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