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| 00:19.29 | gregs | X-Rob: I'm looking at the PJSIP Early deletion thing again. Maybe this is nothing but did you notice that when Asterisk deletes the contact it always had a '/' after the extension number? |
| 00:20.23 | X-Rob | gregs: yes. I'm in the middle of doing a git bisect at the moment. |
| 00:21.05 | gregs | There was a change made in https://issues.asterisk.org/jira/browse/ASTERISK-25826 which introduced some regex comparison functions to make things more efficient along with the contact_expiration_check_interval setting. |
| 00:21.28 | gregs | Wondering if its trying to compare two things that aren't quite the same and then deciding to terminate the contact as a result. |
| 00:22.03 | gregs | Hope that made sense. |
| 00:23.42 | gregs | The changelog has a good summary of what was done... |
| 00:23.46 | gregs | https://www.irccloud.com/pastebin/Z0sMj4Dt/ |
| 00:24.07 | X-Rob | Yep, that's where my first bisect pointed me too |
| 00:24.12 | X-Rob | but haven't confirmed it properly yet |
| 00:25.23 | gregs | Cool. Not sure I'm going to be much help but will keep looking. |
| 00:27.32 | X-Rob | I'm doing a bisect between 13.8.2 and 13.9 at the moment |
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| 00:52.53 | X-Rob | [root@localhost asterisk]# git bisect bad |
| 00:52.53 | X-Rob | Bisecting: 5 revisions left to test after this (roughly 2 steps) |
| 00:52.53 | X-Rob | [3528a9a42e82d1fd6e07ef0b591fa3c81baec3fb] Merge "res_parking: Misc fixes." into 13 |
| 00:52.57 | X-Rob | getting there, gregs 8) |
| 00:53.26 | gregs | ð |
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| 01:07.50 | X-Rob | 7b7e3909e492df29d5d51e885f9328a23e61cb6a] Merge "sorcery/res_pjsip: Refactor for realtime performance" into 13 |
| 01:07.59 | X-Rob | I'm feeling the vibe that this is the one that breaks it |
| 01:10.26 | gregs | I agree. |
| 01:12.11 | gregs | My money's on the changes in res_pjsip_registrar_expire.c |
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| 01:58.04 | gregs | X-Rob: Good to see that has been fixed. Wonder if it will roll into a 13.9.1 or will have to wait for a 13.10 release? |
| 01:58.25 | X-Rob | gregs: this is a pretty bad bug, so I'll be surprised if there's not a 13.9.1 out tomorrow or so |
| 01:58.45 | gregs | X-Rob: Hope so. Thanks for your help tracking it down. |
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| 03:27.00 | DELLtra | hi 0/ |
| 03:27.04 | DELLtra | spanish ? |
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| 07:35.33 | utrack | Arrrr. |
| 07:35.57 | utrack | Is there any way to retrieve sane ANSWEREDTIME for Originate() calls?... |
| 07:36.17 | utrack | My plan works if there's a real call made via softphone (not Originate()d). |
| 07:36.45 | utrack | I found some clues that it's harder than it should be http://forums.asterisk.org/viewtopic.php?p=200368 |
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| 07:53.10 | wyoung | utrack: nah, that module has bipolar |
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| 07:57.44 | utrack | wyoung: but somebody, somewhere got this fixed!... :( |
| 07:58.18 | utrack | that's pretty basic problem tbh, just no clues on how to fix it for newbs |
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| 09:06.19 | utrack | Here's my current dialplan: http://pastie.org/private/2lndhon5bi8rdosoy5iwrg |
| 09:07.00 | utrack | THe call is originated to Local/200@internal and one of XXXXs |
| 09:08.10 | utrack | CDR(billsec) and CDR(duration) contain gibberish (around the time that's needed for call initialization+phone pickup, not sure) |
| 09:08.58 | utrack | However, when I try to place the call from 100 (TestPhone-A) to the external XXXX one - everything works as expected, CDR has everything that's needed, timings are OK |
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| 09:11.43 | utrack | Oopsie, typos - fixed http://pastie.org/private/a08mi9ddnpnrqnchbamog |
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| 11:50.20 | jusle | Any recommendations for SIP providers in (Northern) Europe? |
| 11:51.51 | wyoung | depends where you want to call |
| 11:52.04 | wyoung | if you want to call AU then get pennytel.com :) |
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| 11:59.33 | jusle | I am in Finland and so are most of my customers. My servers are in Germany. Closer the better, I believe? |
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| 16:09.05 | DELLtra | I need to raise an asterisk Server sangoma a102 have a card that is connected to the main line of the operator and the other connected to a PBX telephone . |
| 16:09.05 | DELLtra | The only function that makes Asterisk Server is the etching of the outgoing or incoming calls was shared via Samba |
| 16:10.13 | *** join/#asterisk flTech (FLtech@c-73-23-227-101.hsd1.fl.comcast.net) |
| 16:11.11 | flTech | Hi all. I have a problem where, when calling outbound, recipient voice doesn't reach caller until 16 seconds after establishment of connection. Using Incredible PBX 13.2 |
| 16:12.06 | flTech | I've been asked to try a different sip trunk (original is Comcast through ADTRAN) but I'm not getting sound at all when going through Juniper firewall. |
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| 16:13.37 | flTech | hello. |
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| 17:13.11 | flTech | can anyone assist me? |
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| 17:42.38 | flTech | anyone there? |
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| 17:52.18 | newtonr | flTech, plenty of people are here. Most people won't respond unless they are not busy or someone asks a specific question. :) |
| 17:52.20 | newtonr | howdy! |
| 17:52.39 | newtonr | I'm not exactly free at the moment, but I'll do my best to answer your question once you ask it. |
| 17:54.53 | [TK]D-Fender | flTech, Show the actual call. We need evidence if we're to advise |
| 17:54.58 | [TK]D-Fender | ~pb |
| 17:54.58 | infobot | 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.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 17:54.59 | [TK]D-Fender | ^^^ |
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| 19:36.13 | acidfoo_ | how can I show the "contexts" from the CLI ? |
| 19:36.39 | WIMPy | dialplan show <tab> |
| 19:37.05 | acidfoo_ | ah! good thanks |
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| 20:57.07 | flTech | sorry. I didn't see this response |
| 20:59.18 | flTech | How do I troubleshoot a one sided audio? |
| 21:00.14 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 21:00.48 | flTech | The call connects, the voice outbound is choppy at the recipient end, but no audio from recipient. After 10 seconds, the recipient can hear the call |
| 21:01.08 | flTech | after 16 seconds, the caller can hear the recipient. |
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| 21:54.26 | Synthase_ | I can take a wild guess and say that sounds related to an RTP stream issue via firewall/NAT traversal, but you will need to collect some troubleshooting information before anyone can actually assist you. |
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| 22:26.17 | flTech | but there isn't any firewall |
| 22:34.57 | flTech | what would be troubleshooting information that would be important? I have some sip debug logs... |
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| 23:37.03 | *** join/#asterisk ghoti (~paul@hq.experiencepoint.com) |
| 23:49.19 | *** join/#asterisk MaliutaLap (nikolai@unaffiliated/maliuta) |