We have a range of SIP numbers in CME4 set up as dial-peers (so they can authenticate with their appropriate username / passwords):
dial-peer voice 670583 pots
description x2603 David
authentication username 670583 password xxx
And these are mapped thru to ephone-dn's, such as:
number 670583 no-reg both
description David Deskphone
call-forward noan 7123 timeout 25
However, when an external call comes into this SIP number, the e-phone-dn rings correctly, but upon call-divert, the call is dropped.
x123 is an Asterisk voicemail (SIP client, trunk 7T, ext 123).
All works fine with a similarly configured pots dial-peer for an FXO interface.
Can anyone suggest where I'm going wrong?