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
