[asterisk-users] BYE message not relayed to caller

Vikram Ragukumar vragukumar at signalogic.com
Wed Feb 24 15:56:57 CST 2010


Hello,

I have a setup that includes a cellphone a proxy running Kamailio and
rtpproxy and a SIP server (VoipSwitch/Asterisk). Call flow works well
while using Asterisk, however when VoipSwitch is used i find that the BYE
message from VoipSwitch has an RURI = account at VoipSwitch, so the proxy
ends up repeatedly sending BYE messages to VoipSwitch instead of sending
them to the Cellphone, causing the Cellphone to never hangup. However when
using Asterisk the BYE message is forwarded to the cellphone and both
endpoints of the call hangup. I show below the SIP message flow while
using VoipSwitch.

     Cell Phone     Kamailio       VoipSwitch
          |              |              |
          |INVITE        |              |
          |------------->|              |
          |100 Trying    |              |
          |<-------------|              |
          |              |INVITE        |
          |              |------------->|
          |              |100 trying    |
          |              |<-------------|
          |              |183SessionProg|
          |              |<-------------|
          |183SessionProg|              |
          |<-------------|              |
          |              |    200 OK    |
          |    200 OK    |<-------------|
          |<-------------|              |
          |     ACK      |              |
          |------------->|              |
          |              |     ACK      |
          |              |------------->|
          |              |     BYE      |
          |              |<-------------|<- BYE,RURI=account at VoipSwitch
          |              |     BYE      |
          |              |------------->|
          |              |     BYE      |
          |              |------------->|

Is this issue caused by the SIP server or some other element along the SIP
message flow ? Does anybody know the difference in SIP message handling
between VoipSwitch and Asterisk or can anybody point me to an online
resource ?
-- 
Thanks and Regards,
Vikram Ragukumar.



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