[OpenSIPS-Users] Call sequence in serial forking

[Nguyen Dang Vo]

Hi John

Did you try changing the Call-ID of every INVITE?
"Merged Request" should be generated with requests that have same From tag,
Call-ID, and CSeq
https://tools.ietf.org/html/rfc3261#section-8.2.2.2

Regards,

On Mon, Jan 12, 2015 at 2:09 AM, John Nash <john.nash778 at gmail.com> wrote:

> 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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>


-- 
NguyenVD
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