.59 | jaike | i think the last system to re-register gets the call |
| 11:06.15 | *** join/#asterisk razu (n=razu@tln-kontor.norby.ee) |
| 11:08.03 | trym | what can cause my asterisk to not include the authorization part of a REGISTER? |
| 11:09.03 | jaike | 7pm..time to go home |
| 11:09.44 | jaike | bye all |
| 11:09.49 | vgster | trym: have you compiled and installed zaptel against the kernel you are using? |
| 11:10.03 | trym | yes.. is it known to be a zaptel problem? |
| 11:14.12 | fourcheeze | trym: do you have a ztdummy kernel module somewhere? |
| 11:14.45 | trym | yup.. its loaded |
| 11:15.14 | fourcheeze | ok, I thought you said it wasn't |
| 11:15.39 | trym | fuck me.. its working |
| 11:15.53 | fourcheeze | I'd rather not, if that's OK |
| 11:16.30 | fourcheeze | if you're on debian it's normally easiest to build the debian kernel modules |
| 11:16.42 | fourcheeze | and then install with dpkg |
| 11:16.54 | trym | hmm.. Ill consider it |
| 11:23.57 | *** join/#asterisk chapeaurouge (n=chap@vilhost1.vision.lu) |
| 11:30.41 | Ikarus | Anyone have any documentation to go with POE::Component::Client::Asterisk::Manager it is rather, lets put it softly, terse |
| 11:31.26 | *** join/#asterisk fulgas (n=fulgas@82.102.2.254) |
| 11:32.08 | RoyK | Ikarus: heh |
| 11:32.20 | RoyK | Ikarus: i beleive i got something from the author some time back. i'll check |
| 11:32.27 | Ikarus | RoyK: would be great |
| 11:32.45 | *** join/#asterisk apardo (n=apardo@87.218.44.157) |
| 11:33.35 | *** join/#asterisk coppice (n=chatzill@81.166.17.210.dyn.pacific.net.hk) |
| 11:43.45 | Ikarus | RoyK: anything ? |
| 11:43.56 | Guest^DJ | hey guys, is it possible to have * up and running first follow by installing TE110P later? |
| 11:44.31 | *** join/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it) |
| 11:44.34 | |Vulture| | Guest^DJ: of course, you just cant configure zaptel/zapata for the TE110P |
| 11:44.46 | |Vulture| | or * won't start and fail with Zap errors |
| 11:45.17 | Guest^DJ | so all i have to do is just config zaptel/zapata and * should run ? |
| 11:46.26 | |Vulture| | if you have * running and you want to install a TE110P all you have to do is make sure the zaptel drivers are loaded, then modprobe the device drivers and config zaptel.conf and zapata.conf |
| 11:46.45 | |Vulture| | oh and run ztcfg -vv to verify that it is online before starting * |
| 11:47.39 | Guest^DJ | |Vulture|: thanks, that doesnt sound complicated. for a moment there, i though i need to be a rocket scientist |
| 11:48.24 | |Vulture| | nah its getting easier and easier with every day too |
| 11:49.08 | Guest^DJ | when compiling * without TE110P, do i still need to do modprobe ? |
| 11:49.25 | Guest^DJ | or just make install asterisk |
| 11:49.54 | tzafrir_laptop | BTW: zaptel 1.2.4 comited into Debian. Should be uploaded soon |
| 11:50.33 | tzafrir_laptop | modprobe for what? |
| 11:50.43 | tzafrir_laptop | Is it Debian? |
| 11:50.59 | Guest^DJ | no is White box EL4 |
| 11:51.03 | |Vulture| | Guest^DJ: no, only need to modprobe when installing the TE110P and it can be done whenever |
| 11:51.17 | Guest^DJ | roger that |
| 11:51.31 | |Vulture| | my advice when you install * install the libpri and zaptel drivers |
| 11:52.10 | |Vulture| | then when you put the TE110P in all you have to do is modprobe zaptel;modprobe wcte11xp |
| 11:52.10 | Guest^DJ | install libpri and zaptel without doing modprobe ? |
| 11:52.15 | |Vulture| | yea |
| 11:52.19 | Guest^DJ | ok understood |
| 11:52.28 | Guest^DJ | thanks |
| 11:52.30 | |Vulture| | np |
| 11:52.54 | |Vulture| | just come in when your doing it, you will get a lot of help if you need it |
| 11:53.53 | Guest^DJ | btw, do u recommend 1.2.4 when running EL4 on sata drive or just 1.0 would do |
| 11:54.06 | Guest^DJ | i plan to use with h323 |
| 11:54.21 | |Vulture| | I run 1.2.1 because I havent upgraded I use CentOS 4 but I don't use h323 |
| 11:54.26 | |Vulture| | you integrating with CCM? |
| 11:54.49 | Guest^DJ | no just pure routing, telco uses h323 |
| 11:54.59 | *** join/#asterisk VeNoMouS_ (n=venom@60-234-209-199.bitstream.orcon.net.nz) |
| 11:55.36 | |Vulture| | I am sure 1.2.4 would work well there is a lot changed since 1.0 |
| 11:55.39 | VeNoMouS_ | got an issue, we can call voicemail on asterisk from landline and skype, but we try call it from cell phones, we cant hear anything, but we can see asterisk trying to play the welcome msg for voicemail |
| 11:56.15 | VeNoMouS_ | anyone incounted somehting like this, i did find a random patch for a .uk cell phone service, but its not working here |
| 11:56.46 | |Vulture| | VeNoMouS_: Ive never heard of that, what country is it in? |
| 11:56.51 | VeNoMouS_ | nz |
| 11:56.53 | Guest^DJ | funny thing, when i use cvs checkout -r v1-0 . i always end up with version 1.0 |
| 11:57.00 | VeNoMouS_ | ive personally never come across it myself |
| 11:57.09 | Guest^DJ | never 1.2.4 |
| 11:57.13 | *** join/#asterisk emk (n=ekiara@proxy.usiu.ac.ke) |
| 11:57.22 | |Vulture| | Guest^DJ: dload it off the website |
| 11:57.28 | Guest^DJ | tar ball ? |
| 11:57.31 | |Vulture| | yea |
| 11:57.53 | |Vulture| | VeNoMouS_: the logs look the same on landline calls and cell calls? |
| 11:57.57 | VeNoMouS_ | yea |
| 11:57.59 | VeNoMouS_ | thats the fucked thing |
| 11:58.07 | |Vulture| | and if you call in to talk to someone it works fine? |
| 11:58.10 | VeNoMouS_ | i was even ngrep'ing the sip |
| 11:58.15 | Guest^DJ | ok, thanks agin |Vulture| |
| 11:58.39 | emk | hI ALL. Am I in the wrong place to ask a question about GSM call termination? |
| 11:58.45 | VeNoMouS_ | cant really test that, as main lines go via call manager, we using asterisk as just the vm |
| 11:58.53 | |Vulture| | VeNoMouS_: does it see DTMF? |
| 11:59.10 | VeNoMouS_ | hang lemmie find this url its the exact same problem |
| 11:59.14 | |Vulture| | kk |
| 11:59.24 | |Vulture| | emk: just ask if someone knows they will shout it out |
| 11:59.38 | *** join/#asterisk kippi (n=chris@untrust-gct.equinoxit.net) |
| 12:00.14 | kippi | how well will asterisk run on Fedora Core 4, zaptel etc ? |
| 12:00.52 | |Vulture| | well I believe FC4 is the testing ground for * by Digium so I would immagine quite well |
| 12:01.00 | emk | |Vulture|: ok |
| 12:01.03 | Guest^DJ | bye guys |
| 12:01.20 | VeNoMouS_ | http://esupport.gradwell.net/index.php?_m=knowledgebase&_a=viewarticle&kbarticleid=200&nav=0,9 |
| 12:01.20 | |Vulture| | I use CentOS4 which is pretty much the same thing as RHEL4 and it works well for me |
| 12:01.22 | VeNoMouS_ | thats it there |
| 12:01.25 | |Vulture| | I used FC before |
| 12:01.44 | RoyK | fedora == beta == evil |
| 12:01.46 | VeNoMouS_ | thing is, ours dont even get to 30secs, cause right off the bat there is dead silence on the cells |
| 12:01.51 | VeNoMouS_ | from 2 differnt carrier providers |
| 12:02.07 | |Vulture| | VeNoMouS_: is this coming in over POTs or an E1? |
| 12:02.32 | VeNoMouS_ | e1 |
| 12:02.38 | |Vulture| | RoyK: hey I push everyone I see to CentOS but if they want to use FC it will work... |
| 12:02.45 | |Vulture| | VeNoMouS_: PRI or banked? |
| 12:02.47 | emk | Does anyone have experience with connecting a _lot_ of VoIP calls to a GSM network? |
| 12:02.49 | VeNoMouS_ | pri |
| 12:02.52 | emk | How would you go about it? |
| 12:03.02 | VeNoMouS_ | goes to a cisco box |
| 12:03.07 | |Vulture| | damn I was gunna say try the callprogression switch but I don't think that will work |
| 12:03.15 | VeNoMouS_ | then diverts some away shit to asterisk |
| 12:03.46 | VeNoMouS_ | thing is, landline calls that arnt part this network work fine to the vm |
| 12:03.55 | VeNoMouS_ | if we try the exact same number from a cell it just dont work |
| 12:04.14 | VeNoMouS_ | from vodafone we get "da da da" |
| 12:04.16 | VeNoMouS_ | as in bad # |
| 12:04.16 | |Vulture| | VeNoMouS_: it sounds right, * detects silence and kills the line. Does that sound like whats happening? |
| 12:04.32 | VeNoMouS_ | yea, but it feels like its from the cell provider side |
| 12:04.46 | VeNoMouS_ | telecom dont kill the line |
| 12:04.48 | VeNoMouS_ | vodafone does |
| 12:04.50 | |Vulture| | yea most likley they have some sort of echo can. hardware running |
| 12:04.53 | |Vulture| | oh |
| 12:05.03 | |Vulture| | strange |
| 12:05.06 | VeNoMouS_ | telecom sits there , but it auto terminates on no noise |
| 12:05.13 | VeNoMouS_ | yea i now |
| 12:05.16 | VeNoMouS_ | s/now/know/ |
| 12:05.46 | VeNoMouS_ | <PROTECTED> |
| 12:06.59 | |Vulture| | Im looking through the zapata.conf sample seeing if any options would make sense and Im not seeing them |
| 12:07.40 | |Vulture| | might help to fiddle with the progzone |
| 12:09.26 | VeNoMouS_ | hrmm |
| 12:09.29 | VeNoMouS_ | yea could do |
| 12:09.42 | *** part/#asterisk emk (n=ekiara@proxy.usiu.ac.ke) |
| 12:10.19 | kippi | so simple words, I should be ok with FC4 |
| 12:12.55 | *** join/#asterisk t0ke (n=kaka@194.Red-81-36-121.dynamicIP.rima-tde.net) |
| 12:14.17 | VeNoMouS_ | no, redhat sucks |
| 12:14.17 | VeNoMouS_ | >:P |
| 12:19.29 | vgster | i moved from fc3 to suse |
| 12:19.31 | trixter | no a 50 cent whore at the docks on payday sucks |
| 12:22.22 | vgster | hmm its payday tomorrow |
| 12:22.45 | Skid | everyday is payday! |
| 12:24.02 | vgster | i get taxed everyday i know that |
| 12:26.49 | mover | why SIPGetHeader isnt no longer in trunk? Onlny SIPAddHeader is implemented |
| 12:27.18 | *** join/#asterisk svenna_ (n=svenna@p548D0414.dip0.t-ipconnect.de) |
| 12:29.23 | mover | ah |
| 12:29.25 | mover | see |
| 12:31.32 | *** join/#asterisk apardo (n=apardo@87.218.44.157) |
| 12:32.38 | *** join/#asterisk fulgas (n=fulgas@82.102.2.254) |
| 12:34.41 | *** join/#asterisk _Paulo_ (n=Paulo@200-168-112-132.dsl.telesp.net.br) |
| 12:38.27 | Ikarus | BLEAH, POE::Component::Client::Asterisk::Manager simply does NOTHING |
| 12:38.43 | Ikarus | it never dispatches events |
| 12:39.20 | *** join/#asterisk fafnir (i=hahaha@unaffiliated/fafnir) |
| 12:39.21 | *** join/#asterisk Eitch (n=hugo@unaffiliated/eitch) |
| 12:40.05 | Skid | bleah... that's a new one |
| 12:40.26 | *** join/#asterisk linville (n=linville@azure.tuxdriver.com) |
| 12:42.53 | Ikarus | heh |
| 12:43.01 | Ikarus | I would prefer a working version of that |
| 12:43.12 | *** join/#asterisk _deg_ (n=deg@200.163.193.247) |
| 12:49.38 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
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| 12:57.31 | *** join/#asterisk VeNoMouS_ (n=venom@60-234-209-199.bitstream.orcon.net.nz) |
| 12:58.25 | tzafrir_laptop | Ikarus, any better alternative? |
| 13:00.09 | Ikarus | tzafrir_laptop: I am busy working on it |
| 13:00.27 | *** join/#asterisk oej (n=oej@fuckoff012231-1-3.clients.easynet.fr) |
| 13:01.03 | jbalcomb | bon journo<SP>!! |
| 13:01.08 | tzafrir_laptop | Ikarus, would love to know of one. Especially something with proper handling of replies to commands |
| 13:01.48 | Ikarus | tzafrir_laptop: they just don't work, I am using a blocking IO version for development in the mean time (my application isn't bothered by it), but I will write a working PoCo for it |
| 13:07.30 | *** join/#asterisk tdonahue (n=tdonahue@208.51.101.201) |
| 13:12.13 | FlyboySR22 | Morning All |
| 13:13.22 | *** join/#asterisk Gennaro (n=Gennaro@ppp-62-10-136-139.dialup.tiscali.it) |
| 13:14.00 | Gennaro | si someone know how i can use and Serial modem on tty1 as fxo? |
| 13:14.26 | Gennaro | an fxo cost 80€ of ups..!! |
| 13:14.58 | Ikarus | tzafrir_laptop: I think I will have it done by the end of last week with proper events and a nice API |
| 13:18.12 | _Paulo_ | Gennaro, I thing you cant use a serial modem as fxo. |
| 13:18.37 | Gennaro | argh.. |
| 13:18.54 | Gennaro | i dunno.. because in config. |
| 13:19.03 | Gennaro | that is an modem.config |
| 13:19.47 | Gennaro | and in this file there are some information about isdn or Aopen .. |
| 13:20.08 | Gennaro | so i belive that can be used a channel modem... |
| 13:20.24 | _Paulo_ | E1/T1 can be used for data and voice |
| 13:20.40 | Ikarus | Gennaro: no, it only works on ISDN modems afaik |
| 13:20.47 | Gennaro | ah.. |
| 13:20.50 | Gennaro | tanx.. |
| 13:20.57 | Gennaro | i must pay... |
| 13:21.05 | _Paulo_ | I really dont know what modem.conf is about, but I never heard of serial modem support within * |
| 13:21.15 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 13:21.25 | oej | chan_modem is now removed from Asterisk |
| 13:21.46 | _Paulo_ | Gennaro, I heard that some softmodems can be used as FXO, but never tried. |
| 13:21.56 | coppice | there used to be a kind of half hearted IVR level support with serial voice modems. actual conversions don't work over those modems |
| 13:22.04 | Gennaro | neither i but i'm going to see the price of an fxo card and is very expensive get it with ups. |
| 13:22.37 | coppice | PCI modem cards using the Intel 536 chips work, by replacing the driver with one that uses the hardware not as a modem |
| 13:22.37 | Gennaro | i tried with an intel modem... |
| 13:23.31 | Gennaro | but i need to dissolve an intergrated to let asterisk works with it... but i'm not sure of damage my pc.. |
| 13:23.46 | Gennaro | vendor must be 80xx |
| 13:23.53 | Gennaro | or something like this.. |
| 13:24.02 | Gennaro | either chipset is intel. |
| 13:24.17 | Gennaro | i dont want damage my pc... |
| 13:25.04 | Gennaro | i have an 537ep |
| 13:25.27 | coppice | sorry 537, not 536. |
| 13:25.43 | coppice | load up the wcfxo driver, and it should work |
| 13:26.02 | Gennaro | i tried a lot of time ago but nothing.. |
| 13:26.17 | Gennaro | if i plug it can u help me with this operation?!? |
| 13:27.06 | _Paulo_ | Gennaro, there are recipes at http://www.voip-info.org |
| 13:27.26 | Gennaro | i did it but doen't work so i tried.. |
| 13:27.43 | coppice | lemon chicken, with ginger and FXO port |
| 13:27.59 | Gennaro | in voip info said that i must remove a component for vendor id. |
| 13:28.19 | Gennaro | coppice?!? |
| 13:29.09 | Gennaro | and the starter kit of asterisk cost 100$ |
| 13:29.25 | Gennaro | more 80 $ of ups.. |
| 13:29.31 | dezent | anyone know how to reduce echo in calls ? |
| 13:29.48 | Gennaro | echo training at 800 is good.. |
| 13:30.00 | dezent | ? |
| 13:30.08 | dezent | i allready made a echoservice =) |
| 13:30.24 | dezent | maybe i could call it and ask... |
| 13:32.47 | dezent | http://www.voip-info.org/wiki/view/Asterisk+echo+avoidance |
| 13:32.55 | dezent | says VOIP Phone <-> Asterisk <-> VOIP Phone |
| 13:33.07 | dezent | wont produce any echo in calls |
| 13:33.14 | dezent | noticable |
| 13:33.18 | dezent | but it does |
| 13:33.22 | *** join/#asterisk rogier (n=rogier@16-65-dsl.ipact.nl) |
| 13:33.47 | dezent | when i call another asterisk i dont have the same problem.. its just when its internal in the same asterisk |
| 13:37.43 | *** join/#asterisk mkl1525 (n=daniel@212.80.239.24) |
| 13:39.03 | _Paulo_ | I want to try AGI. Im familiar with both, Any advice about which one is the best (or stable/mature/safe)? |
| 13:39.24 | _Paulo_ | s/both/both perl and php/ |
| 13:40.38 | *** join/#asterisk tzanger (n=tzanger@mixdown.ca) |
| 13:40.50 | tzanger | bah we're back to this stupid +r crap :-( |
| 13:40.54 | mkl1525 | Hi, could anyone tell me what the meaning of the first line (with C: A: SL: values) of the "show queues" command? voip-info.org seems to be down/very slow. |
| 13:42.46 | coppice | _Paulo_: I am looking at the fax files you posted |
| 13:45.54 | _Paulo_ | coppice, thak you! |
| 13:46.07 | _Paulo_ | s/thak/thank/ |
| 13:46.09 | trixter | either scammers are getting dumber or they realized that there are enough dumb people out there they dont have to be clever. I just got two identical emails full of grammatical errors saying that "several credit card BASES have been hacked" and other funny things.. both from aparent zombie machines. at least they are amusing now |
| 13:46.33 | coppice | are the two tarballs a matching set of files from the two ends of the same call? |
| 13:46.48 | _Paulo_ | no... |
| 13:47.16 | _Paulo_ | both are recordings of sendfax operations. |
| 13:47.32 | _Paulo_ | by the receiving side |
| 13:47.35 | coppice | looking at the received audio I see several hiccups. they all appear to be 6ms long |
| 13:48.57 | _Paulo_ | I can be from sync issues with the telco or something like that? |
| 13:49.15 | _Paulo_ | s/I/It/ |
| 13:50.29 | coppice | the problem I thought might be happening is transmit underflow in *. If that is the problem, there is probably a fairly simple cure. however, if underflow at the tx end causes these its puzzling they seem to be con sonsistent length |
| 13:50.54 | coppice | maybe I should send you a test version of app_rxfax anc app_txfax and let you try them |
| 13:51.40 | *** join/#asterisk SibRw0rk (n=SibRw0rk@66.234.235.84) |
| 13:52.57 | _Paulo_ | great. |
| 13:52.59 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm) |
| 13:57.18 | *** join/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com) |
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| 13:58.08 | *** part/#asterisk cogineo (n=cogineo@81.215.105.114) |
| 13:58.11 | _Paulo_ | coppice, You are the boss. I can give you shell access, if you want to watch the action. |
| 14:00.31 | *** join/#asterisk rajo (n=rajo@bfs.cs.uni-sb.de) |
| 14:01.26 | stas_ | 3 4 april |
| 14:02.07 | vaewyn | anyone know if there is a way to boost the gain on the IAXy? |
| 14:07.51 | coppice | _Paulo_ I just e-mailed you some test code. I haven't tried it, but the changes are fairly simple |
| 14:13.25 | _Paulo_ | will try right now. |
| 14:15.05 | *** join/#asterisk my007ms (n=my@213.158.171.162) |
| 14:15.10 | my007ms | hello all |
| 14:15.43 | my007ms | :) i spen many time try to config my new TDM2400 card but there is probelm with PCI |
| 14:16.02 | my007ms | did any one have idea good idea to star from |
| 14:16.15 | my007ms | my server don't see TDM card right |
| 14:16.25 | apardo | http://www.igestec.com/coccinella/jep-iax/jep-iax.html |
| 14:16.36 | apardo | JEP-IAX: Jingle IAX Transport |
| 14:16.49 | my007ms | in intel website i see that my motherebord |
| 14:17.09 | my007ms | have no driver for linux |
| 14:17.52 | my007ms | and lspci see card wrong any one have idea |
| 14:18.10 | trixter | ok so like after 100 minutes of hold time cogent hung up on me, then they call me back because they were on hold |
| 14:18.37 | trixter | as in I put them on hold while I was on hold |
| 14:18.40 | trixter | I think that is funny |
| 14:19.02 | vaewyn | speaking of cogent... they have royally sucked the last 2 days... evil routing crud from them once we hit chicago |
| 14:19.06 | *** join/#asterisk linville (n=linville@nat-pool-rdu.redhat.com) |
| 14:22.53 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
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| 14:23.34 | *** join/#asterisk fugitivo (n=ajf@201.255.177.92) |
| 14:25.24 | *** join/#asterisk Cherebrum (i=cracka@earth.v2business.com) |
| 14:26.05 | Cherebrum | Does anyone know how to limit concurrent calls to a dynamic agent in a queue? I have agents that are getting phone calls while they are allready on a call. |
| 14:27.33 | trixter | tell the agents to work harder so that doesnt happen |
| 14:27.40 | trixter | problem solved |
| 14:28.57 | *** join/#asterisk CosmicRay (n=jgoerzen@2002:452c:8843:1:20e:a6ff:fe66:c5a3) |
| 14:28.59 | _Paulo_ | coppice, t30_stats_t doesnt have a field current_status in my include/spandsp/t30.h... |
| 14:29.51 | coppice | which version are you using? |
| 14:30.22 | CosmicRay | Got a question about realtime (extconfig) voicemail. According to voip-info, once odbc is enabled for any config file (such as voicemail.conf), then the system ignore the voicemail.conf file on disk. Fine, but the example voicemail schemas on the wiki and in README.extconfig don't address how to store the [general] section in a database. Any ideas? |
| 14:30.50 | *** join/#asterisk apardo (n=apardo@87.218.44.157) |
| 14:31.23 | Cherebrum | coppice: 1.2.1 I know it's a little older but upgrading is slightly a big problem right now. |
| 14:32.16 | _Paulo_ | coppice, spandsp-0.0.2pre25, I think. |
| 14:32.34 | _Paulo_ | should i try spandsp-20060205.tar.gz ?? |
| 14:33.05 | coppice | _Paulo_ its my fault. I was looking at the current code, instead of pre25. hang on. |
| 14:33.32 | *** join/#asterisk FlyboySR22 (i=rsears@gateway.adnc.com) |
| 14:34.29 | iCEBrkr | _Sam--: Yo? you around? |
| 14:34.39 | coppice | _Paulo_ just delete the "If" from both files for now |
| 14:36.41 | _Paulo_ | ok. |
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| 14:45.06 | QuAtRo[NL] | Is there a functionallity for making a queu in Asterisk? |
| 14:45.20 | trixter | what type of queue? |
| 14:45.37 | trixter | there is no functionality to make a print queue in asterisk afaik |
| 14:46.02 | trixter | if you want a queue for inbound callers to goto agents then yes |
| 14:46.15 | QuAtRo[NL] | That's what I mean |
| 14:46.20 | trixter | perhaps queues.conf and agents.conf would shed some light on that |
| 14:46.20 | coppice | a bus queue, for the TDM bus, of course |
| 14:46.34 | QuAtRo[NL] | Ok, thank you |
| 14:46.34 | trixter | I used to ride the TDM bus to school |
| 14:50.21 | *** join/#asterisk austinnichols101 (n=austinni@70.46.69.131) |
| 14:53.32 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:53.48 | *** join/#asterisk mazimp (i=nobody@p54AB3D3F.dip0.t-ipconnect.de) |
| 14:55.46 | vaewyn | Nugget |
| 14:55.50 | Nugget | moo |
| 14:55.52 | vaewyn | You didn't get the memo on that? |
| 14:55.54 | vaewyn | :} |
| 14:55.57 | Nugget | :) |
| 14:56.06 | *** join/#asterisk shadebob (n=shadebob@81.192.114.144) |
| 14:56.41 | *** join/#asterisk GerbilNut (i=GerbilNu@65.88.144.41) |
| 14:56.48 | mazimp | having a (maybe simple) problem with wait() and timeouts - can anybody help me? |
| 14:57.11 | _Paulo_ | coppice, I renamed the "t" variable to "ts" ("t" seems to be used someother place) and got it compiled. |
| 14:57.26 | _Paulo_ | testing now. |
| 14:58.16 | shadebob | hi. I need some help. When I have a com between a FXO and a FXS port, hangup is detected for the FXO before the FXS even if it's the FXS wich hangup first. Any idea? |
| 14:59.44 | *** join/#asterisk kpettit (n=keith@69.15.174.114) |
| 15:01.59 | *** join/#asterisk SplasPood (n=jwb@gate.lga2.us.voxel.net) |
| 15:02.00 | fu3 | hello * |
| 15:02.29 | fu3 | Anyone know what a digital pair gain system is? |
| 15:02.43 | fu3 | I just read a completed trouble ticket from Qwest about my gear, and they said they changed one out. |
| 15:03.18 | trixter | pair gain makes a bunch of gear |
| 15:03.24 | trixter | they were the first to do dsl equipment iirc |
| 15:03.37 | trixter | way back in the days when dsl was done over dry pairs instead of a live circuit |
| 15:03.38 | fu3 | it's a company?? oh :) |
| 15:03.46 | fu3 | this sounds like a lie |
| 15:03.48 | trixter | iirc pairgain.com |
| 15:04.05 | *** join/#asterisk kkun (n=none@198.60.73.230) |
| 15:04.06 | mazimp | when I do a wait(15) on an unsanswered CAPI Channel, *1.2.4 goes to Spawn-Extension after 10 seconds altough ResponseTimeout and Digittimeout = 20, any ideas? |
| 15:04.30 | fu3 | well i'll be damned |
| 15:04.33 | fu3 | pairgaincom.com :) |
| 15:04.56 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 15:04.58 | fu3 | the ticket also says "replaced due to being worn or defective" |
| 15:05.48 | vaewyn | yeah... we have some old dry pair stuff from them still in use... will get replaced this summer with LRE units |
| 15:06.03 | fu3 | wtf is that? |
| 15:06.05 | fu3 | dry pair |
| 15:06.09 | GerbilNut | When was Zaptel 1.2.4 released? |
| 15:06.11 | docelm0 | hahaa.. NEWB! |
| 15:06.14 | fu3 | yeah |
| 15:06.15 | fu3 | big time :) |
| 15:06.16 | docelm0 | no dial tone |
| 15:06.24 | trixter | dry pair is a copper pair with no signal on it |
| 15:06.30 | docelm0 | its where DSL or in the old days Alarm circuts were ran on these |
| 15:06.31 | vaewyn | no dial tone... no blocks... no switches... etc.. |
| 15:06.36 | fu3 | ahhh |
| 15:06.42 | fu3 | i've never seen that :) |
| 15:06.47 | docelm0 | How old are you? |
| 15:06.51 | fu3 | 25 |
| 15:06.55 | vaewyn | big long piece of wire they usually used for alarm signals |
| 15:06.56 | docelm0 | thats why |
| 15:07.01 | GerbilNut | i've ran 2mbps + DSL on Telegraph and Alarm circuits |
| 15:07.04 | coppice | i was thinking unlicked breasts, but maybe i should focus on work :-) |
| 15:07.07 | fu3 | I cant wait for another 25 years to go by so I can say the same things to some newb. |
| 15:07.10 | trixter | its useful if you have a bunch of circuits that are within range (depending on application upto a few miles) and you want to not pay the phone company switching fees |
| 15:07.20 | vaewyn | coppice: I'm with you on that... |
| 15:07.26 | docelm0 | fu3 Im only 29 |
| 15:07.38 | docelm0 | Ive just been in Telco for a few years now |
| 15:07.46 | fu3 | cool.. im just getting into it |
| 15:07.50 | coppice | actually dry doesn't mean no signal. it means no DC |
| 15:07.58 | docelm0 | fu3, traditional or voip? |
| 15:08.13 | trixter | basically its directly connected endpoint to endpoint, think payphone -> local office where the local office provides all signalling.. you could use that and a channel bank for example to provide dialtone for payphones that are spread out in the lcoal area.. |
| 15:08.15 | trixter | much cheaper |
| 15:08.26 | fu3 | well i run a traditional system right now, but im planning a migration to voip. |
| 15:08.28 | vaewyn | pain to get though |
| 15:08.37 | vaewyn | at least with SBC land |
| 15:08.40 | trixter | ok, so on an analog line aside from ring what signal is on the line that isnt DC? |
| 15:08.42 | docelm0 | Well the term dry anymore is more or less know as dry copper anymore. . Phone companies dont give them out cause its a bitch to setup and they dont have the pairs to spare |
| 15:08.50 | trixter | onhook/offhook are DC based |
| 15:09.01 | trixter | dtmf is modulated on that DC |
| 15:09.12 | docelm0 | fu3, define traditional? |
| 15:09.19 | mut | we tried to do that with verizon |
| 15:09.23 | jbalcomb | why are my asterisk config files 'executable'? |
| 15:09.27 | fu3 | well.. analog :) |
| 15:09.27 | mut | they were charging us $7/loop/month |
