It is if FS was able to detect NAT. Are you behind PMP or UPnP? Otherwise, no go....
Have you changed the ext-sip-ip too? Regards, JM On Mon, Nov 9, 2009 at 12:32 AM, Mark Campbell-Smith < [email protected]> wrote: > Hi again, > > Actually, changing the <param name="ext-rtp-ip" value="auto-nat"/> to > <param name="ext-rtp-ip" value="$${external_sip_ip}"/> means that I > now see the IP address in the INVITE message: > > v=0 > o=FreeSWITCH 1257711702 1257711703 IN IP4 124.xxx.xxx.xxx > s=FreeSWITCH > c=IN IP4 124.xxx.xxx.xxx > t=0 0 > m=audio 21234 RTP/AVP 0 2 9 8 101 13 > > Why would this be? I thought auto-nat was meant to solve these issues? > > However, I still do not see the TRYING or RINGING messages.... ideas > appreciated. > > Thanks! > > On Mon, Nov 9, 2009 at 1:14 PM, Mark Campbell-Smith > <[email protected]> wrote: > > OK.. thanks Mike. > > > > I assume I am using the Internal profile. I have defined user 2000 > > in the 'directory' using a context called family: switch_ivr.c:1367 > > Transfer sofia/internal/[email protected] to xml[2...@family] > > > > This is an extract from sofia: > > > > sofia status profile internal > > > ================================================================================================= > > Name internal > > Domain Name N/A > > DBName sofia_reg_internal > > Pres Hosts > > Dialplan XML > > Context public > > Challenge Realm auto_from > > RTP-IP 192.168.1.120 > > Ext-RTP-IP 124.xxx.xxx.xxx > > SIP-IP 192.168.1.120 > > Ext-SIP-IP 124.xxx.xxx.xxx > > URL sip:[email protected]:5060 > > BIND-URL sip:[email protected]:5060 > > HOLD-MUSIC silence > > OUTBOUND-PROXY N/A > > CODECS G726-32,G722,PCMU,PCMA > > TEL-EVENT 101 > > DTMF-MODE rfc2833 > > CNG 13 > > SESSION-TO 0 > > MAX-DIALOG 0 > > NOMEDIA false > > LATE-NEG false > > PROXY-MEDIA false > > AGGRESSIVENAT true > > STUN-ENABLED true > > STUN-AUTO-DISABLE false > > CALLS-IN 100 > > FAILED-CALLS-IN 25 > > CALLS-OUT 38 > > FAILED-CALLS-OUT 31 > > > > Registrations: > > > ================================================================================================= > > Call-ID: [email protected] > > User: [email protected] > > Contact: "user" <sip:[email protected]:5060> > > Agent: dunno > > Status: Registered(UDP)(unknown) EXP(2009-11-09 14:58:30) > > Host: freeswitch > > IP: 58.xxx.xxx.xxx > > Port: 5060 > > Auth-User: 2000 > > Auth-Realm: markcs.dyndns.org > > MWI-Account: [email protected] > > > > The internal.xml file has a lot in it, but I guess these are the > > important things for this profile: > > > > <param name="ext-rtp-ip" value="auto-nat"/> > > <param name="ext-sip-ip" value="auto-nat"/> > > > > <param name="sip-port" value="$${internal_sip_port}"/> > > <param name="rtp-ip" value="auto"/> > > > > I will try to change auto-nat to be $${external_sip_ip} > > > > One question though: Any idea why I never see the TRYING or RINGING > > messages? Are tehse related to the RTP IP address or not? Without > > these I assume something is incorrect and I do not hear ringback.... > > > > Thanks! > > > > On Mon, Nov 9, 2009 at 11:41 AM, Michael Jerris <[email protected]> wrote: > >> Your packet traces would disagree with the statements below. It is > >> sending your internal address in rtp, so its not set correctly on > >> whatever profile your using to call out, > >> > >> MIke > >> > >> On Nov 8, 2009, at 4:59 PM, Mark Campbell-Smith wrote: > >> > >>> Hi Mike, > >>> > >>> I should have put that in also. > >>> > >>> I do have external_rtp_ip set in my config. I have it set to my > >>> domain name: > >>> <X-PRE-PROCESS cmd="set" data="external_rtp_ip=host:mydomainname"/> > >>> > >>> I should also mention that if I use flaphone.com (which registers with > >>> an external IP address), then I get audio. In sofia, I see my IP > >>> addresses: > >>> > >>> = > >>> = > >>> = > >>> = > >>> = > >>> = > >>> = > >>> = > >>> = > >>> = > >>> = > >>> = > >>> = > >>> = > >>> = > >>> = > >>> = > >>> = > >>> = > >>> = > >>> = > >>> = > >>> = > >>> = > >>> = > >>> = > >>> = > >>> ====================================================================== > >>> Name internal > >>> Domain Name N/A > >>> DBName sofia_reg_internal > >>> Pres Hosts > >>> Dialplan XML > >>> Context public > >>> Challenge Realm auto_from > >>> RTP-IP 192.168.1.120 > >>> Ext-RTP-IP 124.xxx.xxx.xxx > >>> SIP-IP 192.168.1.120 > >>> Ext-SIP-IP 124.xxx.xxx.x > >>> URL sip:[email protected]:5060 > >>> BIND-URL sip:[email protected]:5060 > >>> HOLD-MUSIC silence > >>> OUTBOUND-PROXY N/A > >>> CODECS G726-32,G722,PCMU,PCMA > >>> TEL-EVENT 101 > >>> DTMF-MODE rfc2833 > >>> CNG 13 > >>> SESSION-TO 0 > >>> MAX-DIALOG 0 > >>> NOMEDIA false > >>> LATE-NEG false > >>> PROXY-MEDIA false > >>> AGGRESSIVENAT true > >>> STUN-ENABLED true > >>> STUN-AUTO-DISABLE false > >>> > >>> On Mon, Nov 9, 2009 at 7:28 AM, Michael Jerris <[email protected]> > >>> wrote: > >>>> You don't have ext-rtp-ip set in your config. > >>>> > >>>> Mike > >>>> > >>>> On Nov 8, 2009, at 6:59 AM, Mark Campbell-Smith wrote: > >>>> > >>>>> Hi! > >>>>> > >>>>> I have FS natted and am connecting with an 'external' extension that > >>>>> is registered to FS. ie the extension 2000 is registered on the > >>>>> internet with a public IP through my router to FS (192.168.1.120 IP > >>>>> address). uPnP works and I see that the extension is registered > >>>>> successfully. > >>>>> > >>>>> The problem is that I do not get any audio > >>>>> > >>>>> When looking at the SIP trace, I see the INVITE but do not see a > >>>>> TRYING or RINGING message. The extension is actually ringing. I > >>>>> modified the RTP port range on the remote end to match the RTP ports > >>>>> of freeswitch. > >>>>> > >>>>> I have put a sip trace in the pastebin at > http://pastebin.freeswitch.org/11035 > >>>>> > >>>>> If anyone has an idea what needs to be set to get audio, help > >>>>> appreciated. > >>>>> > >>>>> Thanks! > >>>> > >>>> > >>>> _______________________________________________ > >>>> FreeSWITCH-users mailing list > >>>> [email protected] > >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > >>>> users > >>>> http://www.freeswitch.org > >>>> > >>> > >>> _______________________________________________ > >>> FreeSWITCH-users mailing list > >>> [email protected] > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > >>> users > >>> http://www.freeswitch.org > >> > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> [email protected] > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > _______________________________________________ > FreeSWITCH-users mailing list > [email protected] > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org >
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