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| 00:27.50 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.7.2 (2016/02/05), 11.21.2 (2016/02/11); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 01:19.19 | snadge | is there a way to packet trace IAX and filter out the rtp packets? |
| 01:19.22 | snadge | i hate iax |
| 01:22.40 | WIMPy | There is no RTP. |
| 01:27.00 | [TK]D-Fender | ^^^^ |
| 01:28.30 | WIMPy | But I think the official answer is don't use IAX any more :-( |
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| 02:02.24 | qloogkm | does anyone know exactly what causes a sip 482 loop detected error in Asterisk chan_sip? |
| 02:03.20 | qloogkm | I'm getting the error on a phone that seems to be sending a valid invite for the second time with the proper nonce, but the response is 482 Loop Detected, and any further call processing stops there |
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| 02:08.49 | [TK]D-Fender | If it detects a loop. |
| 02:08.53 | [TK]D-Fender | Show us an actual call |
| 02:33.55 | snadge | by RTP i meant audio.. IAX does everything over port 4569 |
| 02:34.05 | snadge | anyway.. i ended up using iax2 set debug on.. and just sucking it up |
| 02:34.12 | snadge | instead of packet tracing |
| 02:35.19 | snadge | one of the worst decisions this place ever made, was to support IAX in the beginning.. back when SIP traversal of NAT was more painful than it is now |
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| 02:35.33 | snadge | so of course you have people with pbxs from 8 years ago.. who refuse to update etc |
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| 02:48.40 | pancho_jay | Why Asterisk is trying to save regseconds as character?????? |
| 02:48.50 | pancho_jay | doesn't understand |
| 02:54.50 | Samot | If you don't like IAX just use SIP. |
| 03:01.06 | pancho_jay | <PROTECTED> |
| 03:02.01 | pancho_jay | but Asterisk shows a lots of warnings: |
| 03:02.03 | pancho_jay | WARNING[7621]: res_odbc.c:645 ast_odbc_prepare_and_execute: SQL Execute returned an error -1: HY000: ERROR: column "regseconds" is of type bigint but expression is of type character at character 149; |
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| 03:02.43 | pancho_jay | But regseconds field is supposed to be a big integer |
| 03:03.55 | pancho_jay | Postgres log: ERROR: column "regseconds" is of type bigint but expression is of type character at character 149 |
| 03:05.14 | pancho_jay | STATEMENT: PREPARE "_PLAN0x7f747c1d0ae0"(char, char, char, char, char, char, char, char, char) as UPDATE billing_sipfriends SET ipaddr=$1, port=$2, regseconds=$3, defaultuser=$4, useragent=$5, lastms=$6, fullcontact=$7, regserver=$8 WHERE name=$9;EXECUTE "_PLAN0x7f747c1d0ae0"(<private info>) |
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| 11:53.59 | julius | WIMPy, you can add sip phones |
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| 12:23.10 | julius | ok, looks like the easybox does know sip. but redirecting numbers to a sip account seems impossible |
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| 13:17.22 | WIMPy | So what can you use SIP phones for then? |
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| 13:43.04 | angryuser | Hello, i am trying to to use CUT on this FROM field "TEST TEST" <sip:+35678412348@123.123.208.83>;tag=7ueiin6fq5u2bjge.o" i need to cut only the ;tag=7ueiin6fq5u2bjge.o however when i put ; as a delimiter to CUT, it does not work |
| 13:43.29 | avb | hey guys, any luck to make session-timers via UPDATE to work? |
| 13:43.50 | angryuser | found it in a minute i wrote, the ; in CUT need to be escaped |
| 13:44.00 | angryuser | so the solution is Noop(${CUT(SIP_HEADER(From),\;,1)}) |
| 13:44.07 | angryuser | instead of Noop(${CUT(SIP_HEADER(From),;,1)}) |
| 13:44.30 | avb | im connecting to a weird ims that requires answer as a reinvite without an SDP or an UPDATE |
