[Freeswitch-users] Can I get SIP DID working?

[Ivan C Myrvold]

I put this into internal.xml, and this seems to do the trick:

     <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
     <param name="ext-sip-ip" value="$${external_sip_ip}"/>

Ivan

Den 21. juni. 2008 kl. 15:47 skrev Ivan C Myrvold:

> Yeah, the call arrives nicely now, but the audio is only 1-way.
> I have port forwarded all the ports in my router to FreeSwitch, but
> still experiences just oneway audio, from FreeSwitch to the DID  
> caller.
> Looks like the RTP is not forwarded correctly to FreeSwitch.
> Could it be my Linksys router, which is not forwarding RTP ports
> correctly, or is there still a piece in FreeSwitch I have missed?
>
> Ivan
>
> Den 20. juni. 2008 kl. 22:52 skrev Brian West:
>
>> Good to know it snapped into place now! :P
>>
>> /b
>> On Jun 20, 2008, at 3:49 PM, Ivan C Myrvold wrote:
>>
>>> This is all making sense to me now.
>>
>>
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