----- Original Message ----- > From: "Joshua Colp" <jc...@digium.com> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users@lists.digium.com> > Sent: Monday, May 11, 2015 1:24:53 PM > Subject: Re: [asterisk-users] "Retransmission Timeout" results in dropped > calls after 32 seconds > > > Could this perhaps be because the phone doesn't support "bypass" or > > re-INVITEs? > > Is there a way to disable this functionality and instruct asterisk to just > > stay in the middle of the conversation (bridging or native-bridging) for > > the > > duration of the call? I thought that setting directmedia=no and > > directrtpsetup=no would disable re-INVITEs and force asterisk to use > > bridging > > mode, but perhaps something else is required? > > 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. 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
