[OpenSIPS-Users] Call sequence in serial forking
John Nash
john.nash778 at gmail.com
Mon Jan 12 11:09:01 CET 2015
I am testing one setup where opensips drouting module sends call to
"Freeswitch" and I encountered one situation ...
UA sends Invite to opensips, opensips uses drouting module and sends Invite
to Freeswitch , callee rejects the call and opensips sends ACK to
freeswitch and sends second invite (from failure route). This second invite
(which has same call id but different branch in via) is not treated as
another transaction by freeswitch and it sends back SIP 482 Request merged
response.
I had the same setup tested using SEMS as SBC some times back successfully.
I am not sure which side this issue should be taken care of (opensips or
freeswitch)
I looked in some freeswitch mail archives and in one post I can see someone
suggesting that from opensips side we should increase Cseq in case of
second invite. I think this can be done using script but I am not sure if i
should do or not.
This is the post
http://lists.freeswitch.org/pipermail/freeswitch-users/2013-February/092600.html
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