| 00:00.10 | john-eman0n | anyone have a IAX account i can use temp to test spoofing via IDEFisk? |
| 00:00.11 | lee_is_me | I am just calling the number real quick and hanging up so that might account for the lack of CID |
| 00:00.44 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 00:01.58 | lee_is_me | That fixed it: WaitForRing(1) on first line before Answer() |
| 00:06.36 | *** join/#asterisk [[blah]asfd (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net) |
| 00:06.40 | [[blah]asfd | ok, so why is it that when I set incominglimit=1 it affects the outgoing calls also? |
| 00:07.54 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 00:11.14 | *** join/#asterisk ManxPower (n=manxpowe@dpc67142183150.direcpc.com) |
| 00:14.47 | MrWup | if my extension is exten =>_*1XX,1,Goto(voicemail,s,1) |
| 00:14.58 | MrWup | is there any way i can get the 1XX ? |
| 00:15.04 | MrWup | like the bit after the star? |
| 00:15.15 | MrWup | without having to parse it through some external script? |
| 00:17.10 | [[blah]asfd | ${EXTEN:1} |
| 00:20.21 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 00:20.29 | MrWup | that doesnt do it |
| 00:20.32 | MrWup | just returns blank |
| 00:20.33 | alrs | exit |
| 00:21.16 | MrWup | if someone type *101 and triggers exten =>_*1XX,1, |
| 00:21.19 | MrWup | i want to return 101 |
| 00:23.32 | Nugget | if ${EXTEN:1} doesn't do that you should file a bug with Digium. |
| 00:24.36 | john-eman0n | anyone have a IAX account i can use temp to test spoofing via IDEFisk? |
| 00:26.00 | blitzrage | sure, connect to 127.0.0.1 |
| 00:27.15 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 00:27.19 | *** part/#asterisk [[blah]asfd (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net) |
| 00:28.08 | *** join/#asterisk bkw__ (n=brian@adsl-70-143-39-83.dsl.tul2ok.sbcglobal.net) |
| 00:28.41 | MrWup | oh |
| 00:28.43 | DrukenLPY | anyone got a sipura provisioning server? |
| 00:28.44 | MrWup | the problem is this... |
| 00:29.10 | MrWup | im in context1 and i do SET (__MYVAR="101") then goto (context2,s,1) |
| 00:29.20 | MrWup | then in context2 i try and access ${MYVAR} |
| 00:29.22 | MrWup | and i get blank |
| 00:30.12 | Nugget | that sounds like a completely different problem than the one you were just talking about. |
| 00:30.28 | MrWup | well this one is probably the real one |
| 00:30.45 | MrWup | i.e. ${EXTEN:1} probably works... but for some reason i cant transport it to another context |
| 00:30.46 | shmaltz | MrWup, have you tried ${__MYVAR} |
| 00:31.09 | Nugget | I'm not familiar with your use of __ preceding the variable name, but minimally I'd expect you to have to be consistent. |
| 00:31.21 | Nugget | ${__MYVAR} would perhaps work |
| 00:31.30 | MrWup | nope, same thing |
| 00:31.37 | MrWup | blank when i try and get it from a diff context |
| 00:31.40 | Nugget | perhaps __ is interfering with the Set()? |
| 00:32.06 | MrWup | have tried it without as well |
| 00:32.09 | shmaltz | MrWup, try not putting those __ |
| 00:32.20 | shmaltz | MrWup, what version? |
| 00:32.29 | MrWup | exten =>_*1XX,1,Set(VBOX="101") exten =>_*1XX,2,Goto(voicemail,s,1) |
| 00:32.39 | MrWup | then in [voicemail] i do |
| 00:32.42 | MrWup | ...exten => s,2,Playback(/mnt/telecomstore/voicemail-greetings/${VBOX}) |
| 00:34.01 | Nugget | You've got a long road ahead of you if your plan is to write your own voicemail system. :) |
| 00:34.18 | MrWup | its not that hard |
| 00:34.19 | shmaltz | MrWup try this: |
| 00:34.22 | shmaltz | Set(VBOX=/mnt/telecomstore/voicemail-greetings/101 |
| 00:34.23 | shmaltz | then Playback(${VBOX}) |
| 00:34.26 | MrWup | ok |
| 00:34.27 | bkw__ | MrWup, to do it right it is |
| 00:35.34 | MrWup | shmaltz, now it says playback has no arguments |
| 00:35.51 | Nugget | MrWup: what version of asterisk? |
| 00:35.57 | MrWup | 1.4.` |
| 00:35.58 | MrWup | 1.4.1 |
| 00:35.59 | shmaltz | MrWup, which means it doesnt see the variable |
| 00:36.05 | Nugget | hrm, ok. |
| 00:36.52 | Nugget | it's perfectly stable except when it isn't. :) |
| 00:37.34 | MrWup | should i define vbox as a global? |
| 00:38.11 | Nugget | personally I think you should use comedian mail, but in either case no I wouldn't expect that to make a difference. |
| 00:38.51 | shmaltz | MrWup, why not comedian mail? |
| 00:39.00 | shmaltz | you can use in conjuction with your fancy stuff |
| 00:39.00 | Nugget | to debug you can stick a bunch of NoOp() statements in to help you figure out where asterisk is jumping the tracks and losing your variable. |
| 00:39.26 | shmaltz | for instanse you can always play the recorded name for callers before dialing an extension using vm_path |
| 00:41.00 | MrWup | omg |
| 00:41.01 | MrWup | im so |
| 00:41.02 | MrWup | so |
| 00:41.03 | MrWup | stupid |
| 00:41.08 | MrWup | so stupid |
| 00:41.15 | MrWup | i had an exten of the same number just above |
| 00:41.19 | MrWup | which directed me somewhere else |
| 00:41.22 | shmaltz | rofl |
| 00:41.24 | Nugget | heh |
| 00:42.06 | shmaltz | MrWup, you didn't see the errro on reload???????????? |
| 00:42.27 | aptura | any pros or cons with this wifi router? WRT54GR |
| 00:42.37 | aptura | This unit has a range extender in it. |
| 00:42.40 | MrWup | my voicemail system is going to be really simple actually |
| 00:42.48 | MrWup | just a menu to let people record their greeting |
| 00:42.55 | MrWup | and then a voicemail so people can leave a message |
| 00:43.05 | MrWup | then the message is simply sent as mp3 to the person's email account |
| 00:43.07 | shmaltz | MrWup, and how will you keep track of deleted VM? |
| 00:43.09 | Nugget | That's precisely why we're suggesting you use comedian mail. |
| 00:43.15 | shmaltz | what about vm to email? |
| 00:43.19 | MrWup | vm? |
| 00:43.26 | Nugget | your system is going to be simple and a pain in the ass to use. :) |
| 00:43.39 | Nugget | it's like saying "but mowing the lawn with scissors is simple!" |
| 00:44.02 | MrWup | erm |
| 00:44.10 | MrWup | well |
| 00:44.21 | shmaltz | Nugget, BTW, it is simple to mown the lawn with scissors :P |
| 00:44.23 | MrWup | the way its going to work is... |
| 00:44.39 | MrWup | person leave a vm, the vm is saved to file and indexed in a mysql database |
| 00:44.56 | Nugget | I never said I wasn't. I was just pointing out that often "simple" shouldn't be the goal. |
| 00:44.59 | MrWup | after its been left, it is sent once to the person's email address |
| 00:45.26 | MrWup | and of course the voicemail is also resident in a file on the server indexed by mysql so u admins can check it later for abuse etc |
| 00:45.34 | *** join/#asterisk webman (n=chatzill@200.179.233.220.exetel.com.au) |
| 00:45.59 | MrWup | have been writing this system for several months now |
| 00:46.04 | MrWup | its really fin leet =] |
| 00:46.09 | MrWup | voicemail is just the last step |
| 00:46.14 | JT | and simple, of course |
| 00:46.23 | Nugget | so simple it only took months to write. :) |
| 00:46.30 | MrWup | well |
| 00:46.33 | MrWup | its simple for the users |
| 00:46.35 | MrWup | complex for me |
| 00:46.45 | De_Mon | what are the callerID options ANI DNID and RDNIS used for? |
| 00:46.49 | MrWup | have coded a windows service in delphi |
| 00:46.56 | MrWup | and a win32 client app in delphi |
| 00:47.04 | MrWup | done all the database stuff in the background in mysql |
| 00:47.18 | MrWup | written web management in php |
| 00:47.46 | MrWup | and used agi-bin php scripts to expose certain events |
| 00:47.53 | MrWup | others done through the management interface |
| 00:48.03 | MrWup | its such a leet system =] |
| 00:48.05 | CrashSys | Nice... state of florida doesn't require licensing for datacom... |
| 00:48.24 | *** join/#asterisk iBuMp (n=iBuMp@cpe-66-68-37-190.austin.res.rr.com) |
| 00:49.22 | MrWup | anyway |
| 00:49.25 | MrWup | thanks for the help! |
| 00:49.26 | MrWup | night! |
| 00:52.39 | *** join/#asterisk kavit (n=kavit@ppp167-236-231.static.internode.on.net) |
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| 01:37.41 | _VoiceMeUp_COM | BTW |