| 15:09.30 | trixter | well phone companies when dsl first came out would do stuff like put bridging clips on em and stuff so that range was severly limited |
| 15:09.32 | docelm0 | verizon will not give up shit for loops |
| 15:09.35 | mut | then they found out what we were doing |
| 15:09.37 | *** part/#asterisk mazimp (i=nobody@p54AB3D3F.dip0.t-ipconnect.de) |
| 15:09.38 | coppice | trixter: DC isn't a signal. its a wetting current |
| 15:09.38 | docelm0 | I have tried |
| 15:09.41 | mut | and it was $40 for it |
| 15:09.42 | fu3 | we have 350 lines coming across a fiber, then to a channel bank, to the demarc, to the station. |
| 15:09.52 | trixter | especially in places like NJ where bell atlantic (the ilec at that time) wanted to push their dsl |
| 15:10.02 | docelm0 | mut what were you doing? |
| 15:10.04 | trixter | coppice: symantics |
| 15:10.07 | mut | sdsl |
| 15:10.16 | trixter | and its used as a signal |
| 15:10.17 | jbalcomb | iCEBrkr: PM me eh? |
| 15:10.28 | coppice | trixter: nope. read something about copper pairs |
| 15:10.32 | docelm0 | mut haha.. I tried that in WV before DSL got big there. They knew exactly what I was trying to do.. hehe |
| 15:10.39 | *** join/#asterisk redder86 (n=lee@gateway.howardsilvan.com) |
| 15:10.43 | trixter | for example when the phone is hung up its -48 on ring 0 on tip, but when its picked up it passes that DC at about 7v |
| 15:10.52 | mut | yea |
| 15:10.58 | redder86 | Anyone elaborate (put into english) what "PRI got event: HDLC Abort (6) on Primary D-channel of span 4" means? |
| 15:11.09 | mut | so we put in the paperwork |
| 15:11.14 | mut | and we can lay copper now |
| 15:11.16 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 15:11.19 | coppice | trixter: and try to use the correct word. symantics means meaning, so symantics is the good thing, while you are implying its the bad thing |
| 15:11.19 | mut | TWD communications |
| 15:11.24 | jbalcomb | redder86: do you know what HDLC is? |
| 15:11.31 | redder86 | jbalcomb: yup |
| 15:11.41 | docelm0 | fu3, nice.. I bet your having fun with that.. |
| 15:11.42 | trixter | depending on how you use symantics it can be a good or bad thing |
| 15:11.45 | trixter | so it was the correct word |
| 15:11.54 | trixter | thanks for pointing out that I did use it correctly |
| 15:11.55 | redder86 | jbalcomb: I don't know what an HDLC "abort" is, though. |
| 15:12.10 | fu3 | not particularly.. I am having a hell of a time diagnosing voltage issues. I'm not getting the 500ms battery drop after disconnect and I can't F'ing prove it to qwest. |
| 15:12.15 | jbalcomb | redder86: ok, call your telco and have them monitor the line and make sure the configs match on both sides. |
| 15:12.34 | *** join/#asterisk SplasPood (n=jwb@gate.lga2.us.voxel.net) |
| 15:13.13 | jbalcomb | redder86: http://www.asteriskguru.com/archives/-nocadm-re-asterisk-users-hdlc-bad-fcs-hdlc-abort-vt28241.html?highlight=hdlc+abort++error |
| 15:13.17 | redder86 | jbalcomb: the PRI works fine most of the time... like 99% of the time. I've just got this error once in a while... and too frequently in the last few days (once or twice per day) which almost always seems to precede a red alarm. |
| 15:13.34 | jbalcomb | redder86: Also, punch that msg into http://www.google.com |
| 15:13.49 | GerbilNut | redder, i get that message once every minute |
| 15:14.09 | redder86 | GerbilNut: I presume, then, that you're not getting a red alarm just as much. |
| 15:14.18 | GerbilNut | no alarms on either side |
| 15:14.24 | jbalcomb | redder86: right, so if you are seeing errors then the telco is likely to see them too. Increased activity might be a cause for intrusive testing. |
| 15:14.27 | GerbilNut | I own the switch it's plugging into, all green lights |
| 15:14.54 | GerbilNut | but i couldn't get span 1 up without turning on span 2, even though it had nothing plugged into it |
| 15:14.56 | redder86 | jbalcomb: the telco is going to be monitoring it more closely today |
| 15:15.15 | jbalcomb | redder86: good. whos your provider? |
| 15:16.00 | jbalcomb | iCEBrkr: Hey! PM me before I call god and have him turn the heat off down there. |
| 15:16.19 | redder86 | jbalcomb: global crossing |
| 15:16.21 | jbalcomb | iCEBrkr: Snowurcaine? |
| 15:17.21 | iCEBrkr | lol |
| 15:17.53 | jbalcomb | iCEBrkr 150 MPH hail and 6 FT of snow / hour |
| 15:17.57 | iCEBrkr | jbalcomb: What'chu need? |
| 15:18.05 | jbalcomb | iCEBrkr I need you to PM me |
| 15:18.25 | [TK]D-Fender | Who here has had some real-world experience with the Polycom IP 301? I could use your opinion on its operation aside from the obvious lack of the higer end features : ie things that it offers but is inconvenient that would be overtly detrimental to the users experience. |
| 15:18.31 | iCEBrkr | jbalcomb: I'm not gonna netsex you.... |
| 15:18.56 | fu3 | Are there any security (physical) features for common voip phones? |
| 15:19.06 | fu3 | I want to roll voip here at a college, but am concerned about the theft of phones. |
| 15:20.00 | *** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it) |
| 15:20.14 | GerbilNut | duct tape is a good theft deturent |
| 15:20.54 | iCEBrkr | fu3: if it's not nailed down, someone will take it |
| 15:20.55 | brad_mssw | Darwin35: you still looking for a place to put an east-coast server? |
| 15:20.59 | kkun | I work at a college with 50 + cisco 79XX phones and theft ain't a problem |
| 15:21.07 | fu3 | iCEBrkr.. thats what im concerned about. |
| 15:21.11 | fu3 | kkun.. really? thats cool.. |
| 15:21.22 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 15:21.22 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:21.22 | fu3 | we've had a few projector thefts a few years back |
| 15:21.25 | fu3 | but thats about it. |
| 15:21.29 | fu3 | I'm just trying to cover all the bases :) |
| 15:21.29 | vaewyn | [TK]D-Fender: they are nice phones... they have a little less call quality in the echo handling than the 50x or 60x but barely noticeable |
| 15:21.35 | brad_mssw | Darwin35: if so, my ISP would like to offer to help, they have already tried e-mail David, but got no response |
| 15:21.36 | kkun | Chain (cable) the phones to the desk if you are really concerned |
| 15:21.42 | [TK]D-Fender | fu3 : I've never seen a phone with a Kensington style lock port or anything so what I did here was that in public places I use CHEAPER phones. My regular ones are all polycom 60x's, so I use Uniden UIP-200's for courtesy spots where they could get "molested". |
| 15:21.50 | vaewyn | [TK]D-Fender: but for simple endpoints they are nice |
| 15:21.53 | fu3 | do most actually have a place to run a cable through (fuck I dont want to do that!) |
| 15:21.59 | jbalcomb | kkun: do you use auto-answer at all? |
| 15:22.41 | fu3 | [TK]D-Fender good call. |
| 15:22.41 | [TK]D-Fender | vaewyn : And how is access to common features like missed calls, redial, address book, etc? |
| 15:22.42 | jbalcomb | [TK]D-Fender i'd steal them anyway, cheap or not |
| 15:22.42 | trixter | fu3: you can always get a high capacity fxs server (possibly T1 cards with channelbanks) and use non voip phones, although that may defeat some of the point ... it depends on where exactly the phones will be located as to whether or not theft is a problem |
| 15:23.02 | [TK]D-Fender | jbalcomb : Use Grandstream BudgetTone 101's! Some phones play "hard to get", they play "hard to want"! |
| 15:23.04 | jbalcomb | i wonder about the ELT<SP> on WiFi VoIP phones |
| 15:23.07 | iCEBrkr | file: I HATE HTML EMAIL! |
| 15:23.08 | vaewyn | [TK]D-Fender: little more involved due to the smaller screen... but doable... I would say they are good call center phones... they are not good secretary phones... 500 at least for that kindof app |
| 15:23.17 | *** part/#asterisk redder86 (n=lee@gateway.howardsilvan.com) |
| 15:23.20 | fu3 | trixter.. the point IS going to voip :) |
| 15:23.21 | [TK]D-Fender | fu3 : I'd follow trixter's point on that... |
| 15:23.21 | fu3 | thanks though |
| 15:23.42 | trixter | well it would be voip somewhere just analog to the endpoint in the high risk area |
| 15:23.46 | jbalcomb | [TK]D-Fender yeah, I'm a Rebel Without A Cause though so I'm just making trouble rather than judging the phone. |
| 15:23.49 | fu3 | I want people to be able to move thier phone across campus when they move offices and not have to bother me with the details |
| 15:23.58 | trixter | so that you can get a more secure phone that is less likely to walk off and if it does its $10 or whatever to replace it |
| 15:24.02 | [TK]D-Fender | vaewyn : currently EVERYONE here is on IP60x's I have a 4 person call center I'm looking to "downgrade" to liberate higher end phones for those needing the call handling. |
| 15:24.18 | trixter | ahh that may be more problematic unless you trust the people or can track to whom it was assigned |
| 15:24.22 | trixter | so they can buy it if its missing |
| 15:24.23 | fu3 | trixter.. i see what you mean. Yeah.. they dont need voip in the public spaces. |
| 15:24.27 | trixter | make the individuals responsible for it |
| 15:24.45 | fu3 | plus, I have to keep a few analog lines "just in case" |
| 15:24.49 | kkun | fu3, How large of a deployment are you looking at? |
| 15:24.55 | fu3 | 400 offices |
| 15:24.55 | jbalcomb | trixter that is a terrible idea >morale |
| 15:24.59 | trixter | there are also 'princess' style phones that are coin operated ... |
| 15:25.02 | fu3 | and 15 public phones |
| 15:25.16 | fu3 | 50 of those "offices" are rarely attended to, so they are high risk as well |
| 15:25.25 | [TK]D-Fender | fu3 : for the public phones use ATA's in the back room and 10$ phones from the wall... |
| 15:25.32 | trixter | jbalcomb: well its a college environment I dont know if its students or faculty, if its students I would make them responsible just like the library does with books you check out |
| 15:25.43 | fu3 | I dont know what ATA's are |
| 15:26.19 | kkun | Ata (analog telephone adapter) allows you to use pstn hones with a voip deployment |
| 15:26.26 | fu3 | ah ok.. i've seen those. |
| 15:26.28 | fu3 | they work pretty well eh? |
| 15:26.34 | kkun | usually |
| 15:26.47 | fu3 | excellent. That's actually a great idea. Thanks! |
| 15:26.53 | kkun | That is what plces like vonage give their customers |
| 15:27.06 | fu3 | it would certainly aleviate some concerns in administration |
| 15:27.06 | Hmmhesays | ok i'm having a mental block here |
| 15:27.17 | Hmmhesays | what do you use to break down a ds3 into t1's? |
| 15:27.17 | *** join/#asterisk lorinc (n=ang@caracas-1730.adsl.interware.hu) |
| 15:27.32 | kkun | If you are using asterisk, the iaxy2 is a great ATA |
| 15:27.41 | [TK]D-Fender | Hmmhesays : A de-mux? |
| 15:27.46 | trixter | the digium ds3000 or whatever their model is |
| 15:27.49 | trixter | oh wait that still doesnt exist |
| 15:28.07 | [TK]D-Fender | Or Sangoma's DS3.. oh wait... doesn't do VOICE..... |
| 15:28.12 | *** join/#asterisk apardo (n=apardo@87.218.44.216) |
| 15:28.26 | docelm0 | Say can anyone recommend a decient analog POTS phone? |
| 15:28.26 | trixter | well it was never advertised as a channelized ds3 card |
| 15:28.26 | Hmmhesays | any suggestions? |
| 15:28.33 | [TK]D-Fender | So I gues a straight De-mux is what you need right now... though AudioCodes and others are coming out with that kind of stuff... |
| 15:28.52 | [TK]D-Fender | trixter : I never said it was expected to :) Just pointing to more failures :) |
| 15:28.58 | cpm | Harris TS-22 is a good pots phone. |
| 15:29.04 | [TK]D-Fender | docelm0 : No such thing. |
| 15:29.04 | trixter | I dont consider it a failure to perform as advertised |
| 15:29.08 | fu3 | I like the TS-44 :) |
| 15:29.11 | kkun | Docelmo, for what purpose |
| 15:29.13 | trixter | that is a pretty grim definition of failure |
| 15:29.26 | docelm0 | [TK]D-Fender, I need some for my sales floor |
| 15:29.26 | Hmmhesays | i'd rather stick with the current vendor we are using but they have no ds3 units |
| 15:29.29 | cpm | Yeah, but the 22 is cheaper. Doesn't have the headset option though. |
| 15:29.35 | *** join/#asterisk mkrufky (n=mk@68.160.103.77) |
| 15:29.37 | trixter | cpm: dracon is a big expensive though isnt it :P |
| 15:29.44 | cpm | Not on ebay |
| 15:29.47 | trixter | ha |
| 15:29.59 | trixter | "recently obtained from an unlocked telco van" :P |
| 15:30.14 | cpm | My fave is still the TA312 w/ the TA422 touch tone module. |
| 15:30.17 | docelm0 | Doesnt astra make some? |
| 15:30.22 | [TK]D-Fender | docelm0 : I'd suggest a phone with caller-id and perhaps speakerphone. it should be battery powered and I'd suggest rechargeable NiMH. |
| 15:31.00 | cpm | The panasonics used to be good, but I haven't bought a pots phone in 15 years. |
| 15:31.08 | trixter | speaking of battery I need to replace the ones in my cordless.. I refuse to upgrade it becuase I cant find one with as good of range anymore, got it in 1997 or so and its worked great.. even with diminished battery it still gives me 3+ hours talk time |
| 15:31.24 | *** join/#asterisk lunaphyte_ (n=lunaphye@c-71-193-101-146.hsd1.mi.comcast.net) |
| 15:31.35 | trixter | my cordless is a panasonic gigarange.. dual band (900/2.4G) and works quite well |
| 15:31.42 | Juggie | trixter-> www.batteryguys.com |
| 15:31.56 | trixter | I can geti t local it is just a matter of actually going and getting it |
| 15:32.38 | kkun | radioshack |
| 15:32.58 | trixter | its still a matter of getting it |
| 15:33.30 | trixter | the nearest radio shack (and its not likely to have it due to its size) is 25 miles, in reality odds are I have to go to the nearest major town which is 50 miles.. which I am doing tonight, but odds are I will forget |
| 15:33.44 | cpm | best pots phone around, bar none. http://www.associated-ind.com/images/ta312apt.gif |
| 15:34.07 | Juggie | trixter, all the more reason to order online :P |
| 15:34.20 | trixter | but the phone works great. handles the ring signal (voltage on the line) handles the hangup signal (voltage on the line) etc.. all those fun signals from days gone by |
| 15:34.34 | trixter | shipping for a battery is likely to cost what the battery costs |
| 15:34.38 | trixter | for one that size anyway |
| 15:34.41 | *** join/#asterisk puzzled (n=yeahrigh@62.45.11.228) |
| 15:34.48 | puzzled | afternoon all |
| 15:34.49 | trixter | <-- cheap |
| 15:34.58 | trixter | puzzled: you are 7 hours late! |
| 15:35.11 | puzzled | all depends on your state of mind |
| 15:35.12 | cpm | or 3 hours early, not sure. |
| 15:35.24 | trixter | puzzled: nah I meant you normally sign on 7 hours earlier |
| 15:35.41 | trixter | or have been recently |
| 15:35.47 | GerbilNut | what is libpri used for? |
| 15:35.48 | puzzled | trixter: ah yes you are right. was late this morning so had to rush |
| 15:35.51 | *** join/#asterisk bkw_ (n=bkw_@adsl-70-234-37-160.dsl.tul2ok.sbcglobal.net) |
| 15:35.54 | trixter | hehe |
| 15:36.11 | trixter | GerbilNut: libpri is a library for pri related functions (mostly) |
| 15:36.19 | fu3 | all joking asisde, what IS the best pots phone available? |
| 15:36.27 | cpm | best pots phone around, bar none. http://www.associated-ind.com/images/ta312apt.gif |
| 15:36.28 | trixter | if you dont have a pri and arent likely to get one you probably dont need it |
| 15:36.28 | puzzled | anyone have a clue what this error is all about: "cli.c:49:30: asterisk/version.h: No such file or directory" and something about ASTERISK_VERSION undeclared |
| 15:36.37 | trixter | fu3: that depends on application |
| 15:36.45 | cpm | what do you mean by 'best' ? |
| 15:36.46 | trixter | most pots phones are all about the same, its the 'extras' that matter |
| 15:36.52 | *** join/#asterisk coppice (n=chatzill@37.157.17.210.dyn.pacific.net.hk) |
| 15:36.57 | fu3 | Regular phones w/ answering machines |
| 15:36.59 | fu3 | thats all I need. |
| 15:36.59 | trixter | answering machine (find a cordless today without one!) caller id display, etc |
| 15:37.02 | fu3 | with a flash button :) |
| 15:37.18 | trixter | there are many $10 or less phones that do that |
| 15:37.18 | fu3 | i dont like cordless, i dont need caller id displays, etc./ |
| 15:37.24 | vaewyn | ok... anyone know if you can get an IAXy to bump the gain up a bit? |
| 15:37.25 | trixter | although I bet none are sold retail anymore |
| 15:37.25 | cpm | Well, the panasonics used to be good, but they are all such commodities today. |
| 15:37.26 | fu3 | all the ones that I buy for like $10 I end up replacing every 3-4 months. |
| 15:37.33 | vaewyn | walmart has 7$ callerid phones :P |
| 15:37.34 | trixter | I havent really looked at phones in a decade |
| 15:37.35 | lunaphyte_ | how can i make asterisk tell me more info regarding a sip peer (what address/port the connection is coming from)? |
| 15:37.43 | trixter | really.. ok well there ya go |
| 15:38.02 | fu3 | I hate replacing them ;) |
| 15:38.08 | cpm | don't buy junk then. |
| 15:38.11 | trixter | why do you have to replace them? |
| 15:38.12 | fu3 | thats what im trying not to do! |
| 15:38.19 | trixter | I have had $10 phones for years without failure |
| 15:38.26 | fu3 | because they die.. answering machines stop working, etc |
| 15:38.26 | trixter | I even took em apart and modded em slightly :P |
| 15:38.43 | fu3 | I believe i've got voltage issues here.. it could cause them to fail. |
| 15:39.00 | fu3 | is it normal for a ring to exceed 150VDC? |
| 15:39.00 | QuAtRo[NL] | My inbound-callers do get into the queue but the phones do not ring, do the agenda have to login or something before there phones start ringing? |
| 15:39.04 | trixter | I like the ones with lights, which is technically hearing impaired compatible.. some people have hearing aids that work with the speaker so you may want to ensure that works too (peizo often dont cut it) |
| 15:39.09 | QuAtRo[NL] | agenda = agents |
| 15:39.22 | trixter | the blinky lights tells em to turn their hearing aids back on so they can hear the phone although its more inductance than anything |
| 15:39.25 | coppice | fu3: yes |
| 15:39.27 | trixter | my hearing aid was that way |
| 15:39.34 | fu3 | thanks coppice |
| 15:39.46 | trixter | wow 150vdc ring |
| 15:39.48 | trixter | that is impressive |
| 15:39.53 | fu3 | 153 is my record :) |
| 15:40.01 | trixter | mine is AC not DC and its normally 90vac @ 20hz for me :P |
| 15:40.13 | fu3 | :) |
| 15:40.15 | *** join/#asterisk NexGen (n=me@adsl-70-135-6-65.dsl.tulsok.sbcglobal.net) |
| 15:40.40 | NexGen | anyone using a GXP2000 and have gotten the intercom key to work successfully? |
| 15:40.41 | fu3 | sweet.. the qwest tech just called.. he said he has my problems squared away, and he will be over here soon to talk. |
| 15:40.43 | fu3 | thats a relief. |
| 15:40.49 | trixter | I read somewhere that 'normal' is 90-120vac but I dont know how accurate that is anymore (I read it over 20 years ago) |
| 15:41.02 | coppice | the ring can be quite high, and is on top of -48V DC. the peak negative level is something like -350V, depending on the area |
| 15:41.06 | trixter | I know all mine have been 90 |
| 15:41.13 | trixter | in multiple states with different carriers and different switches |
| 15:42.02 | fu3 | my lines idle at -38 to -41vdc. |
| 15:42.17 | coppice | that's a bit low. |
| 15:42.22 | fu3 | I thought so as well |
| 15:42.53 | fu3 | the qwest guy said the CB was programmed to run that low.. |
| 15:42.56 | fu3 | but no explanation was given |
| 15:43.26 | _Paulo_ | touching nude wires while the phone is ringing is funny :-) |
| 15:43.42 | coppice | modern systems generally run at the same voltage everywhere, until loop is applied |
| 15:43.46 | cpm | stripping wires with your teeth when a call comes in is funnier |
| 15:43.59 | _Paulo_ | ouch! |
| 15:44.06 | fu3 | _Paulo_.. you got it right |
| 15:44.07 | *** part/#asterisk CosmicRay (n=jgoerzen@2002:452c:8843:1:20e:a6ff:fe66:c5a3) |
| 15:44.10 | cpm | it's pretty bad. |
| 15:44.16 | cpm | Jaw ached for days |
| 15:44.54 | coppice | a child with their finger is a telephone socket could be killed |
| 15:44.55 | Seldon1975 | Is anyone here using a TDM2400? |
| 15:44.57 | fu3 | When I have a handset in monitor mode, I can often hear the rings of someone elses call, or multiple rings on top of one another over the handset. According to Qwest, this is 100% normal. Is this true? I dont think I should be hearing that stuff. |
| 15:45.33 | cpm | in monitor mode? yeah, that's pretty normal |
| 15:45.43 | fu3 | why does that happen? |
| 15:45.49 | cpm | Monitor mode is fun, but really really really shouldn't be abused. |
| 15:46.02 | cpm | because there isn't much of a ground reference in monitor mode. |
| 15:46.03 | _Paulo_ | cpm, you are lucky if you dont break any teeth. |
| 15:46.05 | *** join/#asterisk Utah_Dave (n=boucha@0-3pool128-51.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 15:46.14 | cpm | _Paulo_, yeah, I learned. |
| 15:46.41 | fu3 | hmm... ok well thanks for verifying that. |
| 15:46.44 | *** join/#asterisk m-i-a- (n=chaguy42@CPE00304827782b-CM0014f85e8abe.cpe.net.cable.rogers.com) |
| 15:47.13 | docelm0 | NexGen, yes using app_page w/ a mod I wrote |
| 15:47.15 | cpm | now, if you go offhook, and still hear that stuff, it's a problem. |
| 15:48.02 | fu3 | yeah it works fine offhook. |
| 15:49.36 | *** join/#asterisk flujan (n=flujan@internet.nube.com.br) |
| 15:50.25 | flujan | Hi guys, Do you know how to monitor a call not just recording it but also listening it? |
| 15:51.01 | flujan | for instance, My client calls a operator. The manager wants to also listen this call, but without the client or the operator notice. |
| 15:52.31 | Juggie | Monitor() Record() |
| 15:52.40 | Juggie | www.voip-info.org |
| 15:53.22 | trixter | think he wants chanspy |
| 15:53.26 | trixter | yeah monitor |
| 15:53.40 | *** join/#asterisk pointer (i=pointer@aj.catt.com) |
| 15:54.18 | mikefoo | Anyone do asterisk installs for offices on the side? |
| 15:54.28 | mikefoo | or as a main gig, etc.. |
| 15:54.57 | coppice | I've never seen an office on its side :-\ |
| 15:55.07 | *** join/#asterisk Cresl1n (n=matt@gateway.digium.com) |
| 15:55.13 | fu3 | there actually is one in finland or sweden or someplace like that |
| 15:55.23 | lunaphyte_ | no, on the side of the office... |
| 15:55.30 | GerbilNut | After we upgraded to * 1.2.4 we are no longer getting the announcements in the queue. Anyone else seen / fixed this? |
| 15:56.24 | *** join/#asterisk salviadud (n=ralfalfa@dsl-201-129-72-144.prod-infinitum.com.mx) |
| 15:56.44 | coppice | I saw a picture of a house on its side in some kind of theme park in .nz |
| 15:57.00 | djMax | so I have a SIP phone (Polycom IP500) that is set to register every 100 seconds. |
| 15:57.41 | djMax | but after about 30 seconds I get this in the CLI: Really destroying SIP dialog [blahblah] Method: register |
| 15:58.00 | djMax | after which, the IP address of the peer does not show in "sip show peers", which in turn means it can't receive incoming calls. Thoughts? |
| 15:58.25 | djMax | actually, I get one before that about destroying a NOTIFY dialog, not sure which one kills it from peers. |
| 15:59.19 | flujan | Juggie, I saw this features |
| 15:59.20 | Seldon1975 | Is anyone here using a TDM2400? |
| 15:59.34 | flujan | I need something like the MeetMe with the m option You now. |
| 15:59.43 | flujan | ops |
| 15:59.48 | flujan | You know. |
| 15:59.51 | trixter | isnt a tdm2400 more expensive than a T1 card and a channel bank? |
| 16:00.18 | flujan | The monitor just start to record the call, for what I understood. |
| 16:01.05 | trixter | I wonder if the tdm2400 has protection built in so if someone makes all the phones ring at the same time it doesnt try to draw so much current from the power supply it fails.. granted you should have a large enough power supply in the first place but ... |
| 16:01.28 | tzanger | doubtful |
| 16:01.40 | tzanger | but your power supply will fold over and the box will likely spontaneously reboot, so it's protected. :-) |
| 16:01.49 | trixter | that would be a mean worm that infects systems looking for asterisk boxes with tdm2400s and just amkes all the phones ring at the same time ... give it time someone will get bored and write one |
| 16:01.53 | coppice | i thought it took external power for the ringer |
| 16:02.11 | djMax | anybody know what "really destroying sip dialog" is trying to tell me? Is it bad? Normal? |
| 16:02.13 | tzanger | well in that case the power supply'd have to be adequately sized |
| 16:02.19 | trixter | I thought someone recently said they had to hook some weird wiring up to make it work becuae their dell server didnt have the normal drive connectors |
| 16:02.33 | trixter | but I could be mistaken I havent played with a tdm2400 and really dont have any plans to |
| 16:02.34 | tzanger | I think it'd be nice to make sure the traces are capable of supplying enough current to vaporize 26AWG |
| 16:02.44 | tzanger | then you'd definitely find the problem :-) |
| 16:02.58 | cpm | That's because they used a Dell server. |
| 16:02.58 | tzanger | kind of like the old variac trick to find the bad component |
| 16:02.58 | trixter | ha |
| 16:03.03 | trixter | blottobox! |
| 16:03.05 | *** join/#asterisk justinu (n=Justin@cpe-72-129-86-208.socal.res.rr.com) |
| 16:03.15 | *** join/#asterisk juanjoc (n=jcomella@222-32-235-201.fibertel.com.ar) |
| 16:03.16 | tzanger | bring up the supply slowly and soon enough the component will smoke... replace it and you're good to go |
| 16:03.20 | tzanger | trixter: I actually remember that |
| 16:03.23 | delmar | so.. echo at the start of a call on the X100P is pretty common huh? even with echotraining=800 etc? |
| 16:03.25 | coppice | actually 24 ringers isn't that much power |
| 16:03.27 | tzanger | but this is like a 24-port blottobox |
| 16:03.29 | trixter | tzanger: doesnt everyone? |
| 16:03.40 | cpm | Had a dell "workstation" melt the simm slots off the mb on a large render last year. Last dell 'workstation' we're ever going to buy |
| 16:03.46 | delmar | what other tricks if any can be applied to help get rid of the echo in the first 30secs of an incoming call on an X100P? |
| 16:03.47 | coppice | at least compared to some overclocking and dual Nvidia cards :-) |
