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| 00:20.51 | rbd | hey guys, asterisk 10.3.x ... I issue a "reload" cmd through the AMI, I get the following error "ast_careful_fwrite: fwrite() returned error: Broken pipe" ...is this "normal" (the command seems to work) |
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| 01:06.53 | Hunterkll | Hello! |
| 01:07.13 | Hunterkll | I'm working with an existing asterisk config, it has an IVR configured, and one ethernet ata configured for a phone... |
| 01:07.30 | Hunterkll | i've added an identical config in sip.conf to the existing for the new ip/sip phone, however |
| 01:07.37 | Hunterkll | the new phone dosn't ring for inbound calls, but works fine for outbound |
| 01:07.51 | Hunterkll | where would I look for information about how it's routing the call? |
| 01:07.57 | Hunterkll | I want both to ring for inbound at the same time |
| 01:21.30 | Hunterkll | Hm, figured it out. the inbound sip defintion for SIP/8001 in extensions.conf neeeded to be extended with &SIP/8002 |
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| 01:34.23 | rljhaines | hello... i tried perhaps 6 months ago and was not successful... the o'reilly manual i read made me more confident than i should have become that all i needed was asterisk and a sip phone... after several attempts at getting these components properly configured to call landlines, non of them successful, i came across what may be a requirement not covered by o'rieilly... to wit: do i have to have an SIP service provider to use aste |
| 01:34.58 | rljhaines | oops... .insert "to get asterisk working" where it's obviously missing |
| 01:37.14 | rljhaines | (i gave up from then to now for want of clear instructions and due to the SIP service provider requirement, if it is one) |
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| 01:44.39 | LiuYan | do you mean call PSTN phones from your SIP phone? |
| 01:58.30 | rljhaines | i do mean that, yes... |
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| 02:20.21 | rljhaines | and do i necessarily need an SIP service provider for that? |
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| 02:57.10 | LiuYan | I haven't try SIP SP yet, I use a hardware card (which is digium compatible) to connect to PSTN, so I don't need service provider |
| 03:01.02 | LiuYan | this is introduced at chapter 9 - Internationalizaton of the ~book 3rd Edition |
| 03:07.07 | rljhaines | i c... i will look into that, but i don't know if this project is important enough to me to invest much into it... i mean, a $50 card, maybe... more, pretty doubtful... besides, if PSTN connection does require either a hardware card or a SIP SP, there are free sip sp's... possibly a you-get-what-you-pay-for situation, but i will try it first anyway... question: the card you use is *required* for pstn connections, right? (given no |
| 03:07.59 | rljhaines | (i will look in chapter 9, thank you) |
| 03:13.29 | rljhaines | LiuYan, i think i found the answer here... always interesting how having someone to run things past often yields unexpected results: http://www.asterisk.org/hardware |
| 03:15.13 | rljhaines | okay, i gotta turn in... be well... |
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| 04:17.20 | Assid | heya |
| 04:17.35 | Assid | anyone here ever worked with legacy pbx like the NEC SL1000 ? |
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| 08:49.51 | qakhan | anyone have idea of MRCP? |
| 08:49.56 | qakhan | uniMRCP |
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| 09:10.51 | coldfront | Hi I have a Grandstream Handytone 286 ATA box it's currently locked to a VOIP provider, How can I setup a server to use my own SIP server. Because I don't think they will give out the SIP certs. |
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| 15:30.46 | Freeaqingme | digium.com down? |
| 15:31.13 | Freeaqingme | hmz, after ten times refreshing it works now.. |
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| 18:00.48 | DarthExpeditor | Does anyone know how to tell if I'm running 64-bit or 32-bit asterisk? |
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| 18:09.38 | rljhaines | hey, folks... so i got the notion that i need an SIP service provider to call PSTN phones, so i registered with iptel. now asterisk is out of the loop (for now???) as i have my x-lite softphone configured to use the iptel domain... but... now iptel is telling me i can't use their service for PSTN calls. a couple questions: 1) if i was only interested in making VoIP connections, couldn't i just do that with asterisk (though i never got t |
| 18:09.38 | rljhaines | green light from x-lite when configuring, perhaps it's possible and i never got it right); 2) do i somehow need to tie asterisk in with iptel & x-lite to make computer-to-PSTN calls (or perhaps all three are required for getting me only VoIP calls, as considered earlier)? |
| 18:19.12 | WIMPy | You don't even need Asterisk for VoIP. |
| 18:28.40 | rljhaines | right... it's iptel (or any provider) that i didn't want. my original hope was to do VoIP with asterisk, having internet itself as my only service. and i don't know if asterisk can eliminate my dependence on services. i could not get a working configuration of x-lite with asterisk to confirm i could do iP-to-IP connections this way... but, that wasn't what i was ultimately looking for... then i got the notion that iptel would get me the |
| 18:28.41 | rljhaines | IP-to-PSTN i wanted, now i find that's not right, either... |