| 01:37.48 | _VoiceMeUp_COM | check out for a new 514-310 rate center |
| 01:37.54 | _VoiceMeUp_COM | its not on localcalling guides |
| 01:45.06 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 01:45.11 | *** join/#asterisk greendisease (n=jack@fedora/greendisease) |
| 01:45.43 | De_Mon | [TK]D-Fender Finished! thanks again for your help and patience |
| 01:50.47 | john-eman0n | anyone have a IAX account i can use temp to test spoofing via IDEFisk? |
| 01:52.05 | [TK]D-Fender | De_Mon, Good to hear, and great timing |
| 01:53.34 | *** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
| 01:54.43 | mosty | if there is an error while loading chan_zap, asterisk exits. is it possible to get asterisk to continue loading without chan_zap instead of dying? |
| 01:55.01 | JT | noload chan_zap.so |
| 01:55.28 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 01:56.35 | mosty | i don't want it disabled |
| 01:56.49 | mosty | i just want asterisk to continue loading if there is a problem with zap/wanpipe |
| 01:58.11 | mosty | ie, if it loads successfully then great, if not asterisk should still run |
| 01:58.20 | [TK]D-Fender | mosty, Not happening. if it blows, you're toast |
| 02:00.07 | De_Mon | why is my CID working for a landline but the cellphone only shows number? |
| 02:00.32 | De_Mon | im going thru an ITSP for both calls |
| 02:00.46 | De_Mon | (internet telephone service provider) |
| 02:00.52 | De_Mon | *the same* |
| 02:00.56 | mosty | [TK]D-Fender, that's unfortunate. think it's worth submitting a wishlist bug report and/or patch? |
| 02:00.57 | JT | telephony to be exct |
| 02:01.03 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 02:01.03 | JT | exact |
| 02:01.09 | De_Mon | mm |
| 02:01.49 | [TK]D-Fender | mosty, go for it.. it'd be patch, not a "bug" |
| 02:01.51 | JT | mosty: you thinking L1 failover box? |
| 02:01.56 | mosty | De_Mon, i'd say you have to ask your ITSP |
| 02:01.59 | [TK]D-Fender | mosty, So none of us will see it until 1.6 |
| 02:02.26 | mosty | JT, no i just want sip/iax calls to work if chan_zap fails to load |
| 02:02.30 | [TK]D-Fender | JT : no if wanpipe fails to load chan_zap will bomb out taking * with it,. |
| 02:02.46 | [TK]D-Fender | JT : he wants it to skip on over and resume business as usual |
| 02:02.46 | JT | mosty: how often does that happen? |
| 02:03.46 | mosty | JT, as often as someone forgets to recompile wanpipe after a kernel upgrade, heh |
| 02:04.14 | JT | that doesn't sound like a very big need for such a patch in a production environment then ;) |
| 02:05.26 | mosty | i just don't see why asterisk can't print an error message and continue running |
| 02:05.59 | JT | somtimes it needs to be made obvious to people when there's an error, to be honest |
| 02:06.35 | kavit | hey has anyone used the Aastra 55i with Asterisk?? what are the phones like? any caveats? |
| 02:06.40 | mosty | JT, my dialplan tells me if zap calls fail, i can also grep the asterisk log etc |
| 02:09.29 | *** join/#asterisk Coder365_ (i=Coder365@rd02-079.cbpu.com) |
| 02:10.19 | JT | mosty: still, if your case to have this feature is that you sometimes forget to recompile wanpipe, doesn't look very strong ;) |
| 02:11.52 | Coder365_ | okay guys, I'm new at this asterisk thing, and I have successfully managed to get 2 extensions to register...problem is I can't call between the two, any ideas? |
| 02:12.18 | mosty | JT, asterisk still loads if there are errors in sip.conf, same with iax.conf |
| 02:12.33 | JT | an idea would be to tell us what you've done that you hope to allow them to call each other |
| 02:13.22 | Coder365_ | lol, okay... I've added lines in extensions.conf that should say call this number |
| 02:13.28 | ez` | anyone suggest a wakeup call script ? |
| 02:13.53 | mosty | Coder365_, set verbose 3 then look at the console to see what happens when you try calling |
| 02:14.00 | Coder365_ | one sec |
| 02:14.11 | mosty | ez`, call files could work |
| 02:14.24 | ez` | call files ? |
| 02:14.38 | mosty | ez`, google "asterisk call files" |
| 02:14.58 | ez` | yep i am checking .. ;) |
| 02:15.06 | Coder365_ | mosty, I have the verbostiy at 7, how do I punch it down? |
| 02:15.34 | mosty | what do you mean by "punch it down" ? |
| 02:15.44 | Coder365_ | adjust it |
| 02:15.47 | Coder365_ | from 7 to 3 |
| 02:15.51 | Coder365_ | I did asterisk -rvvv |
| 02:16.30 | mosty | just leave it at 7. paste the result on a paste site. ("set verbose 3" at the asterisk console would do) |
| 02:16.37 | Coder365_ | k |
| 02:16.57 | JT | set verbose 10 is all you ever need |
| 02:18.57 | Sweeper | lies |
| 02:19.03 | Sweeper | 100000 is BETTAR |
| 02:19.22 | JT | ;) |
| 02:19.32 | *** part/#asterisk bapril (n=bapril@pool-70-109-158-237.cncdnh.east.verizon.net) |
| 02:21.00 | Coder365_ | mosty any paste sites you know of? |
| 02:21.55 | mosty | google for pastebin or nopaste |
| 02:22.13 | ManxPower | Jun 7 21:01:53 WARNING[2056]: chan_zap.c:1551 zt_enable_ec: Unable to enable echo cancellation on channel 6 |
| 02:22.14 | ManxPower | HUH? |
| 02:22.31 | JT | ~pb |
| 02:22.43 | jbot | i guess pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or at paste.lisp.org |
| 02:24.49 | Coder365_ | masty http://pastebin.4programmers.net/2505 |
| 02:27.40 | *** join/#asterisk mightnare (n=mike@s230165.ppp.asahi-net.or.jp) |
| 02:29.23 | mosty | cader357: set verbose 10 (not verbosity), and also do "sip no debug", then try again |
| 02:29.53 | Coder365_ | k |
| 02:30.49 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 02:31.06 | Coder365_ | nothing |
| 02:31.34 | Coder365_ | nothing popped up |
| 02:31.48 | mosty | is the context in sip.conf correct? |
| 02:32.02 | mosty | it should send calls to a particular context in your dialplan |
| 02:33.12 | Coder365_ | well, I have it set to internal |
| 02:33.17 | Coder365_ | I think, hold on |
| 02:33.22 | Coder365_ | I may not have set that up |
| 02:35.59 | Coder365_ | got it |
| 02:36.00 | Coder365_ | thanks |
| 02:49.55 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 02:50.48 | *** join/#asterisk Idle (n=brian@S010600a024969312.ed.shawcable.net) |
| 02:50.56 | Idle | how do I terminate a call from the CLI app? |
| 02:52.05 | wunderkin | ManxPower, hpec? analog? how many channels? |
| 02:52.07 | *** join/#asterisk Aquavette (n=noone@74-137-181-254.dhcp.insightbb.com) |
| 02:52.23 | Aquavette | Anyone have any recommendations for how to do 56K modem RAS connections through Asterisk |
| 02:53.02 | Idle | heres a better idea... how can I make an extension that hangs up channels |
| 02:53.25 | Aquavette | hows that a better idea? |
| 02:53.55 | Idle | no no |
| 02:53.58 | Idle | not for you, ;) |
| 02:54.01 | Idle | for my question |
| 02:54.01 | Aquavette | sorry mate |
| 02:54.09 | wunderkin | Idle, soft hangup |
| 02:54.23 | Idle | wunderkin: yea, figured that out :P but can I make an extension that does that? |
| 02:55.00 | Idle | so, my parents (who routinely bridge our 2 external phone lines, and yell like banshees when they do), can dial '666' and itll just hangup all channels |
| 02:55.08 | *** part/#asterisk Rusty1 (n=Rusty1@cpe-72-226-96-74.nycap.res.rr.com) |
| 02:56.09 | *** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
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| 03:18.51 | *** join/#asterisk covici1 (n=covici@ccs.covici.com) |
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| 03:21.17 | *** join/#asterisk techman97_andy (n=me@75-134-128-138.dhcp.roch.mn.charter.com) |
| 03:21.33 | techman97_andy | hey all - can anyone give me a recommendation on a 2-port T1 card (PCI or external?) |
| 03:22.00 | JT | external wouldn't be a card.. |
| 03:22.05 | techman97_andy | yeah yeah |
| 03:22.06 | JT | it'd be a PRI to SIP gateway |
| 03:22.06 | techman97_andy | :P |
| 03:22.28 | techman97_andy | my butchering of the terms aside, any recommendations on a gateway / PCI card? |
| 03:22.31 | JT | your main choices on card is digium or sangoma |
| 03:22.43 | techman97_andy | I'm looking at the Sangoma A102? |
