@bazzacad wrote:
We have one user with a virtual extension (no desktop endpoint phone) who uses Find Me/Follow Me to forward her calls to her cell phone. This setup used to work fine. I'm not sure if it broke when I updated from the old PIAF to FreePBX 13, or when I added a new trunk. All of our inbound calls come in on one SIP provider's trunk & most of the outbound calls go to another SIP provider's trunk. I think this might be the issue, but please correct me if I'm wrong. If I call the users extension internally, (x224 --> x249 --> cell phone), it works fine & gets forwarded to her cell phone. But if someone calls in from the outside & the receptionist transfers to her extension, her cell phone never rings. (Outside call --> x221 --> x249 --> cell phone)
I see these warnings in the CLI (when this call\tranfer happens):
[2017-01-31 13:19:06] WARNING[13097][C-00000045]: ast_expr2.fl:470 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected '>', expecting $end; Input:
0
^
[2017-01-31 13:19:06] WARNING[13097][C-00000045]: ast_expr2.fl:474 ast_yyerror: If you have questions, please refer to https://wiki.asterisk.org/wiki/display/AST/Channel+Variables
[2017-01-31 13:19:35] WARNING[13147][C-00000047]: app_dial.c:2455 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
[2017-01-31 13:19:42] WARNING[13156][C-00000047]: app_dial.c:2455 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)And here is the call event log (I've removed the endings of the DIDs)
Any suggestions on how to fix this?
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