----- 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 12:32:06 PM > Subject: Re: [asterisk-users] "Retransmission Timeout" results in dropped > calls after 32 seconds > > Andrew Martin wrote: > > ----- Original Message ----- > > <snip> > > > > > By doing a number of test calls today, I have managed to reproduce this > > while > > sip debugging was on, so I have that information available now as well: > > http://pastebin.com/ZJqzdvY3 > > > > This was a call from 113 to 146 via a queue. Note that the asterisk server > > is > > at 10.10.32.251. I see the following: > > INVITE sip:146@10.10.32.96:5062 SIP/2.0 > > SIP/2.0 180 Ringing > > SIP/2.0 180 Ringing > > SIP/2.0 200 OK > > ACK sip:146@10.10.32.96:5062 SIP/2.0 > > INVITE sip:146@10.10.32.96:5062 SIP/2.0 > > SIP/2.0 200 OK > > ACK sip:146@10.10.32.96:5062 SIP/2.0 > > INVITE sip:146@10.10.32.96:5062 SIP/2.0 > > INVITE sip:146@10.10.32.96:5062 SIP/2.0 > > INVITE sip:146@10.10.32.96:5062 SIP/2.0 > > INVITE sip:146@10.10.32.96:5062 SIP/2.0 > > INVITE sip:146@10.10.32.96:5062 SIP/2.0 > > > > This appears to start out with a successful SIP conversation (ending with > > the > > first ACK), so it is unclear to me why we have two new sets of INVITEs sent > > afterwards. > > Asterisk has sent a re-INVITE to have the media flow directly. The > device (seems) to respond with the 200 OK (you can tell based on the > CSeq) for the initial INVITE, and not for the re-INVITE. As Asterisk > gets no response to its re-INVITE it gives up and terminates the dialog. >
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? 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
