Hello, The guy operating the Asterisk at the far end have been inspecting the traces I've sent him. Hen notes that the ACK relayed doesn't entirely comply to the standard. The values of "branch" in the VIA header of the relayed ACK response incorrectly have the value "0" (also rport in the VIA header added by their Asterisk is modified). He suggest setting the syn_branch parameter to get the "branch" right. I've found this thread about it:
http://blog.gmane.org/gmane.comp.voip.ser/month=20090701 I have put syn_branch=0 in the config and "branch" now have a valid value. I don't quite buy it - that the incorrect value of "branch" should be the culprit. But I'll activate tcpdump again and wait to see if the dropped calls has vanished. /Ole Den 01/07/2010 kl. 09.12 skrev Klaus Darilion: > Hi! > > Kamailio behaves correct in this trace and I couldn't spot an abvious error > in the trace. > > Nevertheless there are 2 problems: > > 1. Asterisk Gateway does not receive/accept the ACK > 2. Asterisk PBX does not retransmit ACK > > reg. 1: ask the gateway operator if he sees the ACK in the Asterisk log, and > if yes the error message of Asterisk why this ACK is not accepted. If the ACK > is not seen in the Asterisk console then maybe it gets dropped by a buggy > firewall. > > reg 2: login to your Asterisk PBX and verify if Asterisk receives the 200 OK, > and maybe spot some log message why it does not trigger ACK retransmissions. > > regards > Klaus > > Am 30.06.2010 23:33, schrieb Ole Kaas: >> >> Den 30/06/2010 kl. 01.23 skrev Iñaki Baz Castillo: >> >>> 2010/6/29 Ole Kaas<o...@tet.dk>: >>>> Hi Klaus, >>>> >>>> I've mailed pcap dump to you directly for further inspection. >>> >>> Hi, it's much better if you capture a trace with "ngrep -Wbyline -t -q >>> port 5060" and paste it in a new mail by replacing your public IP's >>> with other values. Then all the people here could help you rather than >>> asking for private help to a specific member of the maillist. >>> >> >> You are right. But maybe it was something (obvious) that could be resolved >> quickly and I could post an update here on the list. The original log was >> inadequate - Klaus was a great help, with suggestions to obtain new log. So >> here it is attached and anonymized with all ip addresses (and stuff) >> converted to private adresses. The Kamailio server is multi homed and the >> two networks are non-routable (I use rtpproxy to bridge media). Our Asterisk >> PBX is version 1.4.26.1 and the Asterisk Gateway is 1.6.1 (or 1.6.0 - cant >> remember and not under my control). Kamailio is version 3.0.0. >> >> Looking at the trace, it seems the problem starts with the ACK not being >> received by the Asterisk Gateway which then resends the OK. The OK is >> relayed back to the originating Asterisk PBX which seems to ignore the >> retransmission. In fact it seems that Kamailio is routing and relaying the >> sip packets correctly. However, it seems that the problem only exists >> between Asterisk and Kamailio. I have other pbx'es (3CX) connecting to >> Kamailio and I have no evidence that the problem happens with those. I have >> another trace where the call comes from one of the Asterisk Gateways and is >> routed back to one of the other Asterisk Gateways. The result is the same - >> the OK's are ignored by Asterisk. >> >> /Ole >> _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
