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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