| 00:12.19 | *** join/#asterisk pvoigt (~Linux@unaffiliated/pvoigt) |
| 00:55.33 | velix | Ah, interesting: The chan_sip module has no mechanism to unregister/register without doing a sip reload. |
| 00:55.43 | velix | maybe that's I'm getting a forbidden |
| 00:56.37 | [TK]D-Fender | I don't see a relationship yet... |
| 00:58.58 | velix | [TK]D-Fender: Oops sorry, the outgoing TCP SIP I'm connection to is losing connection after a while. |
| 00:59.09 | velix | It timesout and then I'm getting a "forbidden". |
| 00:59.16 | velix | I need to reload SIP module, then it works again. |
| 00:59.37 | velix | let me go though wiresharp |
| 01:01.21 | velix | Interesting. After hundrets of "200 OK", I'm getting a "400 Bad Request". |
| 01:01.33 | velix | After that, acces is forbidden. |
| 01:01.40 | velix | When reloading SIP, I'm getting 200 OK again |
| 01:04.06 | velix | after 66 minutes ... hmm |
| 01:18.47 | velix | Processing REGISTER response 408 from server |
| 01:30.05 | velix | pjsip seems to be pretty buggy |
| 01:45.27 | *** join/#asterisk deavmi (~quassel@165.255.253.203) |
| 02:03.36 | velix | SIP/2.0 403 User agent not registered here |
| 02:24.56 | velix | The "to" is wrong |
| 02:28.15 | velix | anyone with an idea, which setting in pjsip sets the "to" ? |
| 02:28.20 | velix | I'm trying this for a week now |
| 02:54.08 | *** join/#asterisk eharris (~eharris@unaffiliated/eharris) |
| 03:06.07 | velix | Whoever wrote the documentation never used asterisk |
| 03:10.21 | velix | Anyone in here, who could help me to set up PJSIP? I'm able to pay via Paypal. |
| 03:14.24 | Kobaz | velix: 'sip unregister <peername>' |
| 03:18.44 | velix | Kobaz: I'm trying pjsip now, but it's really bad documented. |
| 03:20.22 | Kobaz | I doubt your issue will be fixed by pjsip, it really does look like a problem with the other server |
| 03:20.30 | Kobaz | I sent you some messages |
| 03:20.37 | Kobaz | You can use the pjsip conversion tool |
| 03:20.48 | velix | I tried it, it's totally wrote. It messes up username etc. |
| 03:20.48 | Kobaz | https://wiki.asterisk.org/wiki/display/AST/Migrating+from+chan_sip+to+res_pjsip |
| 03:20.53 | velix | Yeah, I tried it about a week ago |
| 03:20.54 | Kobaz | oh |
| 03:20.56 | velix | and then today again. |
| 03:21.09 | velix | somebody without knowledge of realms wrote it. |
| 03:21.50 | velix | I think, pjsip doesn't know how to deal with outbound proxies. |
| 03:21.57 | Kobaz | yes it does |
| 03:22.02 | velix | When looking at tcpdump, it's completly wrong. |
| 03:22.08 | velix | it uses "route" instead. |
| 03:22.54 | Kobaz | http://dpaste.com/26PTVD2 |
| 03:22.59 | Kobaz | here is an example of a trunk |
| 03:23.29 | Kobaz | type = identify is the inbound handling.. call comes in from match=<hostname> |
| 03:24.28 | Kobaz | type = endpoint is the outbound handling... send calls to the trunk-bob aor at contact= |
| 03:24.38 | Kobaz | aor is basically the outbound proxy |
| 03:25.12 | velix | Yeah, but the registration part is missing and pjsip cannot handle this correctly. |
| 03:25.51 | velix | Kobaz: In your example, where can I set the realm and the proxy? I've got two addresses. |
| 03:27.33 | velix | I'm on this for 15 hours now and I've tried all combinations possible. |
| 03:27.42 | Kobaz | registration... https://wiki.asterisk.org/wiki/display/AST/res_pjsip+Configuration+Examples |
| 03:27.48 | Kobaz | A SIP trunk to your service provider, including outbound registration |
| 03:29.20 | Kobaz | there are realm examples in the pjsip.conf.sample |
| 03:29.39 | Kobaz | [mytrunk_auth] type=auth .... realm=sip.example.com |
| 03:30.44 | velix | Yeah, I've tried this about 8 and 4 hours ago. |
| 03:30.52 | Kobaz | and what happens? |
| 03:31.18 | velix | Nothing. It it times out since pjsip uses "route", which doesn't end anywhere. |
| 03:49.11 | *** join/#asterisk sysgrammer_moe (~sysgramme@50.117.149.117) |
| 03:56.53 | velix | No response received from 'sip:pbx.e-b-z.de' on registration attempt |
