Joaquin Alzola wrote:
Hi Madushan
Maybe I was not clear …. After SIP negotiation and SDP set up on the
VoiceMail Server ….
Is there a file to specify a MGw (the machine that deliver RTP packages
to end user)?
Asterisk does not separate things like this. For media originating from
it the source will always be it. That is if you do a SIP call to
Asterisk then media will come from that same Asterisk.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
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