| 16:04.03 | tzanger | coppice: no, but have the supply able to source enoug current to vaporize 26AWG... I know it's all protected... just remove that protection |
| 16:04.11 | tzanger | delmar: fxotune? |
| 16:04.18 | Seldon1975 | cpm: ahhha |
| 16:04.22 | vgster | i dont get echo on my x100P |
| 16:04.30 | delmar | tzanger, nah thats a feature in the TDM400 fxo's. |
| 16:04.52 | tzanger | delmar: oh I thought it was X100P... sorry |
| 16:04.54 | delmar | vgster, thats nice. what echo-can alg have u set? |
| 16:05.00 | djMax | if a SIP registration is set to expire in 600, and it expires much faster, what might cause that? |
| 16:05.11 | trixter | delmar: 858? |
| 16:05.15 | tzanger | delmar: have you tried the *latest* MG2? They commited code to increase the responsiveness of the echo can |
| 16:05.20 | delmar | tzanger, it is an X100P. the TDM400's do fxotune. not X100's. |
| 16:05.25 | GerbilNut | After we upgraded to * 1.2.4 we are no longer getting the announcements in the queue. Anyone else seen / fixed this? |
| 16:05.36 | salviadud | echo cancelation |
| 16:05.40 | delmar | tzanger, im using KB1 at the moment |
| 16:05.42 | salviadud | its in zapata.conf |
| 16:05.47 | trixter | tzanger: my favourite hack was someone writing code to spin up/down drives in a dual drive workstation (they were side by side, pizza box style but larger system) to make it 'walk' across the desk |
| 16:05.55 | delmar | salviadud, bit beyond that my friend :P |
| 16:06.11 | vgster | delmar: havent changed it |
| 16:06.18 | salviadud | oh... i guess i don't know delmar, sorry |
| 16:06.19 | trixter | there was enough force when they spun up one drive to nudge the box slightly |
| 16:06.19 | tzanger | trixter: hahaha |
| 16:06.36 | delmar | tzanger, ok i might take a look at that other echo-can alg then. |
| 16:06.46 | trixter | delmar: 858? |
| 16:06.51 | tzanger | I wrote an int13 handler that made the heads walk across the platters on EVERY read and write... this was back before the controllers eliminated unnecessary seeking |
| 16:06.53 | delmar | what echo-can algs are other X100 users commonly using? |
| 16:06.54 | *** join/#asterisk heroine (n=heroine@gut75-1-82-67-176-238.fbx.proxad.net) |
| 16:06.57 | heroine | hi |
| 16:06.57 | delmar | trixter, 858? |
| 16:07.01 | trixter | guess not |
| 16:07.02 | delmar | trixter, whats that? |
| 16:07.06 | tzanger | trixter: it took a nice "fast" (at the time) drive and slowed it waaaaaaaaaaaaaaaay down |
| 16:07.12 | trixter | its the area code for del mar california :P |
| 16:07.20 | delmar | trixter, lol |
| 16:07.34 | salviadud | so the station would move |
| 16:07.34 | trixter | wasnt sure if your name was representative of that area or not |
| 16:07.41 | vgster | delmar: KB1 echocancel=yes echotraining=800 or yes thats it |
| 16:07.44 | salviadud | would any ladies get scared with that? |
| 16:07.45 | DeadZen | have the places in california start with del |
| 16:07.48 | DeadZen | or rio |
| 16:07.54 | DeadZen | half* |
| 16:08.02 | *** join/#asterisk Eitch (n=hugo@unaffiliated/eitch) |
| 16:08.08 | trixter | yeah but they all start with [a-z] |
| 16:08.12 | delmar | trixter, long story how this name got stuck to me but its a nick name since D&D high school days after the Dl series.. a toon named Delamaar. |
| 16:08.21 | DeadZen | nod |
| 16:08.24 | trixter | even zyxxy |
| 16:08.38 | DeadZen | i gotta hit the shower and head out to a client |
| 16:08.39 | delmar | vgster, thats what i got... too much echo in the first 30secs.. it settles tho. |
| 16:08.43 | trixter | if I spelled it right, its a real town off the 15 between los angeles and los vegas |
| 16:08.44 | DeadZen | mornin everyone and later |
| 16:08.48 | trixter | near the CA/NV border |
| 16:10.19 | delmar | ok so I can try MG2 echo-can for a laugh... any other suggestions anyone? |
| 16:11.01 | fu3 | I liek Cafe Del Mar |
| 16:11.16 | delmar | rx/txgain are 0.0. txgain applies here of course.. the higher the txgain the worse the echo.. so im keeping that where it is. |
| 16:11.29 | delmar | echocan=yes, echotrain=800 and all that jazz. |
| 16:11.36 | tzanger | I hate echotraining |
| 16:11.41 | tzanger | haven't needed it since I used MG2 |
| 16:11.49 | delmar | tzanger, im not even sure it's working |
| 16:11.55 | justinu | should you not be using echotraining and mg2? |
| 16:12.14 | delmar | justinu, does echotraining not apply for KB1? |
| 16:12.23 | tzanger | delmar: well if you have a 800ms "pause" before you hear anything, it does :-) |
| 16:12.27 | justinu | i dunno... i'm learning about this stuff myself |
| 16:13.08 | delmar | i havent seen anything that says echotraining is specific to a particular echo-can alg. |
| 16:13.38 | delmar | almsot 5:30am and no sleep yet.. brain is hot wired.. body is buggered.. but ill keep at it... |
| 16:13.57 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 16:14.02 | justinu | delmar... tx/rxgain at 0.0 may not be correct for your site... listen to calls, and make sure the received audio isn't overmodulated |
| 16:14.10 | delmar | so just to confirm this cuz im so tired... i dont need to bloody compile Asterisk agai.. just zaptel for the echo-can changes.. and reinsert module blah blah... |
| 16:14.14 | justinu | i started recording calls, and realized the rx audio was way too hot. |
| 16:14.29 | justinu | delmar: yeah, just recompile zaptel, and /etc/init.d/zaptel restart |
| 16:14.51 | delmar | justinu, that part im quite aware of. its the TXgain thats of concern... audio on calls is perfect.. and the other party doesnt hear no echo... its my end that hears echo. |
| 16:14.54 | salviadud | delmar what distro are you using? |
| 16:15.11 | justinu | delmar: k, sounds like your txgain might be too hot |
| 16:15.35 | delmar | salviadud, fresh install of debian sarge and latest 1.2 subversion of *, zap, libpri |
| 16:15.41 | delmar | im good in that department. |
| 16:15.49 | delmar | justinu, u would think. |
| 16:16.00 | delmar | justinu, but if i lower it.. the other end cant hear so well |
| 16:16.05 | justinu | hmm |
| 16:16.20 | delmar | justinu, its where it needs to be as far as volume goes. in fact it probably needs to be at about 1.0 |
| 16:16.25 | justinu | i see |
| 16:16.42 | delmar | compared to the tdm400... x100's are shockingly bad for this sort of problem. |
| 16:16.43 | justinu | i'm kinda working thru the same stuff |
| 16:16.50 | justinu | but I'm using PRI |
| 16:16.55 | delmar | anyway.. ill go fester this zaptel. |
| 16:16.58 | delmar | justinu, lucky u |
| 16:17.05 | delmar | justinu, thats a much better deal :P |
| 16:17.08 | justinu | hah, i suppose... still the same problems |
| 16:17.21 | delmar | justinu, really? |
| 16:17.26 | justinu | yep |
| 16:17.31 | delmar | justinu, i thought Pri was all good? |
| 16:17.38 | delmar | justinu, what hardware talking to the Pri? |
| 16:17.41 | justinu | you can still have echo issues |
| 16:17.46 | justinu | sangoma a101 |
| 16:18.32 | delmar | justinu, good to know the cheaper sangoma cards are like that.. I hear that the digium gear is good.. plus u get better support out of Digium if u bought their gear :P |
| 16:18.59 | justinu | heh, yeah - no one has ever had echo problems with digium cards... sure ;) |
| 16:19.24 | delmar | damn shame tho. be nice to see some competitive product on the market to push prices down.. i mena... i have 2 channel banks here that I scored cheap.. and no T1 card to drive them to even play with... hell if im paying US$500 for one. |
| 16:19.35 | vaewyn | If you have issues with echo via a PRI card it's because of your CB or provider... PRI can't "induce" echo" |
| 16:19.43 | vaewyn | timing issues yes... but not echo |
| 16:19.45 | delmar | justinu, not saying they dont have issues.. just saying there is good support if u do have trouble. |
| 16:20.03 | justinu | sangoma support is good too |
| 16:20.09 | delmar | vaewyn, thats my way of thinking.. its digital...so .. how can it |
| 16:20.09 | justinu | no issue there |
| 16:20.38 | justinu | it can induce echo the same way... by throwing too much energy out into the network w/ incorrect gain settings |
| 16:21.00 | vaewyn | PRI doesn't do "gain" |
| 16:21.03 | mut | i had to buy tellab echo cans for mine |
| 16:21.06 | vaewyn | that is the CB or provider |
| 16:21.11 | justinu | heh |
| 16:21.15 | mut | i couldn't adjust the zap drivers to get rid of echo |
| 16:21.21 | justinu | so why do the rx/tx gain settings work for a T1? |
| 16:21.41 | vaewyn | That's a software thing in * |
| 16:21.42 | justinu | anyway, you have to set the gain correctly on a PRI |
| 16:21.53 | *** join/#asterisk kannan (n=kannan@202.71.149.242) |
| 16:21.58 | vaewyn | only to compensate for a bad provider/CB |
| 16:21.59 | delmar | mut, yeah its pretty obvious digium admit there are echo issues when they start re-releasing their products with the ability to add a hardware echo-can. |
| 16:22.13 | mut | yeh |
| 16:22.33 | *** part/#asterisk flujan (n=flujan@internet.nube.com.br) |
| 16:22.39 | mut | echo only comes in when a call goes out to a verizon pri tho |
| 16:22.45 | mut | it's stupid |
| 16:23.03 | justinu | near end or far end echo? |
| 16:23.06 | mut | verizon wirelss and alltel and local clec calls and internal office calls thru the same pri theres no echo |
| 16:23.15 | mut | near |
| 16:23.27 | justinu | hmm... seems like that's always the issue |
| 16:23.35 | vaewyn | remember... "gain" in PRI is like a "gain" in a network swtich... it doesn't exist |
| 16:23.46 | mut | well |
| 16:23.54 | mut | adjusting the gain on my cisco box got rid of my echo |
| 16:23.55 | vaewyn | is just software covering up an issue somewhere else |
| 16:23.57 | justinu | you're mistaken... all T1 equipment has tx/rx padding settings. |
| 16:24.11 | vaewyn | T1 yes... PRI no |
| 16:24.12 | justinu | everything from DMS switches to dialogic cards |
| 16:24.17 | justinu | yes, PRI runs over T 1 |
| 16:24.21 | delmar | is it a good idea to uncomment /* #define CONFIG_ZAPTEL_MMX */ for an intel processor or just leave it alone? |
| 16:24.38 | vaewyn | ok... "ANALOG T1" yes... PRI no |
| 16:24.54 | justinu | you seem confused about what a PRI is |
| 16:25.03 | justinu | first of all, what's "ANALOG T1"? |
| 16:25.07 | justinu | a T1 is digital, period. |
| 16:25.29 | vaewyn | The signalling is... but the calls can be either |
| 16:25.30 | justinu | PRI is a T1 w/ clear channels on 23 timeslots, and a HDLC data channel on 1 timeslot. |
| 16:25.40 | vgster | delmar: i asked that too |
| 16:25.48 | justinu | the audio data is digital too |
| 16:25.57 | justinu | PCM 8000 samples/sec 8bit |
| 16:25.58 | delmar | vgster, well im sure it wont break if I leave it alone.. for now lol :P |
| 16:26.01 | vaewyn | only on BRI/PRI |
| 16:26.05 | justinu | nope |
| 16:26.06 | *** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka) |
| 16:26.08 | justinu | you're flat wrong... |
| 16:26.12 | vgster | i enabled it |
| 16:26.17 | justinu | covad's here |
| 16:26.29 | tzafrir_laptop | delmar, doesn't the makefile set it automatically by CPU type or anything? |
| 16:27.00 | vaewyn | Ok... so how come i still have 1 old school T that I can actually take one of the copper pairs on it... hook a line set on it and listen to the call on it? |
| 16:27.05 | delmar | tzafrir_laptop, u tell me :P. |
| 16:27.39 | Juggie | vaewyn, you seem to be confused. |
| 16:27.45 | Juggie | a t1 is only a single pair |
| 16:27.57 | Juggie | if its copper. |
| 16:27.59 | vaewyn | These days yes... not in the old days |
| 16:27.59 | docelm0 | A T1 is 2 pair |
| 16:28.17 | mut | a t1 hasn't ever changed.. |
| 16:28.23 | Juggie | vaewyn, then it isnt a t1 |
| 16:28.26 | coppice | it has actually |
| 16:28.27 | vaewyn | old days it was a 24 pair and signalling ran on one... analog calls on the other 23 |
| 16:28.29 | delmar | docelm0, is right. i thought t1/e1 is 4 wires.. 2pair |
| 16:28.46 | *** join/#asterisk flujan (n=flujan@internet.nube.com.br) |
| 16:28.47 | delmar | there we go. that in my book is NOt T1. |
| 16:28.49 | delmar | u see... |
| 16:29.01 | docelm0 | T1 is Pins 1 2 4 5 |
| 16:29.02 | coppice | vaewyn: you're making this up as you go along, aren't you :-) |
| 16:29.02 | Juggie | when a t1 is on copper isnt it just two pairs. |
| 16:29.04 | delmar | t1 itself refers to line speed of what was it.. 1592 or some number? |
| 16:29.05 | Juggie | infact i know it is |
| 16:29.06 | flujan | hi guys, how can I monitor a channel with a call already in progress? |
| 16:29.09 | Juggie | its just a transmit and receive |
| 16:29.15 | djMax | does anybody know what causes a sip registration to be "booted" by asterisk, other than the expiration? |
| 16:29.17 | Juggie | er, one pair. i mean. |
| 16:29.42 | delmar | now.. channelised T1/E1... thats different. is that what u are referring to vaewyn? or even older than that? |
| 16:30.02 | kippi | anyone from digium around? |
| 16:30.02 | Juggie | a t1 definitally only needs one pair, i remember doing this in a t1 course last year. |
| 16:30.02 | vaewyn | Heck if I know... from the 70s |
| 16:30.18 | justinu | t1's need 2 pairs |
| 16:30.22 | justinu | rx pair, and tx pair |
| 16:30.36 | vaewyn | I bet you guys have never seen a T come in over coax either :P |
| 16:30.41 | delmar | Juggie, well conventional T1 connections that I know of .. is 2 pairs.. same as 10/100 ethernet. almost the same shit accept diff. signalling. |
| 16:30.41 | justinu | HDSL T1 does it slightly differently, but still needs to pairs |
| 16:30.54 | jbalcomb | djMax: what phone? |
| 16:30.55 | justinu | i've seen T1s come in over coax... it's called a channelized DS3 |
| 16:31.04 | Juggie | i think your probally right. |
| 16:31.09 | djMax | polycom IP500 |
| 16:31.11 | *** join/#asterisk Egonis (n=chultay@CPE000255fa0fde-CM00407b87dc7b.cpe.net.cable.rogers.com) |
| 16:31.32 | delmar | what anoys me is that the T1/E1 cards u buy are almost the same as 10/100 network cards.. just that the fpga is going a diff job. so why the hell are we paying US$500 for hardware thats physically almos the same as a $20 network card. grrrr. |
| 16:31.32 | Juggie | they used cables that looked a little like 3.5microphone cable |
| 16:31.37 | Juggie | so it was probally a pair inside there. |
| 16:31.46 | Egonis | When I change my vm pin, the log says permission denied on /etc/asterisk/voicemail.conf, although I chown'ed it to asterisk:asterisk. I can't find any help on google |
| 16:31.47 | Juggie | so yah, 2pairs that makes sense. |
| 16:32.00 | Juggie | one pair in each cable. |
| 16:32.06 | jbalcomb | djMax: hrmm.. ok. i believe there is an issues with the GXP-2000 dropping regsitration but not sure about the ip501 |
| 16:32.22 | djMax | I see a registration at 11:32EST for 600 seconds from a particular phone. But I bet that before that 10 minutes, asterisk will lose the IP |
| 16:32.39 | delmar | u know what... someone in here is gonna paste a link to some site that has the T1 cable specification ... google is your friend :P |
| 16:32.48 | jbalcomb | Egonis: is asterisk running as asterisk.asterisk? also, what about the rwx perms? |
| 16:32.50 | docelm0 | delmar, your almost right.. T1 is pins 1 2 4 5 and ethernet is pins 1 2 3 6 |
| 16:33.02 | *** join/#asterisk cian_ (n=cian@g5.cian.ws) |
| 16:33.03 | Juggie | you wanna know whats fun, making cross over t1 cables :P |
| 16:33.03 | justinu | juggie: those are called bantam jacks |
| 16:33.08 | docelm0 | I have cut a MILLION of these cables.. And GOD am I sick of it. |
| 16:33.11 | Juggie | justinu, ahh :) |
| 16:33.17 | jbalcomb | Juggie I just did that this morning |
| 16:33.18 | Egonis | jbalcomb: I tried changing it to 0755 |
| 16:33.20 | docelm0 | Juggie, I have that down to a science |
| 16:33.33 | justinu | juggie: my sunset T1 (test set) has a special RJ45 to two bantam cable |
| 16:33.52 | justinu | connector, that is |
| 16:33.55 | Juggie | i dont know very much about t1, when i rememberedf the course and seeing the two cables |
| 16:33.55 | jbalcomb | Egonis isnt 755 rwxr-xr-x? perhaps 666 would be better? |
| 16:34.02 | Juggie | i neglected to think that one cable = a pair |
| 16:34.08 | GerbilNut | After we upgraded to * 1.2.4 we are no longer getting the announcements in the queue. Anyone else seen / fixed this? |
| 16:34.31 | jbalcomb | GerbilNut: check the logs? |
| 16:34.56 | GerbilNut | i'm not seeing it even try to play the file, even though i have "announce-frequency=90" |
| 16:35.06 | jbalcomb | docelm0: can't you just buy t1 xover connectors? |
| 16:35.18 | Juggie | jbalcomb, they are supprisingly hard to come by |
| 16:35.22 | delmar | jbalcomb, u can make them |
| 16:35.22 | djMax | boom, so 4 minutes passed, and it's dropped the IP for that registration |
| 16:35.33 | jbalcomb | delmar yeah, i made on this morning |
| 16:35.54 | jbalcomb | Juggie that seems odd. i dont do a lot of telco work and i have made enough to want to buy them |
| 16:36.08 | Juggie | jbalcomb, i looked aronud online i didnt find any |
| 16:36.09 | delmar | thankfully T1 uses good old RJ45.. apart from the little pin differences.. its just as simple as any network cross over. |
| 16:36.18 | jbalcomb | djMax no message from asterisk or the phone about it dropping? |
| 16:36.24 | djMax | nope, not a peep. |
| 16:36.33 | delmar | djMax, SIP? |
| 16:36.33 | djMax | asterisk destroyed some SIP dialogs, but none from that IP. |
| 16:36.35 | jbalcomb | djMax are you syslogging from the phone? |
| 16:36.39 | djMax | (one from the asterisk IP) |
| 16:36.41 | delmar | djMax, u doing a debug? |
| 16:36.53 | *** join/#asterisk p0g0__ (n=pogo@madwifi/support/p0g0) |
| 16:36.54 | djMax | just turned on SIP debug, but had debug and verbose way up on CLI |
| 16:36.55 | delmar | djMax, sip debug ip (num) ? |
| 16:36.58 | coppice | actually a T1 uses an RJ48C, but exactly how that is supposed to differ from an RJ45 I have never worked out |
| 16:37.08 | djMax | ooh, didn't know you could limit it. one sec. |
| 16:37.12 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.141.8) |
| 16:37.14 | justinu | delmar: i've seen T1s which use DB15 connectors, DB9, bantam jacks, wirewrap, etc. RJ45 is not the only way they're delivered |
| 16:37.23 | Dr-Linux | hi |
| 16:37.29 | delmar | justinu, indeed. |
| 16:37.38 | Dr-Linux | anybody is using "web Voice mail" ? |
| 16:37.49 | Juggie | t1 is just a signaling protocol |
| 16:37.54 | Juggie | it can arrive on anything |
| 16:37.58 | Dr-Linux | justinu: hey friend, how are you? |
| 16:38.01 | Juggie | its not specific to any cable type |
| 16:38.05 | coppice | t1 is not a signalling protocol at all |
| 16:38.19 | *** part/#asterisk flujan (n=flujan@internet.nube.com.br) |
| 16:38.25 | Juggie | what would you define it as? |
| 16:38.26 | justinu | it's a layer 1 electrical protocol |
| 16:38.27 | *** join/#asterisk websae (n=icechat5@adsl-64-149-206-121.dsl.milwwi.sbcglobal.net) |
| 16:38.49 | delmar | Juggie, exactly. so why are we paying US$500 for an fpga to talk T1 language, instead of ethernet. the programming of the fpga is worth an extra US$480? |
| 16:38.57 | coppice | justinu have a fair description |
| 16:39.02 | Juggie | i've been schooled then :P regardless what i was implying is that t1 can come in on anything |
| 16:39.07 | Juggie | cableing is irelevant |
| 16:39.19 | justinu | voice grade pairs usually aren't good enough |
| 16:39.20 | djMax | ok, so turned on sip debug, it just registered (11:40) for 10 mins. |
| 16:39.26 | justinu | bridge taps and loading coils fuck up T1 |
| 16:39.31 | [TK]D-Fender | delmar : The fact that nobody's doing it in qty's of 1,000,000 allows them to jack it up. |
| 16:39.41 | coppice | delmar: what's the T1 language? weird :-) |
| 16:39.42 | Juggie | yah its all about mass producing |
| 16:39.47 | delmar | [TK]D-Fender, now this is true. |
| 16:40.18 | justinu | coppice: today someone tried to tell me about "ANALOG T1" ;) |
| 16:40.28 | Juggie | hah |
| 16:40.38 | Juggie | you can go totally clear channel, eg 24 b channels |
| 16:40.40 | coppice | yeah. I saw that. wacky, huh? :-) |
| 16:40.41 | Juggie | but theres no analog :P |
| 16:40.44 | rayvd | that would be 5,000 cups attached to a string. |
| 16:40.45 | rayvd | right? |
| 16:40.46 | jbalcomb | Juggie http://www.cablesunlimited.com/products/Prod_Individual3.aspx?groupcode=I3452 |
| 16:41.02 | rayvd | or 10000 cups and 5000 strings :) |
| 16:41.04 | coppice | someone tried to tell me only long distance is digitised the other week :-) |
| 16:41.30 | delmar | coppice, sure.. and your chickens over there are hydropinic too? :P |
| 16:41.30 | Juggie | jbalcomb, thanks. |
| 16:41.37 | delmar | hydroponic* |
| 16:42.05 | Juggie | jbalcomb, find me one that doesnt look like a toy :) |
| 16:42.12 | Juggie | the cross overs i have are peremenant |
| 16:42.14 | delmar | Adapter, T1/E1, Crossover, Dongle, Smartronix Superlooper thingywhatsit trundler. |
| 16:42.19 | Juggie | not something i turn on for a minute |
| 16:42.23 | jbalcomb | Juggie aww. that was my favorite part |
| 16:42.38 | justinu | the adtran t1 equipment usually come with loopback jacks like that |
| 16:42.51 | justinu | except they look less toylike |
| 16:43.04 | jbalcomb | justinu they are xover/loopback? |
| 16:43.25 | delmar | Zaptel Version: SVN-branch-1.2-r962M Echo Canceller: MG2 |
| 16:43.29 | delmar | right. lets see how that goes... |
| 16:43.42 | justinu | yeah, loopback |
| 16:43.47 | jbalcomb | justinu i have 6 adtran CUSs and 3 ACEs and i didn't get anything like that |
| 16:43.50 | justinu | not crossover tho |
| 16:43.51 | delmar | this is the part where i get a call coming in.. and scream "echo bloody echo" down the phone. |
| 16:44.00 | justinu | jbalcomb: you got boned, i think :P |
| 16:44.20 | jbalcomb | justinu so it seems. i have to fight just to have my damn console cable serial connectors |
| 16:44.29 | justinu | heh |
| 16:44.54 | coppice | delmar with 100% humidity for most of the year, pretty much everything stands in water here |
| 16:45.07 | delmar | lol |
| 16:45.25 | jbalcomb | coppice you might consider investing in Arm & Hammer |
| 16:45.45 | delmar | imnot even going to test incoming calls... |
| 16:45.56 | delmar | the echo on out going calls just hit the roof. |
| 16:46.02 | delmar | mg2 = stinky |
| 16:46.35 | fu3 | ahhhhhhhhhhhhhhhHHHHHHHHHHHH!@(*!&@(*!&@(*!&@(*!&@(*!&@(!& |
| 16:46.40 | fu3 | Qwest was just here |
| 16:46.41 | jbalcomb | eh? |
| 16:46.49 | fu3 | they just told me all my problems are because my Centrex is programmed wrong |
| 16:47.05 | cpm | damn, I hate it when I program my centrex wrong. |
| 16:47.05 | fu3 | but the people who program the centrex say it is programmed right and they blame qwest! |
| 16:47.13 | jbalcomb | haha |
| 16:47.13 | fu3 | so I cant get anywhere! |
| 16:47.27 | delmar | fu3, do it yourself. |
| 16:47.28 | delmar | :P |
| 16:47.37 | [TK]D-Fender | centrex = total waste of time. |
| 16:47.37 | fu3 | haha ok.. give me the code to the CO and i'll do it |
| 16:47.44 | fu3 | I am SO trying to get away from the centrex. |
| 16:47.50 | fu3 | as fast as possible (within reason!) |
| 16:47.53 | jbalcomb | whats a centrex? |
| 16:47.57 | cpm | fu3, what's really at issue? |
| 16:48.17 | fu3 | I have lines that can and then cannot dial LD.. lines that can and then cannot transfer calls.. lines that forward to other lines and back, etc. |
| 16:48.28 | jbalcomb | [TK]D-Fender I just got a Telrad Key-BX for handling my faxes. Isn't that nice? |
| 16:48.29 | fu3 | all these things are supposedly programmed into the Centrex, but no one is doing it. |
| 16:48.40 | fu3 | and there are no work orders |
| 16:48.44 | fu3 | so either the hardware is failing, or it's owned and someone's fucking with me nicely. |
| 16:48.53 | cpm | CenTRal office EXchange |
| 16:48.53 | [TK]D-Fender | jbalcomb : If I had any idea of what that usit is for, perhaps! |
| 16:49.00 | justinu | centrex is POTS but with features |
| 16:49.04 | justinu | like transfer, etc. |
| 16:49.07 | fu3 | yeah |
| 16:49.09 | [TK]D-Fender | jbalcomb : What does it do? |
| 16:49.17 | fu3 | and the programming per-line is constantly changing |
| 16:49.25 | fu3 | I have to send in trouble tickets daily to get the programming "fixed" |
| 16:49.29 | cpm | fu3, if you can 'prove' it, it's quests problem without regard to what the centrex is. |
| 16:49.32 | djMax | so I'm seeing some "scheduling destruction of sip dialog in 15000 ms" messages. |
| 16:49.41 | fu3 | cpm.. thats the prob.. I CANT prove it! |
| 16:49.43 | delmar | man that mg2 echo-can was bad... lots worse. |
| 16:49.48 | jbalcomb | [TK]D-Fender its a phone system. i have it in my asterisks PRI card and analog lines out to the fax machines |
| 16:49.50 | justinu | centrex is great if you want to be the telco's bitch |
| 16:49.55 | fu3 | which I dont!~ |
| 16:50.10 | fu3 | I want to be Asterisk's bitch |
| 16:50.18 | cpm | Gotta call that won't do ld, put in a ticket, another call, another ticket, transfer doesn't go through, put in a ticket, rinse and repeat |
| 16:50.19 | delmar | fu3, bend over :P |
| 16:50.26 | fu3 | hahahaha |
| 16:50.30 | cpm | Seriously. |
| 16:50.32 | fu3 | cpm.. thats how it works now. |
| 16:50.40 | cpm | How many tickets a day? |
| 16:50.45 | coppice | centrex makes excellent technical sense - in the minds of the marketing dept |
| 16:50.47 | fu3 | 1 on adverage |
| 16:50.50 | fu3 | some days 0, some days 5 |
| 16:50.55 | cpm | Need 1 or more per hour |
| 16:50.57 | jbalcomb | fu3 I must say though that it seems less likely to be Qwests issue |
| 16:51.09 | fu3 | you think it's mine? |
| 16:51.11 | cpm | Qwest send the bill yes? |
| 16:51.13 | fu3 | yes |
| 16:51.19 | cpm | that's it then. |
| 16:51.38 | fu3 | but its not. They claim they cant program the centrex as only the state of MN phone agency (intertech) can do it. |
| 16:51.52 | fu3 | but they claim that the centrex is programmed right, so they send me to Qwest repair |
| 16:52.01 | fu3 | who say it's a programming issue, so they send me back to InterTech! |
| 16:52.02 | cpm | That's not your problem, you are being billed for service that isn't being delivered. |
| 16:52.07 | jbalcomb | fu3 make them talk to each other |
| 16:52.13 | fu3 | I've been trying and trying! |
| 16:52.20 | fu3 | no one stays around long enough |
