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| 03:43.07 | Lope | I want to configure asterisk to receive calls from my DID number, and forward them to my softphone. I'd like to use IAX2. Can you recommend steps or a guide for doing this on ubuntu 14.04? |
| 04:00.27 | [TK]D-Fender | If you have * installed then your OS doesn't matter |
| 04:00.40 | [TK]D-Fender | As for the rest .... Just DIal() your device |
| 04:00.48 | [TK]D-Fender | ~book |
| 04:00.51 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 04:17.59 | Lope | asterisk is currently listening on my server on 2000 SCCP, 5000 Unistim, 4520 Dundi, 5060 SIP, 4569 IAX2... I only want to receive incoming calls on IAX2 from a DID and forward them to an IAX2 softphone. How can I disable everything that's unnecessary? |
| 04:34.37 | [TK]D-Fender | modules.conf <- |
| 04:34.43 | [TK]D-Fender | noload => |
| 04:34.52 | [TK]D-Fender | for the ones you know you're not going to need |
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| 04:56.40 | Lope | do you think this quickstart guide might be outdated? http://www.asterisk.org/sites/asterisk/files/mce_files/documents/asterisk_quick_start_guide.pdf With the default config I had "2000 SCCP, 5000 Unistim, 4520 Dundi, 5060 SIP, 4569 IAX2" Now I set /etc/asterisk/modules.conf according to the quickstart recommendation, restarted asterisk and now asterisk is not listening on any ports at all :/ hehe |
| 05:18.17 | [TK]D-Fender | Old, but close enough |
| 05:18.58 | [TK]D-Fender | You either disabled too many modules or did another kind of mistake like not autoloading the rest |
| 05:19.07 | [TK]D-Fender | Or broke a config in general |
| 05:19.42 | [TK]D-Fender | THat guide's page for it ... not good |
| 05:19.57 | [TK]D-Fender | because that also doesn't include any of the basic dialplan apps & functions modules, etc |
| 05:20.19 | [TK]D-Fender | That thing is a mess |
| 05:20.37 | [TK]D-Fender | restore it to the way it was and selectively NOLOAD only the channel drivers you don't want at all |
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| 07:14.15 | tripleslash | I have a weird situation. Fresh start of asterisk, voicemail hints are generated as expected. core show hint *98 shows several lines such as *989000@app-dialvm : MWI:9000@default State:Idle Presence:not_set Watchers 1 which is what a person expects. |
| 07:14.20 | tripleslash | Once I do a reload of asterisk (core reload), MWI breaks. |
| 07:14.24 | tripleslash | it only shows as MWI:9000@ |
| 07:14.31 | tripleslash | doing a asterisk core restart now/gracefully/when convenient restores the hints until the next reload |
| 07:14.44 | tripleslash | Any pointers as to how I can trace this down? |
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| 09:01.08 | conosj | I can call out, but cannot receive incoming call. Please help! :) |
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| 09:57.12 | Lope | I'm trying to receive a SIP call from my DID and forward it to my SIP softphone. But when someone calls my DID, I just see asterisk looping around a few times, and then the call is disconnected. The softphone never rings. Any ideas? Here's the debug http://codepad.org/rlylBXQT where 1234567... is my DID number |
| 09:59.53 | Lope | I've just got 7 modules loaded now. |
| 10:00.54 | Lope | res_crypto.so, chan_iax2.so, pbx_ael.so, res_ael_share.so, chan_local.so, chan_sip.so, res_http_websocket.so |
| 10:02.49 | Lope | extensions.ael is only "context guest { 12345678901 => { Dial(testsip00; }; }" |
| 10:04.18 | Lope | Typo correction, extensions.ael is only "context guest { 12345678901 => { Dial(testsip0); }; }" |
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| 10:35.24 | Kunsi | hello, is it possible to allow all sip users from 172.44.32.0/24 to place calls to outside, even if they are not logged in? |
| 10:35.56 | WIMPy | There is no such thing as "logged in". Each single call is authenticated. |
| 10:36.27 | WIMPy | And yes, you can create a peer with 'host' but without 'secret'. |
| 10:36.46 | Kunsi | ah, that's what i wanted to know |
| 10:36.50 | Kunsi | thank you ð |
| 10:47.33 | Lope | extensions.ael contains: context myphones { testsip0 { Dial(SIP/testsip0); }; } |
| 10:47.47 | Lope | NOTICE[28405][C-00000003]: chan_sip.c:25450 handle_request_invite: Call from 'testsip1' (192.228.169.212:1024) to extension 'testsip0' rejected because extension not found in context 'myphones'. |
| 10:53.32 | Lope | I also tried this now, without any improvement in error message: _X./testsip0 => { Dial(SIP/testsip0); }; |
| 10:54.33 | Lope | I did a core reload with verbosity and debug set to 9. No unusual errors. |
| 10:55.32 | WIMPy | Thee are probably not many people who know the ael syntax. Try 'dialplan show' to find out what the result is. |
