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| 00:20.39 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.21.0 (2018/05/01), Standard: 15.4.0 (2018/05/01); 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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| 15:04.19 | dritskalle | is this a good place for a PAMI question? (the php AMI library).. or no? |
| 15:05.27 | [TK]D-Fender | As good as it gets |
| 15:09.59 | dritskalle | :) i have PAMI working, random events are returned by my catch-all event listener, but it does not trigger on any when i send commands |
| 15:10.11 | dritskalle | wondering if i am missunderstanding something is all.. |
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| 15:14.45 | dritskalle | if i just sit and listen - i get all registrations, login, calls, etc as expected.. then i send a command, e.g. MailboxCountAction(123) - i get isSuccess() but never any events with the data |
| 15:15.37 | dritskalle | sniffing i see the data coming - just isnt handled.. |
| 15:16.44 | igcewieling | sorry, I use phpAGI |
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| 19:58.15 | dritskalle | does anyone know what differences in chan_sip vs pjsip are in behavior s- CHANUNAVAIL and CONGESTION? |
| 19:58.45 | dritskalle | i used to get CHANUNAVAIL for a registered endpoint not reachable in chan_sip - it appears i get CONGEST with pjsip |
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| 20:02.19 | [TK]D-Fender | dritskalle, Show us the call attempt |
| 20:15.37 | dritskalle | okay - pasted some here: https://pastebin.com/L15SNASK |
| 20:17.05 | dritskalle | to me it doesnt relaly explain.. basically it dials a SIP softclient - but if that client was not running all the sudden but still registered - before i got CHANUNAVAIL, not i get CONGEST |
| 20:17.40 | [TK]D-Fender | <PROTECTED> |
| 20:17.50 | [TK]D-Fender | this says * actually placed a call |
| 20:18.05 | [TK]D-Fender | which it would not have if it didn't consider a contact valid. |
| 20:18.34 | [TK]D-Fender | Enable pjsip debug and try again |
| 20:18.55 | [TK]D-Fender | we need to be actually looking for an attempt to be placed |
| 20:20.05 | dritskalle | asterisk is not yet aware yet teh client is unreachable - that i do know, so pjsip is trying but fails - and just handles it differently than chan_sip. i can enable debugging, but pretty sure we will just see a an INVITE without a response |
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| 23:25.13 | chris349 | I have an error with inbound SIP trunk. It tells me there is a usernmame missmatch and its mixing up CALLERID with SIP username. How can I force SIP username always be used for SIP authentication and always ignore callerid for SIP authentication? |
| 23:29.34 | chris349 | I try to use fromuser=username but then the callerid gets set to the username! |
| 23:35.24 | Samot | I just answered this |
| 23:35.36 | Samot | 7:26:07 PM <Samot> That's not how that works |
| 23:35.36 | Samot | 7:26:15 PM <Samot> Nothing ever auths by the callerid number. |
| 23:35.36 | Samot | 7:26:25 PM <Samot> Do you have the same extensions on both boxes? |
| 23:36.40 | chris349 | They could overlap, yes. Maybe to someone who writes the asterisk code what you say is technically correct, but my calls are rejected and it says username mismatch, have <CALLERID NUMBER>, digest has <SIP USERNAME> |
| 23:36.55 | chris349 | So in my laymans terms asterisk is indeed trying to use the callerid number for authentication |
| 23:37.10 | Samot | OK. |
| 23:37.34 | Samot | I'm letting you know, nothing auths by the actual CallerID Number. |
| 23:38.01 | Samot | So if ext 250 exists on both boxes.. |
| 23:38.14 | Samot | 250 can't call the other box unless you do some work for it. |
| 23:38.50 | Samot | Because when 250 from Box A sends the call to Box B, Box B is going to go "Oh I have 250 here. Why is this coming from the wrong IP?" |
| 23:39.04 | chris349 | So what is it called? To me a username missmatch would be defined as a step that takes place during authentication. |
| 23:39.24 | Samot | The auth is by username and peer |
| 23:39.40 | chris349 | So how do I tell Asterisk that I have a SIP trunk between two machines and that username and callerid are different things? |
| 23:39.50 | Samot | It's going to see a call from an peer that it believes should be from a different IP. |
| 23:40.31 | Samot | pastebin your trunk settings from both boxes, mask any secrets/passwords. |
| 23:42.35 | chris349 | Heres what I got: https://pastebin.com/4M2PBVaX |
| 23:47.17 | Samot | That's only one of the peers |
| 23:49.10 | Samot | And which box is that from? |
| 23:49.14 | Samot | The one making the call? |
| 23:49.17 | Samot | Or receiving the call? |
| 23:49.24 | Samot | In the issue you are troubleshooting? |
| 23:52.45 | *** join/#asterisk Labyrinth00 (~Labyrinth@64.119.192.198) |
| 23:53.10 | Labyrinth00 | Greetings to all |
| 23:54.39 | Labyrinth00 | I was wondering if someone could point in the right direction. I am trying to find out if I can restrict chosen extensions from dial out on certain phone numbers |
| 23:56.04 | Samot | exten => [pattern]/[callerid],1,NoOP(Do stuff) |
| 23:57.00 | Samot | That will match what they sent with the fromuser field |