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

[Joshua Colp]

Andrew Martin wrote:

<snip>

>>
> 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?

I don't know if chan_sip will allow this, but if it does... it'll keep 
transmitting over and over... it would be better to get to the bottom of 
the problem. Do a packet capture on the machine running Asterisk and see 
where the packet goes. That's the only thing left really. It's also 
possible something got fixed in relation to directmedia between your 
version and latest 11.

-- 
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org