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.




-- ========================================== Rod Bacon - VOIP Systems Engineer Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600 Fax: +613 99401650 ==========================================

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