| 16:52.22 | fu3 | they come, they go. |
| 16:52.28 | delmar | fu3, subpoena time? :P |
| 16:52.33 | fu3 | hahahaha MAYBE! |
| 16:52.34 | jbalcomb | schedule a time for call perhaps and conference? |
| 16:52.42 | cpm | <PROTECTED> |
| 16:52.45 | fu3 | jbalcomb.. last two times I've done that, they cancelled. |
| 16:52.46 | FlyboySR22 | Hey everyone - I need some advice on an IVR application. Basically I need callers to be able to use a local DID number to hit our * box, then have the caller prompted for a user ID and PIN/password, and then verify their callerID is coming from a specific phone number. Once they get through that, there would be two additional questions that they would have to answer and all responses need to be tracked in a MySQL DB.....whew....anyone got ideas...? |
| 16:52.46 | cpm | that's a good idea |
| 16:53.08 | fu3 | I'm about ready to WRITE A LETTER! |
| 16:53.08 | fu3 | :) |
| 16:53.24 | cpm | jbalcomb's approach is the best. |
| 16:53.29 | cpm | Letter comes after |
| 16:53.34 | salviadud | write a letter? |
| 16:53.38 | jbalcomb | FlyboySR22 its all doable and not really hard. you'll find lots of answers for your quest on the voip-wiki sit3e |
| 16:53.41 | salviadud | instead of placing a call? |
| 16:53.41 | delmar | fu3, just head it up ...NOTICE OF IMPENDING LEGAL ACTION. |
| 16:53.51 | fu3 | i ignore phone calls.. I dont ignore personally written letters. |
| 16:53.55 | justinu | fu3: intertech? |
| 16:54.03 | fu3 | InterTech is the state of MN phone "agency" |
| 16:54.07 | cpm | yeah, really, Intertech? |
| 16:54.08 | jbalcomb | fu3 writing a letter is certainly not a bad idea |
| 16:54.16 | FlyboySR22 | jbalcomb, but it would be using the * IVR system..? |
| 16:54.17 | justinu | is that like our public utilities commision? |
| 16:54.18 | fu3 | they are basically a 3rd party that I have to go through before I can call qwest. |
| 16:54.24 | fu3 | www.intertech.state.mn.us |
| 16:54.27 | salviadud | Cemtex is not a bad idea |
| 16:54.32 | delmar | Zaptel Version: SVN-branch-1.2-r962 Echo Canceller: KB1 |
| 16:54.36 | delmar | back to KB1 |
| 16:54.38 | fu3 | if you can get there.. it may be MN only. |
| 16:54.39 | salviadud | or is it called semtex? |
| 16:54.40 | justinu | heh, semtex is what you want |
| 16:54.43 | jbalcomb | FlyboySR22 sure, IVR/ACD with some AGI and DB work |
| 16:54.50 | cpm | yeah, semtex trumps centrex |
| 16:55.08 | salviadud | any anarchists around? |
| 16:55.26 | fu3 | haha |
| 16:55.26 | FlyboySR22 | jbalcomb, IVR and ACD I can do, its the AGI and DB work that I need help on...anyone around that codes for $$$ on * interested in a side project..? |
| 16:55.29 | cpm | but what does jbalcomb call it? |
| 16:55.37 | Egonis | someone updated their vm pin, and voicemail.conf is now 0 lines... !!! Anyone else experience this? |
| 16:55.56 | cpm | restore from tape? |
| 16:55.57 | jbalcomb | FlyboySR22 try [TK]D-Fender, [av]bani, or iCEBrkr. There are other but I know of those three. |
| 16:56.02 | fu3 | Qwest and InterTech also claim they cannot provide a list of my phone numbers with the associated services, and they also cannot provide usage statistics for my phone lines. |
| 16:56.08 | salviadud | egonis that's a horrible bug there |
| 16:56.14 | delmar | wow. huge difference on kb1. its by far the best one yet but..outgoing calls almost 0 echo.. as good as 0 anyway... incoming calls.. echo at the start.. deminishing over about 30seconds. |
| 16:56.24 | jbalcomb | FlyboySR22 the information is on the woip-wiki for the DB functions and isn't particularly difficult. |
| 16:56.39 | Egonis | salviadud: Like, in Asterisk?? |
| 16:56.50 | jbalcomb | delmar are you adjusting any other settings |
| 16:56.52 | delmar | i wonder why ... the echo is so loud at the beginning of the call when its an incoming call.. . |
| 16:56.58 | FlyboySR22 | jbalcomb, Thanks - I will look at that first and see if I can figure it out, I am always interested in expanding my horizons, but I also know my practical limits :-) |
| 16:57.02 | salviadud | yep |
| 16:57.06 | delmar | jbalcomb, nope. just changed the echo-can. |
| 16:57.17 | jbalcomb | delmar echo trainging |
| 16:57.20 | delmar | jbalcomb, tried mg2. terrible. |
| 16:57.21 | salviadud | i think you might need to report it |
| 16:57.25 | delmar | jbalcomb, yep. =800 |
| 16:57.52 | cpm | jbalcomb, nope. just changed the echo-can. |
| 16:57.57 | cpm | dang that echo |
| 16:57.59 | jbalcomb | eh? |
| 16:58.05 | cpm | eh? |
| 16:58.07 | delmar | jbalcomb, now.. if I had an IVR .. answering the phone... by the time it actually hit a real person.. the echo would probably be gone. |
| 16:58.08 | fu3 | eh? |
| 16:58.23 | jbalcomb | delmar yeah, that would probably be part of the echo fading |
| 16:58.24 | cpm | can asterisk get rid of my irc echo? |
| 16:58.33 | GerbilNut | can asterisk get rid of my irc echo? |
| 16:58.38 | *** join/#asterisk rdlang (n=test@s55936e2e.adsl.wanadoo.nl) |
| 16:58.41 | cpm | ccaann aasstteerriisskk ggeett rriidd |
| 16:58.53 | jbalcomb | delmar do you have silence suppresion, comfort noise, and VAD all turned off? |
| 16:58.55 | delmar | in this case the irc echo-can might be mode +b cpm :P |
| 16:59.18 | fu3 | :) |
| 16:59.19 | GerbilNut | so yeah, i can not get the "you are # in line" working in the queues since I upgraded to 1.2.4 |
| 16:59.25 | jbalcomb | delmar have you checked your rxgain & txgain as well? |
| 16:59.37 | delmar | jbalcomb, i think that is all handled by the grandstream phone itself but ill check. this problem occurs on softphones as well tho. |
| 16:59.40 | jbalcomb | GerbilNut possible syntax change? |
| 16:59.44 | rdlang | I hava a serious issue with connecting a TE110p card to an ISDN 10 (isdn15 on friday, now isdn10 for testing) |
| 16:59.55 | rdlang | i am not sure witch cabling to use |
| 16:59.56 | GerbilNut | i checked the sample queue file and it looks identical |
| 17:00.01 | rdlang | i am located in the netherlands |
| 17:00.09 | delmar | jbalcomb, the rxgain is not the issue because the volume of the other end is fine, and they cant hear echo. the txgain is the issue... |
| 17:00.10 | fu3 | that rocks |
| 17:00.13 | *** join/#asterisk dpolitech (n=Owner@207.224.48.130) |
| 17:00.14 | fu3 | I want to go there again this summer |
| 17:00.20 | delmar | jbalcomb, and its at about where it needs to be for volume... |
| 17:00.50 | delmar | jbalcomb, if i mess with it further the echo will start out less... but the call volume will be such that the far end cant hear this end very well... |
| 17:00.53 | rdlang | my telco is KPN (dutch) |
| 17:01.07 | delmar | jbalcomb, so .. rx/tx gains are already at their best. |
| 17:01.23 | FlyboySR22 | Can * handle 600 inbound calls via a SIP trunk (or trunks) on a single server (big/beefey) or is that pushing it too far..? |
| 17:01.33 | *** join/#asterisk muzzz (n=chatzill@218.111.148.137) |
| 17:01.57 | delmar | jbalcomb, if I remove the X100 card.. put in a TDM400.... and dont toucha thing.. other than the module of course.. there is no echo at all :P |
| 17:02.38 | coppice | a TDM400 may give you a better match to the line, but they have lots of other problems |
| 17:02.43 | rdlang | can someone help me with this? => TE110p , kpn (dutch) , ISDN10 / ISDN15 over HDLC, wich cabling? |
| 17:02.49 | fu3 | yeah.. like no groundstart support on TDM400 |
| 17:02.56 | rdlang | i will try to wait for my turn now :) |
| 17:02.57 | delmar | yeah..get this... |
| 17:03.03 | jbalcomb | delmar i have many problems with GXP-2000 that were made much better by reducing my rxgain a bit and txgain a lot. |
| 17:03.26 | delmar | if I increase the TXgain to 4.5, on the TDM400 card...which is requred by the telephone exchange for successful fax transmition/reception.. then there IS some echo... |
| 17:04.00 | delmar | there is a service here to test fax's.. so was was messing with RX/TXFax function.. and the exchange test was saying the transmit dB my end was too low.. |
| 17:04.14 | jbalcomb | delmar yeah, when i dropped the gains for fixing echo it broke my faxing. im implementing a cheap analog pbx to handle faxes outside of asterisk |
| 17:04.29 | delmar | so to get 100% good fax transmition.. i need to increase TXGain to 4.5, but that shits my echo.. so fax transmition in this solution is simply not ever going to happen. |
| 17:05.12 | delmar | jbalcomb, yeah im gonna hook up a fax switch and run it thru a modem on relayfax or some crap. nice and simple. |
| 17:05.37 | jbalcomb | delmar sounds good. we have to have a little more because we have 8 fax lines |
| 17:05.37 | coppice | if you need a few dB gain for FAX to work, something else is really broken in your system |
| 17:06.02 | *** join/#asterisk mrtwister|mobile (n=andrius@cable-10-68.cgates.lt) |
| 17:06.06 | jbalcomb | coppice my gains are -6 and -14 |
| 17:06.09 | delmar | coppice, yeah .. its call Telecom New Zealand.. aint u heard of them? they are a pack of wankers. |
| 17:06.37 | jbalcomb | coppice there are documented studies on noise artifacts and echo cancellation ruining faxing |
| 17:06.54 | coppice | FAX works over a considerable range of recived signal levels. things rarely end up marginal for signal level |
| 17:07.02 | trig_hm | who's chris meredith? |
| 17:07.03 | delmar | my cable distance from the exchange is 1km aparantly. so why I need to increase it so much for fax to work i dont know. |
| 17:07.06 | trig_hm | worry wrong window |
| 17:07.24 | jbalcomb | coppice isnt not a matter of signal range so much as signal loss and artifacts |
| 17:07.34 | coppice | EC which is not port of the modems certainly ruins modem operation, but signal level should not be a problem |
| 17:07.59 | coppice | you shouldn't have artifacts. |
| 17:08.12 | jbalcomb | coppice gain + EC = artifacts |
| 17:08.16 | delmar | Oh here is a new problem.. and is the root cause of my setting up another PC for testing.. the TDM400 card I am using could in fact be faulty...... it started making noises during calls... like.. imagine u are standing in a checkout line at the supermarket and can hear the random "beeps" as things are scanned. not as often as that.. but very much like that. |
| 17:08.20 | fu3 | How can I test to see if the dialtone I am receiving, matches what I am seeing in my indications.conf ? (let me guess, wiki?) |
| 17:08.40 | coppice | modem signals should disable EC |
| 17:08.43 | fu3 | because, if the indications are off, perhaps * doesnt realize a hangup has taken place because the dialtone it receives doesnt match? |
| 17:09.12 | jbalcomb | coppice should as in theory or should as in its a config option that should be turned on? |
| 17:09.19 | delmar | yeah the fax tone turns the echo-can off. |
| 17:09.36 | delmar | by default it works like that in Asterisk |
| 17:09.43 | coppice | if the FAX tones do not disable EC, your setup is broken |
| 17:09.48 | delmar | there is a setting to make the echo can remain ON .. |
| 17:10.02 | delmar | coppice, yep. |
| 17:10.29 | delmar | ./* #define NO_ECHOCAN_DISABLE */ |
| 17:10.55 | Dr-Linux | anybody is using "web voicemail" ?? |
| 17:10.57 | delmar | if its commented like that... the 2100hz tone will turn off the echo-can.. aparantly. |
| 17:11.36 | [TK]D-Fender | jbalcomb : You're better off just leaving them on straight analog lines. |
| 17:11.45 | jbalcomb | Dr-Linux you talking about ARI? |
| 17:12.13 | jbalcomb | [TK]D-Fender How do I have it rollover fax machines on straight analog lines? |
| 17:12.36 | mut | hah |
| 17:12.36 | mut | http://us.gizmodo.com/gadgets/software/windows-vistas-worst-performer-windows-vista-156415.php |
| 17:12.57 | [TK]D-Fender | jbalcomb : have the telco put them on a circular hunt group or something. |
| 17:13.11 | Dr-Linux | jbalcomb: ARI statand for? |
| 17:13.15 | fu3 | umm.. can I record a dialtone in asterisk? |
| 17:13.28 | Dr-Linux | i'm talking about >> http://www.voip-info.org/wiki/view/Asterisk+gui+vmail.cgi |
| 17:13.32 | PupenoL | Hello. |
| 17:13.34 | fu3 | hi |
| 17:13.37 | jbalcomb | coppice delmar anyway to check a call and see if the EC has been turned off because its a fax/modem |
| 17:14.00 | delmar | jbalcomb, surely doing.. zap show channel X |
| 17:14.08 | delmar | jbalcomb, during the fax call |
| 17:14.08 | jbalcomb | Dr-Linux Asterisk Ready/Relatime Interface i think |
| 17:14.43 | DeadZen | info on eagi? |
| 17:14.58 | *** join/#asterisk mmlj4 (n=jkelly@ip70-171-92-106.no.no.cox.net) |
| 17:15.06 | Dr-Linux | jbalcomb: please look at >> http://www.voip-info.org/wiki/view/Asterisk+gui+vmail.cgi |
| 17:15.24 | jbalcomb | delmar damn, i didnt know about that. |
| 17:15.30 | *** join/#asterisk LostFrog (n=LostFrog@c-71-199-110-205.hsd1.pa.comcast.net) |
| 17:15.59 | jbalcomb | Dr-Linux i guess im not using that then |
| 17:16.14 | LostFrog | Is there anything that would be bad about running * on a socket-775 board versus a socket-478 board? |
| 17:16.36 | PupenoL | is it possible to access a postgresql database for realtime without using odbc ? |
| 17:17.07 | Dr-Linux | jbalcomb: i have installed/configure it fine, but i have a little premissions problem. |
| 17:17.18 | jbalcomb | [TK]D-Fender hrmm.. i'll ask the telco based phone guy about that. we only have 8 fax lines so we could have 8 analog lines straigt to the fax machines and just set the hunt groups from the telco? |
| 17:17.51 | jbalcomb | Dr-Linux we have a cron job that runs every five minutes and sets the perms on the vm files so the web interface can make changes |
| 17:18.04 | *** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it) |
| 17:18.26 | jbalcomb | Dr-Linux im building a new asterisk server in the next couple weeks and i am hoping to find a better solution beforehand |
| 17:19.03 | jbalcomb | PupenoL i dont think so. trying google.com or voip-wiki |
| 17:19.48 | jbalcomb | LostFrog it might be best to ask digium directly. they are real sweet about those questions. |
| 17:19.50 | PupenoL | jbalcomb: I have tried and I've found that since cdr can only be accessed by odbc (for postgresql), people tend to use odbc for all postgresql. |
| 17:20.03 | [TK]D-Fender | jbalcomb : Yup... its what we do here. There is no way to jsutify equipment costs for anything else if SpanDSP doesn't cut it for reception. |
| 17:20.25 | jbalcomb | SpanDSP? |
| 17:20.38 | delmar | jbalcomb, thats teh rx/tx fax app |
| 17:20.38 | [TK]D-Fender | jb... you know.. RxFAX...... |
| 17:20.42 | jbalcomb | PupenoL thats probably the way to do it then |
| 17:21.01 | [TK]D-Fender | jbalcomb : if you need it for INCOMING faxes, sure.. I wouldn't sue if for sending though. |
| 17:21.27 | delmar | SpanDSP is a great idea but.. if the rx/txgain needs to be messed with.. which comprimises your echo.. its a waste of time.. and I thought having a nice TDM400 card would mean I could do just that...waste of bloody time. |
| 17:21.44 | delmar | oh and the distinctive ring detection on X100 OR TDM400 is a joke.... |
| 17:21.47 | *** join/#asterisk puzzled (n=yeahrigh@62.45.11.228) |
| 17:21.53 | delmar | so u always fall back on fax tone detection anyway... |
| 17:22.07 | jbalcomb | delmar [TK]D-Fender ah, the fax stuff is still a mystery to me because we have three different setups. one using RxFax, one using ATA, and one being looped out to a copper line |
| 17:22.08 | delmar | and that all works great ... IF you dont mind echo. |
| 17:22.35 | fu3 | so I understand this correctly.. indications.conf has to do with tones generated BY asterisk, and has nothing to do with the tones coming across a line into the FXS port? |
| 17:22.41 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 17:22.43 | delmar | jbalcomb, its not rocket science really.. when u want to look at it.. u will understand it quickly. |
| 17:23.01 | [TK]D-Fender | jbalcomb : Loop-out / ATA = pain. RxFax is mostly decent, but for SEND points I'd go all-analog. Looking at the expense of anything else it just isn't worth it. |
| 17:23.13 | jbalcomb | delmar i hope so. im supposed to have each one moved over to the analog pbx by days end. |
| 17:23.20 | [TK]D-Fender | I use SpanDSP for employee-direct faxes, but not essential stuff like CustomerService |
| 17:23.37 | jbalcomb | [TK]D-Fender the analog pbx we got was only $1300 |
| 17:23.50 | delmar | jbalcomb, piece of cake. if you see anything that says.. "Capt James T Cook was here" ... it will be plain sailing. |
| 17:24.06 | delmar | i swear he owned my fax machine. |
| 17:24.15 | [TK]D-Fender | jbalcomb : And an analog line would cost $20/month? Thats YEARS of service + installation & trouble |
| 17:24.19 | jbalcomb | delmar haha.. i'll keep my eyes pealed. |
| 17:24.43 | jbalcomb | [TK]D-Fender service + installation + trouble = {ME} |
| 17:24.58 | Dr-Linux | jbalcomb: mine everthing is fine .. but i get an error while delete/modify any voicemail |
| 17:25.06 | Dr-Linux | jbalcomb: like >> Unable to open '0001' in '4092', 'INBOX' |
| 17:25.07 | jbalcomb | [TK]D-Fender I feel ya though, if we could have the phone company handle the hunts then I have no idea why we are doing this |
| 17:25.20 | Dr-Linux | Cannot move/delete messages: Unable to lock path |
| 17:25.26 | [TK]D-Fender | jbalcomb : add that cost to your company's expenditures and there's little chance its cost-effective... |
| 17:25.42 | jbalcomb | Dr-Linux ok, i get permissions errors but nothing about lock path. |
| 17:25.45 | [TK]D-Fender | Dr-Linux : Thats a rights problems. |
| 17:25.47 | delmar | i reckon KB1 must be the better echo-can... it works great here on the X100.. cant wait to test it on the TDM400... and also.. i notice kb1 is the echo-can that is set as default in the latest zaptel release. |
| 17:26.12 | jbalcomb | [TK]D-Fender yeah, i understand the cost of my salary being spent on that rather than more progressive things (which is what i'm best at) |
| 17:26.40 | Dr-Linux | [TK]D-Fender: yes sir ofcos its rights problem .. but i'm not getting where is problem. relevant dirs have full permissions |
| 17:26.46 | jbalcomb | delmar you best let me know about that. i just got my new TE411P today and I'm building my new server in the next couple of weeks. |
| 17:27.12 | jbalcomb | Dr-Linux by full permissions do you mean like 777? |
| 17:27.29 | *** join/#asterisk puzzled (n=yeahrigh@62.45.11.228) |
| 17:27.49 | *** join/#asterisk mover (n=dlu@gw-dus-net.dus.de.ncore.net) |
| 17:27.58 | FlyboySR22 | anyone have anything good to say anout a dialtone provider able to provide Canada DIDs..? |
| 17:28.03 | jbalcomb | Dr-Linux sometimes linux is sweet enough to lend a hand and wont let things work that shouldn't, esp. with regard to perms. |
| 17:28.23 | jbalcomb | Dr-Linux perhaps try 775 for dirs and 664 for files |
| 17:28.41 | [TK]D-Fender | FlyboySR22 : Apparently Unlimitel.ca is pretty good |
| 17:28.42 | *** join/#asterisk puzzled (n=yeahrigh@62.45.11.228) |
| 17:28.50 | delmar | jbalcomb, only issues i have with the TDM400 is... to get fax working u need to have rx/txgain tuned right... and if u do that..it can comprimise what tuning u did for echo... doesnt look like u can have it both ways .. not easily... and the other issue i have..might just be the card or PC or hell if I know... but im getting some wird beeds, pips, chirps.. farts.. whatever. :P. as i said earlier.. sounds like a supermarket checkout on a qu |
| 17:28.59 | FlyboySR22 | [TK]D-Fender, Good deal - I will take a look at them - thanks |
| 17:29.02 | Dr-Linux | jbalcomb: well, its test asterisk box, so i set permission 777 .. but .. |
| 17:29.17 | delmar | jbalcomb, so i made this other box here for testing to rule that out. |
| 17:29.20 | [TK]D-Fender | FlyboySR22 : I was about to use them before I decided to use my work as a telco :D |
| 17:29.40 | FlyboySR22 | [TK]D-Fender, :-) |
| 17:29.45 | jbalcomb | delmar gotcha. my analog pbx solves my fax problem for now (i think) so i'm strictly focus on voice call quality. |
| 17:29.54 | delmar | oh and dont use distcc to compile Asterisk stuff. |
| 17:29.56 | delmar | :P |
| 17:30.15 | Dr-Linux | look |
| 17:30.15 | jbalcomb | delmar i dunno what that is. :( why not use it? |
| 17:30.17 | delmar | while distributed compiling is fun.. im not too confident it wasnt the cause of some wierdness i encountered the other day. |
| 17:30.18 | Dr-Linux | [root@I2C-PBX default]# chmod -R a+rw /var/spool/asterisk/voicemail/default/4092/INBOX* |
| 17:30.56 | jbalcomb | Dr-Linux directory = 775; file = 664 |
| 17:30.58 | delmar | jbalcomb, would love to sink a day or two into confirming it but.... i can only suspect distcc was the cause at this stage. |
| 17:31.18 | delmar | jbalcomb, when i get time I will experiment more with it. distcc is important to have working right. speeds things up big time. |
| 17:31.21 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 17:31.47 | jbalcomb | delmar ah, i believe i wont be doing distcc. i assume its not the default method |
| 17:32.23 | delmar | jbalcomb, distcc allows the use of 1 or more remote linux systems to assist with your compile. |
| 17:32.23 | jbalcomb | delmar you have GXP-2000s? |
| 17:32.29 | Dr-Linux | jbalcomb: i have done something else and its working fine now :P |
| 17:32.32 | delmar | jbalcomb, nope. |
| 17:32.41 | jbalcomb | Dr-Linux ok, good. |
| 17:32.49 | Dr-Linux | jbalcomb: but i'll not able to do same at production server |
| 17:32.52 | *** join/#asterisk Dandre (n=Dandre@was59-3-82-236-48-30.fbx.proxad.net) |
| 17:32.53 | Dr-Linux | jbalcomb: you know what i did? |
| 17:33.00 | Dandre | hello all, |
| 17:33.05 | jbalcomb | delmar ok. yeah, i think my dual dual-core 2.8 should manage an asterisk compile alright. ;) |
| 17:33.11 | Dandre | I have a problem with queues, |
| 17:33.16 | jbalcomb | Dr-Linux na-uh |
| 17:33.17 | Dr-Linux | jbalcomb: i did >> [root@I2C-PBX default]# chmod -R 777 /var/spool/asterisk/* |
| 17:33.21 | Dr-Linux | :P |
| 17:33.34 | delmar | jbalcomb, yep. its nice to use when u have alittle p3 or something and wanna do a kernel build or something. |
| 17:33.41 | jbalcomb | Dr-Linux yeah, the executable attribute is real important for directories |
| 17:33.47 | delmar | jbalcomb, distcc burns it up in that case. |
| 17:34.03 | Dr-Linux | jbalcomb: ic |
| 17:34.08 | jbalcomb | delmar haha.. little P3.. I wouldn't even use that for my tftp server. ;) |
| 17:34.13 | cpm | my asterisk itches, too much chili sauce with supper. |
| 17:34.17 | Dr-Linux | jbalcomb: now i need is cron entiry |
| 17:34.29 | Dandre | I have setup a queue with some extensions and I have also prevented in my Dial plan one extension to be called if it is alraedy off hook. |
| 17:34.33 | Nugget | If condition persists, consult your physician. |
| 17:34.41 | cpm | good call. |
| 17:34.43 | delmar | jbalcomb, bah. |
| 17:34.43 | Dandre | But that doesn't work for queues |
| 17:35.03 | jbalcomb | Dr-Linux a better solution im considering is making group asterisk and making the asterisk daemon and the web daeon part of that group |
| 17:35.03 | delmar | anyone here had/seen/played with an IPPH-202 phone? |
| 17:35.19 | Dr-Linux | jbalcomb: sorry, i don't understand |
| 17:35.30 | mrtwister|mobile | anyone know is any driver for CPM cards exist for asterisk? |
| 17:35.39 | mrtwister|mobile | i found, we have some CPM cards |
| 17:35.50 | jbalcomb | delmar bah indeed. i be j/k. im sad for over spending on overkill which stems from lack of understanding. |
| 17:35.55 | cpm | heh |
| 17:36.11 | delmar | jbalcomb, u get that tho. :P |
| 17:36.22 | jbalcomb | Dr-Linux make the asterisk server write the files with a group permission shared by the web app so they can both access the files |
| 17:36.23 | *** join/#asterisk Jetpack (n=plumpdj@dsl-146-159-51.telkomadsl.co.za) |
| 17:36.48 | delmar | jbalcomb, man my gear has really lasted well this round. a few years ago i would have the cutting edge workstation... when P4 2.8 was the top CPU.. I had it... |
| 17:36.51 | delmar | and I still have it :P |
| 17:36.57 | jbalcomb | delmar yeah, i just watched it. we dropped 4K on a new mysql server rather than fix our setup or code. |
| 17:37.29 | delmar | jbalcomb, thats a bit harsh. but sometimes .. for the bean counters.. its a case of .. spend it or loose it. |
| 17:37.36 | jbalcomb | delmar my only itch is planning a box to last three years. |
| 17:37.54 | delmar | jbalcomb, did that with my main server here. |
| 17:38.11 | jbalcomb | delmar /lose it/ is fine for me because i know its going somewhere else in the company and that my dept is operating well. |
| 17:38.28 | *** join/#asterisk chapeaurouge (n=chap@user-85-201-82-146.tvcablenet.be) |
| 17:38.43 | delmar | jbalcomb, i expect it to last 4 years at least. 18months old now and its over kill. runs linux + a vmware gsx virtual host with win2k3 server+exchange+mailmarshal etc. |
| 17:38.52 | jbalcomb | delmar of course, if you have a VERY controlling controller i understand the padding the budget to make sure you have it when you need it |
| 17:39.15 | delmar | jbalcomb, yep. |
| 17:39.17 | *** join/#asterisk Assid (n=assid@203.115.64.13) |
| 17:39.20 | jbalcomb | delmar 3 years is depreciation deadline, losses recouped. |
| 17:39.42 | delmar | 2.4years here i think |
| 17:39.50 | delmar | yeah hardware writes off 100% over 2.4years. |
| 17:39.51 | jbalcomb | canada? |
| 17:39.54 | delmar | NZ |
| 17:39.58 | jbalcomb | wheres that? |
| 17:40.02 | jbalcomb | hahah!!! |
| 17:40.06 | delmar | middle earth |
| 17:40.08 | delmar | :P |
| 17:40.27 | delmar | i actualyl live a little bit south of mt doom :P |
| 17:40.27 | jbalcomb | no really, thats cool. i've heard of you guys. national geographic i think. |
| 17:40.34 | delmar | haha :P |
| 17:40.46 | *** join/#asterisk Seyr (n=Seyr_@pf01.grantgeo.com) |
| 17:41.10 | Seyr | Anyone know how to tell if someone has their phone set DND? |
| 17:41.29 | delmar | Seyr, ring them and ask :P |
| 17:41.42 | Egonis | I am having some severe echo-back on my Sangoma A200, but the person on the other end hears fine, it's just us (the caller) that hears it.. has anyone else experienced this? |
| 17:41.43 | *** part/#asterisk Jetpack (n=plumpdj@dsl-146-159-51.telkomadsl.co.za) |
| 17:41.53 | delmar | Seyr, FXS port or sip phoen client or? |
| 17:41.56 | mrtwister|mobile | Seyr, you mean @home DND commandz? |