| 03:22.44 | *** join/#asterisk martinvandiest (n=chatzill@140.239.173.226) |
| 03:22.49 | JT | you should go for one with hardware echo cancellation |
| 03:22.55 | JT | the A102d has HWEC |
| 03:23.12 | techman97_andy | cool cool |
| 03:23.25 | [TK]D-Fender | techman97_andy, A102d or bust. |
| 03:23.39 | techman97_andy | what's the difference on the "d"? |
| 03:23.42 | JT | gateway will cost more, but should hopefully remove some of the headache |
| 03:23.47 | JT | echo cancellation |
| 03:23.50 | techman97_andy | gotcha |
| 03:23.51 | [TK]D-Fender | techman97_andy, Do it right the first time and you won't be looking to replace it right away |
| 03:24.02 | techman97_andy | [TK]D-Fender: that's why I'm on here asking...=) |
| 03:24.02 | Aquavette | anyone have a suggestion on how to connection a 56k modem to an RAS server through Asterisk |
| 03:25.17 | techman97_andy | rock on guys - thanks for the recommendation |
| 03:25.52 | JT | techman97_andy: gateways also allow you to have more reliable pri termination, in theory |
| 03:26.00 | JT | but they are quite expensive |
| 03:28.26 | martinvandiest | hello, can someone help me with a pri/asterisk voip turnup. pri calls are being show as rejected on the providers side when plugged into our asterisk box but when they plug in their test equipment the call were completed. |
| 03:29.07 | Aquavette | I'm trying to make a data modem call through Asterisk, is this even possible ? |
| 03:29.40 | JT | Aquavette: not really |
| 03:30.13 | Aquavette | damn dial up ras |
| 03:30.25 | mosty | Aquavette, why do you want to use asterisk for this? |
| 03:30.30 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 03:30.43 | Aquavette | my mom has to dialup to her work through RAS |
| 03:30.53 | Marshall- | whats a good provider for sip phones that lets you change CID? |
| 03:31.16 | Aquavette | and I've never had to mess with it before now, because they used to use all VPN, but there business went back to Remote Access Server through Dialup |
| 03:31.30 | Aquavette | and now she needs to dial in to her work, rather than vpn |
| 03:32.22 | mosty | Aquavette, yes but why do you want to use asterisk? |
| 03:32.35 | Aquavette | thats my home phone server |
| 03:32.42 | lee_is_me | Anyone seen wireless sets like those seen in fast food? Maybe something from plantronics or similar company? |
| 03:32.54 | Aquavette | i have a broadvoice phone line |
| 03:33.00 | Aquavette | and I have it connected as a SIP trunk |
| 03:33.10 | JT | Aquavette: yeah won't work |
| 03:33.11 | covici1 | hi anyone can help with a pri termination problem? |
| 03:33.13 | JT | you need a real phone line |
| 03:33.15 | JT | covici1: sure |
| 03:33.27 | mosty | Aquavette, the simplest thing to do would be to unplug the phone line from asterisk, then plug a modem into it instead, use regular ppp software |
| 03:33.50 | covici1 | OK, what happens is that the pri calls are rejected and if you call the number |
| 03:33.55 | Aquavette | i'm not using an FXS or FX0 module |
| 03:34.02 | mosty | Aquavette, you can't do ppp over sip, if that's what you're asking |
| 03:34.09 | Aquavette | yea PPP over SIP |
| 03:34.12 | Aquavette | i didn't think it was possible |
| 03:34.18 | mosty | correct |
| 03:34.19 | Aquavette | I was just confirming my suspicions |
| 03:34.20 | lee_is_me | I need to integrate Asterisk into a existing application using wireless headsets if possible. With what I've read on the lists lately, the result seem to be mixed with wifi... |
| 03:34.22 | covici1 | it says your call cannot be completed as dialed, but when the repair guy |
| 03:34.37 | covici1 | uses his test set you can call the number and his test set will ring. |
| 03:34.50 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 03:36.07 | *** join/#asterisk tsurko (n=tsurko@77.70.24.142) |
| 03:36.16 | JT | covici1: you need to provide a lot more info about your setup |
| 03:37.14 | covici1 | OK, its a rhino t1 card conneted to a cisco router and its actually voip, but its supposed to be pri. |
| 03:37.27 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:39.04 | covici1 | we have signalling=pri_cpe in zapata.conf |
| 03:39.05 | covici1 | we have switchtyhpe=national |
| 03:39.30 | covici1 | we have libpri installed. |
| 03:39.55 | john-eman0n | Anyone know a VOIP provider that allows you to spoof CPN/CID? |
| 03:40.33 | covici1 | the rhino modules is loaded -- how do you load the zaptel pri driver? |
| 03:40.36 | ManxPower | john-eman0n: most of them will let you do CID Number, but because of technical reasons you can't do CID Name. |
| 03:40.52 | ManxPower | Teliax is one the VoIP providers that suck less than most |
| 03:41.26 | ManxPower | PRI is a high level protocol. ASTERISK handles it, Zaptel just handles the low level stuff. |
| 03:41.33 | JT | covici1: not familiar with rhino, but you need to run ztcfg to load zaptel |
| 03:41.50 | covici1 | yep we did that. |
| 03:42.37 | *** join/#asterisk thoughtpolice (n=austin@c75-111-136-171.plaicmtc01.tx.dh.suddenlink.net) |
| 03:42.44 | *** part/#asterisk thoughtpolice (n=austin@c75-111-136-171.plaicmtc01.tx.dh.suddenlink.net) |
| 03:42.56 | covici1 | OK, but how does a call get rejectedand I don't see anyuthing in the logs at all? |
| 03:44.15 | mosty | covici1, do you see the call in the logs/console? |
| 03:44.28 | covici1 | no, that that is what is wierd. |
| 03:45.27 | covici1 | but the call gets dropped in such a way that you get the recordingabout your call cannot be completed as dialed. |
| 03:45.29 | mosty | do you have verbose and debug set to something like 10? |
| 03:45.39 | covici1 | tried that. |
| 03:45.53 | covici1 | I had debug = 2 only. |
| 03:46.00 | JT | covici1: pri intense debug span 1 |
| 03:46.11 | mosty | covici1, make sure verbose is set to 10 |
| 03:46.33 | JT | meh, verbose will not show you much if the card isn't setup right |
| 03:46.39 | covici1 | OK, I will try that. |
| 03:46.43 | JT | covici1: see if anything comes up in a pri debug |
| 03:46.55 | JT | if you, put it on a pastebin site for us to look at |
| 03:47.06 | mosty | JT, i usually check the simple things first |
| 03:47.06 | covici1 | OK, sounds like a plan. |
| 03:47.28 | covici1 | Thanks much . |
| 03:47.32 | JT | he seems to have already done that |
| 03:47.49 | covici1 | I did notknow about pri debug. |
| 03:48.46 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 03:48.57 | mosty | JT, they don't see the call in the logs at all, makes me think that the call might not be going through asterisk |
| 03:48.58 | covici1 | no such command as pri debug |
| 03:49.26 | covici1 | I think asterisk never sees the call. |
| 03:49.36 | JT | i said pri intense debug, but it sounds like chan_zap is not loaded |
| 03:49.53 | JT | what version of asterisk |
| 03:50.07 | covici1 | channel_zap is loaded and its 1.2 |
| 03:50.08 | mosty | covici1, is this call coming in on the pri, or going out on the pri? |
| 03:51.36 | JT | covici1: what is the exact command you typed in that failed? |
| 03:51.36 | covici1 | pri debug and I tried pri intense debug |
| 03:51.36 | JT | grr |
| 03:51.36 | JT | please try reading properly |
| 03:51.36 | JT | < JT> covici1: pri intense debug span 1 |
| 03:51.37 | JT | < JT> covici1: pri intense debug span 1 |
| 03:51.48 | JT | help pri |
| 03:51.53 | covici1 | but there is no pri anything. |
| 03:52.24 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 03:52.30 | JT | covici1: what order did you compile everything in? |
| 03:52.38 | covici1 | it says no such command s pri |
| 03:53.03 | covici1 | compiled zaptel, asterisk, libpri |
| 03:53.03 | JT | covici1: pastebin.ca zapata.conf and zaptel.conf |
| 03:54.45 | covici1 | pastebin.ca? |
| 03:54.57 | JT | type it into a web browser |
| 03:55.24 | covici1 | OK, hang on. |
| 03:58.24 | covici1 | does 1.2 have the pridebug? |
| 03:59.17 | JT | yes |
| 03:59.38 | covici1 | OK, then why do I not see such a thing? |
| 04:00.13 | JT | probably because chan_zap is not loaded or loaded properly |
| 04:00.22 | JT | are you going to pastebin those files for me already? |
