[OpenSIPS-Users] Call sequence in serial forking

[Bogdan-Andrei Iancu]

Hi John,

AFAIK, there is no need to increase the cseq during normal serial 
forking. The only known issue is if you do the serial forking for 
authentication purposes (you received a 401/7 and you do serial forking 
with credentials) - is this case you need to increase the cseq.

But once again, the classic serial forking does not require any change 
in cseq.

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 12.01.2015 12:09, John Nash 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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