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 [email protected]: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
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