| 04:01.31 | Qwell | what's a pride bug? |
| 04:01.46 | JT | lazy space bar |
| 04:03.11 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 04:03.43 | *** join/#asterisk Vorondi1 (n=vorondil@unaffiliated/vorondil) |
| 04:06.10 | *** join/#asterisk Cabal_ (n=Cabal@unaffiliated/cabal) |
| 04:06.15 | covici1 | pastebin.ca/549470 |
| 04:06.22 | Cabal_ | nick Cabal |
| 04:06.58 | JT | covici1: is this connected to a channel bank? |
| 04:07.08 | covici1 | yep. |
| 04:07.27 | JT | rhino cb? |
| 04:07.28 | covici1 | pastebin.ca/549475 |
| 04:08.00 | JT | multiple spans? |
| 04:08.00 | covici1 | yep. |
| 04:08.10 | covici1 | 2 spans the second one is the pri. |
| 04:08.17 | JT | hmrm |
| 04:11.06 | *** join/#asterisk CSD-199 (n=asdf@189.159.57.96) |
| 04:12.30 | JT | covici1: does anything work on the card? |
| 04:13.18 | CSD-199 | Hi. I have a working dial plan, but if I want to transfer a call and make a dial mistake, I loose the call. How can I indicate to asterisk (I think in extensions.conf) to do something in case of a dial mistake? lets say, an not declarated extension |
| 04:14.09 | covici1 | well, when we had cas signalling things worked OK, and the channel bank seems to work ok. |
| 04:15.21 | JT | covici1: umm, so yeah |
| 04:15.40 | JT | covici1: EXACTLY what order did you compile everything in? |
| 04:16.53 | covici1 | first zaptel, then asterisk, then libpri. |
| 04:17.10 | JT | recompile asterisk. |
| 04:17.15 | JT | wrong order |
| 04:17.30 | covici1 | ahhh need libpri first? |
| 04:17.34 | JT | asterisk will not support a pri if libpri is not built |
| 04:17.49 | covici1 | OK. I will try that. |
| 04:17.59 | covici1 | are the configs otherwise OK? |
| 04:18.10 | JT | they're messy, but i maybe |
| 04:18.15 | JT | s/i// |
| 04:20.43 | covici1 | OK, I am recompiling now. |
| 04:21.31 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 04:21.51 | mitcheloc | guys, does this sound valid? GotoIf($["${BridgePIN}" != null]?3:5) |
| 04:22.07 | mitcheloc | i would like to make a jump only if a variable is set/not set |
| 04:23.48 | jcmoore | GotoIf($[!${ISNULL(${BridgePIN})]?3:5) |
| 04:24.36 | *** join/#asterisk brookshire (i=mbrooks@hijacked.us) |
| 04:25.18 | *** join/#asterisk joebob777as7 (n=richard@67-42-53-56.eugn.qwest.net) |
| 04:25.23 | CSD-199 | Hi. I have a working dial plan, but if I want to transfer a call and make a dial mistake, I loose the call. How can I indicate to asterisk (I think in extensions.conf) to do something in case of a dial mistake? lets say, an not declarated extension |
| 04:25.34 | covici1 | I recompiled and reinstalled but still no pri commands and calls stilll get dropped. |
| 04:25.56 | mitcheloc | jcmoore: thanks, testing now |
| 04:29.22 | *** join/#asterisk bkw__ (n=brian@adsl-70-143-39-83.dsl.tul2ok.sbcglobal.net) |
| 04:30.12 | JT | covici1: what did you end up recompiling? |
| 04:31.27 | covici1 | I recompiled asterisk. |
| 04:32.12 | JT | does the span come up in zttool? |
| 04:33.02 | *** join/#asterisk errr (n=errr@fedora/errr) |
| 04:33.20 | covici1 | looking for libnewt maybe I need to recompile. |
| 04:35.30 | lee_is_me | CSD-199: http://www.voip-info.org/wiki/view/Asterisk+standard+extensions |
| 04:36.03 | CSD-199 | :) I'll see, thanks a lot! |
| 04:39.14 | covici1 | recompiled zaptel and now rebooting. |
| 04:39.25 | *** join/#asterisk mbrooks (i=mbrooks@hijacked.us) |
| 04:40.49 | *** join/#asterisk brookshire (i=mbrooks@hijacked.us) |
| 04:41.48 | *** join/#asterisk martinvandiest (n=chatzill@140.239.173.226) |
| 04:43.29 | *** join/#asterisk BugKhaM (n=LAMER@ppp-58.8.6.252.revip2.asianet.co.th) |
| 04:43.48 | *** join/#asterisk bobloblian (n=bob@unaffiliated/bobloblian) |
| 04:43.56 | covici1 | OK, it says no pri running on span 2 |
| 04:44.00 | BugKhaM | anyone using zaptel 1.4.1 on RHEL5? |
| 04:44.22 | BugKhaM | I got some strange errors |
| 04:50.23 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 04:50.24 | mosty | bugkham: ask your question, paste any relevant files/error messages/etc to a paste site |
| 04:51.52 | Marshall- | what's a good IAX2 Provider that allows callerid changing? |
| 04:52.25 | *** join/#asterisk [hC] (n=hardcore@190.10.13.145) |
| 04:56.13 | CSD-199 | oh yes, I got it. |
| 04:56.18 | CSD-199 | Thanks a lot. |
| 04:56.21 | CSD-199 | bye! |
| 04:57.54 | covici1 | OK, not sure what is going on, but it shows no pri running. very strange. I am going to sign off now and go to bed. |
| 04:58.02 | *** join/#asterisk friedrich| (n=friedric@e177249000.adsl.alicedsl.de) |
| 04:58.20 | covici1 | jt thanks for all your help. |
| 04:59.42 | *** part/#asterisk martinvandiest (n=chatzill@140.239.173.226) |
| 05:02.01 | *** join/#asterisk DragonBall-Z (n=Adnan@202.5.145.13) |
| 05:08.26 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 05:10.30 | brookshire | BugKhaM: try upgrading :) |
| 05:10.35 | brookshire | i believe 1.4.1 is out now |
| 05:10.41 | brookshire | i mean.. 1.4.4 |
| 05:10.49 | *** join/#asterisk Marshall- (n=Marshall@cpe-76-181-126-98.columbus.res.rr.com) |
| 05:13.59 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 05:21.29 | BugKhaM | brookshire: zaptel? |
| 05:24.02 | brookshire | Zaptel 1.4.2.1 |
| 05:25.01 | BugKhaM | oh |
| 05:25.56 | BugKhaM | will give it a try then |
| 05:26.35 | brookshire | cory! |
| 05:28.03 | *** join/#asterisk russellb (i=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
| 05:28.03 | *** mode/#asterisk [+o russellb] by ChanServ |
| 05:32.23 | *** join/#asterisk bkw_ (n=brian@adsl-70-143-39-83.dsl.tul2ok.sbcglobal.net) |
| 05:37.24 | *** join/#asterisk Putzz (n=me@CPE001a70d1b3b3-CM0019475d27c2.cpe.net.cable.rogers.com) |
| 05:37.28 | BugKhaM | brookshire: with a bit of work, it compiles with zaptel-1.4.2.1 now |
| 05:37.36 | BugKhaM | brookshire: thanks |
| 05:37.45 | brookshire | np! |
| 05:37.57 | brookshire | i blame Corydon76-work |
| 05:45.07 | russellb | o.O |
| 05:45.17 | Corydon76-home | russellb: !!! |
| 05:45.51 | brookshire | z0mg russell! |
| 05:46.16 | russellb | greetings!!!! |
| 05:46.53 | russellb | sup |
| 05:48.44 | sevard | It could have been disaster |
| 05:48.44 | sevard | Till the pupil found his master |
| 05:48.44 | sevard | Something rhymes with master |
| 05:48.44 | sevard | He's teaching him the ways of rock! |
| 05:48.48 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 05:57.55 | snuffy22 | got a q.. [Jun 8 15:48:21] WARNING[18230]: loader.c:601 inspect_module: Module 'format_mp3.so' does not provide a description. |
| 05:58.17 | snuffy22 | where can i add this 'description into the code so i can recompile the module :) |
| 05:58.28 | snuffy22 | and load it successfully |
| 06:01.51 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 06:01.57 | snuffy22 | anyone? |
| 06:02.29 | *** join/#asterisk [hC] (n=hardcore@190.10.13.145) |
| 06:05.29 | russellb | snuffy22: what version? |
| 06:07.00 | snuffy22 | trunk |
| 06:07.12 | snuffy22 | ast addons trunk as well |
| 06:07.41 | snuffy22 | had same issue with loading g729 codec |
| 06:09.19 | russellb | the code looks right |
| 06:09.44 | russellb | you sure you're not trying to load trunk modules into 1.4 or something? |
| 06:10.05 | snuffy22 | svn checkout http://svn.digium.com/svn/asterisk-addons/trunk asterisk-addons |
| 06:10.23 | snuffy22 | exactly what i do config make make isntall on |
| 06:11.17 | *** join/#asterisk Avochelm (n=damo@gw-morphett.koalatelecom.com.au) |
| 06:13.00 | snuffy22 | hmm.. regot the code after trashing whole dir.. |
| 06:13.06 | snuffy22 | seems to be ok now.. |
| 06:13.37 | *** part/#asterisk flot (n=flot@user183.hovrino.net) |
| 06:17.04 | *** join/#asterisk Strom_M (n=strom@208.47.199.4) |
| 06:19.30 | snuffy22 | my bad |
| 06:22.22 | *** join/#asterisk yassaccan (n=yassacca@admin155.hgo.se) |
| 06:23.05 | *** join/#asterisk snarfer (n=snarfer@unaffiliated/snarfer) |
| 06:23.26 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 06:24.29 | snarfer | I've got a quick question |
| 06:24.36 | snarfer | What is asterisk now? |
