I have never seen a SIP aware firewall work with localnet and externip/externhost. You should try either disabling the SIP fixup on your firewall or remove the localnet/externip from sip.conf.
Carlos Chavez wrote: > I am a bit desperate trying to solve this problem. Sorry if I am > abusing the list a bit with the same king of question. > > The problem I am having is very specific which is why it is very > difficult to diagnose and fix. Basically an Asterisk server is > connected via E1 PRI to an Avaya PBX. The Asterisk server has 45 PAP2T > and 45 SPA-3102 devices connected via the Internet. The Asterisk server > is behind a Fortinet firewall and has all necessary ports redirected to > it. > > By itself, everything is working. I can make and receive calls to all > SIP devices, check voicemail and any other service I configure on the > Asterisk server. I have the relevant parts of NAT configured like > "externip", localnet, nat=yes and canreinvite=no. The problem only > presents itself when a SIP device is trying to call an extension > connected to the Avaya. Since "localnet=192.168.2.0/255.255.255.0" is > defined and the Fortinet firewall rewrites the source IP as its own > "192.168.2.1", I think this may be the cause of my problems but why only > when calling the Avaya and not other SIP extensions or Asterisk > services? > > Since the SPA3102 has Symmetric RTP it works fine. The PAP2T on the > other hand gives one way audio when you call any extension on the Avaya. > The only way I can get the PAP2T to work is to change the localnet to > something else then it works properly but the SPA does not. Any call I > make from the SPA hangs up after a minute or so and any call I make > rings the SPA but I do not get any audio. > > What is the proper NAT setup for something like this? Is it even > possible to work with this type of NAT? Any comment would be truly > appreciated. > > > > ------------------------------------------------------------------------ > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users