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| 18:33.09 | WIMPy | If you want to make or receive calls to/from PSTn you need an ITSP. |
| 18:33.14 | WIMPy | ~itsp |
| 18:33.14 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
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| 18:45.20 | rljhaines | i'll look into that... thank you... |
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| 20:22.40 | afink | hello everyone, I am having a problem with no audio on an external phone. The call is connecting and everything ok, just no audio. I can hear the very first part of the first ring then after that nothing. What are some things I can check? Thanks! |
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| 20:29.30 | cyford | does call-limit in sip.conf limits trunk calls as well |
| 20:31.31 | WIMPy | Sip does not know trunks. And I'm not sue call-limit still does anything at all. |
| 20:37.08 | cyford | what do you mean sip doenst know trunks, thats where all min is located |
| 20:37.48 | cyford | and sip.cof knows all that has to do with sip peers, friends, and users |
| 20:37.52 | WIMPy | ~siptrunk |
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| 20:38.35 | WIMPy | Hmm |
| 20:38.39 | cyford | yes containers for peers right |
| 20:39.07 | cyford | i see |
| 20:39.14 | cyford | call-limit = number : Number of simultaneous calls through this user/peer. |
| 20:39.23 | WIMPy | Anyway: Look at the GROUP* funktions. |
| 20:40.08 | cyford | http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf |
| 20:40.21 | cyford | i am even more lost with that |
| 20:40.57 | WIMPy | You should be aware that much of the stupp on voip-info is very outdated. |
| 20:41.09 | cyford | i just would like to set a limit on users concurant calls without limiting peers |
| 20:42.22 | cyford | yes but much outdated stuff still exsist, or changed into a differnt command |
| 20:42.59 | WIMPy | Or has gone. |
| 20:43.41 | cyford | my problem is with the command just not sure if i set it in genneral and set it too 4 i dont want my peer trunk that every one uses to only limit everyone to 4 calls |
| 20:43.54 | WIMPy | 'core show functions like GROUP' |
| 20:43.59 | cyford | i want every extenstions to have be able to have 4 calls |
| 20:45.45 | cyford | that looks like for information purposes |
| 20:46.21 | cyford | GROUP GROUP([category]) Gets or sets the channel group. |
| 20:46.22 | cyford | GROUP_COUNT GROUP_COUNT([groupname][@category]) Counts the number of channels in the specified group. |
| 20:46.22 | cyford | GROUP_LIST GROUP_LIST() Gets a list of the groups set on a channel. |
| 20:46.22 | cyford | GROUP_MATCH_COUNT GROUP_MATCH_COUNT(groupmatch[@categ Counts the number of channels in the groups matching the specified pattern. |
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| 21:01.50 | afink | <PROTECTED> |
| 21:02.02 | afink | any ideas? |
| 21:09.40 | eZz | possibly nat |
| 21:09.49 | eZz | or codecs |
| 21:24.11 | [TK]D-Fender | afink: PB up a call with SIP debug enabled. |
| 21:24.13 | [TK]D-Fender | ~pb |
| 21:24.13 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
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| 21:50.37 | Freeaqingme | Does anybody know a bastion service I can use for voip (on linux)? |
| 21:56.30 | bluregard | are there any known issues in 10.x.x that causes peers to show up offline after a sip reload? |
| 21:58.31 | bluregard | I've tested this on version 10.0.0, 10.3.1 and 1.8.4.2 using identical sip.conf files. Every time with the 10.x.x versions after I do a sip reload, sip show peers shows all peers offline until they're forced to register which in the Polycom's case is a reboot. |
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| 22:39.09 | qakhan | i m using TDM800p card |
| 22:40.04 | qakhan | when i hangup call * cli shows channel hangup but my phone get still busy |
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| 22:50.56 | WIMPy | Freeaqingme: "bastion"? |
| 22:51.10 | WIMPy | bluregard: Are you using realtime? |
| 22:51.28 | bluregard | WIMPy, nope |
| 22:52.36 | bluregard | WIMPy, I did see something about this or something similar happening with realtime, but I'm just using the old boring sip.conf |
| 22:53.22 | WIMPy | That does not look like a known thing to me then. |
| 22:53.29 | bluregard | once I do a sip reload or core reload all sip peers show up offline and can't receive calls. |
| 22:54.10 | Freeaqingme | WIMPy, http://en.wikipedia.org/wiki/Bastion_host atm looking at opensbc |
| 22:54.13 | bluregard | but then if I restart the polycom or force xlite to reconnect everything is fine. |
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| 22:55.08 | WIMPy | That surely doesn't happen to me. And I can't remember having read about such behaviour. |
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| 23:36.30 | qakhan | i m using TDM800p card |
| 23:36.32 | qakhan | when i hangup call * cli shows channel hangup but my phone get still busy |
| 23:44.55 | bluregard | since 10.0.0 when astdb was converted to sqlite3 it says it should be using /var/lib/asterisk/astdb.sqlite3, yet core show settings has ASTDB = /var/lib/asterisk/astdb |
| 23:47.09 | bluregard | argh! permissions problem on /var/lib/asterisk |
| 23:52.41 | bluregard | WIMPy, asterisk couldn't write to /var/lib/asterisk/astdb.sqlite3. recursive chown on the directory fixed it |
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