| 06:25.12 | JT | a gui |
| 06:25.16 | JT | and a distr |
| 06:25.19 | JT | distro |
| 06:25.47 | snarfer | Tasty. |
| 06:25.51 | snarfer | Thank you. |
| 06:25.55 | *** part/#asterisk snarfer (n=snarfer@unaffiliated/snarfer) |
| 06:26.23 | JT | :o |
| 06:26.38 | snuffy22 | hmm with svn * and svn * addons.. moh in files mode.. keeps restarting music when going from one dial to the next |
| 06:29.29 | Hmmhesays | oh ser driving me nuts |
| 06:37.27 | snuffy22 | anyone got ideas suggestions? |
| 06:37.44 | JT | svn is not very specific, there's 3 svn branches |
| 06:37.51 | *** join/#asterisk porche (n=porche@88.239.132.50) |
| 06:38.06 | porche | hi everybody |
| 06:38.49 | porche | does anybody know the differences in BUSYDETEC_TONEONLY |
| 06:39.02 | snuffy22 | well trunk.. |
| 06:39.13 | snuffy22 | not 1.2 or 1.4 svn |
| 06:39.31 | porche | snuffy is that to me or someone else? |
| 06:39.40 | snuffy22 | someone else.. jt |
| 06:39.50 | porche | oki |
| 06:39.55 | snuffy22 | i have no idea bout that porche |
| 06:40.31 | porche | :) yes, i am digging with tone detection, found the right tones in busy, but cannot work it the right way |
| 06:40.40 | porche | i mean busy detection |
| 06:41.37 | *** join/#asterisk ecoleman (n=ecoleman@cpe-76-50-132-99.buffalo.res.rr.com) |
| 06:41.47 | *** part/#asterisk ecoleman (n=ecoleman@cpe-76-50-132-99.buffalo.res.rr.com) |
| 06:42.18 | *** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl) |
| 06:44.51 | *** join/#asterisk NoCarrier (n=NoCarrie@unaffiliated/badpacket) |
| 06:46.57 | snuffy22 | can't help there.. only have t1 no pstn stuff |
| 06:47.00 | snuffy22 | *e1 |
| 06:47.17 | porche | yep, if i know, i would buy e1 actually |
| 06:47.31 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 06:47.32 | porche | i am in turkey, we dont have polarity reversal on hang up thingo |
| 06:48.15 | porche | since e1 cost arms and legs here, and I needed only 12 lines to start, i bought tdm2400, just i want to make it online, and change it after a while with e1 after it makes money |
| 06:48.29 | snuffy22 | ahh |
| 06:49.21 | porche | this is really a bad condition, timeouts wont work 100% since users will call in to start the process, anyways let's dig more |
| 06:50.12 | *** join/#asterisk euthanasie (n=kvirc@80.146.187.238) |
| 06:50.20 | porche | hey channel, is there anyone here, experienced with analog lines + busy detect where the polariry reversal is not supported? |
| 06:50.20 | euthanasie | hi @ all |
| 06:50.36 | porche | hey euthanasie, solved your problem? |
| 06:51.07 | euthanasie | sry about yesterday porche, got a disconn of my provider, yes problem solved on mysterius way, suddenly it worked |
| 06:51.12 | *** join/#asterisk saftsack (n=oliver@p54A7DA66.dip.t-dialin.net) |
| 06:51.36 | Hmmhesays | kewl start |
| 06:51.37 | porche | got it, np, i was damn digging my problem also, |
| 06:51.48 | porche | hmmheasays, yes, i do use kewlstart |
| 06:51.50 | *** join/#asterisk _BaZaroff (n=BaZ@cvsup1.senergy.ru) |
| 06:52.00 | _BaZaroff | hello all |
| 06:52.05 | euthanasie | hi _BaZaroff |
| 06:52.23 | porche | but on specs kewlstart does support/best with polarity reversal on hangup, which cannot be my option, i dont have it |
| 06:52.40 | _BaZaroff | Did you know, where i can get list of Asteriks Call Manager TELNET commands and replies? |
| 06:52.41 | Nugget | telnet is eeeeeeevil! |
| 06:52.53 | euthanasie | just got another question, how can i submit callerid's between extensions? |
| 06:52.56 | Hmmhesays | now fix my accounting issues in opensers |
| 06:52.58 | porche | i try to detect busydetect, i have the tones on hang up, the issue is asterisk cannot |
| 06:53.43 | porche | _bazaaroff check out http://www.voip-info.org/wiki-Asterisk+manager+API |
| 06:54.23 | _BaZaroff | Ou! Great Thanx!!!! porche |
| 06:55.07 | porche | _bazaroff, always refere to voip-info.org it's a good place to check everything, |
| 06:55.48 | porche | euthanasie, well, isnt there a caller id in the start of the call? |
| 06:58.34 | euthanasie | i mean the number of the caller, in the best way with a prefix, we have 2 asterisk server's calling between works without problems, but i want to see the caller-id on the display of the phne, if anyone from the other server is calling |
| 06:59.14 | porche | ah i see |
| 06:59.27 | porche | does other server send out caller id's? |
| 07:00.23 | euthanasie | don't looks like |
| 07:01.07 | euthanasie | i have to say, im a little noob on asterix, worked with hipath before, but now we migrate |
| 07:01.15 | euthanasie | asterisk |
| 07:01.17 | euthanasie | ^^ |
| 07:02.13 | porche | well have your tried giving callerid to extensions? |
| 07:02.20 | porche | are they sip clients/ |
| 07:02.46 | porche | or just hardware extension? |
| 07:03.36 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 07:04.18 | euthanasie | sip-clients |
| 07:04.27 | porche | check out |
| 07:04.28 | porche | sip.conf |
| 07:04.35 | porche | if the clients have got callerif |
| 07:04.39 | porche | callerid |
| 07:08.04 | euthanasie | ah, thx that works |
| 07:08.30 | euthanasie | but how can i add the prefix, or better, how can i manipulate the CALLERID ? |
| 07:09.03 | porche | which prefix euthanasie |
| 07:11.35 | euthanasie | for example to call in building 2 the ext is _2XX. so i want to add a leading 2 if a call from building to arrives |
| 07:13.39 | porche | you can give any caller id name |
| 07:13.42 | porche | in sip.cof |
| 07:13.46 | porche | sip.conf |
| 07:13.48 | porche | try it |
| 07:14.35 | porche | otherwise you must write some code in dial plan |
| 07:15.02 | porche | which is not a preferred, in this case |
| 07:17.32 | euthanasie | but if i manipulate the callerids it would be shown wrong on the local building. I wanted to change the numbers but i didn |
| 07:17.41 | euthanasie | 't get a ok from my boss |
| 07:18.00 | euthanasie | in his eye all must stay as it is .... |
| 07:18.27 | porche | i see, to my knowladge you need to write a dial-plan rule |
| 07:18.46 | *** join/#asterisk matsk (n=mk@194.68.102.172) |
| 07:18.47 | porche | how do you call other asterisk? |
| 07:18.53 | porche | i mean the command line |
| 07:19.39 | Zeeek | has anyone here been running asterisk successfully for a while? |
| 07:19.54 | mosty | Zeeek, plenty of us have |
| 07:20.04 | Zeeek | I wasn't sure |
| 07:20.06 | *** join/#asterisk JT (n=jon@unaffiliated/jt) |
| 07:20.12 | Zeeek | so who's running 1.4 ? |
| 07:20.18 | jql | quite successfully |
| 07:20.21 | jql | yes, 1.4 |
| 07:20.29 | mosty | i still use 1.2 for production boxes |
| 07:20.32 | porche | zeeek what's the issue specifically, 1.4.4 is very stable |
| 07:20.33 | jql | meeeelions of minutes. wheeee |
| 07:20.53 | euthanasie | exten => _2XX,1,Dial(SIP/${EXTEN:1}@sciencecenter2|120|rtT) ; sciencecenter2 is defined in the sip.conf |
| 07:21.03 | porche | ah sorry, i do also use 1.2 on production, but 1.4.4 currently under development |
| 07:22.08 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 07:22.29 | Zeeek | good. So here's my question. I just installed 1.4 and Echo doesn't work. The same phones can call each other, they're on different LAN behind NAT but this is IAX2 |
| 07:22.45 | Zeeek | why would echo, the wolrd's simplest app with no args not work? |
| 07:23.04 | Zeeek | CLI shows no anomalies |
| 07:23.19 | Corydon76-home | Did you Answer first? |
| 07:23.24 | Zeeek | OMG |
| 07:23.26 | brookshire | did you install your flux capacitor? |
| 07:23.52 | jql | 1.21 jiggabps |
| 07:24.36 | oej | morning! |
| 07:24.43 | Zeeek | my working echo test on 1.2 works with no answer, but there is a playback before. That must be what answered |
| 07:25.11 | Corydon76-home | Correct, Playback answers the channel if not answered and you don't explicitly tell it not to |
| 07:25.24 | JT | Always Answer before you Echo. TM |
| 07:25.30 | Zeeek | yes this is now painfully obvious |
| 07:25.41 | porche | euthanasie |
| 07:25.44 | Zeeek | you almost never have to answer |
| 07:25.45 | porche | can you add this line |
