----- Original Message ----- > From: "Andrew Martin" <[email protected]> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <[email protected]> > Sent: Monday, May 11, 2015 4:18:58 PM > Subject: Re: [asterisk-users] "Retransmission Timeout" results in dropped > calls after 32 seconds > > ----- Original Message ----- > > From: "Andrew Martin" <[email protected]> > > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > <[email protected]> > > 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. > Joshua,
As a mitigation for this problem, could I increase the "timerb" option in sip.conf to a large value, say 1 hour (instead of the default 32 seconds)? What other consequences would there be from this change? 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
