Hi there,
I just installed a new VM with a public IP, latest Debian+FreePBX+Asterisk, all by the book.
I set Extensions and Trunks up, and got calls running across the dialplan smoothly.
The problem started when I tried to make a call deflection by sending a 302 Moved Temporarily message to my trunk via a Transfer(PJSIP/sip:[email protected]:port)
It actually does nothing: no reaction in the logs, no errors, no 302 Message is generated and only the TRANSFERSTATUS variable gets a FAILURE value (if I strip the PJSIP from the Transfer app parameters, I get UNSUPPORTED instead).
I’ve looked everywhere, but cannot find anything related to this (I was expecting for some headers to be set prior to the Transfer(), but nothing is documented anywhere). I’d really appreciate any hints on what could be the problem here…
Tested so far:
- adding allow_transfer=yesto the endpoint at pjsip_custom.conf
- adding canreinvite=yes to the endpoint at pjsip_custom.conf (I suppose that is not even supported on Asterisk 21)
- adding redirect_method=pjsip_uri to the endpoint (same place as before, but not showing when a “pjsip show endpoint my_endpoint” is issued).
- setting direct_media = yes on the FreePBX GUI (not really related, I know, but…)
- All kind of variations on the Transfer parameters (with/without specifying Technology, using SIP URIs, number@trunk as detinations, marking with <>…
Thanks for any ideas! (really running out of them right now…)