We recently upgrade from Asterisk 1.4 to 1.8.15 and experiencing an issue. Searched the web with no luck except to find others with the same experience
When a user sets call forwarding from their desk phone, and an external call comes through, the external call times out and hangs up eventually.
Temporary workaround is to have the call “Answer” before ringing to the original extension
Any fixes to this? Or any way to solve this?
Machine is not behind NAT and dont see any codec issues
EDIT - I realize that the calls are forwarded to Loal/XXX which is causing issues. I found that promiscredir=yes should switch the “Local” to “SIP”, which is does. Except it goes to a weird string:
Now forwarding SIP/XXX-0000000f to 'SIP/208::::UDP@pbx01.xx.xx
Not sure what the “::::UDP” stuff is, but if I can get that out, I think that would help
If anyone knows how, this would be great