| 03:57.43 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 03:58.09 | *** join/#asterisk NightMonkey (~NightMonk@pdpc/supporter/professional/nightmonkey) |
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| 05:53.35 | *** join/#asterisk CatCow97 (~mine9@c-73-96-109-206.hsd1.or.comcast.net) |
| 06:07.00 | velix | How can I spy into a connection, which has been created by originate ? |
| 06:10.50 | wyoung | Wireshark |
| 06:13.01 | [TK]D-Fender | Depends on your definition of "spy" |
| 06:15.19 | velix | I am using ChanSpy() for my SIP clients. |
| 06:15.28 | velix | But I'm directly connectiong two external numbers. |
| 06:15.38 | [TK]D-Fender | Which changes nothing. |
| 06:15.43 | [TK]D-Fender | those are channels just the same |
| 06:15.55 | velix | But how can I find the connection? |
| 06:16.03 | [TK]D-Fender | Chanspy still does the exact same thing |
| 06:16.15 | [TK]D-Fender | See the channel name? Done. |
| 06:16.18 | velix | My callfile is initiating the call from my trunk [university] to the external number |
| 06:16.33 | velix | They're all going off from the same account. |
| 06:16.42 | velix | Since there are 20 lines on it. |
| 06:16.46 | velix | channels* |
| 06:17.31 | [TK]D-Fender | then do something to make them identifiable |
| 06:17.40 | [TK]D-Fender | like setting a channel variable you'll attempt to look up, etc |
| 06:18.08 | [TK]D-Fender | but this is going to take more scripting skill |
| 06:18.13 | velix | But how can ChanSpy() find those? I mean, I have to look them up somehow. |
| 06:18.25 | [TK]D-Fender | You don't go searching with chanspy obviously |
| 06:18.32 | [TK]D-Fender | to tell it exactly which channel to spy on |
| 06:19.21 | velix | Maybe it works with SPYGROUP ? |
| 06:21.04 | [TK]D-Fender | That's an idea.... |
| 06:21.45 | velix | damn you, echo service. I can't stand my voice anymore :D |
| 06:25.48 | velix | It seems to work, but I can hear the voice saying "SIP" ;) |
| 06:27.55 | velix | Interesting, where does this come from? |
| 06:28.54 | [TK]D-Fender | As in? |
| 06:31.43 | velix | When I'm ChanSpying, it says "SIP" |
| 06:31.49 | velix | Whenever I initiate a call :D |
| 06:32.01 | velix | A female voice on the spy site. |
| 06:32.35 | [TK]D-Fender | How is "where" a question? |
| 06:33.17 | [TK]D-Fender | because * supports pile of different techs and since youcould be putting something vague in for your search you should know the difference ebtween the channels you're cycling through |
| 06:35.21 | velix | Ok. I just never heard it before :) |
| 06:35.45 | velix | I just realied I cannot listen to an established call this way: exten => 9000,1,ChanSpy(,g(123),q) |
| 06:36.34 | [TK]D-Fender | You're already inventing a 3rd parameter there |
| 06:37.06 | velix | meh ;) |
| 06:38.38 | velix | Yeah, but listening to an established call doesn't work :/ |
| 06:38.40 | velix | Interesting |
| 06:39.37 | [TK]D-Fender | I'm thinking you did something else wrong there... |
| 06:40.01 | [TK]D-Fender | Because that sounds incredibly backwards |
| 06:41.39 | velix | This is my callfile: https://bpaste.net/raw/KVLQ |
| 06:42.30 | velix | When one phone rings, the call to the second gets established. Uses two channels, but works. |
| 06:43.01 | [TK]D-Fender | ... |
| 06:43.09 | *** join/#asterisk tomaluca95 (~quassel@kde/developer/tomaluca) |
| 06:43.24 | velix | This is for external users, who don't have a SIP compatible phone. |
| 06:43.41 | [TK]D-Fender | That means nothing so far... |
| 06:44.22 | velix | And I'm using exten => 9000,1,ChanSpy(,g(123)) to listen to this group. |
| 06:44.34 | velix | When I *first* listen and the call comes in, it works. |
| 06:44.44 | velix | But when the call is established already, I don't hear anything. |
| 06:45.15 | [TK]D-Fender | If you are allowing REINVITES then the audio wouldn't be passing through * |
| 06:45.34 | [TK]D-Fender | You'd better be blocking those.... |
| 06:46.53 | velix | canreinvite=no ? |
| 06:47.12 | [TK]D-Fender | directmedia=no |
| 06:48.35 | velix | Hmm, no change :/ |