| 17:42.15 | _Sam-- | Egonis: i dont have an answer, but am just curious if you got the A200 w/ hardware echo cancellation? |
| 17:42.19 | jbalcomb | delmar haha.. i tried to oder a pizza from some place that delivers pizza all over new zealand. thier damn webpage didn't suppose US address format. |
| 17:42.31 | Egonis | _Sam--: Nope |
| 17:42.47 | Seyr | no, setting DND on a Cisco phone |
| 17:42.52 | Seyr | how can I tell in the dialplan? |
| 17:42.55 | delmar | jbalcomb, oh? was it.. hells pizza? |
| 17:42.57 | Seyr | whats it return? |
| 17:42.57 | _Sam-- | did you mess with the echo settings in zapata.conf? |
| 17:43.03 | jbalcomb | Egonis yeah, a lot with the GXP-2000. bad mic with too much gain. had to turn down txgain a lot. |
| 17:43.06 | jbalcomb | delmar YEAH! |
| 17:43.24 | delmar | jbalcomb, their pizza's fuckin own. |
| 17:43.28 | mrtwister|mobile | Seyr, anyway, you can use varyous notificators and even make extension like exten => s,1, check-dnd-on-off; exten => s,2, background(user-busy-if-you-sure-that-want-reach-press-one-now) ... :) |
| 17:43.33 | jbalcomb | delmar i loved the site. incredible that a business could have that image. |
| 17:43.40 | delmar | jbalcomb, but i learnt how to make my own.. cream cheeze and avocado :P |
| 17:44.05 | jbalcomb | delmar if i wasn't going to be making $900/Hr consulting internationally I would so damn fer sure open Hells Pizza USA. |
| 17:44.10 | [av]bani | ... |
| 17:44.47 | Seyr | mrtwister|mobile: thats if you set the var DND |
| 17:44.47 | delmar | jbalcomb, heh. the "group" are based out of napier and own hells, but also a massive chain of pubs and other eateries.. they are well networked and heavily into VoIP ..networked well with Asterisk themselves :P |
| 17:44.52 | Seyr | thats not from phone :-) |
| 17:44.54 | jbalcomb | [av]bani oh sorry, bani, didn't I set off your child_safe_channel script? |
| 17:45.00 | Dandre | Is there any possibility to define a member of a queue with only his extension? |
| 17:45.29 | tzanger | I tagged my code at point-in-time A with tag BEFORE_I_BEGIN_DEBUGGING |
| 17:45.36 | tzanger | changed a ton of shit but found the bug. |
| 17:45.38 | jbalcomb | delmar haha.. maybe i could consult for them. VoIP and Pizza delivery package |
| 17:46.11 | jbalcomb | hrmm.. PoIP or PizzoIP |
| 17:46.22 | delmar | jbalcomb, their IT guy is a good mate of mine. was almsot engaged to my wifes sister. |
| 17:47.01 | jbalcomb | delmar right on. |
| 17:47.19 | jbalcomb | so you gonna same day air me some Hell's Pizza? |
| 17:47.39 | delmar | jbalcomb, if u are paying.. im sure I can arrange that :P. |
| 17:48.15 | *** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 17:48.17 | delmar | its not hard to make how they make it tho. |
| 17:48.18 | jbalcomb | yeah, i can't consider how much that would be. $500.00 pizza? |
| 17:48.25 | delmar | hrm. |
| 17:48.49 | delmar | well.. probably cant even do it... I mean... customs here wouldnt allow u to make me a pizza and ship it im sure.. all kindsa restrictions. |
| 17:49.19 | delmar | so id have to take u a photo and email it.. that would be the best I could offer.. pure torture tho :P |
| 17:49.35 | jbalcomb | delmar we dont tell them what it is |
| 17:50.01 | fu3 | I remember back in London, Ont.. a pizza place that delivered marijuana with your order |
| 17:50.03 | fu3 | they didnt last too long :/ |
| 17:50.34 | delmar | heh. would need to get them to make it.. not cook it.. snap freeze it.. drive to the airport and hand it to the nice airline going your way.. and get them to put it in the freezer.. then special pickup at the airport there :P |
| 17:50.37 | bronze | "the veggie supremo" |
| 17:50.41 | delmar | it would work im sure... :P |
| 17:50.58 | *** join/#asterisk stoffell (n=stoffell@d5153FE41.access.telenet.be) |
| 17:51.49 | *** join/#asterisk wolfmat99 (n=wolfmat9@209.194.32.60) |
| 17:52.21 | fu3 | well lads.. i think im going to get my head out of this telco business for a day or so. Let things cool down. |
| 17:52.35 | fu3 | ttyl all.. and thanks for the advice |
| 17:52.41 | delmar | fu3, l8r |
| 17:53.02 | *** join/#asterisk iq (n=iq@71-214-4-12.omah.qwest.net) |
| 17:53.12 | iq | Hi all |
| 17:53.15 | webmind | hi |
| 17:53.19 | sevard | I think I'm going to report this Navy spam as phishing. |
| 17:54.38 | iq | I'm trying to install TE110P. When I do 'zap show status' It gives me Digium Wildcard TE110P T1/E1 Card 0 UNCONFIGUR 0 0 0 |
| 17:54.49 | sevard | They ask everything from full name, and street address to phone number, DOB, and GPA |
| 17:54.52 | *** join/#asterisk baahabaa (n=bbabahaa@125.22.72.197) |
| 17:54.56 | *** part/#asterisk Dandre (n=Dandre@was59-3-82-236-48-30.fbx.proxad.net) |
| 17:54.58 | Darwin35 | wow my 533 p3 with 256 megs ram still kicking but 20 calls at 1 time no issue |
| 17:55.13 | Darwin35 | but going to up the ram |
| 17:55.16 | delmar | Darwin35, what codecs ? |
| 17:55.19 | iq | Does it mean not configured properly? |
| 17:55.26 | wolfmat99 | hi all. |
| 17:55.27 | Darwin35 | g729 ilbc adn gsm |
| 17:55.31 | delmar | iq, have u defined the channels etc? |
| 17:55.42 | delmar | Darwin35, what runs mostly tho? |
| 17:55.49 | Darwin35 | g729 |
| 17:55.49 | iq | delmar: I tried. But I dont think i defined them properly. |
| 17:56.06 | delmar | Darwin35, 20calls at once g729? not bad. MMX optimisation enabled too? |
| 17:56.09 | Darwin35 | on asterisk 1.2.4 on freebsd |
| 17:56.14 | wolfmat99 | got a question on a Polycom 501 and a line appearance on multiple units. I want two phones to share the same extension, where can I do this? |
| 17:56.15 | delmar | iq, thats what I reckon. |
| 17:56.17 | Darwin35 | yes |
| 17:56.22 | sevard | freebsd :S |
| 17:56.44 | delmar | Darwin35, not bad at all. |
| 17:57.34 | *** join/#asterisk spatulamaan (n=ggilmore@ip66-107-33-196.z33-107-66.customer.algx.net) |
| 17:57.51 | iq | delmar: should i meanually edit the .conf file or use ztcfg? |
| 17:57.57 | *** join/#asterisk asterisk99 (n=astguy@modemcable169.194-130-66.mc.videotron.ca) |
| 17:58.21 | *** part/#asterisk Seyr (n=Seyr_@pf01.grantgeo.com) |
| 17:58.25 | Darwin35 | wolf exten => 2345,s,1,DIal(SIP/phone1&SIP/phone2|20|tr) |
| 17:59.34 | Darwin35 | Wolfe you have to setup multi sip cridentials |
| 17:59.46 | delmar | iq, u need to define stuff in your /etc/zaptel.conf &/or /etc/asterisk/zapata.conf |
| 17:59.51 | wolfmat99 | Darwin, so this is an extension command, and since I amusing AMP, I should do that there? |
| 17:59.55 | Darwin35 | and link line 1 to the 1st sip cridential on each phone |
| 17:59.58 | delmar | iq, so whats the card attached to? |
| 18:00.22 | iq | delmar: Robbed-bit T1 |
| 18:00.31 | Darwin35 | I dont use am so dont know |
| 18:00.40 | Darwin35 | I use straight asterisk |
| 18:00.41 | iq | delmar: CAS |
| 18:00.44 | *** join/#asterisk steve___ (n=steve@store-fw.porchlight.ca) |
| 18:00.53 | [TK]D-Fender | wolfmat99 ...... |
| 18:00.54 | [TK]D-Fender | ~amp |
| 18:00.56 | jbot | well, amp is NOT supported here! people using it should join #amportal |
| 18:01.08 | delmar | iq, take a look on the wiki there is lots of info on setting up your T1 channels. |
| 18:01.51 | wolfmat99 | understood.. not meaning to offend, just confirming that this is extension information, not phone configuration and I should manage that using the tools I have. |
| 18:01.52 | iq | delmar: I'll go back to wiki now. Would you know why should I go with ISDN T1 instead of CAS T1? |
| 18:02.36 | Darwin35 | delmar next is to get it working in realtime on postgress |
| 18:02.49 | *** join/#asterisk ReD-MaN (i=redman@dhcp-0-2-b3-9a-4a-5b.cpe.quickclic.net) |
| 18:03.03 | delmar | iq. nope. havent had the good fortune to be able to play with such toys as T1 cards. too damn expensive. |
| 18:03.15 | iq | delmar: thanks. :) |
| 18:03.16 | Darwin35 | yes they are |
| 18:03.29 | Darwin35 | I play with the at work but cant afford 1 here |
| 18:03.43 | Darwin35 | boss to cheap to buy me one also |
| 18:03.51 | delmar | yeah. i have 2 channels banks here. no T1 cards :( |
| 18:04.06 | sevard | I'm going to get a couple of channel banks and a TDM400P soon to play with them |
| 18:04.14 | sevard | Hard to set up? |
| 18:04.39 | delmar | sevard, channel banks asside.. easy.. |
| 18:04.55 | delmar | sevard, dont have any T1 cards... so havent got to play with my channel banks. |
| 18:05.05 | Darwin35 | man I work for a voip provider . yet if I had the money I be my own boss |
| 18:05.23 | Darwin35 | what the hell is up with cogent |
| 18:05.46 | rayvd | Cogent is feeling magnificent! |
| 18:05.57 | mikefoo | Darkhalf: who you work for if you don't mind me asking. |
| 18:06.02 | tzanger | I dunno but I caused a routing loop that took down our 100meg cogent feed a few days ago |
| 18:06.36 | Darwin35 | day 3 of issues with cogent nd horriable ping times |
| 18:06.41 | sevard | either of you want to hook me up with a trial voip line with pstn termination to test out? :) |
| 18:08.10 | mikefoo | heh |
| 18:09.52 | sevard | mikefoo: I'd promise to only dial 1-800s |
| 18:11.19 | *** join/#asterisk bronze (n=clark@c-24-91-157-212.hsd1.ma.comcast.net) |
| 18:12.02 | *** join/#asterisk Morex (n=blah@host86-132-151-106.range86-132.btcentralplus.com) |
| 18:12.07 | Morex | Hey folks |
| 18:12.12 | Morex | Got a weird one for you |
| 18:12.23 | sevard | eww |
| 18:12.29 | *** join/#asterisk trelane_ (n=trelane@209.43.90.13) |
| 18:12.32 | Morex | I've got * hooked up to an Avaya InDEX legacy switch over E1 PRI |
| 18:12.38 | Morex | All working fine EXCEPT |
| 18:13.15 | Morex | When launching a call out through Asterisk, the first 30 keypresses on the phone result in silence. Only after the 31st are DTMF tones actually produced/received |
| 18:13.23 | Morex | Consequently, my client can't dial extensions... |
| 18:13.42 | Morex | Their Avaya consultant is swearing blind it must be something to do with * |
| 18:13.46 | Morex | Any suggestions? |
| 18:13.54 | mikefoo | sevard: we only have on-net customers that we have private t1's to. |
| 18:14.05 | sevard | mikefoo: sad :( |
| 18:14.14 | mikefoo | yah.. b2b voip |
| 18:14.15 | russellb | Morex: consistently 30? |
| 18:14.30 | iq | Any idea what do we set 'dchan' to when using CAS/Robbed-bit T1? |
| 18:14.34 | Darwin35 | back ltr with new hardware... |
| 18:14.38 | Darwin35 | and haircut |
| 18:14.47 | sevard | makeup too. |
| 18:15.16 | brad_mssw | Darwin35: still looking for a place to put an east coast server |
| 18:15.20 | delmar | right. just realised the sun is up so im off to hide in my coffin with the wife. |
| 18:15.24 | delmar | catch u all later. |
| 18:15.32 | jbalcomb | [TK]D-Fender why are some of my asterisk config files set to executable? |
| 18:16.56 | *** join/#asterisk robin_sz (n=yeah@host-212-18-247-190.static.mailbox.co.uk) |
| 18:17.00 | robin_sz | meep? |
| 18:17.18 | robin_sz | ok, so ... how do I dial from the console on a particular sip acccount? |
| 18:17.24 | robin_sz | like one acount is 8001 |
| 18:17.38 | robin_sz | dial 004412345678@8001 doesnt do it ... |
| 18:17.53 | robin_sz | dial 004412345678/sip/8001 ... nop |
| 18:18.01 | robin_sz | sigh ... brain fade |
| 18:19.10 | iq | robin_sz: did you tri 'dial 8001' ? |
| 18:19.38 | robin_sz | ummm ... no |
| 18:19.51 | robin_sz | because that wont work |
| 18:19.56 | iq | robin_sz: I'm not sure if thats what u r trying to do. but try it anyways |
| 18:20.22 | robin_sz | its OK, it not what I ma trying to do .. that would dial extension 8001 in the "local" context |
| 18:21.12 | iq | any idea what does this mean: line 0: Unable to open master device '/dev/zap/ctl' |
| 18:21.53 | robin_sz | it means you didnt do "insmod zaptel" probably |
| 18:23.07 | iq | robin_sz: you mean insmod wcte11xp |
| 18:23.42 | [TK]D-Fender | jbalcomb : They shouldn't be. unless maybe you shared them through Samba or something.... |
| 18:24.13 | robin_sz | sigh .. I just need to check these 6 sip accounts are working ok before trying to route cllas down them for "real" |
| 18:24.32 | [TK]D-Fender | jbalcomb : IIRC you only e-mailed them to for my evaluation for "mass-renovation" which we have not followed up on, so I am certainly not involved in their current state. |
| 18:24.34 | jbalcomb | [TK]D-Fender ok, probably some misshap managed by the previous admin. |
| 18:24.40 | luke-jr_ | What might be responsible for Asterisk just stopping in the middle of a dialplan? |
| 18:24.54 | [TK]D-Fender | luke-jr_ : Hanging up would do it :) |
| 18:25.24 | luke-jr_ | [TK]D-Fender: it's during routing the call initially =p |
| 18:25.43 | luke-jr_ | it gets into macro route-home, sets the argument variables, and just stops |
| 18:25.55 | [TK]D-Fender | luke-jr_ :pastebin it. (the call & the code) |
| 18:25.56 | jbalcomb | [TK]D-Fender i do understand. i am still working on the new system. as soon as that is purchased and set up i will contact you further regarding the complete overhaul of our configuration. |
| 18:26.01 | luke-jr_ | that's from an incoming call; a local call to route-home works |
| 18:26.23 | robin_sz | sigh ... dial sip/8001/0044123456789 doesnt work either |
| 18:26.31 | [TK]D-Fender | jbalcomb : Door is still plenty open. I'm surprised you are still going through the process of the upgrade though. Its been a while now... |
| 18:26.59 | *** join/#asterisk fulgas (n=fulgas@82.102.2.254) |
| 18:27.29 | luke-jr_ | [TK]D-Fender: http://pastebin.ca/42913 |
| 18:28.27 | *** join/#asterisk ke4qqq (n=chatzill@68-115-212-155.static.spbg.sc.charter.com) |
| 18:29.13 | luke-jr_ | http://pastebin.ca/42914 for the 'disassembly' too |
| 18:31.16 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:32.24 | *** join/#asterisk kpettit (n=keith@69.15.174.114) |
| 18:33.17 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 18:35.15 | [TK]D-Fender | AEL... ick.... |
| 18:35.22 | luke-jr_ | ick? |
| 18:35.39 | luke-jr_ | anyway, from the disassembly, it looks like the AEL rendering isn't at fault |
| 18:36.25 | jbalcomb | [TK]D-Fender yes, i have the PRI card on my desk but the PO for the server has not been approved |
| 18:37.49 | *** join/#asterisk _deg_ (n=deg@200.163.193.247) |
| 18:37.51 | Egonis | what's the diff between voicemail.conf voicemail.conf.save, etc? how do I set it up? |
| 18:38.05 | robin_sz | ick |
| 18:38.46 | sevard | Egonis: pretty sure voicemail.conf.save is a backup of your voicemail.conf |
| 18:38.53 | luke-jr_ | Egonis: I think *.save is what happens when you terminate an editor the wrong way |
| 18:39.06 | Egonis | sevard: b/c when someone tries to change their pin for vm, it tries to save to voicemail.conf.save |
| 18:39.15 | [av]bani | \o> |
| 18:39.17 | [av]bani | <o/ |
| 18:39.31 | sevard | heh. |
| 18:40.07 | Egonis | is that a config option I should change? |
| 18:41.02 | [TK]D-Fender | luke-jr_ : just not sure... |
| 18:41.16 | [TK]D-Fender | jbalcomb : Which card? |
| 18:41.22 | jbalcomb | te411p |
| 18:41.22 | luke-jr_ | [TK]D-Fender: it works fine from a local phone |
| 18:45.51 | *** join/#asterisk paulhuynh (n=paul@c-68-37-18-82.hsd1.de.comcast.net) |
| 18:46.20 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 18:46.52 | *** join/#asterisk justinu (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 18:46.53 | *** join/#asterisk krischnoff (n=chatzill@209.77.205.12) |
| 18:46.58 | paulhuynh | help me please i need a little asterisk help with the sip.conf |
| 18:47.03 | paulhuynh | this is what i have |
| 18:47.04 | luke-jr_ | does Asterisk have a max macro depth or something? |
| 18:48.01 | paulhuynh | asterisk 2 nic. Nic one go to our t1 static IP, Nic 2 go to our office router 192.168.1.254/255.255.255.0 but no gateway was set on it |
| 18:48.27 | justinu | just as I was plugging my new covad circuit in, the cable internet went down again |
| 18:48.28 | justinu | lol |
| 18:48.48 | paulhuynh | http://pastebin.ca/42917 |
| 18:50.05 | paulhuynh | i have 6 phone from with in the office and want those phone to use internal lan to talk to asterisk |
| 18:51.14 | paulhuynh | sip provider complain that http://pastebin.ca/42918 |
| 18:52.39 | shido6 | zZZzZZ |
| 18:52.48 | shido6 | whats wrong paulhuynh |
| 18:53.40 | paulhuynh | i need a correct setup for my sip.conf |
| 18:54.12 | paulhuynh | provider can't delievery the call to us because mu asterisk register with two ip instead of on |
| 18:54.14 | paulhuynh | one |
| 18:55.28 | *** join/#asterisk linville (n=linville@nat-pool-rdu.redhat.com) |
| 18:58.17 | *** join/#asterisk lalito (n=erg@201.102.4.195) |
| 18:58.28 | *** join/#asterisk rdlang (n=test@s55936e2e.adsl.wanadoo.nl) |
| 18:58.47 | rdlang | any dutch ppl here? |
| 18:59.30 | paulhuynh | any one |
| 18:59.35 | paulhuynh | to help me here? |
| 18:59.38 | rayvd | my family moved from holland in the 1700's does that count? |
| 19:00.28 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 19:00.47 | robin_sz | i visted Holland 20 years ago .. does that count? |
| 19:00.54 | Nugget | Ik ben een vliegende koe. |
| 19:01.50 | rdlang | hehehe |
| 19:01.57 | Nugget | boe boe! |
| 19:02.00 | paulhuynh | help |
| 19:02.02 | paulhuynh | anyone? |
| 19:02.04 | paulhuynh | please |
| 19:02.18 | rdlang | i am looking for someone with experience in connecting an E1 line to a TE110p digium card |
| 19:02.23 | paulhuynh | i just need alittle help with asterisk sip.conf |
| 19:04.27 | paulhuynh | is anyone here running asterisk with two nic? |
| 19:04.32 | *** join/#asterisk SplasPood (n=jwb@gate.lga2.us.voxel.net) |
| 19:04.33 | Nugget | I am |
| 19:04.40 | paulhuynh | nugget |
| 19:04.45 | paulhuynh | can you help me please |
| 19:04.56 | Nugget | Not unless you actually ask a damn question, no. |
| 19:05.07 | paulhuynh | OK |
| 19:05.10 | paulhuynh | sorry |
| 19:05.22 | *** join/#asterisk NovceGuru (n=asdf@oh-71-53-48-11.dhcp.sprint-hsd.net) |
| 19:05.35 | rdlang | noone here with E1 experience? |
| 19:05.37 | paulhuynh | I have t1 go to the first nic cable-router-asterisk2nic |
| 19:05.53 | paulhuynh | the cable-router-phone as well |
| 19:06.22 | *** part/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it) |
| 19:06.28 | paulhuynh | i just want asterisk to use the first nic to talk to the outside and 2 nic to the lan so that the internal phone can work with it |
| 19:06.33 | *** join/#asterisk kpettit (n=keith@69.15.174.114) |
| 19:06.39 | Nugget | sounds like your setup is almost exactly like mine. I've got two NICs in my asterisk server, one on my public network (with a "real" IP) and one on my private network (with an RFC1918 IP). |
| 19:06.41 | paulhuynh | so what do you thikn nugget |
| 19:06.56 | paulhuynh | yup |
| 19:06.59 | rdlang | any more asterisk ir channels i can visit? preferably dutch orientated |
| 19:07.07 | paulhuynh | second nic is 192.168.1.254 |
| 19:07.11 | Nugget | I've got phones that talk SIP to the server on the private space, and IAX peers (my voip providers) that talk over the "real" IP |
| 19:07.21 | paulhuynh | yup |
| 19:07.26 | paulhuynh | that is what i want |
| 19:07.26 | Nugget | I've never had any problems running that way, and I don't recall having had to do anything special to make it work |
| 19:07.43 | paulhuynh | did you see my current sip.conf that i have posted |
| 19:07.53 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 19:08.27 | Nugget | I guess the big difference is that I use an IAX provider, not a SIP provider. |
| 19:08.28 | paulhuynh | <PROTECTED> |
| 19:08.36 | Nugget | I'm not doing any registering to SIP peers |
| 19:09.12 | paulhuynh | i didn't even include the gateway address for the second nic setup to insure it doesn't go out that way |
| 19:09.53 | Nugget | perhaps try setting externip= to your external IP. |
| 19:10.18 | paulhuynh | then what happen to all my internal phone woudl it still work? |
| 19:10.21 | *** part/#asterisk pointer (i=pointer@aj.catt.com) |
| 19:10.25 | Nugget | yes |
| 19:10.36 | justinu | i remember hearing about problems with using asterisk in a multiple network interface system |
| 19:11.00 | justinu | like asterisk will take the IP from the wrong interface, and tell the SIP phones to contact it on that interface |
| 19:11.13 | Nugget | I know that's the case for aliases, but not for separate NICs. |
| 19:11.16 | justinu | ah |
| 19:11.37 | Nugget | but I would say that this sort of use is not very well-tested or explored. |
| 19:11.47 | justinu | i don't have the details, but I remember anthm telling me about it |
| 19:11.51 | luke-jr_ | interesting |
| 19:11.57 | luke-jr_ | Zapateller(nocallerid|answer) doesn't work |
| 19:12.18 | *** join/#asterisk vgster (n=vg@spc1-ledn1-3-0-cust194.seac.broadband.ntl.com) |
| 19:12.30 | paulhuynh | i did that and all phone in the office which talk to asterisk via 192.168.1.254 stop working |
| 19:12.40 | Nugget | paulhuynh: make sure you also define a localnet= |
| 19:12.42 | justinu | make sure localnet are set |
| 19:12.56 | paulhuynh | how do you do that |
| 19:13.01 | justinu | in sip conf |
| 19:13.03 | paulhuynh | can you gave me an example? |
| 19:13.06 | justinu | uncomment the example lines |
| 19:13.10 | Nugget | there's an example in sip.conf |
| 19:13.15 | justinu | localnet=192.168.0.0/255.255.0.0 |
| 19:14.15 | paulhuynh | i get it |
| 19:14.37 | paulhuynh | cool |
| 19:15.30 | paulhuynh | never mind that didn't work |
| 19:15.37 | Nugget | dang |
| 19:16.02 | justinu | can you pastebin the entire SIP dialog from asterisk to phone? |
| 19:16.05 | *** join/#asterisk JCC_ (n=john@207.41.92.131) |
| 19:16.06 | paulhuynh | do i need to add the local net stuff to all the phone that inside the local network (there sip.conf) |
| 19:16.11 | justinu | nop |
| 19:17.01 | paulhuynh | http://pastebin.ca/42925 |
| 19:17.31 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
| 19:17.42 | justinu | that's not what I asked for |
| 19:18.23 | paulhuynh | oh ok |
| 19:18.23 | paulhuynh | i know what you mean let me correct it real quick |
| 19:18.35 | justinu | what I want is the entire output of sip debug |
| 19:18.42 | justinu | from invite to bye |
| 19:19.23 | paulhuynh | so i correct the local net stuff to localnet=192.168.0.0/255.255.0.0 |
| 19:19.35 | paulhuynh | but that didn't help |
| 19:19.36 | justinu | that shouldn't matter |
| 19:19.47 | paulhuynh | let me try sip debug |
| 19:20.24 | iCEBrkr | oh hell. |
| 19:20.44 | iCEBrkr | Is there a query length limit in the func_odbc stuff? Kinda looks as if there is one. |
| 19:20.57 | Qwell[] | iCEBrkr: 512 |
| 19:21.03 | Qwell[] | I had to expand it... |
| 19:21.09 | iCEBrkr | <PROTECTED> |
| 19:21.09 | iCEBrkr | <PROTECTED> |
| 19:21.11 | iCEBrkr | grr |
| 19:21.17 | iCEBrkr | Qwell[]: hehe, yea, I see that now :) |
| 19:21.17 | Qwell[] | there are about 6 places, iirc |
| 19:21.20 | iCEBrkr | ok |
| 19:21.33 | *** join/#asterisk websae (n=websae@h69-129-132-18.69-129.unk.tds.net) |
| 19:21.43 | Qwell[] | bug Corydon-w to "fix" it in trunk :P |
| 19:21.54 | paulhuynh | justinu |
| 19:22.01 | iCEBrkr | Qwell[]: he doesn't like it when I bug him :) |
| 19:22.06 | Qwell[] | indeed |
| 19:22.20 | paulhuynh | what do i type into asterisk -r for it to just debug ext 104 |
| 19:22.24 | Corydon-w | Why the hell do you have SQL queries bigger than 512 bytes? |
| 19:22.37 | Corydon-w | especially in the dialplan? |
| 19:22.40 | Qwell[] | Corydon-w: I Already showed you mine.. |
| 19:22.47 | iCEBrkr | Corydon-w: joins? |
| 19:22.49 | Qwell[] | perfectly valid |
| 19:22.49 | rayvd | many joins! |
| 19:22.56 | Corydon-w | Qwell[]: yes, but you're demented |
| 19:22.57 | iCEBrkr | tablename.column name. |
| 19:22.59 | Qwell[] | ;] |
| 19:23.04 | justinu | paulhuynh: sip debug peer 104 |
| 19:23.07 | Qwell[] | 512 really isn't hard to hit |
| 19:23.12 | Corydon-w | iCEBrkr: use table aliases |
| 19:23.26 | jbalcomb | why are there 0 decent ring tones out there |
| 19:23.27 | iCEBrkr | Corydon-w: oh come'on make it 1024 :) |
| 19:23.28 | rayvd | you could use mysql 5.x and use a stored procedure :) |
| 19:23.29 | *** join/#asterisk Eggplant (i=No@208.193.217.216.cascadeaccess.com) |
| 19:23.37 | Corydon-w | Besides, I'm in the middle of updating Directory |
| 19:23.53 | Qwell[] | don't make me email you a patch :p |
| 19:23.55 | iCEBrkr | Corydon-w: Then finish Directory, which is more important!! |
| 19:23.56 | [TK]D-Fender | jbalcomb : What do you mean? I can sample anything for mine :) I currently use the Cisco "24" ring..... |
| 19:24.20 | *** join/#asterisk Utah_Dave (n=boucha@0-3pool128-60.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 19:24.21 | [TK]D-Fender | jbalcomb : Or maybe I can sign you up on the mailing lists that keep offering me ringtones ;) |
| 19:24.29 | jbalcomb | [TK]D-Fender aw man, that sounds nice. you got the bin files for the GXP-2000 or maybe just wav files you could email me? |
| 19:26.39 | paulhuynh | Registered SIP '104' at 192.168.1.101 port 5060 expires 50 |
| 19:26.39 | *** join/#asterisk Dr-Linux (n=Dreamer@host202-147-168-130.lhr.dancom.net.pk) |
| 19:26.39 | paulhuynh | that is when i havd bindaddress= 0.0.0.0 and no localnet stuff |
| 19:26.39 | *** part/#asterisk austinnichols101 (n=austinni@70.46.69.131) |
| 19:26.54 | paulhuynh | else i get can't get ip address from peer 104 |
| 19:31.11 | iCEBrkr | Damnit, I suppose I can just SELECT * |
| 19:31.35 | paulhuynh | justinnu |
| 19:31.55 | paulhuynh | can you look at the pastebin really quick and tell me what happen |
| 19:32.17 | paulhuynh | http://pastebin.ca/42929 |
| 19:32.29 | [TK]D-Fender | jbalcomb : I have CCITT ULAW & MP3 if you like and you can then remix as needed. |
| 19:33.35 | jbalcomb | [TK]D-Fender that would be swell, in a zipped archive perhaps. you still have my address? |
| 19:33.59 | [TK]D-Fender | jbalcomb : ULAW is the only one I have handy. Will that do? |
| 19:34.16 | mzo | someone fix my FWD? please? =( |
| 19:34.24 | jbalcomb | iCEBrkr yeah, do that SELECT * and then use PHP to parse the actual data. |
| 19:34.26 | *** join/#asterisk skkip (n=Skipper@216.160.91.91) |
| 19:34.45 | jbalcomb | iCEBrkr you know, like back at MIS, when I had to kill bogus queries and slap programmars. |
| 19:35.00 | jbalcomb | [TK]D-Fender yeah, that'd be just fine |
| 19:35.03 | iCEBrkr | *eye roll* |
| 19:35.11 | jbalcomb | haaha |
| 19:35.12 | paulhuynh | can someoine explain to me with my sip debug and tell me why i can recieve call from my sip people |
