Is there a way to determine if FS has detected nat? I am behind UPnP and I can see on the router the mappings for Freeswitch.
2009/11/9 João Mesquita <[email protected]>: > 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 > > > _______________________________________________ > 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
