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...w 711, UDP, ip address & port, etc), and a route pattern exists to take calls to the right trunk. The system will let me complete the dialing sequence to the Asterisk server, but as soon as I enter the last digit I get a busy signal. Thoughts anyone? Here's my sip.conf if that helps... [callman] type=peer context=incoming insecure=very host=(ip of my call manager server) disallow=all allow=ulaw allow=alaw nat=no canreinvite=yes qualify=yes Thanks! Aaron -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachm...

Dear All : I need to mask the caller ID and pretend to make a transfer call from another extension : exten => 558,1,Answer exten => 558,2,Playback(soundclip) exten => 558,3,Dial(SIP/472@callman) The scenario is like this : Someone is calling 558 at my company - he will hear a soundclip voice message then I will direct it to extension 472 I need 472 to not see the extension of the caller-ID I need him to see it coming from another extension let us say : 111 ... How can I do this...

I have Asterisk 1.4 tied via SIP to a Cisco Callmanager 6.1 system. Calls between the systems (ie. extension to extension) work perfectly. However when I attempt to make an outside call from an Asterisk extension through Call Manager to the outside world, it connects but only for a few seconds, and on the Asterisk console I get: Got SIP response 5...