IRC log for #asterisk-bugs on 20081228

01:18.29*** join/#asterisk-bugs adker_ (n=chatzill@74-33-209-209.dr02.glvv.ny.frontiernet.net)
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12:38.34*** join/#asterisk-bugs asterisk_user2 (n=jim@195-240-249-117.ip.telfort.nl)
12:38.38asterisk_user2hello
12:38.52asterisk_user2is there a developer online ?
12:40.46asterisk_user2i want to report a bug when letting asterisk connect to SER
12:40.57mvanbaakgood morning :)
12:41.13asterisk_user2good morning :)
12:41.17mvanbaakwe collect bugreports on http://bugs.digium.com
12:41.53asterisk_user2mvanbaak kom je toevallig uit nederland ? :)
12:42.04mvanbaakyup
12:42.19asterisk_user2ik weet niet 100% zeker of het een asterisk bug
12:42.40asterisk_user2misschien ik het even uitleggen voordat ik dit instuur als asterisk bug
12:42.41mvanbaakhhmm, maybe you can explain a bit what's going wrong
12:42.45mvanbaakyup :)
12:43.13asterisk_user2probleem is dat bij nieuwe versies van asterisk na versie 1.2.14
12:43.48asterisk_user2de verbinding bij inkomende oproepen zomaar wordt verbroken
12:43.53asterisk_user2na ongeveer 30 sec.
12:44.06asterisk_user2dit probleem doet zicht niet altijd voor
12:44.14asterisk_user2maar bij 2 van de 10 inkomende oproepen
12:44.19asterisk_user2in het logboek staat dan
12:44.33asterisk_user2maximum retries exceeded on transmission
12:44.41asterisk_user2en bij nieuwe versies staat daaronder :
12:44.44asterisk_user2hanging up call ...
12:45.02asterisk_user2daarom zitten bij "vast" aan de oude versie
12:45.21asterisk_user2misschien ligt dit probleem wel bij de voip provider
12:45.22asterisk_user2SER
12:46.00mvanbaakdid you try a sip debug ?
12:46.05asterisk_user2yes
12:46.14mvanbaakanything usefull in there ?
12:46.28asterisk_user2no
12:46.52mvanbaakcan you put it online somewhere ?
12:46.57asterisk_user2yes
12:47.17asterisk_user2if found a site where users had the same problem
12:47.20asterisk_user2as i have
12:47.30asterisk_user2one moment please
12:47.36asterisk_user2i'll give you the link
12:47.40mvanbaaksure thing. it's sunday, no need to rush
12:47.41mvanbaak:)
12:47.56asterisk_user2yes i know :)
12:48.53asterisk_user2---------------
12:49.24asterisk_user2I've encountered a similar problem with Cisco equipment.  The Cisco proxy
12:49.24asterisk_user2was not replying to Asterisk with an ACK after * sent an OK.
12:49.24asterisk_user2Since version 1.2.14, * was changed so that not receiving an ACK to an OK is
12:49.24asterisk_user2considered a FATAL error.
12:49.24asterisk_user2The specific change that causes this problem is in sip_answer() in
12:49.25asterisk_user2chan_sip.c:
12:49.27asterisk_user2res = transmit_response_with_sdp(p, "200 OK", &p->initreq, 2);
12:49.29asterisk_user2Changing the 2 to a 1 will probably fix it.  Note that this is NOT a bug in
12:49.31asterisk_user2* but improper implementations--either caused by latency, or a software bug
12:49.33asterisk_user2(not sending an ACK).  Perhaps it might be beneficial to have an option in
12:49.35asterisk_user2sip.conf to change how * handles not receiving an ACK?  I know... it's
12:49.37asterisk_user2someone else's problem, but might help those of us stuck with buggy
12:49.39asterisk_user2implementations in production environments. :)
12:49.41asterisk_user2Brian.
12:49.43asterisk_user2LINK:
12:49.45asterisk_user2http://lists.digium.com/pipermail/asterisk-users/2007-April/184875.html
12:50.57asterisk_user2-
12:50.58asterisk_user2-
12:50.59asterisk_user2-----
12:53.02*** join/#asterisk-bugs asterisk_user2 (n=jim@195-240-249-117.ip.telfort.nl)
12:53.10asterisk_user2what happend ??
12:53.17asterisk_user2i disconnected ???
12:53.27mvanbaakyup
12:53.32mvanbaakyou pasted too many lines
12:53.37asterisk_user2ok :)
12:53.39mvanbaak~pb
12:53.40infobot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding 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://www.rafb.net/paste, or , http://bin.cakephp.org/
12:53.54asterisk_user2http://lists.digium.com/pipermail/asterisk-users/2007-April/184875.html
12:54.32asterisk_user2what do you think ?
12:54.40asterisk_user2is it a asterisk problem ?
12:54.49mvanbaaknope
12:54.54mvanbaakit's a problem on the other side
12:54.59mvanbaakit should send the ACK
12:55.12asterisk_user2hmmm ok
12:55.19asterisk_user2i can't solve that
12:55.52mvanbaakhave you tried what Brian mentioned in his mail ?
12:55.56asterisk_user2yes
12:55.59mvanbaakpatch chan_sip.c
12:56.00asterisk_user2that worked
12:56.16asterisk_user2but i can't patch the new asterisk version 1.4.x
12:56.17mvanbaakif that works, you can run a newer version of asterisk with that patch applied
12:56.22mvanbaakwhy not ?
12:56.34asterisk_user2i couldn't find the line
12:57.05asterisk_user2res = transmit_response_with_sdp(p, "200 OK", &p->initreq, 2);
12:57.11asterisk_user2i couldnt find the above line
12:58.30asterisk_user2but i does work in all the 1.2 versions of asterisk
12:58.59mvanbaakok, easy
12:59.09mvanbaakso, in the latest 1.4 subversion branch
12:59.24mvanbaakit's line 3726 in channels/chan_sip.c
12:59.32asterisk_user2ok thank you !!!
12:59.53asterisk_user2i'm not a programmer no i could be wrong :)
13:00.05mvanbaakchange the XMIT_CRITICAL into XMIT_UNRELIABLE
13:00.13asterisk_user2ok
13:00.26asterisk_user2i'm going to try it right now :)
13:04.11asterisk_user2i'm looking in channels/chan_sip.c (Version 1.4.22)
13:04.17asterisk_user2could it be line
13:04.31asterisk_user26174 ?
13:05.07mvanbaakres = transmit_response_with_sdp(p, "200 OK", &p->initreq, XMIT_CRITICAL);
13:05.09mvanbaakthat line
13:05.46asterisk_user2ok that's line number
13:05.52asterisk_user23706
13:05.58asterisk_user2thank you
13:06.15mvanbaakwelcome :)
13:08.50asterisk_user2maybe it's a good idea to make a switch option in sip.conf
13:08.55asterisk_user2for people like me
13:09.00asterisk_user2:)
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