| 07:26.03 | porche | SetCallerId(2111) |
| 07:26.12 | porche | b4 you call other box? |
| 07:26.33 | Zeeek | porche you can't use 2111. That's my extension |
| 07:26.56 | *** join/#asterisk elmargol (n=elmargol@82.48.60.52) |
| 07:26.57 | JT | also, that's well and truly deprecated, pr |
| 07:26.58 | porche | sorry zeeek, this was to euthanasie |
| 07:27.00 | JT | porche: |
| 07:27.11 | euthanasie | ^^ just found this command in our asterisk-book |
| 07:27.31 | porche | JT, he must change the caller id 4 calling another box |
| 07:27.34 | JT | Set(CallerID(num)=) |
| 07:27.38 | porche | the simplest way to set yes |
| 07:27.45 | porche | JT, well true |
| 07:27.45 | JT | porche: yes, so don't use outdated functions |
| 07:27.48 | Corydon76-home | JT: almost |
| 07:27.49 | elmargol | Hi, I search a cheap way to record my phone calls. |
| 07:27.57 | Corydon76-home | Set(CALLERID(num)=) |
| 07:28.14 | Corydon76-home | Functions are case-sensitive |
| 07:28.15 | elmargol | What kind of hardware do i need for this? |
| 07:28.27 | Zeeek | I am case sensitive |
| 07:28.58 | porche | btw, I have got a question |
| 07:29.14 | porche | is busydetect enabled by default, on standard compilation of asterisk? |
| 07:29.43 | Hmmhesays | my openser will not account anything damnit |
| 07:29.52 | Corydon76-home | busydetect is never enabled unless you tell it explicitly to |
| 07:29.53 | Zeeek | Corydon76-home you would know this: when I was making zaptel I kept getting the error: autoconfig: no input file in the menuselect dir. If I went to menuselect (or is it selectmenu) and make then go back zaptel makes. However, make clean, bring it back to the error |
| 07:30.16 | Corydon76-home | Zeeek: don't know |
| 07:30.55 | Zeeek | ok |
| 07:31.15 | Zeeek | anyway, "use what works" is my technique, so it works |
| 07:31.28 | porche | is it, ./configure --enable-busydetect ? |
| 07:31.39 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 07:32.07 | Corydon76-home | No, it's 'busydetect=yes' in zapata.conf |
| 07:32.30 | porche | no i mean during the compile operation of asterisk |
| 07:32.38 | Zeeek | there's a crapload of info on the mailing list about these busydetect and hangup detect problems |
| 07:32.39 | porche | yes i know zapata.conf option |
| 07:32.42 | Corydon76-home | It's compiled in by default |
| 07:32.51 | euthanasie | got it |
| 07:33.12 | porche | zeeek, yes that crapload, does not answer my problem, i dont have a polarity reversal on hang up |
| 07:33.14 | *** join/#asterisk JustAGuy1 (n=majortom@ip68-4-192-190.oc.oc.cox.net) |
| 07:33.36 | euthanasie | exten =>_X.,1,Set(CALLERID(num)=2${CALLERID(num)}) |
| 07:33.42 | euthanasie | thx porche |
| 07:33.47 | Zeeek | the crapload often does anwer these questions since there are usually hundreds of people who have the same problem |
| 07:33.54 | porche | sure, authanasie |
| 07:33.55 | mosty | porche, with some phone service providers you have to ask them to turn that on |
| 07:34.05 | JustAGuy1 | Anyone here connect AIM phoneline to Asterisk? |
| 07:34.08 | elmargol | what kind of hardware do i need to record phone calls on ISDN? |
| 07:34.18 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 07:34.20 | Zeeek | y |
| 07:34.27 | porche | mosty, talked with the telco, they dont have it, i am in turkey, so, |
| 07:35.09 | Zeeek | porche have you searched the mailing list for "turkey" by any chance? |
| 07:35.17 | Zeeek | you never know |
| 07:35.17 | mosty | elmargol, any isdn card of the correct format that works with asterisk will do |
| 07:35.48 | porche | yes searched, even talked with a guy who says he solved the same problem two years ago by playing |
| 07:36.09 | mosty | porche, maybe you can ask the digium distributor in turkey, the distributor in my country has lots of knowledgable people you can ask |
| 07:36.26 | Zeeek | or even maybe digium themselves? |
| 07:36.29 | porche | dsp.c, enum busy_detect structure |
| 07:36.58 | mosty | porche, what card are you using? |
| 07:36.58 | porche | no distributor here, well can be best to ask digium directly |
| 07:37.04 | porche | tdm2400 |
| 07:38.25 | Zeeek | porche Digium will help you if your distrib can't |
| 07:38.25 | mosty | porche, have you considered upgrading to isdn? not sure if that's affordable in turkey |
| 07:38.31 | porche | mosty, the issue is i am stuck with the card, yes i will be changing this to e1 |
| 07:38.47 | porche | which will be a better option even from isdn |
| 07:38.56 | mosty | porche, E1 is ISDN |
| 07:39.17 | porche | i mean, here we also have bri/pri |
| 07:39.25 | JT | E1 is usually PRI |
| 07:39.27 | JustAGuy1 | E1 is not ISDN. |
| 07:39.35 | JT | JustAGuy1: it *can* be |
| 07:39.40 | Hmmhesays | you can get a cas E1 I believe |
| 07:39.42 | porche | by isdn we usually mean, single line |
| 07:39.52 | JT | Hmmhesays: i don't think cas E1s exist, but data E1s do |
| 07:39.54 | porche | i mean, here the telcon means |
| 07:40.09 | JT | indegrated services digital network |
| 07:40.16 | mosty | well anyway once you move to E1 you won't have to worry about recognising hangup/busy etc |
| 07:40.26 | Hmmhesays | JT: aye |
| 07:40.30 | mosty | whatever you want to call it |
| 07:41.08 | porche | yes seems so, only clue i have currently is the dsp.c busy_detect structure, anyways |
| 07:41.37 | JustAGuy1 | Well, E1 is just a european format T1, PRI is available both in European format and NA format. |
| 07:42.47 | JT_ | err, an E1 is just the serial line standard |
| 07:42.58 | JT_ | JustAGuy1: E1 is the international stndard |
| 07:43.02 | JT_ | T1 for north america |
| 07:43.25 | jql | yeah, gsm is also an international standard. funny stuff about the us |
| 07:44.42 | JustAGuy1 | Actually, both formats are international standards, recognized by the ITU. |
| 07:44.52 | JT_ | sure |
| 07:44.54 | JustAGuy1 | As is CDMA, and TDMA. |
| 07:45.18 | JT_ | JustAGuy1: anyway, T1 is mostly only used in nort america |
| 07:45.22 | JT_ | north |
| 07:46.09 | porche | good, i have a tdm2400 at the end of the day, stuck :) |
| 07:46.16 | JustAGuy1 | Mostly used in NA, but not exclusively. |
| 07:46.23 | JT_ | yes |
| 07:46.46 | jql | same with good ol' ulaw, for much the same reason |
| 07:46.56 | JT_ | but what i am trying to get through to you is E1 is the one generally used globally |
| 07:49.46 | JustAGuy1 | That is true. I called it the European format, as that is where it originated. AT&T developed t1 and the Europeans standardized on 2048 for their T1s. To differentiate, we started calling that E1 for European T1. |
| 07:50.48 | JT_ | so even if they're both "recognised", it's the responsibility or ulaw countries to transcode at their international gateways |
| 07:50.57 | JT_ | into alaw |
| 07:53.13 | elmargol | mosty: I have more than one phone here, can I just set a asterisk "box" between ISDN and my phones? |
| 07:53.13 | JustAGuy1 | Actually, that is negotiated by the carriers. |
| 07:53.13 | elmargol | Or do I have to manage phone calls trough asterisk? |
| 07:53.13 | JT_ | if you say so |
| 07:53.40 | JT_ | elmargol: you can't just put it on the same S-bus as your phones and phone line |
| 07:53.58 | elmargol | Is there a way to do this? |
| 07:53.58 | JT_ | elmargol: you need to run ports with NT mode for phones |
| 07:54.01 | JT_ | TE mode to exchang |
| 07:54.03 | JT_ | e |
| 07:55.37 | elmargol | I already have an expensive siemens exchange thing... for 14 phones |
| 07:55.46 | Zeeek | elmargol we left an emergency phone connected in case asterisk or even DSL became unavailable, but you wouldn't want to leave asterisk and normal phones on the same line. Asterisk can't tell when the line is in use |
| 07:55.51 | elmargol | sorry for my poor english |
| 07:56.13 | JT_ | elmargol: yes asterisk can act as a gateway |
| 07:57.52 | *** join/#asterisk s0ck (n=m@unaffiliated/s0ck) |
| 08:00.56 | *** join/#asterisk porche (n=porche@88.239.132.50) |
| 08:09.19 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 08:09.57 | *** join/#asterisk saftsack (n=oliver@p54a7c81e.dip.t-dialin.net) |