| 19:35.15 | paulhuynh | http://pastebin.ca/42929 |
| 19:35.38 | iDunno | if you can receive a call, what's the problem? |
| 19:35.38 | jbalcomb | SELECT * FROM * WHERE * = * |
| 19:35.50 | paulhuynh | FUNNY |
| 19:35.55 | Qwell[] | no, no, no |
| 19:36.00 | iDunno | jbalcomb: that's a very broken query. |
| 19:36.01 | paulhuynh | sorry bad english can not recieve call |
| 19:36.02 | Qwell[] | WHERE * LIKE %* |
| 19:36.02 | iCEBrkr | jbalcomb: I had more fun when Ron spammed 25,000 emails and killed the email server. |
| 19:36.32 | jbalcomb | iCEBrkr oh yeah! umm.. wasn't part of his /persona/ endeavor? |
| 19:36.57 | iCEBrkr | jbalcomb: I dunno, but I really really wanted to walk over there and act like him saying "Who do I have to fire?" |
| 19:37.15 | iCEBrkr | jbalcomb: It was always funny when HE screwed shit up, but when we DID it. It was the end of the world. |
| 19:37.36 | paulhuynh | anyone have a min to look at the sip debug yet? |
| 19:37.39 | jbalcomb | iCEBrkr yeah, i kinda just wanted to straight bitch slap him and not actually discuss the matter |
| 19:38.28 | jbalcomb | iCEBrkr 8:00, just back from vacation, "You need to fix the mail server!" |
| 19:39.01 | websae | an yone know a good 800 DID provider for us 48 and canada? |
| 19:39.06 | websae | and maybe mexico |
| 19:39.48 | [TK]D-Fender | jbalcomb : Sent |
| 19:40.03 | jbalcomb | [TK]D-Fender right on, you rawk. thanks. |
| 19:41.05 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-225-183.claranet.co.uk) |
| 19:41.11 | paulhuynh | help please |
| 19:41.18 | paulhuynh | here are my pastebin |
| 19:41.37 | paulhuynh | <PROTECTED> |
| 19:43.45 | *** join/#asterisk crich1999 (n=crich@port-212-202-0-5.dynamic.qsc.de) |
| 19:43.50 | JCC_ | Is anyone here able to comment on pros and cons of asterisk with the ZyXEL Prestige 2000W v2 wireless sip phone? Thanks. |
| 19:43.52 | LostFrog | I wish all three of my phones would break for a couple of days. |
| 19:45.07 | [TK]D-Fender | JCC_ : All Wi-fi SIP phones are "toys" these days. Most public hot-spots may require HTTP to set up access and their range/battery life is normally deplorable. If you are thinking home/office use, go with an ATA |
| 19:46.23 | justinu | agreed, but apparently the hitachi wifi sip device is slightly better |
| 19:46.25 | luke-jr_ | [TK]D-Fender: I imagine they could be of use as a replacement for wireless analog phones |
| 19:46.39 | websae | JCC: it sucks |
| 19:46.43 | *** join/#asterisk kpettit (n=keith@69.15.174.114) |
| 19:46.44 | justinu | the ZyXEL phones have poor battery life and configuration sucks |
| 19:46.47 | websae | JCC: I returned mine |
| 19:46.47 | tzanger | has anyone used 2 or more polycom phones (501) through NAT to a * box? |
| 19:46.54 | justinu | it works, but it's not a very nice solution |
| 19:46.59 | [TK]D-Fender | luke-jr_ : possible, but not cost-effective |
| 19:47.03 | websae | get a cisco |
| 19:47.07 | justinu | tzanger: yep |
| 19:47.08 | websae | that's only $500 ish |
| 19:47.12 | luke-jr_ | [TK]D-Fender: are any digital phones cost-effective? |
| 19:47.20 | [TK]D-Fender | tzanger : You're in trouble because Polycom didn't plan their phones with NAT in mind.... |
| 19:47.20 | justinu | the gxp2000 is cost effective |
| 19:47.28 | [TK]D-Fender | luke-jr_ : not that I've seen. |
| 19:47.31 | tzanger | justinu: how did you work with NAT? I don't see any way to set RTP port ranges or anything |
| 19:47.32 | websae | and works nice...i have the gxp2000 |
| 19:47.38 | tzanger | [TK]D-Fender: yeah that's what I thought |
| 19:47.41 | JCC_ | well... those are some pretty definitive comments :-) I was thinking of replacing my SP3K in the SOHO, but I think not. Thanks. |
| 19:47.45 | justinu | tzanger: i haven't had any issues really |
| 19:48.00 | tzanger | justinu: ?? no special config or port forwarding?? |
| 19:48.01 | justinu | tzanger: i have an office behind NAT that connects to a hosted * machine |
| 19:48.08 | justinu | they run about 10 501s |
| 19:48.10 | luke-jr_ | justinu: how so? |
| 19:48.18 | tzanger | weird |
| 19:48.21 | tzanger | which router? |
| 19:48.21 | justinu | nat=yes in asterisk |
| 19:48.28 | justinu | linux firewall/NAT |
| 19:48.28 | luke-jr_ | re cost effectiveness |
| 19:48.37 | justinu | luke-jr_: they're 4 line phones for $85 |
| 19:48.45 | tzanger | justinu: any special NAT helpers? |
| 19:48.47 | [TK]D-Fender | justinu : And what about ports? |
| 19:49.12 | justinu | tzanger: nope... the phone registers with *, and the qualify keeps the nat bindings open |
| 19:49.26 | luke-jr_ | justinu: can only use 1 line at a time... |
| 19:49.29 | justinu | canreinvite=no, of course |
| 19:49.36 | |Vulture| | anyone know why * wont let users change VM passwords? I have voicemail.conf set to 777 for testing |
| 19:49.37 | luke-jr_ | why should a digital phone have a line limit? |
| 19:49.41 | tzanger | yeah but a call would want to allocate 2 RTP ports that the firewall wouldn't have any clue about |
| 19:49.47 | mzo | i tried the xyzel phone nad it's a huge POS |
| 19:49.55 | mzo | it sits in the charger all day, because I would never use that |
| 19:49.56 | justinu | tzanger: symmetric RTP |
| 19:50.01 | tzanger | justinu: ? |
| 19:50.04 | |Vulture| | ah nvm |
| 19:50.29 | justinu | tzanger: http://www.voip-info.org/wiki-RTP+Symmetric |
| 19:50.43 | justinu | the phone sends RTP to * on the same port combination it expects to receive from * |
| 19:50.47 | justinu | so the NAT binding is taken care of |
| 19:51.05 | justinu | anyways, for my setup "it just works" |
| 19:51.08 | tzanger | hmm |
| 19:51.32 | justinu | but i don't use any of the NAT settings on the phones |
| 19:51.36 | justinu | (what little there is) |
| 19:51.46 | justinu | i let asterisk take care of all the SDP/SIP IP rewriting |
| 19:51.50 | tzanger | crazy... and that just works |
| 19:51.53 | justinu | yep |
| 19:52.02 | tzanger | all you have in sip.conf is nat=yes and (can)reinvite=no |
| 19:52.05 | justinu | right |
| 19:52.11 | tzanger | do you have a sample sip.conf entry I can base mine off of? |
| 19:52.14 | justinu | sure |
| 19:53.11 | *** join/#asterisk apardo (n=apardo@87.218.44.213) |
| 19:53.34 | justinu | http://pastebin.ca/42930 |
| 19:53.58 | tzanger | justinu: that seems WAY too simple |
| 19:54.00 | tzanger | :-) |
| 19:54.22 | justinu | i know |
| 19:54.29 | justinu | but I swear it works |
| 19:54.36 | justinu | my * box is on a public IP |
| 19:54.54 | justinu | no devices in front of it, so it's a single NAT, which makes life easier |
| 19:55.35 | *** join/#asterisk apardo (n=apardo@87.218.44.213) |
| 19:55.53 | tzanger | yeah |
| 19:56.27 | tzanger | on the polycom you just have the "line" configuration done then? Nothing in outbound proxy/server1/2/etc.? |
| 19:57.51 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 19:59.05 | tzanger | wow it does owrk as simply as you say |
| 19:59.13 | justinu | heh |
| 19:59.14 | tzanger | I am quite beside myself |
| 20:02.25 | *** join/#asterisk iamjeff (n=iamjeff@jumpgate2.kingston.net) |
| 20:04.01 | iamjeff | hi all, can someone lead me in the direction for documentation on call queues. I have read through the voip-info pages but I am still running into problems. :P |
| 20:07.13 | fugitivo | what problems_ |
| 20:07.46 | Hymie | anyone here using hardware sip phones? and recommendations? |
| 20:07.48 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 20:07.48 | Hymie | as in, ethernet-built-in sip phones |
| 20:08.04 | fugitivo | polycom |
| 20:08.10 | GerbilNut | we've got some snom 360s, no complaints |
| 20:08.29 | Hymie | 360s? GerbilNut: who makes them? |
| 20:08.34 | GerbilNut | Snom |
| 20:08.37 | Hymie | hmm |
| 20:08.38 | iamjeff | i am setting up two basic call queues, one for support and one for billing. I have the agents log in (that is working), but when a caller enters the queue, they get directed to the voicemail of the logged in user, not the voicemail for the support queue. |
| 20:08.48 | Hymie | GerbilNut: looking for em on google now... |
| 20:09.00 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.cust.bezeqint.net) |
| 20:09.03 | fugitivo | Hymie: polycom |
| 20:09.12 | GerbilNut | iamjeff, paste your extension.conf, agents.conf, and queues.conf to pastebin.ca and let us look at them |
| 20:09.26 | GerbilNut | remove whatever password or personal info you don't want there |
| 20:09.29 | Hymie | fugitivo: thanks |
| 20:09.30 | Hymie | fugitivo: take a gander at that next |
| 20:09.40 | iamjeff | Hymie: we are using Polycom 300/500/600's, plus cisco 7960's. Both work great. We tried the Snom 190's but they are just not enough for our business needs. |
| 20:10.39 | Hymie | iamjeff: thanks... I take it that the polycoms have transferm, etc buttons built in? |
| 20:11.28 | Hymie | actually, it doesn't have to be sip, just asterisk.. so they could be IAX... the client asked for SIP.. but it may be a case of people just thinking that sip is what they need ;) |
| 20:11.40 | GerbilNut | Hymie, http://www.abptech.com/mainpages/products/snom360.html |
| 20:13.08 | Hymie | GerbilNut: thanks, looks featureful |
| 20:13.09 | justinu | the ip301 has no transfer button :( |
| 20:13.18 | justinu | you have to use the softbuttons to do it |
| 20:13.33 | justinu | kinda disappointed with that |
| 20:13.41 | iamjeff | Hymie: the 300 does not, the 500 & 600's have hard buttons. |
| 20:13.46 | [TK]D-Fender | iamjeff : If you're using agen callbacklogin be careful to not use an exten with VM of its own atached to it... |
| 20:13.57 | Hymie | iamjeff: cool |
| 20:14.05 | [TK]D-Fender | justinu : I only use the soft-key on my IP60x's :) so.... |
| 20:14.17 | iamjeff | GerbilNut: i am just cleaning the file and will post them. Should I post 3 separate files or one big? |
| 20:14.20 | justinu | yeah, but your 601 displays the transfer softkey on the display always |
| 20:14.26 | justinu | the 301, you have to press "more" |
| 20:14.33 | [TK]D-Fender | justinu : "OH" |
| 20:15.20 | [TK]D-Fender | in call it should have xfer & conference all the time as the primary options since it has a fixed hold key. |
| 20:15.32 | [TK]D-Fender | justinu : What are the default softkeys in call? |
| 20:15.42 | *** join/#asterisk redder86 (n=lee@gateway.howardsilvan.com) |
| 20:15.43 | GerbilNut | iamjeff, seperate them with ------'s and you can do one |
| 20:16.01 | redder86 | stevek: are you there? |
| 20:17.07 | *** join/#asterisk ceeto (i=cio@adsl-072-149-159-016.sip.bhm.bellsouth.net) |
| 20:17.11 | GerbilNut | Hymie, very featurefull, i like them |
| 20:17.52 | justinu | d-fender: i'll take a look at that |
| 20:19.17 | justinu | i've still got this nagging issue where I try to make an outbound call from the polycom, and the phone reboots shortly after |
| 20:19.23 | justinu | can't figure out how to reproduce that |
| 20:19.39 | ceeto | Hi all. Just ran into an interesting problem and I think I know the solution but wanted to get some opinions. We have an 8-line asterisk box (1.7/zap1.9) and two 4-port TDM400P's to handle 8 inbound pots lines. We use the inbound faxing on all channels with spandsp/etc. It has been failing randomly and I thought that the implementation was just flakey. Come to find out, if no lines are in use, the faxes come through everytime. |
| 20:19.42 | justinu | i think it might have something to do with the MGCP dial string |
| 20:20.04 | justinu | 1.7/1.9? |
| 20:20.12 | justinu | we're on asterisk 1.2.4 and zap 1.2.4 |
| 20:20.25 | ceeto | Yea, but not everybody upgrades just for the sake of ugprading... |
| 20:20.32 | ceeto | It's a great working system, why mess with it. |
| 20:20.33 | justinu | do you mean 1.1.7? |
| 20:20.37 | ceeto | Yea, sorry. |
| 20:20.37 | ceeto | heh |
| 20:20.38 | justinu | k |
| 20:20.39 | iamjeff | GerbilNut: take a look at wtc-extensions.conf |
| 20:20.39 | ceeto | :( |
| 20:20.44 | fugitivo | ceeto: try zaptel 1.0.8 |
| 20:20.53 | *** join/#asterisk Z_God (n=Z_God@jabber.xs4all.nl) |
| 20:20.54 | justinu | i'm not one of the upgrade for the same of upgrading guys, just trying to understand what you're running |
| 20:20.57 | iamjeff | the agents and queue are tucked at the bottom of the file. |
| 20:20.58 | justinu | s/same/sake/ |
| 20:20.58 | ceeto | We're running zap 1.0.9 right now, downgrade? |
| 20:21.13 | GerbilNut | iamjeff, where? i see no link |
| 20:21.13 | ceeto | heh, got ya!!! :) |
| 20:21.35 | iamjeff | http://pastebin.ca/42933 |
| 20:21.41 | Z_God | is there a config guide anywhere on how to configure Asterisk for a SIP conference room? |
| 20:21.41 | justinu | ceeto: if there's any break in audio on your fax calls, the faxes give up |
| 20:21.43 | Hymie | heh, polycom doesn't know how to design web pages ;) |
| 20:21.51 | *** join/#asterisk TallAndy (n=dfa@81-178-201-109.dsl.pipex.com) |
| 20:21.52 | ceeto | sfugitivo: Are you suggestion a downgrade from 1.0.9 to 1.0.8 to solve the issue? |
| 20:22.03 | justinu | ceeto: so maybe an IRQ conflict |
| 20:22.05 | ceeto | justinu: yea, I figured that. Do you think it's wiring related? |
| 20:22.07 | Hymie | Z_God: eh? a conference room is a conference room, SIP or not |
| 20:22.21 | fugitivo | ceeto: you can try, i have a working box with fax with 1.0.8, but the tdm400 is known to have problems with faxing |
| 20:22.23 | Hymie | Z_God: there were issues that I saw with 1.2.2 and 1.2.0 with Asterisk and Meetme |
| 20:22.32 | Hymie | Z_God: but they seem to be gone with 1.2.4 |
| 20:22.36 | justinu | ceeto: doubtful... what terminates the faxes? |
| 20:22.37 | Z_God | yeah, I've already defined 1234 as a conference room I think |
| 20:22.38 | ceeto | justinu: Well, I don't see how an IRQ conflict could cause crosstalk (if that's what it is)... I'll try that, though, thanks! |
| 20:22.39 | justinu | spandsp? |
| 20:22.49 | Z_God | but now I just need the sip config |
| 20:22.52 | justinu | ceeto: not cross talk, but small (i mean less than 50ms) breaks in the audio |
| 20:22.55 | ceeto | justinu: sxfax.o, rxfax.o |
| 20:22.56 | justinu | portions of silence |
| 20:23.06 | Z_God | I've got: |
| 20:23.06 | Z_God | [1234] |
| 20:23.06 | Z_God | type=peer |
| 20:23.06 | Z_God | now |
| 20:23.14 | justinu | fax protocols use those periods of silence to denote phases of the call |
| 20:23.20 | justinu | pretty stupid, but that's how it works |
| 20:23.34 | Z_God | I just want any SIP user to be able to call 1234@myserver and be in my conference room then |
| 20:23.39 | Hymie | Z_God: you need to setup a sip account, add it to your dialplan, so it can reach the meetme |
| 20:23.39 | ceeto | Ahh, got ya. interesting. |
| 20:24.07 | justinu | ceeto: if your zaptel module isn't providing a constant stream of audio to/from rxfax, you could end up with failed faxes |
| 20:24.19 | Z_God | hmm, I seem to have asterisk version 1.2.1 (from debian etch) |
| 20:24.19 | *** join/#asterisk Samoied (n=Samoied@popeye.opens.com.br) |
| 20:24.39 | justinu | z_god: you also need ztdummy loaded if you have no zap hardware |
| 20:24.42 | justinu | for conferencing |
| 20:24.48 | ceeto | justinu: Ok.. got ya there... But if I hear faint fax noises in the background of external calls during a fax receive, wouldn't it more likely be in the wiring? |
| 20:24.55 | mzo | the only time i mess up is when i scream into my phone. i get echo then :P |
| 20:24.56 | Hymie | Z_God: debian has 1.2.4 in unstable now |
| 20:25.02 | justinu | ceeto: yeah, you're having crosstalk? |
| 20:25.11 | justinu | ceeto: what kind of device do you hear the cross talk on, sip phone? |
| 20:25.11 | Hymie | Z_God: meetme probelms were with choppy voice |
| 20:25.17 | Hymie | Z_God: (and sip) |
| 20:25.35 | Hymie | Z_God: you might have problems with asterisk without a digium card.. they use it as a dongle for asterisk |
| 20:25.39 | ceeto | justinu: Yep... which makes me think I have a wiring problem and hence the extra noise on the fax line is causing them to fail. Like I said, when ther eis nobody on the phone, faxes come through every time. |
| 20:25.47 | Z_God | dongle?? |
| 20:25.50 | ceeto | justinu: Yep, but our inbound lines are all pots. |
| 20:25.50 | Hymie | Z_God: they use it for a timer.... |
| 20:25.55 | justinu | ceeto: i think your reasoning is probably sound |
| 20:26.08 | Hymie | Z_God: so there may be issues with meetme, there is a bit of info at voip-info.orh |
| 20:26.09 | Z_God | hmm, I have used asterisk with only SIP & ISDN before, seemed to work well then |
| 20:26.09 | Hymie | org |
| 20:26.16 | Z_God | k |
| 20:26.17 | justinu | meetme works fine. |
| 20:26.19 | ceeto | I hope so. I guess I'll make a call to the phone company and have them come check it out. Thanks for your help!!! |
| 20:26.22 | justinu | make sure that ztdummy is loaded |
| 20:26.29 | justinu | and make sure your phones use 20ms RTP packet size! |
| 20:26.41 | Hymie | cool |
| 20:26.42 | ceeto | justinu: Is that to me? |
| 20:26.46 | Hymie | there you go Z_God ;) |
| 20:26.49 | justinu | no, to z_god |
| 20:26.54 | Hymie | wisdom from justinu ;) |
| 20:27.01 | justinu | ceeto: another thing you could try. plug one of your pots line directly into fax |
| 20:27.14 | Z_God | but what kind of SIP account do I need to set up, most stuff seems to be about acting as a SIP server (for connecting SIP clients) or acting as a SIP client itself |
| 20:27.19 | Z_God | thx for the info :) |
| 20:27.30 | justinu | sip is peer to peer |
| 20:27.38 | justinu | UAs are both client and servers |
| 20:28.10 | *** part/#asterisk wolfmat99 (n=wolfmat9@209.194.32.60) |
| 20:29.11 | Z_God | I've got exten => 1234,1,Meetme(1234) |
| 20:29.11 | Z_God | in my extensions.conf now |
| 20:30.16 | *** join/#asterisk lalito (n=erg@201.102.4.195) |
| 20:30.17 | *** join/#asterisk wolfmat99 (n=wolfmat9@209.194.32.60) |
| 20:30.59 | *** part/#asterisk wolfmat99 (n=wolfmat9@209.194.32.60) |
| 20:31.16 | Z_God | ah, http://www.voip-info.org/wiki-Asterisk+timer+ztdummy |
| 20:31.17 | Z_God | indeed looks good |
| 20:31.22 | *** join/#asterisk nagl (n=nagl@vie-086-059-104-148.dsl.sil.at) |
| 20:31.45 | justinu | z_god, if you hadn't compiled zaptel before you compiled asterisk, you won't even get app_meetme |
| 20:31.53 | justinu | so make sure that you recompile asterisk after compiling/installing ztdummy |
| 20:32.15 | Z_God | ah, I'm using the debian package currently |
| 20:32.29 | justinu | you really oughta be compiling this stuff yourself |
| 20:32.33 | justinu | packages... meh |
| 20:32.48 | Z_God | hehe I'll do that then :) |
| 20:33.00 | Z_God | compiling only works for me on i686 btw |
| 20:33.07 | *** part/#asterisk redder86 (n=lee@gateway.howardsilvan.com) |
| 20:33.08 | justinu | ? |
| 20:33.09 | Z_God | on i586 and lower it fails for me |
| 20:33.36 | Z_God | I've tried that before on an older comp, seems to be a gcc problem |
| 20:33.36 | justinu | i686 kernel? |
| 20:33.40 | Z_God | I'll look up the bug |
| 20:33.42 | justinu | oh, how old? |
| 20:33.54 | justinu | i did an asterisk installation on a dual P3 600 |
| 20:33.57 | justinu | no problems |
| 20:34.05 | Z_God | http://gcc.gnu.org/bugzilla/show_bug.cgi?id=25912 |
| 20:34.11 | justinu | also compiled it on my powermac G3 |
| 20:34.25 | Z_God | yep, i686 works fine for me too |
| 20:34.38 | Z_God | just on a i586 it seems to break |
| 20:38.18 | justinu | any of you guys use public key auth w/ putty? |
| 20:39.28 | dpolitech | yes |
| 20:43.09 | *** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 20:44.17 | a1fa | _Sam-- |
| 20:44.19 | a1fa | sup |
| 20:44.49 | [av]bani | <o/ |
| 20:45.10 | a1fa | guys, i want my phone to ring twice before asterisk picks it up.. but |
| 20:45.25 | a1fa | asterisk picks it up first, then waits for the appropriate ammount of seconds |
| 20:45.26 | a1fa | exten => s,1,Wait,2 |
| 20:45.26 | a1fa | exten => s,2,Answer( ) |
| 20:47.01 | a1fa | bastard picks-up immediatley |
| 20:47.12 | a1fa | i want it to pickup after two rings |
| 20:48.01 | Math` | wait waits 2 seconds, not two rings |
| 20:48.10 | justinu | heh |
| 20:48.12 | a1fa | Math` : i know |
| 20:48.22 | a1fa | i put it to ~9s~2 rings |
| 20:48.29 | [av]bani | s,1,Playtones(ring) s,2,Wait,2 s,3,Stopplaytones() s,4,Answer |
| 20:48.38 | a1fa | it still answers immediatley |
| 20:48.54 | a1fa | but waits 9s, then "answers" |
| 20:48.58 | [av]bani | its probably your gateway picking up the line |
| 20:49.21 | a1fa | thou shall not judge my gayway :P |
| 20:49.33 | a1fa | so Playtones(ring) will work? |
| 20:50.09 | [av]bani | try it |
| 20:50.28 | *** join/#asterisk GoRK (n=GoRK@amarillo.energynet.com) |
| 20:51.25 | GoRK | is there an option to stop asterisk from sending OPTIONS requests on a SIP channel? |
| 20:51.26 | Z_God | do I need chan_zap.c to compile? |
| 20:51.30 | Z_God | it gives a compile error |
| 20:51.48 | justinu | gork: qualify=no |
| 20:51.52 | sivana | is res_mp3 in 1.2.4? |
| 20:51.59 | a1fa | Z_God : if you dont have zaptel.. then you dont need it |
| 20:52.05 | Z_God | k :) |
| 20:52.12 | GoRK | justinu: oh ok that will do; thanks; searching got me into a loop -- 'options option' hehehe |
| 20:52.28 | *** join/#asterisk Navman (n=icechat5@62.108.206.82) |
| 20:53.25 | a1fa | Z_God : you may want to fix the dependency missing tho |
| 20:54.41 | Z_God | is that libpri? |
| 20:55.08 | a1fa | dont know |
| 20:55.10 | a1fa | lol |
| 20:55.12 | a1fa | look it up |
| 20:55.16 | sivana | anyone know how to convert .mp3 to .ulaw using sox? |
| 20:55.58 | _Sam-- | a1fa : i dont think it will make a difference, but i use wait(2) |
| 20:56.26 | _Sam-- | that is how the wiki page shows it wait() |
| 20:56.30 | _Sam-- | http://www.voip-info.org/wiki-Asterisk+cmd+Wait |
| 20:58.07 | *** join/#asterisk astar` (n=astar@ANantes-154-1-45-53.w81-53.abo.wanadoo.fr) |
| 20:58.14 | [av]bani | <o/ |
| 20:58.23 | a1fa | _Sam-- : i was on efnet the other day.. looks like boys in motorcycles have a lot to say |
| 20:58.44 | [av]bani | _Sam--: tried 1.0.2.13 yet? |
| 20:58.48 | _Sam-- | of course they do, i was one of them until i started kneedraggers. |
| 20:58.56 | _Sam-- | i have some friends still over there, and one employee there |
| 20:59.10 | a1fa | yeah |
| 20:59.11 | _Sam-- | [av]bani: nah not yet, saw it, but the changes werent enough to make upgrade aagian |
| 20:59.15 | [av]bani | _Sam--: o_O |
| 20:59.25 | _Sam-- | how is the dhcp stuff? |
| 20:59.31 | [av]bani | it is fine |
| 20:59.32 | a1fa | _Sam-- : they are not very friendly sometimes.. i guess you need to get to know them |
| 20:59.52 | a1fa | i am going to use this ring |
| 21:00.23 | *** join/#asterisk fourcheeze (n=rich@westbury.doilywood.org.uk) |
| 21:00.41 | _Sam-- | a1fa : do you normally spend any time there on #motorcycles efnet? |
| 21:00.44 | fourcheeze | trying to use func_odbc to get a value from database |
| 21:00.46 | _Sam-- | or this was like the first time? |
| 21:01.05 | fourcheeze | according to show modules: func_odbc.so ODBC lookups 0 |
| 21:01.24 | fourcheeze | I have a conf for it |
| 21:01.37 | fourcheeze | I'm trying out quick count of my cdr |
| 21:01.45 | fourcheeze | mysql log shows that it was counted |
| 21:01.48 | fourcheeze | but nothing is returned |
| 21:01.59 | fourcheeze | I'm doing this |
| 21:02.00 | fourcheeze | exten => _9200XXXXX,1,Set(somewhere=${ODBC_LOCATION('${EXTEN:1}')}) |
| 21:02.05 | a1fa | _Sam-- : i do.. once a while |
| 21:02.08 | fourcheeze | oops wrong line |
| 21:02.13 | fourcheeze | exten => 100,1,Set(something=${ODBC_SQL('SELECT COUNT(*) FROM cdr')}) |
| 21:02.14 | fourcheeze | there |
| 21:02.24 | a1fa | _Sam-- : but everytime its the same story.... there is that group of power trippers |
| 21:02.29 | fourcheeze | in the console it shows an empty string |
| 21:02.35 | a1fa | _Sam-- : i guess the motorcycle elite.. |
| 21:02.39 | _Sam-- | yeah, thats why i stopped going there. |
| 21:02.48 | a1fa | they make fun of everybody, and they do not allowed to be made fun of |
| 21:02.52 | _Sam-- | that and the fact then when i wouldnt give everyone discounts they were a bunch of jackasses... |
| 21:03.03 | a1fa | :P |
| 21:03.05 | _Sam-- | but like i said, i still have like 5 people im friends with there.... |
| 21:03.10 | a1fa | cool |
| 21:03.13 | a1fa | that guy moike is cool |
| 21:03.21 | a1fa | i am cool with him.. but not with the rest of them |
| 21:03.22 | _Sam-- | cyron, ryanbktm, mototoad, descart3s |
| 21:03.29 | fourcheeze | no-one out there using func_odbc ? |
| 21:03.43 | _Sam-- | ive met most of those guys IRL |
| 21:03.56 | _Sam-- | my right hand man at kneedraggers came from #motorcycles |
| 21:03.58 | a1fa | fourcheeze : what is your question |
| 21:04.11 | a1fa | _Sam-- : yeah... i met him too :P |
| 21:04.12 | fourcheeze | query executes but nothing returned |
| 21:04.14 | [av]bani | _Sam--: so none of your gxp's had the lcd corruption? |
| 21:04.24 | fourcheeze | alfa query is logged in mysql.log |
| 21:04.28 | _Sam-- | a1fa : you met exis? |
| 21:04.31 | a1fa | fourcheeze : maybe there is nothing to be returned? |
| 21:04.35 | _Sam-- | [av]bani : nope none |
| 21:04.36 | fourcheeze | there is |
| 21:04.38 | a1fa | _Sam-- : nope |
| 21:04.42 | a1fa | fourcheeze : are you sure? |
| 21:04.45 | fourcheeze | yep |
| 21:04.51 | fourcheeze | I put the same query into mysq |
| 21:04.53 | fourcheeze | l |
| 21:04.56 | _Sam-- | exis is the right hand man at KD...he knows all. |
| 21:05.00 | _Sam-- | from #Motorcycles |
| 21:05.01 | a1fa | ok.. go run sql manually and run that query |
| 21:05.09 | a1fa | see if anything returns |
| 21:05.14 | fourcheeze | alfa: exten => 100,1,Set(something=${ODBC_SQL('SELECT COUNT(*) FROM cdr')}) |
| 21:05.16 | fourcheeze | that's what I'm donig |
| 21:05.24 | a1fa | go to mysql console and run it there |
| 21:05.43 | *** join/#asterisk Tamarisk (n=adrian@user-2899.lns1-c8.dsl.pol.co.uk) |
| 21:06.02 | fourcheeze | a1fa: | 2373 | |
| 21:06.06 | _Sam-- | hey bani, you are still originating through teligax? |
| 21:06.17 | _Sam-- | just wondering if there has been improvement at all |
| 21:06.24 | a1fa | ok.. now see if you can play something |
| 21:06.33 | a1fa | or SetCallerId($something) |
| 21:06.39 | fourcheeze | a1fa: console shows -- Executing Set("SIP/20060001-2cd9", "something=") in new stack |
| 21:06.46 | a1fa | ok |
