Hi Brian. Basically it becomes handy if you can change contact to be a SIP proxy in front of FS. I've a scenario with kamailio being the SIP proxy in front of FS both on the same server. Kamailio handles 5060/5061 UDP/TCP/TLS and FS listen on 5070/5080 (internal/external) both FS profiles have outbound_proxy enabled and pointing to Kamailio. In this scenario FS sets contact to: <contact_user>@ext-sip-ip:ext-sip-port if i have ext-sip-ip, ext-sip-port defined in my sip profiles otherwise it will be <contact_user>@sip-ip:sip-port. When SIP passes through kamailio it will be using the Contact set in FS and will present it to the endpoints and some endpoints answer back using the contact and not the original source_ip/source_port. When this happens and my firewall prevents FS from being reached directly from the outside world i start experiencing issues like drop calls etc. With that being said it would be useful if i could tell FS to rewrite the contact with a sip_uri of my own, something simillar to what contact-user/ext-sip-ip/ext-sip-port profile variable do but without making FS bind to that IP:PORT which prevents me from setting the same variables in more than one sip profile. Looking forward to hear from you. Best Regards, -- *Nuno Miguel Reis* | *Unified Communication** Systems* M. +351 913907481 | nreis@wavecom.pt WAVECOM-Soluções Rádio, S.A. Cacia Park | Rua do Progresso, Lote 15 3800-639 AVEIRO | Portugal T. +351 309 700 225 | F. +351 234 919 191 *GPS <http://maps.google.com/maps/ms?msa=0&msid=202333747613191340808.0004b4b227a6144f0df88> | www.wavecom.pt <http://www.wavecom.pt/>** <http://www.wavecom.pt/>* [image: Description: Description: WavecomSignature]<http://www.wavecom.pt/pt/wavecom/premios.php> [image: Publicity] <http://www.wavecom.pt/pt/mail_eventos.php> On Tue, May 20, 2014 at 4:48 PM, Nuno Reis <nreis@wavecom.pt> wrote:

Hi! I've been looking at FS source code and realized that at least the invite_contact can be changed by declaring the contact-user, ext-sip-ip and ext-sip-port profile variables. I would like to know what other option do i have to change contact(all the sip_uri string) in the different stages of a call: REGISTER/INVITE/SESSION_IN_PROGRESS, ACK, OK, etc... I would also like to be able to change contact in runtime dynamically, what option do i have in this case?

Thanks.

--

*Nuno Miguel Reis* | *Unified Communication** Systems* M. +351 913907481 | nreis@wavecom.pt WAVECOM-Soluções Rádio, S.A. Cacia Park | Rua do Progresso, Lote 15 3800-639 AVEIRO | Portugal T. +351 309 700 225 | F. +351 234 919 191 *GPS <http://maps.google.com/maps/ms?msa=0&msid=202333747613191340808.0004b4b227a6144f0df88> | www.wavecom.pt <http://www.wavecom.pt/>** <http://www.wavecom.pt/>*

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