Hi all,
we trying to integrate a Cisco Call Manager with the Asterisk PBX.
We are running this configuration: the Call Manager is connected via SIP to Asterisk as a peer, and handles on its own a number of telephones.
When the Call Manager receive a call the its own IVR starts the playback but no audio is heard to the caller. I also noticed the RTP stream in this stage is not up, and the connection is still in ringing state on the Asterisk side. Then after a while, the Cisco IVR transfers the call to an telephone, where picking up the conversation is heard.
Any ideas or suggestions?