[Freeswitch-users] Extension: No audio
Michael Jerris
mike at jerris.com
Sun Nov 8 16:41:42 PST 2009
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 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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>
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