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| 03:19.04 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.18.2 (2017/11/10), Standard: 15.1.2 (2017/11/10); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 09:16.21 | RomanDcoz | Hello, I am behind a NAT, when try to register gateway showing UNREACHABLE, Is there any solution how to make it as OK status, i tried to add Extip and localIP in SIp configuration |
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| 09:35.15 | RomanDcoz | ANyone who can guide me for NAT registration issue for net2phone sip trunk ? |
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| 10:40.54 | [[thufir]] | what does it mean that fastAGI replaces the dialplan? |
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| 13:10.15 | lolose | Hello. How do I make pjsip to forget old registrations? It keeps denying an endpoint to register because the contact address has changed and it would exceed max contact of 1 |
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| 13:18.33 | [TK]D-Fender | lolose, there is a parameter to have it replace the previous one.... |
| 13:19.54 | lolose | [TK]D-Fender, Could you please elaborate a little further? Is that a setting in pjsip.conf that I should be using? |
| 13:20.08 | [TK]D-Fender | yes |
| 13:20.26 | lolose | I'll look it up. thanks |
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| 15:21.46 | BR-Saldanha | Hi everyone, i'm getting this error on my Inbound Calls: Connected to Asterisk 11.25.0 currently running on issabel (pid = 14952) |
| 15:21.46 | BR-Saldanha | [2017-11-20 13:18:00] NOTICE[15002][C-00000001]: chan_sip.c:25906 handle_request_invite: Failed to authenticate device "031987997411" <sip:031987997411@XXX.XXX.XXX.XXX:5060>;tag=556741892 |
| 15:22.16 | BR-Saldanha | My trunk has this configurations: host=10.0.15.228 |
| 15:22.16 | BR-Saldanha | type=peer |
| 15:22.16 | BR-Saldanha | context=from-internal |
| 15:22.16 | BR-Saldanha | nat=no |
| 15:22.16 | BR-Saldanha | qualify=yes |
| 15:22.28 | BR-Saldanha | Anyone can help me? |
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| 15:38.54 | [TK]D-Fender | ?pb |
| 15:39.01 | [TK]D-Fender | www.pastebin.com <------------ |
| 15:39.17 | [TK]D-Fender | And go enable SIP debug and look at the full call attempt |
| 15:53.05 | BR-Saldanha | Can you explain me how I can debug sip ? My log show only this: : chan_sip.c:25906 handle_request_invite: Failed to authenticate device "031987997411" <sip:031987997411@10.0.15.227:5060>;tag=198882740 |
| 15:53.40 | [TK]D-Fender | "asterisk cli enable sip debug" <- Google it |
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| 16:05.02 | mn1 | I'm using MixMonitor on Asterisk 11 with the 'b' option but i'm ending up with some stereo recordings where the channels aren't lined up properly. The reading I did online suggested the 'b' MixMonitor option should've taken care of this. Is this a known issue still? Any known workaround? |
| 16:05.11 | mn1 | I'm merging the recordings with a sox -M |
| 16:05.47 | [TK]D-Fender | 11 = dead |
| 16:05.55 | [TK]D-Fender | 13 or 15 are the only supported branches |
| 16:06.28 | mn1 | yeah for sure, we've got plans to upgrade in the future. just trying to see if there's a workaround for this while we're on 11 still. |
| 16:06.43 | mn1 | is this issue known to be fixed in 13+? |
| 16:09.19 | [TK]D-Fender | Check the tracker |
| 16:09.53 | [TK]D-Fender | https://issues.asterisk.org/jira/secure/Dashboard.jspa |
| 16:25.26 | matt_ | is there an 'ignoresdpversion' flag for pjsip? |
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| 20:33.05 | hans109h | After upgrade to Ubuntu 17.10 the only way that I can get my DAHDI channel to work is to first unload the DAHDI and WCFXO, then reload DAHDI and WCFXO then restart Asterisk. Any suggestions on why the channel might not load on boot? The error I get from asterisk is: ERROR[2049] chan_dahdi.c: Unable to open channel 1: Invalid argument |
| 20:33.05 | hans109h | here = 0, tmp->channel = 1, channel = 1 |
| 20:50.54 | Samot | Not that I use DADHI |
| 20:51.08 | Samot | But Ubuntu 17.10 is 2 months old. |
| 20:51.12 | Samot | In stable release. |
| 20:51.41 | Samot | It is possible that there could be something with DAHDI/DAHDI-tools that doesn't jive in 17.10 |
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| 21:45.24 | AsteriskLearner | hello people |
| 21:47.46 | AsteriskLearner | can somebody please help me one second... I registred the sip account that I got from vitelity and I want that sip to call the Agi script... I'm new with asterisk and I will apreciate if somebody will help me than you |
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| 21:49.30 | AsteriskLearner | :-/ |
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| 22:13.40 | asteriskmonkey | anyone seen this before? Command 'module load res_http_media_cache.so' failed. |
| 22:17.41 | asteriskmonkey | nm |
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| 22:21.57 | [TK]D-Fender | AsteriskLearner, it's your dialplan, go match the inbound call and call your AGI |
| 22:36.26 | hans109h | Samot, that was my first thought, it's just odd that it all works correctly after an unload/load, so it seems like it might be something really simple like modifying some systemd requirements, I"m just not sure which ones. |
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