Hi Klaus, I've mailed pcap dump to you directly for further inspection.

Den 30/06/2010 kl. 01.23 skrev Iñaki Baz Castillo: 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

Hi, I have a similar issue,

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<obk(a)tet.dk>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 >

> > Hi, I have a similar issue, > > It is not possible to debug this issue without full SIP trace! > > ngrep -Wbyline -t -d any port 5060

Ok, here it goes again. Pay attention to the "200: OK" response from the PSTN-GW. I think its Contact header should be read by the Cisco GW (calling party), and then send its ACK-request with R-URI based on that Contact. But it doesn't happen. Then here i think this is the failure. (rfc3261, section 12.1.1 and 12.2.1.2). May i am wrong.. What do you think? I'd appreciate your comments.