[Asterisk-Users] RTP not being sent by asterisk

[Rod Bacon]

 From my understanding, * uses the incoming RTP stream itself as a 
timing source for sending it's outgoing stream, hence the reason * 
doesn't like/support silence suppression.

In other words, if there's no RTP headed back to *, then it won't send 
anything.

(Someone please correct me if I'm talking crap!)

I don't know if this is relevant to your situation in any way, but it's 
worth consideration.




trixter http://www.0xdecafbad.com wrote:
> On Thu, 2005-04-14 at 16:15 +1000, Rod Bacon wrote:
> 
>>Are the calls coming from SIP or PSTN?
> 
> 
> from sip, and I can see packets going from sip -> asterisk just nothing
> outside of sip going from asterisk -> sip phone. 
> 
> Its like there is a blocking issue, although I dont know why this would
> have happened.  
> 
> 

-- 
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Rod Bacon - VOIP Systems Engineer
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