FreePbx B2BUA - Skip confirm call

Hi, everyone

I am not familiar with the rules of this forum.
Maybe I need to paste the whole problem description here instead of putting the link here…
So, I will do both things.

Here is the link to SO, I have asked the same question there, but no luck yet:

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 [[email protected]: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 [[email protected]:11] Set("SIP/InTrunk-000013aa", "CALLERID(number)=+18578888888") in new stack
[2017-09-06 15:25:41] VERBOSE[16408][C-0000a153] pbx.c: Executing [[email protected]:12] Set("SIP/InTrunk-000013aa", "CALLERID(name)=+18578888888") in new stack
[2017-09-06 15:25:41] VERBOSE[16408][C-0000a153] pbx.c: Executing [[email protected]: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'

Hope, someone can help me with this.
Thanks in advance!

please check the callee side, and make sure route to right direction.

The callee side seems to answer the call, according to this line:

[2017-09-06 15:25:23] VERBOSE[16408][C-0000a153] app_dial.c: SIP/OutTrunk-000013ab answered SIP/InTrunk-000013aa

And as far as I understand, this line means that he does not press anything after hearing the confirmation question:

[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

Obviously, he can not respond anything in our case due to our integration conditions…

Do you think this line means that call did not reach callee?

you also check the callee(remote side), what happened from callee? nothing from callee?

Unfortunately, I can find this out only in the morning…