[asterisk-users] VoiceMail Audio playing

Joshua Colp jcolp at digium.com
Fri Jul 15 07:14:39 CDT 2016

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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