I am using Asterisk via FreePbx to implement B2BUA. As part of my task, I have created an Inbound Route, and set the Destination=Trunk, and selected one of my trunks with correct SIP credentials.
Everything seems to work fine except one sad issue. When Asterisk dials the target SIP trunk, it prompts "Confirm Call" there, asking the destination side to press 1 to accept the call. I need to remove this prompt.
It sounds stupid, but I can not find a way to do it anywhere in FreePbx Web GUI or in FreePbx online documentation.
Can someone suggest a solution to turn this Confirm Call feature Off for my FreePBX SIP trunk?
Some destination trunk settings
Asterisk Trunk Dial Options:
SIP/[email protected]
Sip Settings/Outgoing/Peer Details:
host=hostname.com
username=username
type=peer
port=5060
transport=tcp
tcpenable=yes
privacy=off
Piece of log showing the problem
[2017-09-06 15:25:21] VERBOSE[16408][C-0000a153] pbx.c: Executing [tdial@ext-trunk:10] Dial("SIP/InTrunk-000013aa", "SIP/OutTrunk/username,300,SIP/[email protected]") in new stack
[2017-09-06 15:25:21] VERBOSE[16408][C-0000a153] app_dial.c: Privacy DB is 'tdial', clid is '+18578888888'
[2017-09-06 15:25:21] VERBOSE[16408][C-0000a153] netsock2.c: Using SIP RTP TOS bits 184
[2017-09-06 15:25:21] VERBOSE[16408][C-0000a153] netsock2.c: Using SIP RTP CoS mark 5
[2017-09-06 15:25:21] VERBOSE[16408][C-0000a153] app_dial.c: Called SIP/OutTrunk/username
[2017-09-06 15:25:21] VERBOSE[16408][C-0000a153] res_musiconhold.c: Started music on hold, class 'none', on channel 'SIP/InTrunk-000013aa'
[2017-09-06 15:25:21] WARNING[16408][C-0000a153] format_wav.c: Read failed (type)
[2017-09-06 15:25:21] WARNING[16408][C-0000a153] file.c: Unable to open format wav
[2017-09-06 15:25:21] WARNING[16408][C-0000a153] res_musiconhold.c: Unable to open file '/var/lib/asterisk/moh/.nomusic_reserved/silence': No such file or directory
[2017-09-06 15:25:21] VERBOSE[16408][C-0000a153] res_musiconhold.c: Stopped music on hold on SIP/InTrunk-000013aa
[2017-09-06 15:25:21] VERBOSE[16408][C-0000a153] app_dial.c: SIP/OutTrunk-000013ab is ringing
[2017-09-06 15:25:23] VERBOSE[16408][C-0000a153] app_dial.c: SIP/OutTrunk-000013ab answered SIP/InTrunk-000013aa
[2017-09-06 15:25:23] VERBOSE[16408][C-0000a153] file.c: <SIP/OutTrunk-000013ab> Playing 'priv-callpending.ulaw' (language 'en')
[2017-09-06 15:25:27] VERBOSE[16408][C-0000a153] file.c: <SIP/OutTrunk-000013ab> Playing 'priv-callerintros/+18578888888.slin' (language 'en')
[2017-09-06 15:25:32] VERBOSE[16408][C-0000a153] file.c: <SIP/OutTrunk-000013ab> Playing 'priv-callee-options.ulaw' (language 'en')
[2017-09-06 15:25:41] VERBOSE[16408][C-0000a153] file.c: <SIP/OutTrunk-000013ab> Playing 'screen-callee-options.ulaw' (language 'en')
[2017-09-06 15:25:41] VERBOSE[16408][C-0000a153] file.c: <SIP/OutTrunk-000013ab> Playing 'vm-sorry.ulaw' (language 'en')
[2017-09-06 15:25:41] VERBOSE[16408][C-0000a153] file.c: <SIP/OutTrunk-000013ab> Playing 'priv-callerintros/+18578888888.slin' (language 'en')
[2017-09-06 15:25:41] VERBOSE[16408][C-0000a153] file.c: <SIP/OutTrunk-000013ab> Playing 'priv-callee-options.ulaw' (language 'en')
[2017-09-06 15:25:41] VERBOSE[16408][C-0000a153] file.c: <SIP/OutTrunk-000013ab> Playing 'screen-callee-options.ulaw' (language 'en')
[2017-09-06 15:25:41] VERBOSE[16408][C-0000a153] file.c: <SIP/OutTrunk-000013ab> Playing 'vm-sorry.ulaw' (language 'en')
[2017-09-06 15:25:41] VERBOSE[16408][C-0000a153] file.c: <SIP/OutTrunk-000013ab> Playing 'priv-callerintros/+18578888888.slin' (language 'en')
[2017-09-06 15:25:41] VERBOSE[16408][C-0000a153] file.c: <SIP/OutTrunk-000013ab> Playing 'priv-callee-options.ulaw' (language 'en')
[2017-09-06 15:25:41] VERBOSE[16408][C-0000a153] file.c: <SIP/OutTrunk-000013ab> Playing 'screen-callee-options.ulaw' (language 'en')
[2017-09-06 15:25:41] VERBOSE[16408][C-0000a153] file.c: <SIP/OutTrunk-000013ab> Playing 'vm-sorry.ulaw' (language 'en')
[2017-09-06 15:25:41] NOTICE[16408][C-0000a153] app_dial.c: privacy: no valid response from the callee. Sending the caller to voicemail, the callee isn't responding
[2017-09-06 15:25:41] VERBOSE[16408][C-0000a153] pbx.c: Executing [tdial@ext-trunk:11] Set("SIP/InTrunk-000013aa", "CALLERID(number)=+18578888888") in new stack
[2017-09-06 15:25:41] VERBOSE[16408][C-0000a153] pbx.c: Executing [tdial@ext-trunk:12] Set("SIP/InTrunk-000013aa", "CALLERID(name)=+18578888888") in new stack
[2017-09-06 15:25:41] VERBOSE[16408][C-0000a153] pbx.c: Executing [tdial@ext-trunk:13] Hangup("SIP/InTrunk-000013aa", "") in new stack
[2017-09-06 15:25:41] VERBOSE[16408][C-0000a153] pbx.c: Spawn extension (ext-trunk, tdial, 13) exited non-zero on 'SIP/InTrunk-000013aa'
By default freepbx not do confirm call. Check privacy settings,check "dial options" on trunk.
[2017-09-06 15:25:21] VERBOSE[16408][C-0000a153] app_dial.c: Privacy DB is 'tdial', clid is '+18578888888'
You log is usless, use "core set verbose 3" to see dialplan transitions.