I have a Sip trunk between a PiaF Green w/Asterisk 11.5.1 and FreePBX 2.11.0.23 and an Avaya Session Manager. The calls from the PiaF to the Asterisk work great. When I status the signaling-group in the SM it shows Far-End Bypass. Both PBXs are 1n the same network. This is what the trunk in the PiaF looks like.
host=x.x.x.115
type=friend
bindport=5060
disallow=all&all
allow=ulaw,alaw
context=from-internal
qualify=yes
The System Manager is v 6.2. Communication Manager is v 5.2
Any suggestions or ideas?
host=x.x.x.115
type=friend
bindport=5060
disallow=all&all
allow=ulaw,alaw
context=from-internal
qualify=yes
The System Manager is v 6.2. Communication Manager is v 5.2
Any suggestions or ideas?