----- Original Message ----- > From: "Andrew Martin" <amar...@xes-inc.com> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users@lists.digium.com> > Sent: Monday, May 11, 2015 1:35:07 PM > Subject: Re: [asterisk-users] "Retransmission Timeout" results in dropped > calls after 32 seconds > > > That should be all that is required. If that were broken I'd expect > > issue reports to implode - what's the configuration? > > > > Here's the sip.conf (only showing a single extension since they're all the > same): > [general] > directmedia=no > directrtpsetup=no > dtmfmode=rfc2833 > context=asterisk-internal > allowsubscribe=no > qualify=no > disallow=all > allow=ulaw > allow=alaw > allow=gsm > localnet=10.10.32.0/255.255.248.0 > localnet=192.168.32.0/255.255.255.0 > > [146] > secret= > host=dynamic > type=friend > > From the aforementioned sip debug capture, 146 is on the 10.10.32.0/21 > network > and 113 is on the 192.168.32.0/24 network (these are directly route-able so > no > NAT is involved). However, I have now been able to reproduce the problem > between > two devices directly on the 10.10.32.0/21 network as well. >
I've gathered the log for this dialog from the SIP phone: http://pastebin.com/aAWs4j6i What I see is that there's an INVITE for CSeq 103, then an OK for CSeq 103, then another INVITE is received for CSeq 103, at which point the phone reports an error: <0> | ERROR | receive a request with same cseq?? >From the asterisk side, it never seems to receive this OK for CSeq 103, hence the reason it sends out the INVITE again. Thanks, Andrew -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
