[asterisk-users] "Retransmission Timeout" results in dropped calls after 32 seconds

[Andrew Martin]

----- Original Message -----
> From: "Joshua Colp" <jcolp at digium.com>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at 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 at 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 at 10.10.32.96:5062 SIP/2.0
> > INVITE sip:146 at 10.10.32.96:5062 SIP/2.0
> > SIP/2.0 200 OK
> > ACK sip:146 at 10.10.32.96:5062 SIP/2.0
> > INVITE sip:146 at 10.10.32.96:5062 SIP/2.0
> > INVITE sip:146 at 10.10.32.96:5062 SIP/2.0
> > INVITE sip:146 at 10.10.32.96:5062 SIP/2.0
> > INVITE sip:146 at 10.10.32.96:5062 SIP/2.0
> > INVITE sip:146 at 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