Hi there,
I’m setting up a private SIP server with the FreePBX distro (STABLE SNG7-PBX-64bit-2008-1, FreePBX 15/Asterisk 16).
It’s currently working normally, we can call each other and be called, using pjsip channel. I would like to receive calls on my 3G phone (using external SIP URI) when I’m not on my desk, so I looked on the forum for various ways to ring another custom extension when unavailable, none are currently working.
I have IPPI on my iPhone (and Linphone, but let’s concentrate on IPPI first) and I tried to create a “custom extension”, with PJSIP/[email protected]
in the Dial input, but when I call with an internal extension, I’m immediately disconnected (no error voice message like when I call an inexistant extension), and the logs show:
[2020-09-17 14:48:55] WARNING[9310][C-00000001] app_macro.c: Macro() is deprecated and will be removed from a future version of Asterisk.
[2020-09-17 14:48:55] WARNING[9310][C-00000001] app_macro.c: Dialplan should be updated to use Gosub instead.
[2020-09-17 14:48:55] WARNING[9310][C-00000001] app_dial.c: Unable to create channel of type ‘PJSIP’ (cause 3 - No route to destination)
I tried to call the ippi URI (nickname@sip.ippi.com) from linphone, it rings, so the URI is correct.
I tried SIP/…, PJSIP/…, doesn’t work. Any idea if I’m missing something? I don’t have any trunk for this call as I’m trying to make Asterisk make a p2p call to IPPI.
Thanks for your help