[asterisk-dev] SIP INVITE retransmission, erroneous

[Nikolaos Markolefas]

Hi * world,

I have an issue where an asterisk server communicates over SIP with 
another SIP server (not asterisk). An audio call is established 
successfully; when * side issues a reINVITE to the remote side, the 
remote server takes some time to respond. Configuration for "qualify" is 
"yes", so after 2seconds, * issues a second reINVITE. Problem is that 
this latter reINVITE is not identical to the reINVITE previously sent. 
Second reINVITE's CSeq header value is incremented by 1, as well as the 
SDP session version, which leads the remote server to the assumption 
that the second reINVITE triggers as a new transaction, while the first 
transaction (initiated by the first reINVITE) has not closed. Thus, 
remote side releases the call.

I tried to crosscheck with the bugs archive, though did not find 
anything like this. I'm running * v1.2.4 on gentoo.
As a newcomer to the * community, it could be that I missed something, 
so has anybody come across such a case? If yes, could you point me to 
the relevant bug, or should I open a new one?

Best regards.