Hi,
My original post was here BLF attended transfers with transfer callbacks but it closed with inactivity as I was pulled back into another project at that time.
I have my custom dial plan built and tested by dialling *2 for an attended transfer, however when I move to using a BLF attended transfer, it doesnt use the same context as a dtmf *2 attended transfer. Can anyone tell me how or where I can get the BLF transfer to pick up my custom dial plan?
Hopefully this is something simple, but having looked through lots of other posts I cant see anywhere that has an answer. Some people setup BLFs and DTMF buttons but having to setup and look after two sets of buttons is not somethign we can do as we dont have enough buttons on the Sangoma 500’s, plus i’m sure the client would think it strange we cant do the same attended transfer in the BLF as by pressing *2.
I know we can do extra things in the BLF like short and long presses but I would expect an attended transfer (by whatever means, BLF or *2) to go to the same transfer_context. Can I do something funky in the basefile edit template or any other file to tweak the BLF transfer context?
I am using a new FreePBX install
FreePBX 14.0.13.4
Current Asterisk Version: 13.22.0
My config edit file changes -
globals_custom.conf
TRANSFER_CONTEXT = custom-test_transfer
extensions_custom.conf
[custom-test_transfer]
exten => _X.,1,NoOp(Entering Daves custom-test_transfer)
this is not the full dial plan, this is just to show in the logs below the noop line during a dtmf attended transfer, and that it doesnt show during a BLF attended transfer.
Logs during a DTMF Attended transfer using *2 EXT #
[2019-09-06 08:48:21] VERBOSE[5620][C-0000001e] app_dial.c: PJSIP/4001-00000038 is ringing
[2019-09-06 08:48:22] VERBOSE[5620][C-0000001e] app_dial.c: PJSIP/4001-00000038 answered IAX2/sbc3-2340
[2019-09-06 08:48:22] VERBOSE[5622][C-0000001e] bridge_channel.c: Channel PJSIP/4001-00000038 joined ‘simple_bridge’ basic-bridge <7bdc2e82-aaa2-456d-af5d-fa141463d643>
[2019-09-06 08:48:22] VERBOSE[5620][C-0000001e] bridge_channel.c: Channel IAX2/sbc3-2340 joined ‘simple_bridge’ basic-bridge <7bdc2e82-aaa2-456d-af5d-fa141463d643>
[2019-09-06 08:48:26] VERBOSE[5622][C-0000001e] bridge_basic.c: Channel PJSIP/4001-00000038: Started DTMF attended transfer.
[2019-09-06 08:48:26] VERBOSE[5622][C-0000001e] file.c: <PJSIP/4001-00000038> Playing ‘pbx-transfer.ulaw’ (language ‘en_GB’)
[2019-09-06 08:48:26] VERBOSE[5620][C-0000001e] res_musiconhold.c: Started music on hold, class ‘default’, on channel ‘IAX2/sbc3-2340’
[2019-09-06 08:48:31] VERBOSE[5625][C-0000001e] bridge_channel.c: Channel Local/4000@custom-test_transfer-0000004f;1 joined ‘simple_bridge’ basic-bridge <365c2d13-a28d-4c97-8987-9e82c9f6f199>
[2019-09-06 08:48:31] VERBOSE[5624][C-0000001e] pbx.c: Executing [4000@custom-test_transfer:1] NoOp(“Local/4000@custom-test_transfer-0000004f;2”, "Entering Daves custom-test_transfer") in new stack
[2019-09-06 08:48:31] VERBOSE[5624][C-0000001e] pbx.c: Executing [4000@custom-test_transfer:2] NoOp(“Local/4000@custom-test_transfer-0000004f;2”, “The caller id number before dial is: 4001”) in new stack
Logs during a BLF Attended transfer to the same extension - we dont see it picking up the custom-test_transfer
[2019-09-06 08:51:06] VERBOSE[6110][C-00000021] app_dial.c: PJSIP/4001-0000003e is ringing
[2019-09-06 08:51:08] VERBOSE[6110][C-00000021] app_dial.c: PJSIP/4001-0000003e answered IAX2/sbc3-6413
[2019-09-06 08:51:08] VERBOSE[6112][C-00000021] bridge_channel.c: Channel PJSIP/4001-0000003e joined ‘simple_bridge’ basic-bridge <4b2ab841-5127-458d-92f1-be575877046f>
[2019-09-06 08:51:08] VERBOSE[6110][C-00000021] bridge_channel.c: Channel IAX2/sbc3-6413 joined ‘simple_bridge’ basic-bridge <4b2ab841-5127-458d-92f1-be575877046f>
[2019-09-06 08:51:13] VERBOSE[6110][C-00000021] res_musiconhold.c: Started music on hold, class ‘default’, on channel ‘IAX2/sbc3-6413’
[2019-09-06 08:51:13] VERBOSE[15638] pbx_variables.c: Setting global variable ‘SIPDOMAIN’ to ‘servername.domain.co.uk’
[2019-09-06 08:51:13] VERBOSE[15638] netsock2.c: Using SIP RTP Audio TOS bits 184
[2019-09-06 08:51:13] VERBOSE[15638] netsock2.c: Using SIP RTP Audio TOS bits 184 in TCLASS field.
[2019-09-06 08:51:13] VERBOSE[15638] netsock2.c: Using SIP RTP Audio CoS mark 5
[2019-09-06 08:51:13] VERBOSE[6113][C-00000022] pbx.c: Executing [4000@from-internal:1] GotoIf(“PJSIP/4001-0000003f”, “1?ext-local,4000,1:followme-check,4000,1”) in new stack
[2019-09-06 08:51:13] VERBOSE[6113][C-00000022] pbx_builtins.c: Goto (ext-local,4000,1)
[2019-09-06 08:51:13] VERBOSE[6113][C-00000022] pbx.c: Executing [4000@ext-local:1] Set(“PJSIP/4001-0000003f”, “__RINGTIMER=30”) in new stack
[2019-09-06 08:51:13] VERBOSE[6113][C-00000022] pbx.c: Executing [4000@ext-local:2] Macro(“PJSIP/4001-0000003f”, “exten-vm,4000,4000,0,0,0”) in new stack
[2019-09-06 08:51:13] VERBOSE[6113][C-00000022] pbx.c: Executing [s@macro-exten-vm:1] Macro(“PJSIP/4001-0000003f”, “user-callerid,”) in new stack
Thanks for your help,
Regards
David