7 Feb
2007
7 Feb
'07
2:08 p.m.
Hi Michel,
Thank you for your advice.
And finally I successed to get BYE message from Cisco AS 5300! :-)
Cisco setting as follows...
MYAS5300#show run
Building configuration...
(omitted)
!
voice call disc-pi-off
voice call carrier capacity active
voice rtp send-recv
!
(omitted)
Here is an grobal configuration for 5300 voice service.
And I have no change for my ser.cfg.
(I appreciate for your advice, thanks Michel)
So my SER and AS 5300 call scenario makes BYE each other
when call is done (hooked on).
Very thanks.
Sahria
2007/2/7, Miklos Tirpak <miklos(a)iptel.org>rg>:
You may see some hint in the log of Cisco AS if you
enable more detailed
debug, for example:
debug sip all
debug voip ccapi inout
Just be prepared for the huge about of log and system slowdown!
Miklos
Michal Matyska wrote:
--
----------
shosuke
msn : anseie(a)hotmail.co.jp
email : sahria.hao(a)gmail.com
Hi,
rport missing is no problem at all, default 5060 is then used.
The branch=0 in the ACK might be the problem (if Cisco does not match it
with the INVITE (but then it should retransmit the 200 OK reply)).
You can try add syn_branch=0 to the ser.cfg to have the branch id
calculated for ACKs too.
Michal
On Wed, 2007-02-07 at 13:02 +0900, Sahria Hao wrote:
I got an additional SIP message log from my
5300.
Scenario:
[UA] => [Asterisk] => [SER] => [CISCO AS5300] => [PSTN]
[UA] "12"
[Asterisk] 0355558888(a)MY.ASTERISK.IP.ADDRESS
[SER] MY.SER.IP.ADDRESS
[CISCO AS5300] MY.AS5300.IP.ADDRESS
[PSTN] T1 Line
And maybe I found little bad Via header in
ACK message from Asterisk (via SER) to AS5300.
Feb 7 03:22:34.401: Received:
ACK sip:056708077771111@MY.AS5300.IP.ADDRESS:5060 SIP/2.0
Record-Route: <sip:MY.SER.IP.ADDRESS;ftag=as2ad98fe4;lr>
Via: SIP/2.0/UDP MY.SER.IP.ADDRESS;branch=0
Via: SIP/2.0/UDP
MY.ASTERISK.IP.ADDRESS:5060;branch=z9hG4bK0a0daaf9;rport=5060
...
Why Via header looks like no branch value and port value...?
And all SIP messages have no Route header.
I attached AS5300 SIP message log (see below).
Any ideas?
Sahria
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