| 08:13.03 | *** join/#asterisk samarora (i=minesh@203.88.149.166) |
| 08:13.57 | samarora | any body can tell me how to make "call pickup" and "call transfer" in asterisk |
| 08:14.09 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 08:14.28 | Zeeek | samarora what is the exact outcome you are trying to accomplish? |
| 08:15.28 | samarora | i would like to enable call pickup facility in my group in asterisk |
| 08:16.11 | samarora | i have enabled "pickupexten" in features.conf |
| 08:16.17 | samarora | but not able to accomplish the same |
| 08:17.22 | *** join/#asterisk fbffff (n=fbffff@c-67-167-98-42.hsd1.il.comcast.net) |
| 08:17.49 | *** join/#asterisk af_ (n=getsmart@81-174-9-192.f5.ngi.it) |
| 08:18.59 | Zeeek | samarora what is the line you added to features.conf ? |
| 08:19.36 | *** part/#asterisk JustAGuy1 (n=majortom@ip68-4-192-190.oc.oc.cox.net) |
| 08:19.59 | samarora | i have enebled "pickupexten = *8" in features.con |
| 08:21.21 | Zeeek | are you declaring pickgroups? |
| 08:22.05 | samarora | that is what my confusions.. |
| 08:22.09 | key2 | in asterisk 1.2, whats the CLI command for originating a call ? |
| 08:22.13 | mosty | samarora, transfer is atxfer and blindxfer in features.conf |
| 08:22.27 | samarora | where to define pickgroups |
| 08:23.14 | Zeeek | samarora look for ther info on the wiki |
| 08:23.57 | Zeeek | look how easy this is: http://www.voip-info.org/tiki-index.php?page=Channels%20and%20Groups |
| 08:25.40 | *** join/#asterisk waptaxi (n=cahe@stat-5-160.e-sky.ru) |
| 08:30.47 | *** join/#asterisk Dibbler_ (n=Dibbler@host217-45-198-229.in-addr.btopenworld.com) |
| 08:31.22 | cy303 | man, getting that SPA3102 going as an ATA was kind of a pain in the ass |
| 08:31.23 | cy303 | heh |
| 08:31.32 | Zeeek | why? |
| 08:31.56 | cy303 | I just had some trouble with the configuration |
| 08:32.02 | *** join/#asterisk fbffff (n=fbffff@c-67-167-98-42.hsd1.il.comcast.net) |
| 08:32.09 | Zeeek | what was the trouble? |
| 08:32.12 | cy303 | mostly I was trying to use the ethernet port on my LAN instead of the Intarweb port I guess |
| 08:32.29 | cy303 | works like a charm now tho |
| 08:32.38 | cy303 | two hours later, heh |
| 08:32.55 | *** part/#asterisk porche (n=porche@88.239.132.50) |
| 08:32.57 | Zeeek | and what was the secret? |
| 08:33.35 | cy303 | use the port labeled internet instead of the port labeled ethernet and enable the web gui on the internet port |
| 08:34.03 | Zeeek | seems obvious now |
| 08:34.06 | cy303 | indeed |
| 08:37.05 | *** part/#asterisk samarora (i=minesh@203.88.149.166) |
| 08:44.28 | *** join/#asterisk Marshall- (n=Marshall@cpe-76-181-126-98.columbus.res.rr.com) |
| 08:45.18 | *** join/#asterisk john-eman0n (n=eman0n@cpe-76-181-126-98.columbus.res.rr.com) |
| 08:55.29 | *** join/#asterisk CBU[^_^]M`` (n=love@210.213.139.239) |
| 08:59.29 | *** join/#asterisk canberk (n=canberk@212.156.213.131) |
| 09:00.17 | canberk | hello, is it possible to have a x brand sip server and have asterisk work in different networks and let the x branded sip server to use asterisk's features such as voicemail, meetme... |
| 09:01.47 | mosty | the sip server can make a sip call to asterisk |
| 09:02.00 | mosty | and access whatever any sip client can access |
| 09:02.39 | *** join/#asterisk sasch (n=sasch@host102-30-static.107-82-b.business.telecomitalia.it) |
| 09:02.57 | sasch | hi all I try to reinstall asterisk ... i have one problem with zaptel |
| 09:03.09 | canberk | mosty, shouldn't the client have an account on asterisk in order to access to voicemail for ex |
| 09:03.11 | sasch | when i try to lunch ./configure i recivie that |
| 09:03.52 | mosty | canberk, yes. or you can accept unauthenticated calls. depends you you want to configure the asterisk box |
| 09:04.24 | canberk | i mean how is asterisk going to save the voicemails without having an account associated to them |
| 09:05.22 | mosty | you specify the voicemail account when you use the Voicemail command in the asterisk dialplan |
| 09:05.55 | canberk | the fact is i am not using the voicemail feature at the moment and i don't know how it works, trying to find out by asking here |
| 09:07.04 | mosty | canberk, http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+VoiceMail http://www.voip-info.org/wiki/index.php?page=Asterisk+config+voicemail.conf |
| 09:22.33 | *** join/#asterisk matsk (n=mk@194.68.102.172) |
| 09:26.02 | *** join/#asterisk Marshall- (n=Marshall@cpe-76-181-122-205.columbus.res.rr.com) |
| 09:31.14 | *** join/#asterisk mvanbaak (n=mafkees@vanbaak.xs4all.nl) |
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| 09:36.20 | key2 | JT ? |
| 09:36.21 | key2 | there ? |
| 09:39.21 | JT | a bit |
| 09:39.28 | *** join/#asterisk nitram (i=foo@superblob.com) |
| 09:39.52 | *** join/#asterisk Paytats_Pingvin (n=77889789@cpe-76-173-56-41.socal.res.rr.com) |
| 09:39.53 | Paytats_Pingvin | hi |
| 09:40.15 | Paytats_Pingvin | im running Asterisk 1.4.4 |
| 09:40.31 | Paytats_Pingvin | and everything is really choppy, is it because im running it off of vmware? |
| 09:46.13 | *** join/#asterisk saftsack (n=saftsack@pD9E07D67.dip.t-dialin.net) |
| 09:46.53 | *** join/#asterisk friedrich| (n=friedric@e177249000.adsl.alicedsl.de) |
| 09:50.26 | JT | key2: ? |
| 09:55.23 | *** join/#asterisk Polis_ttt (n=your@194-237-172-225-no48.business.telia.com) |
| 09:55.23 | key2 | JT: when using bristuff |
| 09:55.23 | key2 | if you want to do a round robbin for the 4 different span, you put them all the the same group and dial(zap/r1/number) ? |
| 09:58.34 | JT | yes |
| 09:58.37 | JT | R or r |
| 09:58.42 | JT | depending on which order |
| 09:58.46 | JT | you want... |
| 09:59.32 | Paytats_Pingvin | anyone can tell me if asterisk on vmware is a good idea or not? |
| 10:02.01 | matsk | Asterisk and VMware is not a good solution, often shoppy sound |
| 10:02.21 | Paytats_Pingvin | matsk yeah thats my problem now |
| 10:02.24 | Paytats_Pingvin | any fix to it? |
| 10:02.41 | matsk | Noop |
| 10:14.36 | key2 | JT: I didnt know round robin had order ;) |
| 10:14.48 | key2 | u mean either it starts from the span number 4 or 1 ? |
| 10:17.00 | JT | read the docs |
| 10:17.04 | JT | http://www.voip-info.org/wiki/view/Asterisk+ZAP+channels |
| 10:20.49 | key2 | JT: for some reason when I place a call from no matter what span, it gets hang up |
| 10:20.56 | key2 | JT: but I can receive calls |
| 10:21.50 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 10:30.29 | *** join/#asterisk Here_And_There (n=oldfart2@pool-68-238-252-162.phlapa.fios.verizon.net) |
| 10:30.45 | Here_And_There | join #trixbox |
| 10:30.55 | *** part/#asterisk Here_And_There (n=oldfart2@pool-68-238-252-162.phlapa.fios.verizon.net) |
| 10:32.03 | tzafrir_laptop | key2, and if you call a specific channel it works? |
| 10:33.54 | Paytats_Pingvin | hey guys whats a good endpoing maanger? |
| 10:39.41 | *** join/#asterisk s0ck (n=m@unaffiliated/s0ck) |
| 10:43.13 | key2 | tzafrir: what u mean a specific channel ? |
| 10:43.24 | key2 | if I do a Dial(zap/1-1/number) ? |
| 10:43.49 | Paytats_Pingvin | what about running asterisk on Parallels? how will that work? |
| 10:46.45 | *** join/#asterisk j23 (n=jerms@static-213-115-44-90.sme.bredbandsbolaget.se) |
| 10:46.48 | j23 | hi |
| 10:47.31 | tzafrir_laptop | key2, Dial(Zap/1) |
| 10:47.40 | tzafrir_laptop | key2, Dial(Zap/1/number) |
| 10:48.01 | tzafrir_laptop | what do you mean by ims? |
| 10:48.38 | tzafrir_laptop | j23, what do you mean by ims? |
| 10:49.34 | key2 | tzafrir: ssame |
| 10:52.00 | *** part/#asterisk robin_z (n=robin@rszemeti.gotadsl.co.uk) |
| 10:58.25 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 11:01.07 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 11:08.25 | tzafrir_laptop | key2, I don't know PRI well enough. But mayb try messing with stuff related to the pridialplan? |
| 11:10.12 | *** join/#asterisk canberk (n=canberk@212.156.213.131) |
| 11:10.15 | canberk | hello |
| 11:10.19 | canberk | is it possible to merge calls:? |
