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SIP call doesn't disconnect

Pasha Teplitsky

Hi guys,

i'm doing some pre sale testing for a customer. he wants to use cisco router as a sip proxy. either in CME mode or as a ip2ip gw.

i tried 2 configurations with the clients SIP softphone.

1. CME config while i register the sip softphones.

2. ip2ip gw while i use dial peer voip with session protocol sipv2 to point to the ip add for the extension.

tests were doing great up untill the calee tried to hang up.

when the callee hangs up, a bye msg is sent to the router and the callee leg disconnects but the bye msg never gets to the caller!

and this leg stays up.  i run several debugs including debug ccsip all and saw no errors. it seems like the router is fine with everything.

i never have a problem while the caller hangs up. the session ends just fine on both call legs.

this happens both in CME and ip2ip configurations.

i've tried the same with another softphone and everything was working great!

the only difference i saw was this:

on the other softphone i used, the BYE message that was generated by the callee had a CSEQ value greater than the INVITE (generated by the caller).

on the clients softphone the CSEQ of the BYE message (from the callee) was initial. ie- 1.

i've done some research on the issue and everything works out just fine by IETF rules.

since the INVITE came from the caller with CSEQ 1, if the caller would send the BYE it should have had CSEQ 2, but since

its the callee that sends the BYE and this is the first initiation from its behalf, CSEQ indeed should be 1.

any thoughts about this?

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