I notice that since the SIP update Asterisk seems to crash every
few hours. I haven't managed to get a core dump yet but I will post
the backtrace as soon as I can get one...
In case I need to back down to a known good version, does anyone know
what the version of channels/chan_sip.c before the
I've also noticed that my SIP phones (snom [12]00) seem to deregister
themselves after some time, and not be able to re-register until
Asterisk is restarted. This problem only manifested with the latest
CVS.
-- Luke
--
Luke Howard | PADL Software Pty Ltd | www.padl.com
Turn off reinvites and that will likely fix this.
Notice how the first invite is totally ignored, and then for some reason
the second gives us the 481.
Mark
On Sat, 29 Mar 2003, Luke Howard wrote:
This seems to fix incoming calls but outgoing calls terminate
immediately, at least for me,
Turn off reinvites and that will likely fix this.
Thank you; it does.
-- Luke
--
Luke Howard | PADL Software Pty Ltd | www.padl.com
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This Helps :)
Index: chan_sip.c
===
RCS file: /usr/cvsroot/asterisk/channels/chan_sip.c,v
retrieving revision 1.16
diff -c -r1.16 chan_sip.c
*** chan_sip.c 29 Mar 2003 00:42:16 - 1.16
--- chan_sip.c 29 Mar 2003 02:20:46 -
This seems to fix incoming calls but outgoing calls terminate
immediately, at least for me, with a
481 Call Leg/Transaction Does Not Exist
from the SIP phone. Here's the SIP debug output (NB: IP addresses
have been changed). It *used* to work.
-- Luke
-- Attempting native bridge of
My Patch fixes only the 15Sec Channeldestroy bug, for incomming/outgoing
calls but your error is an other.
Andre
- Original Message -
From: Luke Howard [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, March 29, 2003 3:59 AM
Subject: Re: [Asterisk-Users] SIP Retransmission Patch
