03-29-2007 09:19 AM
My customer would integrate its CallManager 4.1(3) platform with a SIP Trunk using Asterix.
The incoming calls seems busy......maybe I wrong configuring Trunk SIP on CallManager.
Someone can help me ?
03-30-2007 07:20 AM
We did the integration by means of H323 Protocol try configuring the Asterisk Server as h323 Gateway in the CM,Configure the Route-pattern pointing to the Asterisk.
This solution is working with us in production.
regards,
Anis Faruqui...
04-01-2007 03:13 AM
hi
search in this site
http://www.voip-info.org
GD
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