[OpenSIPS-Users] ACK bug?
Bogdan-Andrei Iancu
bogdan at voice-system.ro
Tue Jun 16 19:05:14 CEST 2009
Hi Charles,
It seams the ACK is looping on opensips server... and this because when
doing "loose_route", the opensips is doing strict route instead of
loose_route..:
Jun 15 14:09:55 server231 /usr/sbin/opensips[2331]:
DBG:rr:after_strict: Next hop: 'sip:X.X.X.231;lr' is loose router
I guess you have a misconfiguration in your script as the X.X.X.228:5059
is recognized as a local address (to opensips). Please check if you
haven't set this IP as alias or as domain in "domains" table....
Regards,
Bogdan
Charles Solar wrote:
> Thanks for the replys, I uploaded my config here:
> http://pastebin.com/m75846e3d
> Its a pretty standard config for the most part, I picked some parts
> out of a book, some parts off this list, some parts on other forums.
> This ack thing is just stretching me thin here.
>
> I did throw some log messages in there to help debug in syslog, here
> is the log stuff I was trying to look through yesterday.
> http://pastebin.com/m75e67226
>
> I just noticed while pasting that log, on line 44 there I think shows
> the real problem. The server keeps sending the ack to itself over and
> over.
>
> Oh, here are some complete sip messages too: http://pastebin.com/m21d11a25
> I marked on that paste which message correspond to messages in my
> original picture
>
> Thank you
>
> On Mon, Jun 15, 2009 at 8:23 PM, Alex Balashov
> <abalashov at evaristesys.com <mailto:abalashov at evaristesys.com>> wrote:
>
> Can you paste your OpenSIPS config? It may be that the ACK is not
> being properly routed in all circumstances.
>
> Charles Solar wrote:
>
> I am experiencing an ack bug in opensips I believe.
> I have a caller register to a server, call it 231, and I have
> 231 send invites to 228 which processes the route and does lcr.
> 228 sends calls to the best gateway, which in my tests is just
> one asterisk server (also on 228, port 5059).
>
> I have 231 and asterisk record their route, 228 does not show
> up in the route header.
>
> The problem comes in when asterisk sets up a call it tries to
> bridge the caller and callee with reinvites. I see the 200 OK
> message and my caller sends a ACK back, but opensips does not
> forward the ACK properly.
>
> This is a wireshack graph of the conversation from 231's
> perspective
> http://img197.imageshack.us/img197/7889/sshot2mfv.png
>
> I have tried shifting through the debug messages in syslog but
> all I can tell is that 231 is trying to forward the ACK to
> itself.
> Has anyone else experienced this problem or know whats going on?
>
> Thank you for your time
>
>
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> --
> Alex Balashov
> Evariste Systems
> Web : http://www.evaristesys.com/
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> Direct : (+1) (678) 954-0671
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