| 10:56.24 | Lope | '_X.' (CID match 'testsip0') => 1. Dial(SIP/testsip0) |
| 10:56.37 | Lope | is CID the caller ID of the incoming caller? |
| 10:56.46 | WIMPy | yes |
| 10:56.56 | Lope | okay, then that's wrong |
| 10:57.03 | WIMPy | So testsip0 is allowed to dial any number. |
| 10:58.07 | Lope | yeah that's fine |
| 10:58.13 | Lope | it's still not working |
| 10:58.24 | Lope | basically I want testsip1 to be able to dial testsip1 |
| 10:58.38 | Lope | but I keep getting the above NOTICE |
| 10:58.39 | WIMPy | Not allowed |
| 10:58.55 | Lope | WIMPy: what do you mean, not supported by asterisk? |
| 10:58.56 | WIMPy | So remove that CID matching part. |
| 10:59.04 | Lope | okay |
| 10:59.10 | WIMPy | By your configuration. |
| 11:00.11 | Lope | now show dialplan says: 'testsip0' => 1. Dial(SIP/testsip1) 'testsip1' => 1. Dial(SIP/testsip0) |
| 11:00.52 | Lope | Warning. no application Dial for extension myphones, testsip0, 1 |
| 11:00.53 | WIMPy | Looks a little mixed up. |
| 11:01.10 | Lope | Yeah I just pasted one line after the next, people on IRC don't like floods of lines. |
| 11:01.28 | Lope | so maybe I need to load some module to dial? |
| 11:01.37 | WIMPy | Looks like you need to load some modules. Unless you know better, enable autoload. |
| 11:01.58 | WIMPy | No, the Dials don't match the extensions. |
| 11:02.02 | Lope | WIMPy when I autoload them asterisk listens on so many ports that I don't want it to |
| 11:02.06 | Lope | I'd rather just load what I need? |
| 11:02.46 | Lope | No application 'Dial' for extension (myphones, testsip0, 1 == Spawn extension (myphones, testsip0, 1) exited non-zero on 'SIP/testsip1-00000000' |
| 11:04.47 | Lope | app_dial.so was not loaded. |
| 11:06.10 | Lope | lol, ok I successfully called myself, it seems, from testsip1 to testsip1... seems a little mixed up |
| 11:06.25 | WIMPy | That's what I said. |
| 11:07.03 | Lope | WIMPy yes, you did. I obviously interpreted the params wrong. |
| 11:08.49 | Lope | Ok I managed to connect a call, but there's no audio |
| 11:09.30 | WIMPy | Did you load rtp modules? |
| 11:09.35 | Lope | DTMF works from testsip0 > testsip1 but does not work from testsip1 > testsip0 |
| 11:09.46 | Lope | WIMPy: yes, I did load all the RTP modules |
| 11:11.11 | WIMPy | The joys of SIP. You should read up on SIP and NAT to get a understanding of all the things that can go wrong, including the things that go wron when they should work, because of clever thing to make |
| 11:11.17 | WIMPy | it work when it shouldn't. |
| 11:12.27 | Lope | gonna test the mic on my testsip0 because the DTMF from testsip0 goes through to testsip1. But no audio. |
| 11:13.03 | Lope | WIMPy: yeah, I don't like the SIP protocol, I prefer IAX2. but my provider DIDX wasn't playing nice with me :/ now I'm using DIDforsale and they only support SIP. |
| 11:13.23 | Lope | well, didx was just not working, website errors etc. |
| 11:18.14 | Lope | Okay cool. Pulseaudio wasn't working, on my laptop (testsip0) but Alsa does. |
| 11:18.24 | Lope | So I've got DTMF and voice going one way. |
| 11:18.31 | Lope | So how can I debug this shiz? |
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| 11:19.17 | WIMPy | hates pulseaudio. |
| 11:19.25 | WIMPy | What do you want to debug? |
| 11:19.41 | Lope | I need sound to go from testsip1 > testsip0 |
| 11:21.03 | Lope | testsip0 and testsip1 are identical in sip.conf. allow=!all,ilbc,g729,g723,ulaw |
| 11:22.37 | Lope | I tried allow=all. That didn't help |
| 11:30.08 | Lope | okay I've loaded every codec module that asterisk has. |
| 11:30.49 | Lope | all 10 of them. if I choose allow=all then I lose audio going from 0 > 1 |
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| 17:26.20 | coreyfarrell | anyone good with SELinux know the correct way to permit sendmail from asterisk (for app_voicemail)? I know I can repeatedly use audit2allow to get each permission that is needed, but I'd rather use a proper policy interface macro. using Centos7 with asterisk 13 based on fedora RPM's. |
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| 20:02.36 | Tagor | what causecode on hangup should be used for 'nobody picked up the phone'? 18? 21? 16? 17? |
| 20:06.39 | [TK]D-Fender | To do what? |
| 20:09.40 | Tagor | [TK]D-Fender: PSTN call > VOIP > Hangup() after 30 seconds if phone is not picked up |
| 20:10.22 | [TK]D-Fender | If you let the call fall through It'll already carry the status from the dial |
| 20:11.26 | Tagor | [TK]D-Fender: ok, thanks and happy new year :) |
| 20:38.08 | WIMPy | Tagor: 19, but app_dial should hopefully set that for you. |
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| 23:26.01 | WebSprocket | Hi, We have a box behind a DMZ but all external calls come in showing the following channel Answer("SIP/10.99.1.79-00000011", "") |
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