| 13:44.49 | avb | angryuser: asterisk sometimes makes your life interesting :) |
| 13:44.54 | angryuser | avb: what do you mean Answer ? |
| 13:45.30 | avb | angryuser: a keepalive message which should be sent after certain period |
| 13:45.37 | avb | to confirm that call is still alive |
| 13:45.41 | angryuser | i think i had a doc on asterisk session timers somewhere |
| 13:45.49 | avb | a default one? |
| 13:46.08 | avb | ive googled all over and checked the source |
| 13:46.13 | angryuser | pdf ish, i've got from somewhere |
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| 13:46.29 | avb | oh, if you can upload somewhere that will be nice |
| 13:46.54 | angryuser | avb: that's the one https://issues.asterisk.org/jira/secure/attachment/28201/AsteriskSipSessionTimers.pdf |
| 13:48.12 | angryuser | i must admit that its not the most exiting part of sip and * |
| 13:48.35 | angryuser | if you are forced to search that, generally you are allready in a deep hole |
| 13:48.46 | WIMPy | Is there anything that is not exciting? |
| 13:55.36 | avb | angryuser: yeh, ive seen that. |
| 13:56.10 | avb | WIMPy: :) well, the most interesting part that some gateways requires an answer to come as a reinvite *without* SDP |
| 13:56.46 | avb | and as appeared SIP spec do not count that as required |
| 13:56.57 | avb | so its ok for asterisk not to do so |
| 13:57.12 | avb | but not for my trunk provider :) |
| 13:57.21 | avb | isnt thats not exciting :) |
| 13:58.13 | WIMPy | Well, most things SIP aren't really specified. That's why it's so crappy. |
| 13:58.13 | avb | and second exciting thing that providers is refusing my UPDATEs because asterisk doesnt state that it support UPDATEs in Supports header |
| 13:58.26 | avb | i hear ya |
| 13:58.37 | WIMPy | Or there are many different specifications to choose from. |
| 13:59.30 | WIMPy | Last month I heard the first one proclaim SIP as dead. |
| 14:06.07 | avb | well, if telepresence systems decided to adopt h323 that says a lot |
| 14:06.18 | avb | which is old and creepy |
| 14:06.27 | avb | but 'it works' :) |
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| 14:10.32 | WIMPy | It's old, but a lot less creepy than SIP. |
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| 14:43.05 | saint_ | I can t wait for the asterisk definitive guide to be updated for release 13 with pjsip |
| 14:45.32 | WIMPy | Did I miss any plans to revive the book? |
| 14:45.41 | file | you may be waiting quite... awhile... |
| 14:46.01 | Samot | for the book or pjsip? |
| 14:46.49 | file | book. |
| 14:46.55 | Samot | :) |
| 14:48.17 | saint_ | i am pretty sure it s going to be a while. it s just a pain to switch from chansip to pjsip since asterisk is not something i do everyday ... |
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| 14:55.09 | [TK]D-Fender | PJSIP is still missing a few things and is a bitt of a pain. |
| 14:55.59 | [TK]D-Fender | almost good enough to replace chan_sip, and then beyond. |
| 14:56.06 | saint_ | [TK]D-Fender pure pain, for someone not used to it :/ I'm thinking about simply switch back to chansip and at least have my release 13 up and running ... |
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| 15:02.45 | mub | I've been using pjsip and chan_sip interchangeably for months now. Works just fine here |
| 15:02.48 | mub | and I'm an idiot |
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| 15:04.36 | saint_ | mub which one do you like the most ? |
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| 15:05.51 | mub | pjsip. Simply because I'm more familiar with the pjsip command |
| 15:05.57 | saint_ | ok |
| 15:06.00 | mub | pjsip show endpoints is prettier |
| 15:07.43 | [TK]D-Fender | And is missing ACLS & extternaddr equivalent |
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