| 00:10.09 | *** part/#asterisk-dev kharwell (kharwell@nat/digium/x-nshbhbzqyddgfezb) |
| 01:16.29 | Kobaz | what's a good thing to look for when your chan_sip dialogues are getting autodestroyed before their time? (Asterisk sends a BYE to due some internal happenings during call setup) |
| 01:16.46 | Kobaz | [2017-11-20 12:25:09.991] WARNING[7309] chan_sip.c: Autodestruct on dialog '1926a28b52eefd9d4e3af4c740e23a8a@192.168.50.1:5060' with owner SIP/104-00000ce3 in place (Method: BYE). Rescheduling destruction for 10000 ms |
| 01:20.59 | coreyfarrell | are you using SIP over TCP/TLS/WS? |
| 01:21.17 | Kobaz | nope |
| 01:23.47 | coreyfarrell | I'd probably start with the SIP debug so I can trace the send/receive related to that dialog. |
| 01:24.01 | Kobaz | yeah |
| 01:24.14 | Kobaz | the problem is, it's very random |
| 01:24.30 | Kobaz | although i did turn on debug 3 for chan_sip... so i can wait for it to happen again |
| 01:25.29 | coreyfarrell | 'sip debug on' so you get SIP traffic in the logs |
| 01:25.35 | Kobaz | that too |
| 01:28.22 | coreyfarrell | Kobaz: it could turn out to be harmless message caused by special timing. if the same dialog gets rescheduled multiple times that would be more concerning. |
| 01:28.59 | Kobaz | well the issue i'm having is that the call starts and then just dies |
| 01:29.06 | Kobaz | and then you call back in and then it works fine |
| 01:29.53 | coreyfarrell | ah, that's different. what version asterisk? |
| 01:30.01 | Kobaz | 11.25.3 |
| 01:30.26 | Kobaz | i promise we'll be upgrading to 15 soon |
| 01:30.54 | Kobaz | we just have a serious amount of infrastructure that relys on tracking channels via masquerades and all that, that needs a lot of testing |
| 01:34.45 | coreyfarrell | Kobaz: between 11.25.3 and latest 13 about 4000 lines of chan_sip.c are different.. zero chance of anyone looking into an issue with chan_sip on 11. |
| 01:34.50 | Kobaz | yeah |
| 01:35.20 | Kobaz | just was curious as to getting pointed in the right direction, with respect to authdestruct |
| 01:35.26 | Kobaz | *auto |
| 01:35.44 | Kobaz | this could very well affect a newer version.... who knows |
| 01:36.42 | coreyfarrell | you need dialog history to determine why the autodestruct was scheduled in the first place. |
| 01:38.22 | Kobaz | k |
| 01:48.24 | Kobaz | random different topic |
| 01:48.30 | Kobaz | how would i get the channel associated with a thread |
| 01:48.48 | Kobaz | trying to use instance->chan, but it's not defined |
| 01:48.52 | Kobaz | [2017-11-20 20:47:05.442] > [C-0000001c] <NoChannel> 0x879b4b8 -- Strict RTP switching to RTP remote address 192.168.50.116:2234 as source |
| 01:49.10 | Kobaz | but immediately prior.. the previous line: [2017-11-20 20:47:05.376] -- [C-0000001c] <SIP/1010-0000002b> Internal Gosub(CallRouter_Pickup,s,1) complete GOSUB_RETVAL= |
| 01:49.22 | Kobaz | we know the channel we're running on, but the rtp instance does not.... yet |
| 01:50.14 | Kobaz | i figure if [C-0000001c] can introspect itself and get the channel it's currently handling... that would be useful |
| 01:50.38 | Kobaz | well, the thread that's handling id C-0000001c |
| 01:58.38 | coreyfarrell | you can't get a pointer to the ast_channel from a thread. you can use the logger callid to associate messages in the logger. |
| 01:59.54 | Kobaz | right, you can visually associate.... what if i wanted to get the owner channel from an rtp instance |
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| 02:03.40 | coreyfarrell | you can't. the rtp instance doesn't know anything about the channel it belongs to. |
| 02:05.40 | Kobaz | hmm |
| 02:07.58 | snuff-work | mm.. has there be any change to pjsip timer related code in the last 1-2 months.. wondering if i should try re-test my issue with asterisk terminating calls |
| 02:08.07 | snuff-work | prematurally |
| 02:11.25 | Kobaz | naming question for you guys |
| 02:12.27 | Kobaz | Call from SIP/itsp-00000040 -> SIP/1010-00000041.... SIP/1010-00000042 then calls SIP/1001-00000043 |
| 02:13.10 | Kobaz | and then you do a sip replaces transfer... |
| 02:13.26 | Kobaz | The caller=SIP/itsp-00000040. The callee=SIP/1010-00000041 and transfer target=SIP/1001-00000043 |
| 02:13.33 | Kobaz | that leaves SIP/1010-00000042... what would you call this channel? |
| 02:14.11 | Kobaz | i have weird names for this stuff and it's been always confusing |
| 02:14.21 | Kobaz | transferer, transferee |
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| 16:17.44 | coreyfarrell | kharwell: ended up with only a tiny change to the cli_completion commit. it was getting too large so I just continued with additional commits. the commits on gerrit complete the CLI core completion work. |
| 16:22.30 | Kobaz | what sort of exciting stuff are you working on in the cli? |
| 16:25.46 | coreyfarrell | Kobaz: I was working on performance issues with CLI completion. at this point all work to CLI/console itself is on gerrit. only thing remaining is to use the new completion function from poorly performing completion routines. |
| 16:25.59 | Kobaz | ah |
| 16:28.58 | kharwell | coreyfarrell: ack |
| 16:29.25 | Kobaz | does ctrl-d work in trunk? |
| 16:29.35 | Kobaz | to delete-forward |
| 16:33.18 | coreyfarrell | I have no idea, try it. but maybe that's a readline feature rather than an asterisk feature? |
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| 16:34.23 | file | pretty sure it is |
| 16:35.17 | Kobaz | that should be in readline, yeah, but classicly it hasn't worked |
| 16:38.52 | coreyfarrell | file: do you have an opinion on what load prioirity res_parking should use on master? discussion at https://gerrit.asterisk.org/7312 |
| 16:46.04 | coreyfarrell | Kobaz: asterisk source contains an (ancient) embedded copy of readline at main/editline. I think it's used automatically if you don't have editline headers on your system. |
| 16:52.40 | Kobaz | ah |
| 16:52.49 | file | coreyfarrell: I haven't looked, I'm out |
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