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 <mcampbellsmith@gmail.com> 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/1000@192.168.1.120 to XML[2000@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:mod_sofia@192.168.1.120:5060 BIND-URL sip:mod_sofia@192.168.1.120:5060 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: 68534BBA9B461526@58.169.138.53 User: 2000@192.168.1.120 Contact: "user" <sip:2000@58.xxx.xxx.xxx: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: 2000@192.168.1.120
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 <mike@jerris.com> 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:mod_sofia@192.168.1.120:5060 BIND-URL sip:mod_sofia@192.168.1.120:5060 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 <mike@jerris.com> 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!
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