| 21:07.01 | a1fa | it is taking something literally |
| 21:07.19 | fourcheeze | ok |
| 21:07.49 | a1fa | fourcheeze : look into the synthax |
| 21:08.19 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 21:08.34 | fourcheeze | a1fa: that example is straight out of the bugzilla |
| 21:08.51 | fourcheeze | mantis or whatever it is |
| 21:09.07 | a1fa | ok |
| 21:09.08 | a1fa | lemme see |
| 21:09.10 | *** join/#asterisk WasPhantom (n=neil@203-86-192-98.tasman.net) |
| 21:09.26 | fourcheeze | a1fa: http://bugs.digium.com/view.php?id=5055 |
| 21:10.35 | fourcheeze | if I put a $ directly in front of ODBC it returns 0, which is wrong but is at least a value |
| 21:10.48 | [av]bani | _Sam--: havent been originating or terminating at all lately, busy with other stuff |
| 21:10.58 | a1fa | ok |
| 21:11.00 | a1fa | i got it |
| 21:11.03 | fourcheeze | Oo |
| 21:11.51 | a1fa | did u set func_odbc.conf |
| 21:11.59 | fourcheeze | yep |
| 21:12.00 | _Sam-- | [av]bani : too busy? hahaha! you had plenty time to add some bug reports the first day the gxp firmware came out :) |
| 21:12.12 | fourcheeze | a1fa: the query is exectuted - it's in the mysql log |
| 21:12.44 | a1fa | Set(something=${ODBC_SQL('SELECT COUNT(*) FROM cdr')}) |
| 21:13.07 | fourcheeze | a1fa: yep |
| 21:13.10 | Qwell[] | eh? |
| 21:13.10 | a1fa | what version? |
| 21:13.25 | Qwell[] | What does your read= line look like, and how are you calling it? |
| 21:13.33 | fourcheeze | a1fa: in mysql.log : 060223 21:12:16 309 Query SELECT COUNT(*) FROM cdr |
| 21:13.46 | fourcheeze | a1fa: 1.2.4 |
| 21:13.56 | Qwell[] | func_odbc isn't in 1.2.4 |
| 21:14.22 | fourcheeze | it was in mine |
| 21:14.33 | Qwell[] | shouldn't be |
| 21:14.39 | Qwell[] | in fact, I'd bet money on it |
| 21:14.44 | *** join/#asterisk RoyK (n=roy@216-99-29.0506.adsl.tele2.no) |
| 21:15.13 | *** join/#asterisk ke4qqq (n=ke4qqq@68-115-212-155.static.spbg.sc.charter.com) |
| 21:15.15 | [av]bani | yay 7970 shipped |
| 21:15.28 | kuku5 | 7970? |
| 21:15.30 | [av]bani | chan_sccp here i come |
| 21:15.31 | fourcheeze | Qwell: it is :-) |
| 21:15.45 | kuku5 | Qwell: So nothing beats your own pri origination ? |
| 21:15.50 | [av]bani | kuku5: http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheet09186a00801c9638.html |
| 21:15.52 | kuku5 | ( as far as quality ? ) |
| 21:16.01 | fourcheeze | Qwell: |
| 21:16.03 | fourcheeze | find asterisk-1.2.4 -name func_odbc.c |
| 21:16.03 | fourcheeze | asterisk-1.2.4/funcs/func_odbc.c |
| 21:16.09 | [av]bani | 13:15 -!- Qwell[] [i=north@unaffiliated/qwell] has quit ["My damn controlling terminal disappeared!"] |
| 21:16.26 | a1fa | fourcheeze : patch it with the newest version |
| 21:16.39 | fourcheeze | a1fa: where do I find that? |
| 21:16.47 | kuku5 | [av]bani: its just color |
| 21:17.16 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
| 21:17.24 | fourcheeze | a1fa: do I just grab one from cvs? |
| 21:17.34 | [av]bani | kuku5: and backlight, touchscreen |
| 21:17.42 | Qwell[] | woops |
| 21:17.43 | *** join/#asterisk Majestik (n=Majestik@S0106000024c058cc.ed.shawcable.net) |
| 21:17.46 | kuku5 | ... and what else - thats not enough |
| 21:17.53 | Qwell[] | What did I miss? |
| 21:18.00 | [av]bani | kuku5: 8 lines, xml, poe, dual 10/100 |
| 21:18.32 | kuku5 | 7960g has it |
| 21:18.39 | justinu | not real PoE |
| 21:18.44 | [av]bani | kuku5: not 8 lines |
| 21:18.45 | kuku5 | fake ? |
| 21:18.53 | justinu | Cisco's idea of PoE |
| 21:18.54 | kuku5 | they faked poe |
| 21:19.01 | kuku5 | so different pairs ? |
| 21:19.09 | justinu | different negotiation as well, iirc |
| 21:19.13 | justinu | it uses CDP |
| 21:19.20 | [av]bani | justinu: afaik everything above 7940 uses 802.3af |
| 21:19.38 | justinu | my friends 7960s won't plug into an 802.3af switch |
| 21:19.43 | justinu | well, they will, but they get no power |
| 21:19.46 | Qwell[] | G? |
| 21:19.50 | Qwell[] | I think that matters |
| 21:19.50 | justinu | i believe so |
| 21:19.53 | kuku5 | So can I use the Dell switches |
| 21:19.54 | a1fa | ok |
| 21:19.55 | a1fa | home time |
| 21:19.57 | justinu | i'll clarify that... |
| 21:20.06 | jbalcomb | [av]bani the 7940G requires a magic cable to work with standard PoE |
| 21:20.07 | Qwell[] | I think G does 802.3af |
| 21:20.22 | [av]bani | jbalcomb: yes, which is why i said everything above 7940 |
| 21:20.23 | justinu | he had to buy some magic 802.3af to cisco PoE adapters |
| 21:20.24 | jbalcomb | [av]bani i believe the 7960G is the same |
| 21:21.18 | kuku5 | justinu: the non G's have wording on the buttons, g's have an image ( message, services, etc ) |
| 21:21.18 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 21:21.18 | jbalcomb | [av]bani only real difference between 7940G and 7960G is 2 lines |
| 21:21.18 | kuku5 | I got the injectors for like $20 |
| 21:21.18 | Qwell[] | 4 lines |
| 21:21.18 | Qwell[] | 7940 has 2, 7960 has 6 |
| 21:21.19 | kuku5 | 4 lines |
| 21:21.19 | [av]bani | bleh, 7960 is cisco poe -_- |
| 21:21.49 | justinu | what's the other diff between 7960 and 7960g? |
| 21:21.50 | jbalcomb | Qwell[] ah, page i saw said 2/4 |
| 21:21.50 | Qwell[] | the buttons on the G are pictures |
| 21:21.50 | justinu | that's it? |
| 21:21.51 | *** join/#asterisk rajiv (n=irc@gentoo/developer/rajiv) |
| 21:21.51 | GerbilNut | if an agent is in the queue, and they have their dnd button pushed, its going directly to their voicemail instead of ringing everyone else and skipping them. Anyone know how to get around this? |
| 21:21.51 | jbalcomb | G is for 'Global' so no English |
| 21:21.52 | kuku5 | juanjoc: g - global |
| 21:21.52 | justinu | ah |
| 21:22.05 | [av]bani | 7941 supports 802.3af |
| 21:22.12 | Qwell[] | all the 79x1's will |
| 21:22.23 | jbalcomb | price difference? |
| 21:22.31 | Qwell[] | between 79x0 and 79x1? |
| 21:22.34 | jbalcomb | yeps |
| 21:22.37 | [av]bani | ~phones |
| 21:22.38 | jbot | from memory, phones is at http://bani.anime.net/phones/ |
| 21:22.40 | Qwell[] | very little, unless you get a -ge |
| 21:22.50 | [av]bani | ahh but 79x1 only do sccp ... |
| 21:22.56 | Qwell[] | yeah, so? |
| 21:22.56 | jbalcomb | bad |
| 21:22.57 | kuku5 | Anyone using voipsupply.com power supplies? |
| 21:23.09 | [av]bani | Qwell[]: so if you want to talk with sip pbx, too bad |
| 21:23.21 | Qwell[] | [av]bani: Like CCM or Asterisk? |
| 21:23.28 | [av]bani | Qwell[]: like say a nortel |
| 21:23.39 | Qwell[] | Why would I plug a cisco into a pile of shit like Nortel? |
| 21:23.48 | stoffell | lol |
| 21:23.49 | jbalcomb | like say.. why would want to run SCCP? |
| 21:23.52 | [av]bani | because it wasnt your decision? |
| 21:23.56 | Qwell[] | jbalcomb: Because sccp rocks |
| 21:24.00 | Qwell[] | [av]bani: Sucks to be you then... |
| 21:24.06 | [av]bani | Qwell[]: not me, just an example |
| 21:24.08 | russellb | Qwell[]: liar |
| 21:24.10 | jbalcomb | [av]bani that doesn't happen. we are all in full control of our destinys. |
| 21:24.14 | kuku5 | qwell: you use sccp over sip ? |
| 21:24.18 | Qwell[] | russellb: I wuv sccp :D |
| 21:24.20 | Qwell[] | kuku5: Yes |
| 21:24.28 | [av]bani | sccp over sip? wtf? |
| 21:24.35 | jbalcomb | silly kids |
| 21:24.40 | Qwell[] | It's better for my needs |
| 21:25.03 | kuku5 | Qwell: is there anything that i can reat that will compare those two ? Im interested |
| 21:25.07 | jbalcomb | better for a working VoIP setup that relatively affordable? |
| 21:25.17 | Qwell[] | kuku5: RFC and the spec for sccp? |
| 21:25.24 | kuku5 | no no |
| 21:25.26 | kuku5 | not that specifi |
| 21:25.28 | kuku5 | more general |
| 21:25.32 | Qwell[] | [av]bani: I bet I can plug them in, and they just work |
| 21:25.40 | Qwell[] | so...go ahead and ship them over |
| 21:26.02 | jbalcomb | is that because they are SCCP or because they are Cisco? |
| 21:26.12 | Qwell[] | jbalcomb: both |
| 21:26.22 | Qwell[] | cisco did auto config the right way |
| 21:26.49 | jbalcomb | wow. well, i have a hard time with rediculously pointless, narrowminded conversations so.. see ya'll tomorrow. |
| 21:27.02 | kuku5 | hm |
| 21:27.05 | kuku5 | dumbass :) |
| 21:27.09 | Qwell[] | indeed |
| 21:27.09 | *** join/#asterisk Dr-Linux (n=Dreamer@host202-147-168-130.lhr.dancom.net.pk) |
| 21:27.23 | justinu | i don't think there's any RFC on SCCP |
| 21:27.36 | Qwell[] | justinu: No, there isn't. That's why I said the sccp spec :p |
| 21:27.55 | kuku5 | is thre a sccp spe |
| 21:28.08 | Qwell[] | thought so |
| 21:28.10 | kuku5 | if yes - where |
| 21:28.17 | Qwell[] | got me. google? |
| 21:28.19 | [av]bani | wouldnt be suprised if sccp is patented |
| 21:28.26 | [av]bani | cisco likes to do that kind of thing |
| 21:28.27 | *** join/#asterisk nguyep (n=chatzill@64.34.203.231) |
| 21:28.51 | znoG | business wise, it probably makes sense for them. |
| 21:29.01 | nguyep | does n1 has Access211 ATA firmware? |
| 21:29.03 | jontow | i didn't mind SCCP at all |
| 21:29.04 | justinu | the best spec is probably the chan_sccp code |
| 21:29.06 | [av]bani | well, theyve patented other protocols and stuff, which prevents people from using it |
| 21:29.11 | justinu | or whatever sccp channel driver actually works |
| 21:29.11 | Qwell[] | justinu: likely so |
| 21:29.25 | jontow | my problem was shoddy support at the time i tried it, and later, not enough time to make chan_sccp.so compile instead of using chan_skinny.so |
| 21:29.44 | jontow | the protocol itself makes a lot of sense; when you need utter control over the actions a phone is attempting |
| 21:29.55 | Qwell[] | sccp can do far more |
| 21:29.55 | justinu | agreed |
| 21:30.01 | justinu | it's better in a star topology |
| 21:30.11 | *** part/#asterisk nguyep (n=chatzill@64.34.203.231) |
| 21:31.10 | *** join/#asterisk Katty (n=angela@64.82.232.54) |
| 21:31.15 | Katty | allo. |
| 21:31.22 | justinu | herro |
| 21:31.29 | Qwell[] | how're the teeth? |
| 21:31.32 | [av]bani | does Qwell have any non-cisco phones? |
| 21:31.37 | Qwell[] | [av]bani: No |
| 21:31.39 | Katty | Qwell[]: the teeth are gone. |
| 21:31.47 | Katty | Qwell[]: my gums, however, still ache. |
| 21:31.48 | Qwell[] | how're the gums? |
| 21:31.58 | Qwell[] | still no meds? |
| 21:32.05 | Katty | i'm taking tylenol. |
| 21:32.11 | justinu | so according to my friend the 7960g's don't do 802.3af either |
| 21:32.14 | justinu | need an adapter |
| 21:32.22 | [av]bani | justinu: yes, i just corrected the page |
| 21:32.33 | Juggie | i dont think thats right. |
| 21:32.45 | Juggie | later models of the 7960 do both power standards |
| 21:32.51 | [av]bani | justinu: 7961 |
| 21:33.02 | [av]bani | s/justinu/juggie/ |
| 21:33.22 | justinu | 7961 exists? |
| 21:33.26 | justinu | haven't seen those yet |
| 21:33.38 | [av]bani | neither have i, but its on ciscos product list |
| 21:33.38 | Qwell[] | justinu: The line buttons have lights |
| 21:34.09 | Juggie | what are the two power standards, theres the cisco one, and then 802.3af? |
| 21:34.14 | GoRK | If my provider does not provide CNAM and only calling number, can I purchase CNAM on my own and get it out of band? |
| 21:34.23 | Qwell[] | "All Cisco IP Phones support the Cisco Inline Power variant, and the 7970G model offers both Cisco Inline Power and 802.3af support." |
| 21:34.32 | justinu | juggie: there's more than a few proprietary PoE implementations, i think |
| 21:34.33 | Qwell[] | I do believe Juggie is right though |
| 21:34.50 | Juggie | i'm pretty sure the 7960 i have is using 802.3af |
| 21:34.58 | GoRK | juggie: the only one that I'd call a "standard" is 802.3af.. CDP is supported by a lot of things but is being supplanted as fast as it can be by everyone (including Cisco) |
| 21:35.08 | *** part/#asterisk iamjeff (n=iamjeff@jumpgate2.kingston.net) |
| 21:35.22 | [av]bani | ah, 7960 = cisco 7960g = 802.3af |
| 21:35.30 | GoRK | there are several switches that will do 802.3af or CDP |
| 21:35.30 | *** join/#asterisk ManxPower (n=ewieling@stirprop-s4-0-0-21.ndcr2.datasync.net) |
| 21:35.35 | Juggie | then i must have 7960g's |
| 21:35.45 | Juggie | GoRK, i have a cisco switch which probally does both. |
| 21:35.52 | Juggie | not sure i havnt looked at it lately |
| 21:35.56 | Juggie | and switches arnt my thing |
| 21:36.32 | justinu | bani: according to my friend with 7960g's that's not true |
| 21:36.32 | Katty | 1111 new missed calls |
| 21:36.46 | justinu | he has a 802.3af switch, and they don't take power from it |
| 21:36.48 | kuku5 | Qwell: can you use the auto config thing with * and cisco's sccp |
| 21:37.01 | [av]bani | justinu: http://www.voipuser.org/review_18.html |
| 21:37.22 | GoRK | manxpower: you are special, at least you must be to polycom. they are so freakin stingy with it. They wouldnt even let me download the firmware that came preinstalled on a phone i purchased |
| 21:37.29 | justinu | *shrug*... i'd have to believe my friend over that page. |
| 21:37.41 | [av]bani | http://www.cisco.com/en/US/netsol/ns340/ns394/ns147/ns412/networking_solutions_white_paper09186a008026641c.shtml |
| 21:37.51 | [av]bani | http://www.cisco.com/en/US/products/hw/phones/ps379/products_qanda_item0900aecd8039de5e.shtml |
| 21:38.03 | Juggie | sigh |
| 21:38.08 | Juggie | let me log into the catalyst and look |
| 21:38.17 | GoRK | i can confirm that nearly all of the cisco phones and specifically including the 7960 are not 802.3af |
| 21:38.25 | justinu | GoRK: how about 7960g? |
| 21:38.59 | Qwell[] | all new cisco phones do 802.3af... |
| 21:39.07 | Qwell[] | 79x1, 7911, etc |
| 21:39.24 | _Sam-- | as the dumb here,...what is the advantage of running sccp vs. sip? |
| 21:39.24 | [av]bani | http://www.cisco.com/en/US/netsol/ns340/ns394/ns147/ns412/networking_solutions_white_paper09186a008026641c.shtml <- if the 7960g doesnt support 802.3af, why does cisco list it on their 802.3af page |
| 21:39.25 | GoRK | highly doubtful on the 7960g either but depending on the 802.3af switch, you might be able to get it to work - hang on and ill get you the doc |
| 21:39.37 | [av]bani | is cisco in error? |
| 21:39.46 | *** join/#asterisk skac (n=matilda@unaffiliated/acidchild) |
| 21:40.08 | skac | whats the name of that box that fits in your hub? and then your phone goes in to it? |
| 21:40.31 | _Sam-- | skac : ata (analog terminal adapter) ? |
| 21:40.32 | cpm | it's the phonehubbox |
| 21:40.48 | *** part/#asterisk Tamarisk (n=adrian@user-2899.lns1-c8.dsl.pol.co.uk) |
| 21:41.22 | justinu | my friend was not able to make his 7960g's work with 802.3af switches until he bought these: http://voipstore.atacomm.com/Shops/ViewItem.aspx/27934028032-44101559296.htm |
| 21:41.27 | cpm | skac, http://www.voip-info.org/wiki/view/ATA |
| 21:41.28 | skac | _Sam--: whats a good one? |
| 21:41.30 | skac | kk |
| 21:41.31 | skac | thx |
| 21:41.40 | GoRK | here is a way that some 802.3af switches can provide power to 7940/7960(G) : http://www.voip-info.org/wiki-Cisco+POE |
| 21:41.52 | _Sam-- | justin/qwell/bani ...what is the advantage of sccp? |
| 21:41.52 | justinu | GoRK: or the device I just linked to |
| 21:42.00 | [TK]D-Fender | What *I* want to know is.... why isn't the power consumpion of that guitar listred!??!?!? CONSPIRACY I SAY! |
| 21:42.05 | [av]bani | _Sam--: sccp gives qwell a stiffy |
| 21:42.09 | justinu | advantage is the iPBX controls the behavior of the phone |
| 21:42.10 | skac | whats a good ATA? |
| 21:42.18 | skac | i forgot the name of the one someone told me in here before. |
| 21:42.20 | justinu | skac: SPA-1000 |
| 21:42.21 | skac | began with a 's' |
| 21:42.24 | justinu | sipura |
| 21:42.25 | skac | thats the one!! |
| 21:42.25 | cpm | The IAXy is good because it costs a whole lot of money. |
| 21:42.27 | [TK]D-Fender | skac : Depending on your needs, Linksys' are pretty good. |
| 21:42.28 | cpm | So it must be good |
| 21:42.47 | znoG | PAP2-NA are as good as the Sipuras, infact they use the same soft and cost about the same |
| 21:42.50 | JCC_ | or the spa-3000, its worked well for me |
| 21:42.53 | [TK]D-Fender | SPA-1000 = waste. SPA-2002 is the only one in its class/bracket worth considering. |
| 21:42.59 | [av]bani | justinu: so apparently, cisco's own website is in error |
| 21:43.05 | justinu | ok, well.. all of the spa products are decent |
| 21:43.07 | GoRK | justinu: sure I guess a $20 power converter is always an option if the cable trick doesnt work.. $20 per phone is quite a chunk if just making a cable would do the trick |
| 21:43.09 | justinu | bani: i would think so |
| 21:43.12 | znoG | when the IAXy comes down in price, people might consider it a viable option |
| 21:43.14 | [av]bani | justinu: spa-841 ... |
| 21:43.16 | justinu | GoRK: agreed... |
| 21:43.19 | [TK]D-Fender | SPA-3000 is a great little box, but its more than just an ATA. If you need PSTN connectivity to boot its a great start. |
| 21:43.26 | justinu | bani: ok, all of sipura's software is decent |
| 21:43.27 | [av]bani | [TK]D-Fender: the spa-3000 echo canceller is poo |
| 21:43.37 | justinu | they have nice fast web interfaces |
| 21:43.37 | cpm | I have some IAXys and I really like them, but they are just way too much money. |
| 21:43.41 | justinu | and they boot up quick |
| 21:43.44 | [TK]D-Fender | [av]bani : Its PASSABLE poo for the price :) |
| 21:44.05 | [av]bani | [TK]D-Fender: it is cheap poo, sure. but i have a zoom 5801 on order i will compare with |
| 21:44.12 | JCC_ | yeah, echo comes and goes and tat's a pain |
| 21:44.14 | GoRK | btw if anyone is interested, D-link's got a universal 5v/12v switchable 802.3af converter.. it is a very handy unit and cheap too. I use them with access points but no reason they wouldnt work with any phone that takes 5v or 12v DC |
| 21:44.27 | Juggie | grrr i hate this cisco network assistant |
| 21:44.27 | justinu | ciscos want 48vdc |
| 21:44.30 | Juggie | where do i find the power stuff |
| 21:44.35 | Juggie | i dont see it |
| 21:44.38 | justinu | so does my aastra 480i |
| 21:44.46 | cpm | Juggie, don't use it. |
| 21:44.55 | [av]bani | [TK]D-Fender: the spa-3000 only supports g165 to 8ms echo tail. the zoom 5801 does g168 out to 50ms |
| 21:45.09 | znoG | GoRK: whats the name of DLinks product? |
| 21:45.34 | [av]bani | [TK]D-Fender: zoom also has t38 for the 5801... :O~~~ |
| 21:45.34 | GoRK | znoG: D-Link DWL-P50 |
| 21:45.39 | znoG | GoRK: ta |
| 21:45.53 | Juggie | cpm, i know almost nothing about this box :) where do i enable telnet? |
| 21:46.18 | Juggie | i had console via serial |
| 21:46.21 | GoRK | znoG: 12V at up to 1A.. They work great plugged into linksys access points though I feel kind of funny powering a linksys AP with a D-Link power converter |
| 21:46.23 | [TK]D-Fender | [av]bani : Could be interesting. |
| 21:46.46 | [av]bani | justinu: i only wish sipura's software was decent. its buggy :| |
| 21:46.55 | znoG | GoRK: hehe yeah |
| 21:47.01 | justinu | is it? |
| 21:47.06 | justinu | i haven't noticed anything |
| 21:47.11 | [TK]D-Fender | GoRK : Thats ok, I use my Polycom IP600 power supplies to power my Plantronics headset amplifiers :) Plantronics are cheap asses and charge you for it seperately.... |
| 21:47.22 | justinu | i can't figure out why my polycoms crash occasionally when they try and dial |
| 21:47.36 | justinu | gets caught in some kind of infinite messaing loop w/ * and then they crash |
| 21:47.38 | [av]bani | justinu: at least the spa3k firmware is |
| 21:47.45 | znoG | GoRK: so on one end you stick the P50, and how do you then split the power/data on the Linksys end? |
| 21:48.04 | *** join/#asterisk websae (n=icechat5@adsl-64-149-206-121.dsl.milwwi.sbcglobal.net) |
| 21:48.12 | [av]bani | justinu: maybe the ata's and the 941's are ok |
| 21:48.18 | GoRK | znoG: the 802.3af switch is on the other end. This is a splitter not an injector |
| 21:48.31 | justinu | except for the buttons and the unreadable LCD, i liked the SPA841 |
| 21:48.32 | *** join/#asterisk tangle_eye (n=alexthew@rrcs-24-227-188-230.sw.biz.rr.com) |
| 21:48.33 | Darwin35 | bucks |
| 21:48.41 | Darwin35 | and it works great |
| 21:48.45 | [av]bani | justinu: and the shoddy construction? |
| 21:48.46 | justinu | nice deal |
| 21:48.46 | xachen | where from? :o |
| 21:48.49 | [TK]D-Fender | justinu : Hva eyou tried changing the firmware? |
| 21:48.56 | justinu | bani: it felt better than the gxp, actually |
| 21:49.02 | justinu | yeah, i did a firmware flash on it |
| 21:49.10 | Darwin35 | a small mom and pop computershop |
| 21:49.22 | [TK]D-Fender | justinu : happens on multiple revisions? |
| 21:49.25 | [av]bani | justinu: the 841 was compared with cheap chinese clones, lots of people were very unhappy. the 941 made everyone \o/ |
| 21:49.40 | justinu | d-fender: yeah, from 1.5.x to the latest 1.6 releases |
| 21:49.42 | GoRK | znoG: run a cable from your 802.3af switch to the DWL-P50 then run power and ethernet out of the P50 to the device.. it comes with the necessary cable -- if the barrel plug fits the device anyway and it's center positive. If either are not true you'd have to rewire or build your own little power jumper cable |
| 21:49.43 | cpm | [TK]D-Fender, are they being cheap? Or are they (plantronics) having been in this biz since the day, assuming -48vdc everywhere, and don't charge you for the worthless wall wart you don't need if you are actually doing telephones? |
| 21:49.45 | jontow | indeed.. i'd still like to get ahold of a 941 as well |
| 21:49.47 | Darwin35 | going to make it my new asterisk box |
| 21:50.02 | [TK]D-Fender | justinu : could just be a flakey phone.... RMA it. |
| 21:50.08 | justinu | multiple phones |
| 21:50.19 | [av]bani | what? a flaky polycom? |
| 21:50.20 | [av]bani | O_O |
| 21:50.30 | justinu | i think it has something to do with the dialplan I've setup, and the MGCP dial string |
| 21:50.39 | justinu | but I can't pinpoint it, because I can't make it happen all the time |
| 21:50.44 | tangle_eye | Anyone have trouble recording out-bound calls on demand with A@H 2.2? |
| 21:50.47 | znoG | GoRK: ah i see, nice... I'm not familiar with 802.3af though |
| 21:50.52 | jontow | man, my IP600 just rebooted out of the blue once last week while i was sitting here hacking on a webserver.. freaked me out |
| 21:51.01 | znoG | GoRK: i presume its a PoE switch |
| 21:51.16 | jontow | my cow-orkers don't often hear me say "what the fuck?!" in such a loud tone during the business day |
| 21:51.18 | [av]bani | jontow: well see there you go, you're using a crappy phone. you need a ip601 |
| 21:51.21 | znoG | GoRK: i mean 802.3af means PoE |
| 21:51.22 | [TK]D-Fender | cpm : Not including the wart for a wired amplifier is crazy. The price they CHARGE for it seperately is WORSE. Fortunately I had 26 extra compatible PS's from my Polycom's |
| 21:51.32 | jontow | hehehe |
| 21:51.50 | Juggie | this cisco 3750 i have does both cisco pre-draft and 802.3af |
| 21:51.54 | jontow | nah, i want the SPA-941 to replace my IP600 |
| 21:51.55 | jontow | :) |
| 21:52.05 | jontow | i don't use most of the advanced features.. am not on the phone that often |
| 21:52.06 | stoffell | jontow, you think that's a good replacement? ;) |
| 21:52.09 | jontow | i gave my boss my 7960 |
| 21:52.11 | jontow | i have no idea |
| 21:52.14 | justinu | jontow: i just expect stuff like that now |
| 21:52.16 | jontow | but i just need a phone.. |
| 21:52.19 | justinu | rebooting phones |
| 21:52.25 | justinu | rebooting anything |
| 21:52.27 | jontow | hah |
| 21:52.46 | znoG | GoRK: DLINK probably sell a pair of devices to convert standard ethernet to PoE and PoE to ethernet/power, right? |
| 21:53.48 | Darwin35 | now to load fbsd on this puppy |
| 21:54.03 | jontow | Darwin35: :D |
| 21:54.15 | jontow | how are the zaptel drivers these days for wct1xxp cards on freebsd? |
| 21:54.18 | jontow | i haven't tried mine in quite some time.. |
| 21:54.25 | jontow | spend too much time in gentoo :( |
| 21:54.26 | [TK]D-Fender | jontow : Are you on crack? 941 ranks inferior to an IP501, and its not even a dream against the 600 |
| 21:54.32 | justinu | heh |
| 21:54.39 | Darwin35 | heaqd seems like they should work |
| 21:54.53 | Darwin35 | i only us ztdummy for timing |
| 21:54.55 | jontow | [TK]D-Fender: nah.. just like to swap out my phone once every 3-6months to keep things fresh.. and to learn the intricacies of a new piece of hardware |
| 21:55.08 | jontow | i don't use it a lot :) lots of speakerphone test calls to various PBX's around here though |
| 21:55.17 | jontow | i'd like to replace my SPA841 at home with a 941 too |
| 21:55.24 | justinu | true... i've gone thru a lot of phones |
| 21:55.28 | justinu | just to learn em |
| 21:55.31 | GoRK | znoG: they sell poe switches. I have a DWL-1526 switch that I like a lot |
| 21:55.35 | znoG | jontow: perhaps you should look into something a little more exciting than changing phones every 3-6 months :) |
| 21:55.37 | cpm | [TK]D-Fender, I dunno, it might stem from making a good product in the days of half-assed crap, and having to compete on a per unit cost basis. The fact that you need power comes as an ugly suprise. I have a plantronics headset that is older than I am, and that's saying something, I'm an old fart. |
| 21:55.38 | justinu | i have a des-1526 |
| 21:55.39 | jontow | (but.. you know.. that can come AFTER i convince our router admin to enable QoS on the friggin' router) |
| 21:55.46 | justinu | 24 port PoE linksys |
| 21:56.03 | justinu | err dlink |
| 21:56.09 | jontow | znoG: i'm expected to know what's-what on a regular basis for stations.. part of my job is choosing the right hardware for the job.. that means matching features against configurability and price, etc etc. |
| 21:56.11 | GerbilNut | I have a snom 360 with the ringer volume turned all the way down, but it's still way too loud, anyone have experience with those? |
| 21:56.16 | Darwin35 | it does the job |
| 21:56.18 | GoRK | yes des-1526 not dwl-1526.. it is a good switch |
| 21:56.27 | GoRK | for the price |
| 21:56.32 | justinu | i made it freak out a few times by playing with the vlan stuff |
| 21:56.35 | justinu | scared me |
| 21:56.43 | jontow | GerbilNut: i have a bunch of 320s.. i just plugged in a short clip of bob dylan and johnny cash singing 'ring of fire'.. coolest. ringtone. evar. |
| 21:56.44 | justinu | (the des1526) |
| 21:56.45 | GoRK | i touched mine one time and it rebooted |
