[asterisk-users] Asterisk codec negotiation and canreinvite=no

[Ira]

Not that it has anything to do with this, but after having tried and 
failed to use many 1.8 betas and most of the 1.8 release versions, 
yesterday I followed the instructions on how to get trunk and my 
problem with 1.8 is fixed.

It involved this error:

WARNING[24384] chan_sip.c: Retransmission timeout reached on 
transmission 6cdb5a2f6cbe781b3a2553745a92dcf0 at 192.168.2.2:5060 for 
seqno 102 (Critical Request) Packet timed out after 20672ms with no response

Which happened when I made an outing call on a DAHDI POTS line back 
to my other POTS line and asterisk tried to ring my internal SIP 
phones and failed with that message.

So, if anyone was tracking that error, it seems to be fixed.

Ira

At 10:38 AM 4/11/2011, you wrote:
>The code you are talking about underwent a complete rewrite [1] and 
>has already been merged into trunk[2].  Not that it helps you now, 
>but you may want to try testing with trunk (will become Asterisk 
>1.10) and see if you have the same issues.
>
>This is one of the major milestones for Asterisk 1.10, and I'm sure 
>any feedback in testing will be much appreciated.
>
>[1] https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal
>[2] http://svn.digium.com/view/asterisk?view=revision&revision=306010
>--
>Paul Belanger