| 07:02.54 | [TK]D-Fender | Nothing to look at and bed-times here ... maybe later... |
| 07:03.39 | velix | good night ;) |
| 07:15.04 | *** join/#asterisk life_of_e (~life_of_e@108-95-189-245.lightspeed.irvnca.sbcglobal.net) |
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| 10:10.56 | velix | How many RTP ports are needed? |
| 10:26.52 | *** join/#asterisk DodgeThis (~DodgeThis@2001:818:df17:3000:1c17:7161:fefd:78b8) |
| 11:36.56 | sibiria | 2 per call |
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| 12:33.45 | velix | sibiria: I've just read 4 per call is better |
| 12:37.56 | sibiria | only if you insist on not using symmetric rtp |
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| 12:39.09 | *** join/#asterisk twanny796 (~user@antazzo.com) |
| 12:39.47 | file | it's 2 per stream on normal endpoints, or 1 for all if using WebRTC |
| 12:43.01 | Kobaz | velix: sounds like you're making calls... did you figure out the pjsip issue? |
| 12:43.21 | velix | Kobaz: Nope. SIP is working great. |
| 12:44.00 | velix | Kobaz: tried it 4 hours after you left. |
| 12:44.02 | velix | gave up |
| 12:44.11 | Kobaz | so if sip is working. you're not blocked |
| 12:44.22 | Kobaz | basically, compare the packet dumps of the REGISTER |
| 12:44.29 | Kobaz | chan_sip vs pjsip.. and see what's missing |
| 12:44.35 | velix | Kobaz: They're completly different :( |
| 12:44.47 | velix | I'll do this in the next days, I'm too tired :( |
| 12:44.47 | Kobaz | but will have the same core components |
| 12:45.18 | velix | A friend said, don't use pjsip, better go for FreeSWITCH |
| 12:45.35 | sibiria | those are two different pieces of software |
| 12:45.49 | velix | dunno ;) |
| 12:46.24 | Kobaz | i like pj quite a lot, but it could use some improvements |
| 12:46.30 | *** join/#asterisk zapata (~zapata@2a02:1748:fad4:7260:a98d:289a:9413:3f28) |
| 12:46.45 | Kobaz | it'll give me an incentive to work on asterisk development again |
| 12:47.37 | velix | oh my: 3CX turned free for 3 years today. |
| 12:47.38 | velix | unlimited |
| 12:51.00 | velix | Kobaz: I really wonder, why the converted has problems with my working SIP.conf |
| 12:51.43 | Kobaz | the conversion isn't perfect |
| 12:51.58 | Kobaz | i had to fix some items when i did a conversion for one system |
| 12:52.32 | velix | In my eyes, my config isn't that special. I only have an internal and an external IP. |
| 12:53.56 | Kobaz | well you have realm and some sort of server identity that's different from your sip proxy |
| 12:54.01 | Kobaz | is my understanding |
| 12:56.12 | Kobaz | oh |
| 12:56.16 | Kobaz | i see the problem |
| 12:56.58 | Kobaz | my only excuse was that it was super late and I was exhausted |
| 12:58.04 | velix | same here, I'm awake for 21 hours again. |
| 22:47.50 | *** join/#asterisk infobot (ibot@c-174-52-60-165.hsd1.ut.comcast.net) |
| 22:47.50 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.32.0 (2020/03/12) 16.9.0 (2020/03/12) Standard: 17.3.0 (2020/03/12); DAHDI: 3.0.0 (2018/11/15); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 |
| 23:01.53 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 23:39.08 | *** join/#asterisk DannyA (~DannyA@cpe-74-64-125-9.nyc.res.rr.com) |
| 23:39.30 | Nivex | and I may have just figured out the wizard reload issue (direct_media => endpoint/direct_media) |
| 23:39.45 | DannyA | hey all. i want to execute a cURL command after every phone call, to do my own CDR logging. is there a way to continue in the dialplan, or call a sub, when either party hangs up? |
| 23:40.42 | file | https://wiki.asterisk.org/wiki/display/AST/Hangup+Handlers |
| 23:42.53 | DannyA | i was just reading that. and that will happen when either party hangs up? |
| 23:43.24 | file | it's attached to a channel |
| 23:43.29 | file | when that channel is hung up, it is executed |
| 23:44.17 | DannyA | gotcha. and i was having a hard time understanding how it's different from using an "h" extension |
| 23:46.35 | file | it follows the channel, no matter what |