[Freeswitch-users] Extension: No audio

[Mark Campbell-Smith]

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 at 192.168.1.120:5060
BIND-URL                sip:mod_sofia at 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 at 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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