| 11:11.52 | JT | key2: pastebin.ca zaptel.conf and zapata.conf |
| 11:13.45 | *** join/#asterisk Dibbler_ (n=Dibbler@host217-45-198-229.in-addr.btopenworld.com) |
| 11:14.58 | canberk | is it possible to merge calls, for ex i am talking to someone, then i park that call and call another person and join these two calls without hanging up |
| 11:15.04 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 11:15.32 | tzafrir_laptop | canberk, transfer the calls to a meetme room? |
| 11:19.37 | canberk | maybe yeah |
| 11:19.42 | canberk | is it like n-way call? |
| 11:25.47 | *** join/#asterisk ccesario (n=ccesario@200-158-226-61.dsl.telesp.net.br) |
| 11:36.48 | *** join/#asterisk kkeil (n=kkeil@p54978D5A.dip0.t-ipconnect.de) |
| 11:37.45 | *** join/#asterisk Polis_ttt (n=your@194-237-172-225-no48.business.telia.com) |
| 11:47.08 | *** join/#asterisk penguinFunk (n=penguin@87.224.86.46) |
| 11:47.43 | penguinFunk | hi, i upgraded my kernel via yum and now zttranscode will not compile |
| 11:48.18 | penguinFunk | it compiled fine with 2.6.20-1.2933 but will not compile since moving to 2.6.20-1.2952 |
| 11:48.21 | penguinFunk | howcome? |
| 11:48.26 | penguinFunk | how can i fix this please? |
| 11:51.22 | *** join/#asterisk coppice (n=chatzill@10.198.17.210.dyn.pacific.net.hk) |
| 11:55.06 | tzafrir_laptop | penguinFunk, rpm -qa | grep kernel |
| 11:55.20 | tzafrir_laptop | maybe your kernel and kernel-devel don't match? |
| 11:57.26 | penguinFunk | kernel-devel-2.6.20-1.2952.fc6 |
| 11:57.26 | penguinFunk | kernel-headers-2.6.20-1.2952.fc6 |
| 11:57.26 | penguinFunk | kernel-2.6.20-1.2952.fc6 |
| 11:57.40 | penguinFunk | here are my compile errors: http://pastebin.ca/550321 |
| 11:57.53 | penguinFunk | asterisk was working fine for months, until this new kernel came along :( |
| 11:58.03 | s0ck | can i ask why you upgraded |
| 11:58.25 | penguinFunk | is that you mike? |
| 11:58.41 | tzafrir_laptop | s0ck, because that is Fedora. The "stable" distribution that upgrades kernel in its lifespan |
| 11:59.06 | s0ck | my name is mike, yeh |
| 11:59.09 | s0ck | who dat? :P |
| 11:59.11 | tzafrir_laptop | A better question would be: why run a server on Fedora and expect it not to break |
| 11:59.12 | tzafrir_laptop | ? |
| 11:59.28 | penguinFunk | well s0ck, i did a yum update to update the whole dist |
| 12:00.11 | penguinFunk | i been trying to get pandora working and it was complaining about old perl modules so i thought i would update the dist |
| 12:00.22 | *** join/#asterisk saftsack (n=saftsack@pD9E07D67.dip.t-dialin.net) |
| 12:00.33 | penguinFunk | tzafrir: i never had trouble with fedora before |
| 12:00.47 | tzafrir_laptop | someone I know (who runs Fedora for his home systems, including home servers) compared Fedora, change-wise, to Debian-Testing |
| 12:00.49 | penguinFunk | these auto dist updaters are no good :( |
| 12:01.00 | tzafrir_laptop | It had some QA, but it keeps changing under your feet |
| 12:01.15 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 12:01.24 | tzafrir_laptop | penguinFunk, the auto-upgrade is generally a good idea, if the codebase doesn't change much |
| 12:01.36 | tzafrir_laptop | look at RHEL/centos, or Debian Stable |
| 12:02.03 | penguinFunk | well if zttranscode.c has not changed and was 'compilable' before |
| 12:02.04 | tzafrir_laptop | I can normally almost blindly trust upgrades from debian-stable |
| 12:02.10 | penguinFunk | why would a new kernel not compile it |
| 12:02.11 | penguinFunk | ? |
| 12:02.12 | penguinFunk | fs |
| 12:03.00 | penguinFunk | so many people say debian is better, i'm yet to be convinced |
| 12:03.07 | tzafrir_laptop | penguinFunk, please try latest zaptel. |
| 12:03.11 | penguinFunk | ok |
| 12:03.21 | tzafrir_laptop | e.g: 1.4.2.1 |
| 12:03.24 | penguinFunk | does this mean i have to use new version of asterisk to match? |
| 12:03.33 | tzafrir_laptop | such compilation errors don't last long |
| 12:03.48 | s0ck | penguinFunk: whereabouts you from |
| 12:03.58 | penguinFunk | s0ck: hah, i freaked you out yet? |
| 12:04.07 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 12:04.13 | s0ck | :( :P |
| 12:04.25 | penguinFunk | been a v long time, put it like that |
| 12:04.33 | penguinFunk | howz the misses / daughter? |
| 12:05.17 | penguinFunk | tzafrir: does asterisk version have to match zaptel version? |
| 12:06.47 | s0ck | perhaps you have me confused with someone else? :) |
| 12:07.24 | penguinFunk | maybe ;) |
| 12:09.46 | *** join/#asterisk Daejeo1 (n=chatzill@124.62.150.49) |
| 12:09.58 | *** part/#asterisk frocos11292 (n=ask@firewall.vipvoz.com) |
| 12:10.46 | tzafrir_laptop | no. But there were quite a few fixes to zaptel since 1.4.0 . try 1.4.2.1 |
| 12:11.07 | tzafrir_laptop | And actually latest asterisk will need zaptel >= 1.4.2 for chan_zap to build |
| 12:11.40 | Daejeo1 | where can I download MP3 music for asterisk (except digium). |
| 12:11.55 | penguinFunk | ok tzafrir_laptop :) |
| 12:11.58 | tzafrir_laptop | Daejeo1, moh does not need to be mp3 |
| 12:12.26 | penguinFunk | hmm, ztranscode built |
| 12:12.32 | penguinFunk | zttranscode* |
| 12:12.33 | Daejeo1 | tzafrir_laptop: what is the format? |
| 12:12.38 | tzafrir_laptop | and anyway, use anything you like. There are tons of places from which you can get "elevator music" |
| 12:12.48 | penguinFunk | Daejeo1: use wav? |
| 12:13.02 | Daejeo1 | bitrate? |
| 12:13.15 | Daejeo1 | I guess I can't play any bitrate |
| 12:13.24 | Daejeo1 | should be something specificv |
| 12:14.28 | Daejeo1 | something like " PCM Encoded, 16 Bits, at 8000Hz" |
| 12:14.29 | penguinFunk | 8Khz mono |
| 12:14.43 | Daejeo1 | now where can I find? |
| 12:14.50 | penguinFunk | find any music |
| 12:14.56 | Daejeo1 | I don't have encoder |
| 12:14.56 | penguinFunk | rencode it yourself |
| 12:15.07 | penguinFunk | windows sound recorder will change any wav format into any wav format |
| 12:16.44 | Daejeo1 | did you ever use? |
| 12:19.55 | *** join/#asterisk DEsotosa (i=d46d6d64@gateway/web/cgi-irc/ircatwork.com/x-2372ea1e3280b838) |
| 12:20.26 | DEsotosa | hi all anyone know vtiger asterisk intigration |
| 12:22.39 | *** join/#asterisk Strom_M (n=strom@208.47.199.4) |
| 12:22.57 | DEsotosa | hi all anyone know vtiger asterisk intigration |
| 12:23.36 | *** join/#asterisk ccesario (n=ccesario@201-0-124-117.dsl.telesp.net.br) |
| 12:27.20 | *** join/#asterisk ToyMan (n=Stuart@72.168.167.241) |
| 12:28.24 | penguinFunk | tzafrir: asterisk is up and running, zaptel and all the module are compiled and loaded |
| 12:28.24 | penguinFunk | zapata.conf is configure |
| 12:28.24 | penguinFunk | d |
| 12:29.04 | penguinFunk | yet...No channel type registered for 'Zap' AND Unable to create channel of type 'Zap' (cause 66 - Channel not implemented |
| 12:29.32 | tzafrir_laptop | Daejeo1, wow, is it that hard to find music? |
| 12:29.48 | penguinFunk | whats that command again... module load chan_zap |
| 12:29.50 | penguinFunk | or something |
| 12:30.04 | tzafrir_laptop | now, if you want music with proper license, http://search.creative-commons.com/ |
| 12:30.36 | Daejeo1 | tzafrir_laptop: it is |
| 12:33.03 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:33.09 | penguinFunk | http://pastebin.ca/550408 |
| 12:33.21 | penguinFunk | any ideas tzafrir_laptop ? |
| 12:33.41 | Strom_M | penguinFunk: looks like you didnt load teh modules correctly |
| 12:33.52 | Strom_M | what happens when you run ztcfg -vv at the bash prompt? |
| 12:34.55 | penguinFunk | Strom_M: http://pastebin.ca/550411 |
| 12:34.55 | tzafrir_laptop | "no such device or address". zaptel modules not loaded? |
| 12:34.59 | tzafrir_laptop | ls -l /dev/zap/1 |
| 12:35.12 | penguinFunk | # lsmod | grep zaptel |
| 12:35.12 | penguinFunk | zaptel 202216 2 wct4xxp,zttranscode |
| 12:35.12 | penguinFunk | crc_ccitt 10817 2 zaptel,hisax |
| 12:35.26 | penguinFunk | and... |
| 12:35.26 | penguinFunk | # ls -l /dev/zap/1 |
| 12:35.26 | penguinFunk | ls: /dev/zap/1: No such file or directory |
| 12:36.03 | Daejeo1 | tzafrir_laptop: can you point me to music link? |