Hello

When I try to toggle call flow from fop2 plugin I get error:

[2017-09-06 17:35:57] NOTICE[23396][C-00000eed]: chan_local.c:985 local_call: No such extension/context s@custom-fop2-callflow while calling Local channel
[2017-09-06 17:35:57] NOTICE[23396][C-00000eed]: channel.c:5747 __ast_request_and_dial: Unable to call channel Local/s@custom-fop2-callflow
[2017-09-06 17:35:58] NOTICE[23396][C-00000eed]: chan_sip.c:30073 sip_request_call: Asked to get a channel of unsupported format (nothing) while capability is (gsm|ulaw|alaw|g726)
[2017-09-06 17:35:58] WARNING[23396][C-00000eed]: app_dial.c:2455 dial_exec_full: Unable to create channel of type 'SIP' (cause 58 - Bearer capability not available)
[2017-09-06 17:35:58] WARNING[23396][C-00000eed]: channel.c:5362 set_format: Unable to find a codec translation path from (nothing) to (gsm|ulaw|alaw|g729|g722)
[2017-09-06 17:35:58] WARNING[23396][C-00000eed]: file.c:1017 ast_streamfile: Unable to open all-circuits-busy-now (format (nothing)): Function not implemented
[2017-09-06 17:35:58] WARNING[23396][C-00000eed]: app_playback.c:484 playback_exec: ast_streamfile failed on OutgoingSpoolFailed for all-circuits-busy-now&please-try-call-later, noanswer
[2017-09-06 17:35:58] WARNING[23396][C-00000eed]: channel.c:5362 set_format: Unable to find a codec translation path from (nothing) to (ulaw|alaw|g729|g722)
[2017-09-06 17:35:58] WARNING[23396][C-00000eed]: file.c:1017 ast_streamfile: Unable to open please-try-call-later (format (nothing)): Function not implemented
[2017-09-06 17:35:58] WARNING[23396][C-00000eed]: app_playback.c:484 playback_exec: ast_streamfile failed on OutgoingSpoolFailed for all-circuits-busy-now&please-try-call-later, noanswer
[2017-09-06 17:35:58] WARNING[23396][C-00000eed]: channel.c:5362 set_format: Unable to find a codec translation path from (nothing) to (slin)
[2017-09-06 17:35:58] WARNING[23396][C-00000eed]: indications.c:157 playtones_alloc: Unable to set 'OutgoingSpoolFailed' to signed linear format (write)
[2017-09-06 17:35:58] WARNING[23396][C-00000eed]: channel.c:4861 ast_prod: Prodding channel 'OutgoingSpoolFailed' failed
[2017-09-06 17:35:58] WARNING[23396][C-00000eed]: channel.c:4675 ast_indicate_data: Unable to handle indication 8 for 'OutgoingSpoolFailed'
[2017-09-06 17:35:58] WARNING[23396][C-00000eed]: channel.c:5362 set_format: Unable to find a codec translation path from (nothing) to (slin)
[2017-09-06 17:35:58] ERROR[23396][C-00000eed]: channel.c:8832 ast_channel_start_silence_generator: Could not set write format to SLINEAR

Any suggestions ?

Hi guys
First of all great product !.
We have a lot of attended transfer from the telephone operator. In the call history of my extension in Fop2 in does show only the number of the operator. My extension does show operator, and after the transfer it show the CID. (PAI option in asterisk)
Is there a way to show the original CID. It is in the PAI header of sip.

KR
LM