Hello,
can you post the entire call flow, from initial invite to to the bye.
There is some mistake done somewhere in the routing elements. The sip
trace will help to identify where.
Cheers,
Daniel
On 02/18/2010 12:48 AM, Vikram Ragukumar wrote:
Hello,
I also want to add that, this behavior is not seen while using Asterisk
instead of VOIPSwitch. While using Asterisk, the BYE message has a request
URI corresponding to the cellphone, so the proxy forwards the BYE message
to the cellphone which causes the cellphone to hang up.
Thanks in advance for your help,
Vikram.
Hello,
My setup includes a cellphone that uses Wi-Fi to hookup to the internet
with IP 192.168.1.101. Cellphone app, tries to register with VOIPSwitch
through an outbound proxy server(Kamailio + rtpproxy), which rewrites SDP
to force rtp through an rtpproxy.
I can make a call from the cellphone to a PSTN number. When the PSTN
endpoint hangs up, VOIPSwitch relays a BYE message to the proxy that looks
like :
No. Time Source Destination Protocol
Info
27 18.383929 IPaddr_VoipSwitch IPaddr_proxy SIP
Request: BYE sip:1...@ipaddr_voipswitch
Frame 27 (420 bytes on wire, 420 bytes captured)
Ethernet II, Src: Unispher_40:b5:39 (00:90:1a:40:b5:39), Dst:
Supermic_bd:b9:bc (00:30:48:bd:b9:bc)
Internet Protocol, Src: IPaddr_VoipSwitch , Dst: IPaddr_proxy
User Datagram Protocol, Src Port: sip (5060), Dst Port: 7160 (7160)
Session Initiation Protocol
Request-Line: BYE sip:1...@ipaddr_voipswitch SIP/2.0
Method: BYE
[Resent Packet: False]
Message Header
Route:<sip:IPaddr_proxy:7160;lr=on;nat=yes>
CSeq: 1 BYE
Via: SIP/2.0/UDP IPaddr_VoipSwitch:5060;
branch=z9hG4bk180242100434182912333954
From: sip:5...@ipaddr_voipswitch;tag=18024210041929123187506289
Call-ID: LUOyIDm-ecy36NcUfj5G3v1lM6g5snkM
To: "1234"
<sip:1...@ipaddr_voipswitch>;tag=9XHJmiuf58vTrF7.vwflEK-G63JVPQSi
Content-Length: 0
The request URI in the above message is the IP address of VOIPSwitch
itself, which causes the proxy to forward the BYE message back to
VOIPSwitch. The cellphone never receives the BYE message and consequently
never hangs up.
Any suggestions on how i can get around this problem ?
Thanks and Regards,
Vikram.
_______________________________________________
Kamailio (OpenSER) - Users mailing list
Users@lists.kamailio.org
http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
--
Daniel-Constantin Mierla
SIP Server Professional Solutions
* http://www.asipto.com/
_______________________________________________
Kamailio (OpenSER) - Users mailing list
Users@lists.kamailio.org
http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
