[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 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