Re: [OpenSIPS-Users] Call sequence in serial forking

2015-01-12 Thread 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  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
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>


-- 
NguyenVD
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users


Re: [OpenSIPS-Users] Call sequence in serial forking

2015-01-12 Thread 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


___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users


___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users


[OpenSIPS-Users] Call sequence in serial forking

2015-01-12 Thread John Nash
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
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users