Hi,
I’m trying to interconnect 2 offices with FreePBX (Office A) and Grandstream UMC6102 (Office B).
Office B has no problem calling Office A but I can never call the SIP Trunk of Office A to call Office B. I keeps getting that “All circuit is busy…” prompt when I dial 3XX
This are what I have configured on the FreePBX:
Outbound Routes:
(prepend)prefix| [3XX]
Log:
[2019-01-07 21:56:35] VERBOSE[10909][C-000009fd] pbx.c: Executing [s@macro-dialout-trunk:23] GotoIf("PJSIP/288-00000f21", "0?customtrunk") in new stack
[2019-01-07 21:56:35] VERBOSE[10909][C-000009fd] pbx.c: Executing [s@macro-dialout-trunk:24] Dial("PJSIP/288-00000f21", "PJSIP/307@PBXTW,300,T") in new stack
[2019-01-07 21:56:35] ERROR[30517] res_pjsip.c: Endpoint 'PBXTW': Could not create dialog to invalid URI 'PBXTW'. Is endpoint registered and reachable?
[2019-01-07 21:56:35] ERROR[30517] chan_pjsip.c: Failed to create outgoing session to endpoint 'PBXTW'
[2019-01-07 21:56:35] WARNING[10909][C-000009fd] app_dial.c: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination)
[2019-01-07 21:56:35] VERBOSE[10909][C-000009fd] app_dial.c: Everyone is busy/congested at this time (1:0/0/1)
[2019-01-07 21:56:35] VERBOSE[10909][C-000009fd] pbx.c: Executing [s@macro-dialout-trunk:25] NoOp("PJSIP/288-00000f21", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 3") in new stack
[2019-01-07 21:56:35] VERBOSE[10909][C-000009fd] pbx.c: Executing [s@macro-dialout-trunk:26] GotoIf("PJSIP/288-00000f21", "0?continue,1:s-CHANUNAVAIL,1") in new stack