| 21:56.49 | znoG | GoRK: i was thinking more along the lines of a non-PoE switch, where a port plugs into a device that outputs ethernet/power over ethernet and on the other end, splits it into power/ethernet again.. make sense? |
| 21:56.49 | justinu | lol |
| 21:56.50 | [TK]D-Fender | cpm : Yeah I never said it wasn't a GOOD product, just a slap at trying to save 3$ while charging 100$ for a friggen cheap amp... |
| 21:56.57 | GoRK | and reset to factory defaults |
| 21:56.58 | [av]bani | GerbilNut: replace the ringtone |
| 21:57.02 | justinu | gork: :( |
| 21:57.05 | GoRK | znoG: they make injectors that are 802.3af compliant |
| 21:57.07 | cpm | Yeah, I hear ya. |
| 21:57.10 | jontow | plus, that way you can mess with the volume of the clip itself.. and then deal with it on the phone when you need to. |
| 21:57.11 | GerbilNut | yea, i have Blues Travelers - Run Around, but mine is the only one in the office that does that, and it annoys people |
| 21:57.15 | znoG | GoRK: yeah i figured they make them |
| 21:57.19 | jontow | hehehe |
| 21:57.24 | GoRK | znoG: but they cost as much as the low end switches that do PoE |
| 21:57.37 | znoG | GoRK: a low end switch still probably costs a bit though |
| 21:57.38 | justinu | the des1526 is quite the bargain at this point |
| 21:57.44 | justinu | 400 USD, i think |
| 21:58.01 | GoRK | znoG: the des1526 is about 400.. 24 port injectors are around that price too if not a lot higher |
| 21:58.11 | cpm | And you've got a point, I've got a plantronics wall wart in my hand right now, the only thing plantronics about it is the sticker. It's a typisch pos. Should be included. |
| 21:58.11 | jontow | i mean, seriously, getting a call with your boss standing there talking to you and all of a sudden "i went down, down, down and the flames went hiiigher, and it burns, burns, burns.. that ring of fire... that ring of fire" |
| 21:58.21 | GoRK | heck 10 cisco wall warts will cost you more than a des1526 |
| 21:58.22 | justinu | 400 for a vlan/snmp/poe 24 auto mdi ports |
| 21:58.26 | justinu | 2 gig uplinks |
| 21:58.40 | [TK]D-Fender | I run 2 x DES-1526 here for my Polycom IP 600's and am very happy with them. |
| 21:58.45 | cpm | btw, jic, phones should NOT NEED wallwarts! |
| 21:58.56 | justinu | anyone using VLANs on the DES-1526? |
| 22:00.43 | GoRK | i have the phones on the 1526 and the uplink port is plugged into a gige switch on a port that is on a vlan, so i am sort of using vlans with 1526 but not mixing vlans on the 1526.. the cpu in that switch is really slow; i would not trust it with a lot of vlans to be honest |
| 22:01.09 | justinu | yeah, i wanted to do .q trunks |
| 22:01.21 | justinu | didn't seem to work |
| 22:01.44 | justinu | i got it to work, then it stopped working, couldn't troubleshoot it |
| 22:01.49 | jontow | :/ |
| 22:04.19 | Darwin35 | ok fbsd loaded |
| 22:04.22 | [av]bani | GoRK: vlan (802.1q) should be done at the hardware level by the chips, not cpu |
| 22:04.26 | jontow | gotta love that ;) |
| 22:04.53 | jontow | 11min |
| 22:05.04 | *** part/#asterisk Samoied (n=Samoied@popeye.opens.com.br) |
| 22:05.08 | justinu | huh? |
| 22:05.27 | jontow | 16:54 < Darwin35> now to load fbsd on this puppy |
| 22:05.31 | jontow | 17:05 < Darwin35> ok fbsd loaded |
| 22:05.36 | justinu | oh |
| 22:05.36 | jontow | :) |
| 22:05.58 | justinu | i do centos diskless installs (over the internet) in 20 |
| 22:06.06 | justinu | no media required |
| 22:06.08 | Darwin35 | nonow I have to start doing okgs |
| 22:06.29 | [av]bani | interweb |
| 22:06.37 | justinu | yes, it's http |
| 22:06.48 | [av]bani | pxe ftw |
| 22:06.51 | Darwin35 | I use a 40 mb boot only cdrom |
| 22:06.52 | justinu | yep |
| 22:06.56 | justinu | pxelinux |
| 22:06.58 | Darwin35 | and do net installs |
| 22:07.07 | justinu | needing a CDrom is lame |
| 22:07.21 | *** join/#asterisk miller7 (n=none@gige-2.office-nl.irismedia.gr) |
| 22:08.48 | Darwin35 | I can do it of usb disk on a cchip if i new where I placed it |
| 22:09.20 | justinu | you can do it with just a PXE network card |
| 22:09.30 | Darwin35 | 64 meg diskon chip |
| 22:09.41 | Darwin35 | yeah |
| 22:10.03 | Darwin35 | not had time to learn/setup for that |
| 22:10.10 | xachen | sir |
| 22:10.14 | *** join/#asterisk FLeiXiuS (n=fleixius@c-68-50-206-161.hsd1.md.comcast.net) |
| 22:10.18 | xachen | If you think you can get ztdummy working good on BSD |
| 22:10.22 | xachen | then you are indeed smoking crack :) |
| 22:10.34 | justinu | haha |
| 22:10.39 | justinu | lol |
| 22:10.43 | Darwin35 | ztdummy works great on fbsd and has for over a year now |
| 22:10.47 | xachen | I crash my kernel everytime I load up ztdummy |
| 22:10.51 | xachen | and I even used the ports collection |
| 22:11.04 | Darwin35 | your not using heads |
| 22:11.08 | Darwin35 | head |
| 22:11.18 | xachen | i compiled it |
| 22:11.19 | Darwin35 | ver 11 inports is crap |
| 22:11.28 | xachen | manually that is |
| 22:11.31 | Darwin35 | you have to grab the cvss |
| 22:11.35 | xachen | and loaded it. didn't work, went to unload and my kern crashed |
| 22:11.38 | xachen | see the pattern? :p |
| 22:11.53 | Darwin35 | mine works fine and I have 11 servers all running it |
| 22:12.02 | jontow | xachen: i had no problems with ztdummy |
| 22:12.07 | justinu | heh |
| 22:12.09 | jontow | it was wct1xxp that i had problems with.. |
| 22:12.16 | justinu | freebsd users unite! |
| 22:12.19 | jontow | i ran ztdummy for 6 months on netbsd, and a year on freebsd |
| 22:12.28 | justinu | what? no openbsd? |
| 22:12.39 | jontow | don't really like openbsd all tha tmuch |
| 22:12.46 | Darwin35 | I will open yuo bsd if you want |
| 22:12.48 | justinu | hmm, whats wrong with it? |
| 22:12.57 | jontow | too obfuscated ;) |
| 22:13.02 | xachen | openbsd I love |
| 22:13.02 | justinu | i'm ignorant when it comes to bsd |
| 22:13.06 | FLeiXiuS | Is there a special configuration you have to do to get callerid (alpha names) working |
| 22:13.29 | Darwin35 | yes you have to have a s7 backbone to use cid on pstn |
| 22:13.43 | justinu | that's basically true |
| 22:13.51 | FLeiXiuS | I'm using SCCP2 and I can't seem to grasp why it's not pulling the information off the telco. |
| 22:13.56 | Darwin35 | and a database agreement with thte phone company |
| 22:13.56 | ManxPower | FLeiXiuS, incoming or outgoing? |
| 22:14.07 | FLeiXiuS | Incoming, sorry I forgot to specify |
| 22:14.32 | ManxPower | what type if interface to the PSTN? VoIP, POTS, PRI, CT1? |
| 22:15.02 | FLeiXiuS | POTS |
| 22:15.14 | ManxPower | FLeiXiuS, What country? |
| 22:16.09 | FLeiXiuS | USA.. My analog line runs to my Cisco 3640 where it's sent via SIP to my PBX box. From there the PBX establishes the call to my 7960's via SCCP. |
| 22:16.12 | FLeiXiuS | General layout.. |
| 22:16.38 | ManxPower | FLeiXiuS, that's much more complex that I feel like helping with |
| 22:16.47 | FLeiXiuS | Ha :-P |
| 22:17.02 | *** join/#asterisk Igbothom_III (n=HiltonT@office.quarkit.com.au) |
| 22:17.17 | FLeiXiuS | the only problem I'm having is getting the CID to display on the phones. The router is receiving them... |
| 22:17.26 | GerbilNut | goooo Teliax and acidently porting our main number over early |
| 22:20.31 | Darwin35 | ok cvsup done |
| 22:20.42 | Darwin35 | now for make buildworld |
| 22:21.22 | *** join/#asterisk remiss (i=bofh@129.84-48-68.nextgentel.com) |
| 22:21.35 | remiss | bah |
| 22:22.47 | *** join/#asterisk rakanuj (i=rakanuj@64.207.60.100) |
| 22:22.48 | Katty | humbug. |
| 22:23.29 | Darwin35 | I have to figure otu he math on this server 768m ram 1ghz cpu =# concurrent calls |
| 22:23.49 | Darwin35 | g729 |
| 22:24.01 | xachen | Feb 23 17:23:52 WARNING[32293]: chan_zap.c:915 zt_open: Unable to open '/dev/zap/pseudo': Permission denied |
| 22:24.09 | xachen | that is what I get when I do get ztdummy working |
| 22:24.30 | *** join/#asterisk mexuar-tim (n=mexuar-t@future-is.orange.co.uk) |
| 22:24.42 | Darwin35 | how are you loading it |
| 22:25.14 | Darwin35 | and do you have the zaptel.conf file in/usr/local/etc |
| 22:25.24 | xachen | make ztdload |
| 22:25.32 | Darwin35 | that warning means normaly the file is missing |
| 22:25.42 | xachen | no zaptel.conf file |
| 22:25.48 | Darwin35 | thats why |
| 22:25.59 | xachen | How do I generate one? |
| 22:26.03 | Darwin35 | you need the zaptel.conf |
| 22:26.21 | Qwell[] | do a `make install` |
| 22:26.24 | Darwin35 | get the one from linux |
| 22:26.38 | Darwin35 | no they forgot to add it to the bsd cvs tree |
| 22:26.45 | Darwin35 | I willl add it this weekend |
| 22:26.53 | Qwell[] | why are you using ports? |
| 22:26.58 | xachen | I'm not now |
| 22:26.58 | Qwell[] | compile the real source |
| 22:27.09 | Qwell[] | good, so...`make install |
| 22:27.12 | Darwin35 | and use /usr/local/etc/rc.d/zaptel.sh to load the drivers at boot time |
| 22:27.12 | xachen | I did :P |
| 22:27.20 | xachen | yeah |
| 22:27.23 | xachen | I'll create a bootup script |
| 22:27.28 | Darwin35 | no just svn the src and get the 1 file |
| 22:28.43 | Darwin35 | zaaptel.conf |
| 22:28.57 | Darwin35 | loadzone = us |
| 22:29.03 | Darwin35 | defaultzone = us |
| 22:29.06 | Darwin35 | thats all |
| 22:29.14 | tangle_eye | I got outbound call recording to work with A@H 2.2, by adding a "W" in the [macro-dialout-trunk], but recordings are not showing up in the ARI, anyone have a suggestion? |
| 22:29.15 | Darwin35 | thats needed for ztdummy |
| 22:31.46 | Darwin35 | PREFIX=/usr/local |
| 22:31.46 | Darwin35 | LIBDIR=${PREFIX}/lib/zaptel |
| 22:31.46 | Darwin35 | case "$1" in |
| 22:31.46 | Darwin35 | <PROTECTED> |
| 22:31.49 | Darwin35 | <PROTECTED> |
| 22:31.51 | Darwin35 | <PROTECTED> |
| 22:31.53 | Darwin35 | <PROTECTED> |
| 22:31.54 | Qwell[] | ~pb |
| 22:31.56 | jbot | well, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 22:31.56 | Darwin35 | <PROTECTED> |
| 22:31.57 | Darwin35 | <PROTECTED> |
| 22:31.59 | Darwin35 | <PROTECTED> |
| 22:32.01 | Darwin35 | <PROTECTED> |
| 22:32.03 | Darwin35 | <PROTECTED> |
| 22:32.05 | Darwin35 | <PROTECTED> |
| 22:32.07 | Darwin35 | <PROTECTED> |
| 22:32.09 | Darwin35 | <PROTECTED> |
| 22:32.11 | Darwin35 | <PROTECTED> |
| 22:32.13 | Darwin35 | <PROTECTED> |
| 22:32.15 | Darwin35 | <PROTECTED> |
| 22:32.17 | Darwin35 | <PROTECTED> |
| 22:32.21 | Darwin35 | <PROTECTED> |
| 22:32.23 | Darwin35 | <PROTECTED> |
| 22:32.25 | Darwin35 | <PROTECTED> |
| 22:32.27 | Darwin35 | <PROTECTED> |
| 22:32.29 | Darwin35 | <PROTECTED> |
| 22:32.30 | *** kick/#asterisk [Darwin35!n=russell@asterisk/developer-and-stable-maintainer/drumkilla] by russellb (russellb) |
| 22:32.37 | jontow | woi, paste-o |
| 22:32.43 | *** join/#asterisk Darwin35 (n=Darwin@c-24-9-75-234.hsd1.co.comcast.net) |
| 22:32.45 | Darwin35 | <PROTECTED> |
| 22:32.47 | Darwin35 | esac |
| 22:32.51 | Darwin35 | <PROTECTED> |
| 22:32.53 | Darwin35 | rc.d/zaptel.sh |
| 22:32.55 | Darwin35 | shit |
| 22:32.56 | justinu | nice |
| 22:32.57 | Darwin35 | sorry |
| 22:32.59 | Darwin35 | I thought it all went pvt |
| 22:33.03 | shmaltz | wtf? |
| 22:33.13 | FLeiXiuS | dubass. |
| 22:33.16 | FLeiXiuS | dumbass* |
| 22:34.33 | justinu | lol |
| 22:34.45 | remiss | To: <sip:1@10.45.0.3:5060>;tag=as26483eee |
| 22:35.01 | remiss | should'nt this be going to extension 1 in the context specified for the sip-device? |
| 22:35.21 | remiss | i just get 404 not found from asterisk :-/ |
| 22:36.21 | jontow | hmm |
| 22:37.32 | *** join/#asterisk AlexCTI (i=AlexCTI@30.sub-70-219-121.myvzw.com) |
| 22:38.32 | AlexCTI | Hi, someone has used GNUDialer? |
| 22:39.46 | lunaphyte | how can i make asterisk tell me more about a sip connection when it occurs? |
| 22:39.51 | *** join/#asterisk Sternn (n=sternn@user-0c938ku.cable.mindspring.com) |
| 22:39.52 | remiss | sip debug |
| 22:40.43 | IronHelix | or sip debug peer (peername) |
| 22:40.49 | IronHelix | or sip debug ip (address) |
| 22:40.59 | lunaphyte | perfect. thanks. |
| 22:41.15 | justinu | remiss, you can use that to debug your problem too |
| 22:41.31 | lunaphyte | hmm. sip looks like smtp headers. |
| 22:41.39 | justinu | it's an HTTP style protocol |
| 22:41.45 | remiss | justinu: all i can see of reasonable things from that is the 404 |
| 22:42.02 | IronHelix | also if you dont already have it, set verbose 3 can be most helpful |
| 22:42.06 | justinu | there's a debug line output that states what context it's looking for what extension |
| 22:42.36 | justinu | it gets output somewhere in between the invite, and the 404 |
| 22:43.37 | *** join/#asterisk kpettit (n=keith@69.15.174.114) |
| 22:43.43 | lunaphyte | who sends INVITE? the client or server? |
| 22:43.56 | justinu | depends on which way you're calling |
| 22:44.04 | dezent | this is really frustrating.. i got problems with echo.. when using meetme i get like 2 seconds lag and echo making it impossible to use.. its both sip->asterisk<-sip and pstn->asterisk<-pstn does anyone have and idea what could be done ? |
| 22:44.23 | justinu | dezent make sure you're using 20ms RTP packets |
| 22:44.24 | lunaphyte | it's a sip call coming from a did provider to exten@myasteriskbox |
| 22:44.34 | dezent | justinu: rtp.conf ? |
| 22:44.45 | IronHelix | also check your endpoints- possibly is the phone creating echo on the handset (grandstream)? |
| 22:44.47 | justinu | no, it's a setting on the sip UA |
| 22:45.09 | remiss | Looking for 1 in phone (domain 10.45.0.3) |
| 22:45.21 | remiss | okay... |
| 22:45.35 | dezent | justinu: ok |
| 22:46.04 | justinu | lunaphyte: your did provider sends you the invite when someone calls your DID |
| 22:46.05 | remiss | damn.. is it possible... |
| 22:46.09 | remiss | [phone] |
| 22:46.09 | remiss | include = phone |
| 22:46.12 | remiss | *stupid* |
| 22:46.16 | justinu | lol |
| 22:46.52 | lunaphyte | justinu: what are they "inviting" me to do by sending that? |
| 22:47.01 | justinu | inviting you to accept a phone call |
| 22:47.05 | lunaphyte | ah |
| 22:49.42 | *** part/#asterisk tangle_eye (n=alexthew@rrcs-24-227-188-230.sw.biz.rr.com) |
| 22:51.23 | lunaphyte | is it 'typical' that the source ip/port are consistent when a did provider connects to me? |
| 22:53.21 | *** join/#asterisk xachen (i=mo202@pdpc/supporter/student/xachen) |
| 22:53.22 | remiss | justinu: thanks, champ :) |
| 22:53.25 | xachen | aha |
| 22:53.25 | xachen | found the problem |
| 22:53.37 | xachen | root@magnum:/usr/local/etc# kldunload wcte11xp.ko |
| 22:53.41 | xachen | and now the box has crashed |
| 22:55.05 | lunaphyte | if i'm only getting inbound called from a did provider, i don't need/want it to register with me, do i? |
| 22:55.14 | lunaphyte | s/called/calls/ |
| 22:55.31 | lunaphyte | that's cool. |
| 22:55.55 | justinu | remiss: np |
| 22:56.09 | justinu | lunaphyte: not necessarily typical |
| 22:56.16 | lunaphyte | ok |
| 22:56.43 | justinu | your provider doesn't need to register with you to make calls to you |
| 22:58.01 | *** join/#asterisk tdonahue (n=tdonahue@208.51.101.201) |
| 22:59.55 | *** join/#asterisk ibob63 (n=hp@88-96-66-62.dsl.zen.co.uk) |
| 23:00.35 | ibob63 | does anyone know any distros which come with 1.2 rather than 1.0 ? |
| 23:00.36 | Darwin35 | youhave toregister with the provider to get calls from the provider |
| 23:01.29 | Darwin35 | dont ask @home questions |
| 23:01.35 | Darwin35 | use real asterisk |
| 23:01.41 | *** join/#asterisk Tamarisk (n=adrian@user-2899.lns1-c8.dsl.pol.co.uk) |
| 23:01.51 | Darwin35 | and 1.2.4 iss the best |
| 23:01.52 | MGSsancho | lol |
| 23:04.28 | Tamarisk | Basic simple question I hope! After downloading and installing asterisk,,, I then find out about asterisk@home which has a web base GUI. Is the a web GUI for asterisk that will help a mere beginner? |
| 23:04.44 | IronHelix | the web gui is a crutch |
| 23:04.52 | IronHelix | you will learn more by mucking through the config files |
| 23:04.54 | IronHelix | also read the |
| 23:04.56 | IronHelix | ~book |
| 23:04.58 | jbot | somebody said book was a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 23:05.18 | Tamarisk | OK keeping it simple for me will I have crutch with 1.2.4 or can I download seperatly |
| 23:05.21 | dezent | Tamarisk: better to learn first and use gui later.. http://www.voip-info.org/wiki/ |
| 23:05.41 | Tamarisk | I have the book |
| 23:05.42 | dezent | Tamarisk: after you learned you probably wont need a gui |
| 23:05.44 | IronHelix | aah is nice if you want a system up and running really really fast and dont want to 'deal with anything', but if you watn to learn i highly recommend learning the config files |
| 23:05.45 | Tamarisk | <PROTECTED> |
| 23:06.02 | IronHelix | agreed dez |
| 23:06.03 | Tamarisk | I would like to have some success early to help keep me going |
| 23:06.44 | dezent | i dont think ive been reading this much howtos since i started with linux =) |
| 23:07.54 | Tamarisk | I am reading the book, lots of pages and I am sure it will be a very good ref. |
| 23:08.11 | Tamarisk | So If I have eread] |
| 23:08.15 | Katty | hmm. |
| 23:08.21 | Tamarisk | read your comments correctly |
| 23:08.27 | dezent | work will bye me one next month |
| 23:08.30 | IronHelix | as soon as you get the whole context-extension-priority-application thing you're awesome |
| 23:08.34 | dezent | really looking forward to it |
| 23:08.54 | dezent | hmm buy |
| 23:09.03 | dezent | hihi |
| 23:10.36 | Katty | who wrote jbot? |
| 23:10.37 | Tamarisk | So there is crutch and aah, both gui's is that correct ? |
| 23:10.45 | Katty | anthm: did you write jbot? |
| 23:11.00 | Qwell[] | Katty: it's an infobot...owned by tim riker |
| 23:11.09 | Qwell[] | ~timriker |
| 23:11.10 | jbot | timriker is, like, my owner http://rikers.org/ mailto:Tim@Rikers.org mailto:TimR@Debian.org maintainer of BZFlag, member of a ton of open source projects http://www.advogato.com/person/timriker/ http://sourceforge.net/users/timriker/ the guy who GPL'd SCO's ABI files, giving every Linux user the right to use them ;-), or a very cool guy. |
| 23:11.11 | Katty | so the /whois says. |
| 23:11.34 | Qwell[] | ~jbot |
| 23:11.35 | jbot | i heard jbot is only marginally useful at best, or a silly little bugger |
| 23:11.35 | Katty | then i suppose he isn't in the channel. |
| 23:11.49 | Qwell[] | yeah...no clue...never seen him |
| 23:12.03 | *** join/#asterisk tdonahue (n=tdonahue@208.51.101.201) |
| 23:12.18 | Qwell[] | ~darthclue |
| 23:12.34 | Qwell[] | jbot darthclue is such a newb! |
| 23:12.35 | jbot | okay, Qwell[] |
| 23:12.38 | Qwell[] | ;] |
| 23:12.39 | Katty | DarthClue: :>> |
| 23:12.50 | dezent | hehe |
| 23:12.56 | *** part/#asterisk dpolitech (n=Owner@207.224.48.130) |
| 23:15.34 | [TK]D-Fender | *yawn* |
| 23:15.38 | [TK]D-Fender | Katty: Mew. |
| 23:15.48 | Katty | mew. |
| 23:16.26 | *** part/#asterisk JCC_ (n=john@207.41.92.131) |
| 23:16.54 | *** join/#asterisk Eggplant (n=none@208.193.217.216.cascadeaccess.com) |
| 23:21.51 | dezent | if i throw a smoothie out the window it would be ice before hiting the ground |
| 23:24.13 | cypromis | than just don'tr |
| 23:24.58 | dezent | good advice |
| 23:25.02 | dezent | thank you |
| 23:26.24 | DarthClue | it would melt here. I think I prefer the melting. |
| 23:26.26 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 23:26.35 | dezent | me too |
| 23:26.48 | dezent | i think its -15c here |
| 23:28.22 | cypromis | just -10c here |
| 23:28.36 | DarthClue | about 74f here...23c |
| 23:29.30 | dezent | aww just about 4 month away... 23C that is |
| 23:30.20 | dezent | better spend 4 month playing with * |
| 23:31.00 | dezent | to bad there isnt any nice looking iax softphones for osx |
| 23:31.13 | dezent | would be nice with eyebeam talking iax |
| 23:32.04 | DarthClue | it'll be close to 38C in another 4 months. |
| 23:32.16 | dezent | thats to hot =) |
| 23:32.48 | DarthClue | not with a good a/c unit. besides, it'll be a dry heat which is better than what I'm used to. |
| 23:32.54 | *** join/#asterisk CrashHD (n=crashhd@c-24-7-168-46.hsd1.ca.comcast.net) |
| 23:33.10 | dezent | where do you live ? |
| 23:33.53 | DarthClue | 1 Hour from the Texas/Mexico border in the middle of nowhere. |
| 23:34.06 | dezent | sweet |
| 23:34.27 | tuxinator_linux | DarthClue: people really live there? |
| 23:35.13 | DarthClue | Yeah. Although it is more like hiding out than living. Civilization stops before it ever begins. |
| 23:35.48 | dezent | haha |
| 23:35.52 | *** join/#asterisk g4m (n=g4m@dsl253-014-003.sba1.dsl.speakeasy.net) |
| 23:36.25 | dezent | aslong as you got adsl there is civilization |
| 23:36.38 | *** part/#asterisk Tamarisk (n=adrian@user-2899.lns1-c8.dsl.pol.co.uk) |
| 23:36.40 | g4m | how would you rewrite your outbound CallerID on a PRI. Do you have to create a zapata group that is used by the Dial? |
| 23:36.52 | DarthClue | i have to dodge donkeys, cows, chickens, coyotes, rabbits, skunks, horses, jack-rabbits, deer, and a few other critters while driving. |
| 23:37.21 | dezent | we only got elks and badgers |
| 23:37.23 | dezent | almost |
| 23:37.43 | tuxinator_linux | DarthClue: you forgot 'border-crossers' |
| 23:37.49 | DarthClue | that is one of the few advantages, I actually get decent speeds cause of my proximity to the local carrier station. |
| 23:38.11 | DarthClue | we don't dodge them, they don't hurt the bus like the other things do. |
| 23:38.11 | dezent | how fast do you get ? |
| 23:38.18 | tuxinator_linux | DarthClue: ;-) |
| 23:38.35 | dezent | i got about 2.1MB / sec down =) |
| 23:38.42 | dezent | 300 meters from the station |
| 23:38.47 | DarthClue | I get close to 3 / sec down |
| 23:39.22 | tuxinator_linux | DarthClue: that's a lot of border-crossers you take down |
| 23:39.54 | DarthClue | at $500 a head, I'm raking in the dough too. |
| 23:40.10 | tuxinator_linux | hey Katty, thanks for the hug ;-) |
| 23:40.17 | Katty | (= |
| 23:40.19 | *** join/#asterisk sfChrisJacob (n=cjacob@70-32-0-48.frdrmd.adelphia.net) |
| 23:40.28 | dezent | :/ |
| 23:40.32 | tuxinator_linux | Katty: Feeling better? |
| 23:40.32 | Katty | :> |
| 23:40.35 | Katty | no :< |
| 23:40.44 | Katty | i shan't for a long time though |
| 23:40.57 | tuxinator_linux | DarthClue: If I die from a katty hug, I'd die happy |
| 23:41.06 | dezent | awww |
| 23:41.10 | DarthClue | tuxinator_linux, we all would. |
| 23:41.21 | Katty | DarthClue: i had my wisdom teeth cut out friday. |
| 23:41.26 | *** part/#asterisk ibob63 (n=hp@88-96-66-62.dsl.zen.co.uk) |
| 23:41.28 | Katty | DarthClue: all of them |
| 23:41.41 | Katty | i'm in post-chipmunk recovery stage. |
| 23:42.01 | DarthClue | I bet they won't be calling you about the windoze servers this week...if they are smart. |
| 23:42.08 | Katty | i'm at work. |
| 23:42.25 | sfChrisJacob | Hey guys, quick question about PRI cableing.... I have the "box" from Verizon at the dmarc... I plug in a CAT5 cable with a normal plug on one side and punch the oter into a 66 block. At the 66 block on the other end I do the reverse to the jack on the Asterisk box. Will normal eathernet cabling work? |
| 23:42.34 | Katty | just not talking much..or singing at my desk... |
| 23:42.42 | Katty | or wandering around the office spreading joy, etc. |
| 23:42.54 | Katty | DarthClue: oh, and i got my hair cut. |
| 23:42.57 | Katty | DarthClue: 11.5 inches. |
| 23:43.05 | DarthClue | off or left on? |
| 23:43.09 | Katty | off |
| 23:43.13 | Katty | i donated it. |
| 23:43.40 | tuxinator_linux | Short haired katty now |
| 23:43.44 | Katty | not really. |
| 23:43.49 | Katty | it's to my shoulderblades. |
| 23:43.58 | tuxinator_linux | that was pretty long before |
| 23:44.08 | Katty | it was to my hips before :P |
| 23:44.19 | tuxinator_linux | crazy, fun to wash I bet |
| 23:45.20 | Katty | mm, subway. |
| 23:45.30 | Katty | i miss subway......and regular food :< |
| 23:45.34 | [av]bani | o_O |
| 23:45.36 | [av]bani | O_o |
| 23:45.43 | sfChrisJacob | I have been googleing around, but I can't find an answer... can someone point me in the right direction? |
| 23:45.49 | Katty | sfChrisJacob: left. |
| 23:45.51 | Qwell[] | Katty: Got a blender? |
| 23:45.57 | Katty | Qwell[]: no |
| 23:46.03 | Qwell[] | no subway for you then... ;/ |
| 23:46.05 | Katty | Qwell[]: i have a fork and a potato masher! |
| 23:46.10 | Katty | Qwell[]: :<< |
| 23:46.12 | DarthClue | i don't miss subway, but i do miss real food. |
| 23:46.19 | tuxinator_linux | mm, taters |
| 23:46.22 | Qwell[] | I hate you guys |
| 23:46.24 | Qwell[] | I really do |
| 23:46.27 | Katty | Qwell[]: don't hate. |
| 23:46.32 | Qwell[] | now I have to go to subway tonight |
| 23:47.00 | Katty | you're clearly suffering from post-operative lack-of-hug syndrome. |
| 23:47.18 | Katty | also known as polohs. |
| 23:47.21 | Qwell[] | lack-of-subway syndrome |
| 23:47.39 | *** join/#asterisk MRH2 (n=Mr_happy@fcirc-adsl.demon.co.uk) |
| 23:47.39 | Qwell[] | DarthClue: eh? |
| 23:47.41 | Katty | Qwell[]: eat a veggie one for me :> |
| 23:47.47 | Katty | Qwell[]: no cheese. |
| 23:47.53 | Qwell[] | Katty: meatball...with extra meat ;/ |
| 23:47.58 | Katty | Qwell[]: with vegetable oil, oregano, and black pepper. |
| 23:48.03 | Qwell[] | my wife might though :p |
| 23:48.27 | MRH2 | is cdr_psql hosed in 1.2 stable branch? |
| 23:48.33 | Katty | Qwell[]: k :>> |
| 23:48.38 | MRH2 | cdr_pgsql |
| 23:49.15 | Katty | i should go home |
| 23:49.42 | DarthClue | you should, really, now, leave already, take the blender and go home. leave the salt. |
| 23:49.57 | Katty | i can't leave the salt. i have to rinse with that stuff. |
| 23:50.05 | Qwell[] | salt + open wounds...FUN! |
| 23:50.21 | Katty | salt water is actually nice on it. |
| 23:50.47 | MRH2 | my bet is this http://lists.digium.com/pipermail/svn-commits/2006-February/010923.html |
| 23:50.58 | [av]bani | Qwell[]: salt water doesnt hurt... thats part of why you use salt |
| 23:51.15 | MRH2 | undefined symbol crypt on the console |
| 23:51.23 | [av]bani | distilled water hurts |
| 23:52.10 | Katty | these heels hurt |
| 23:56.18 | *** join/#asterisk voip470 (n=A_mail@71.244.98.67) |
| 23:59.09 | rajiv | odd problem. inbound call via zap to a sip phone. no prob. second inbound call via sip to the same sip phone. call waiting beep. press flash. i cannot hear the calling party but they can hear me. any ideas why? |
