Hello Olivier,
On 04.11.2009 12:25 Uhr, olivier.tay...@gmail.com wrote:
Hi Daniel, long time, I am on Kamailio(1.5) now :)
not for long time :-) , 3.0 is around the corner...
ok, first of all, sorry I forgot the record-route :((
Now I have it, but the proxy doenst do anything when the Bye arrives
from the callee.
What action do I have to take if any?
You have to do loose_route() and if true then relay. The default config
file has sample routing of in-dialog requests (when to tag is set).
Cheers,
Daniel
What I receive for the BYE:
BYE sip:997321...@xxx.xxx.67.183:5060 SIP/2.0.
Via: SIP/2.0/UDP yyy.yyy.123.88:5060;branch=z9hG4bK4dd17330;rport.
Route:
<sip:0485336...@yyy.yyy.123.83:5060;nat=yes;ftag=5672;lr=on>,<sip:xxx.xxx.67.183;lr=on;ftag=5672;my_param=AAAAAAAAAAAAAAAACgBUWFxPXVpFWhhVSzE4Mw-->.
From: <sip:0485336...@xxx.xxx.67.183>;tag=as3e2ef419.
To: "997321079" <sip:997321...@my.be>;tag=5672.
Contact: <sip:0485336...@xxx.xxx.123.88>.
Call-ID: 1257324791-1991-macbook%20de%20olivier%20tay...@192.168.2.125.
CSeq: 102 BYE.
User-Agent: Phonext B2Bua.
Max-Forwards: 70.
Content-Length: 0.
Daniel-Constantin Mierla a écrit :
Hello,
On 04.11.2009 11:58 Uhr, olivier.tay...@gmail.com wrote:
Hello,
I use uac_replace_from to setup a call and everythings are ok.
When the caller send a Bye, no problem.
When the caller send a CANCEL, no problem.
But when callee send a BYE to stop the call after the call has been
setup, the bye is never sent back to caller...
I also don't see any record-route header...
Any idea?
What did I forget?
if you do record routing and the bye from callee does not have it,
then seems to be a broken UA at the callee side.
Have you tried with different UAs?
Cheers,
Daniel
--
Daniel-Constantin Mierla
* http://www.asipto.com/
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