Problem with features

Asterisk Asterisk Support
1

Hi,
In my features.conf, i have this :
[featuremap]
;blindxfer => #1 ; Blind transfer (default is #) – Make sure to$
;disconnect => *0 ; Disconnect (default is *) – Make sure to se$
;automon => *1 ; One Touch Record a.k.a. Touch Monitor – Make$
atxfer => ** ; Attended transfer – Make sure to set the T a$
;parkcall => #72 ; Park call (one step parking) – Make sure to$
;automixmon => *3

But when I do, nothing happens.

Is there any other adjustment to be made?

2

Precision :
This is with a user in Queue, so I can not add the T on the Dial.

3

Queue takes the same options.

Incidentally, enabling features inhibits direct media!

4

Why can’t you? The Queue application accepts both a ‘t’ and ‘T’ option to control attended transfers.

It’s documented in the Wiki

https://wiki.asterisk.org/wiki/display/AST/Asterisk+14+Application_Queue

5

exten => 1,1,Queue(technique,[Tt])
OR
exten => 1,1,Queue(technique,Tt)

Don’t work

6

The first format is invalid. Square brackets are meta characters indicating optional parts of the command.

As the second is valid, you will need to:

  1. provide DTMF logging;

  2. give details of the configuration with respect to the handling of DTMF on the channel on which which you attempted to request the transfer;

  3. describe the exact user experience, and provide verbose logging of the call at the point where the transfer was attempted.

7

I do not know how to apply 1 and 2.

What the user does:
The user calls it is placed in a queue.
The agent wants to transfer it, it does **.
Nothing is happening.

8

  1. https://wiki.asterisk.org/wiki/display/AST/Logging+Configuration

  2. Typically it is dtmfmode on the Asterisk side. However, if you need to ask, you really need to read up more on basic configuration.

9

In my sip.conf in [general], i have :
dtmfmode=inband

And with your link, my “full” file log :
https://pastebin.com/nMDPC6bY

10

The log stops before you would expect to see the ‘*’.

The only DTMF digit I see is the 2 for the IVR.

You are using chan_sip, and don’t seem to have a native bridge.

I can’t tell which codec you are using, but note that using dtmfmode=inband is unusual and generally only works reliably with ulaw or alaw.

[2017-06-18 19:24:49.768] Asterisk 14.5.0 built by root @ vps364285.ovh.net on a x86_64 running Linux on 2017-06-08 19:52:48 UTC
[2017-06-18 19:24:49.768] VERBOSE[18657] config.c: Parsing '/etc/asterisk/logger.conf': Found
[2017-06-18 19:24:49.768] VERBOSE[18657] logger.c: Asterisk Queue Logger restarted
[2017-06-18 19:24:59.851] VERBOSE[7545][C-000066d0] netsock2.c: Using SIP RTP CoS mark 5
[2017-06-18 19:24:59.852] VERBOSE[18859][C-000066d0] pbx.c: Executing [0186954041@trunk-ovh:1] Goto("SIP/siptrunk.ovh.co.uk-0000003c", "ivr-voice,s,1") in new stack
[2017-06-18 19:24:59.852] VERBOSE[18859][C-000066d0] pbx_builtins.c: Goto (ivr-voice,s,1)
[2017-06-18 19:24:59.852] VERBOSE[18859][C-000066d0] pbx.c: Executing [s@ivr-voice:1] Answer("SIP/siptrunk.ovh.co.uk-0000003c", "") in new stack
[2017-06-18 19:25:00.353] VERBOSE[18859][C-000066d0] pbx.c: Executing [s@ivr-voice:2] Set("SIP/siptrunk.ovh.co.uk-0000003c", "TIMEOUT(response)=2") in new stack
[2017-06-18 19:25:00.353] VERBOSE[18859][C-000066d0] func_timeout.c: Response timeout set to 2.000
[2017-06-18 19:25:00.354] VERBOSE[18859][C-000066d0] pbx.c: Executing [s@ivr-voice:3] BackGround("SIP/siptrunk.ovh.co.uk-0000003c", "IVR-001") in new stack
[2017-06-18 19:25:00.354] WARNING[18859][C-000066d0] mp3/interface.c: Junk at the beginning of frame 49443303
[2017-06-18 19:25:00.354] VERBOSE[18859][C-000066d0] file.c: <SIP/siptrunk.ovh.co.uk-0000003c> Playing 'IVR-001.slin' (language 'en')
[2017-06-18 19:25:00.385] VERBOSE[18859][C-000066d0] res_rtp_asterisk.c: 0x7f19343d07d0 -- Probation passed - setting RTP source address to 91.121.129.169:36996
[2017-06-18 19:25:02.325] DTMF[18859][C-000066d0] channel.c: DTMF begin '2' received on SIP/siptrunk.ovh.co.uk-0000003c
[2017-06-18 19:25:02.325] DTMF[18859][C-000066d0] channel.c: DTMF begin ignored '2' on SIP/siptrunk.ovh.co.uk-0000003c
[2017-06-18 19:25:02.625] DTMF[18859][C-000066d0] channel.c: DTMF end '2' received on SIP/siptrunk.ovh.co.uk-0000003c, duration 280 ms
[2017-06-18 19:25:02.625] DTMF[18859][C-000066d0] channel.c: DTMF end passthrough '2' on SIP/siptrunk.ovh.co.uk-0000003c
[2017-06-18 19:25:02.626] VERBOSE[18859][C-000066d0] pbx.c: Executing [2@ivr-voice:1] GotoIfTime("SIP/siptrunk.ovh.co.uk-0000003c", "9:00-23:59,mon-sat,*,*?technique-ouvert,1,1") in new stack
[2017-06-18 19:25:02.626] VERBOSE[18859][C-000066d0] pbx.c: Executing [2@ivr-voice:2] Goto("SIP/siptrunk.ovh.co.uk-0000003c", "technique-fermer,1,1") in new stack
[2017-06-18 19:25:02.626] VERBOSE[18859][C-000066d0] pbx_builtins.c: Goto (technique-fermer,1,1)
[2017-06-18 19:25:02.626] VERBOSE[18859][C-000066d0] pbx.c: Executing [1@technique-fermer:1] Queue("SIP/siptrunk.ovh.co.uk-0000003c", "technique,Tt") in new stack
[2017-06-18 19:25:02.626] VERBOSE[18859][C-000066d0] res_musiconhold.c: Started music on hold, class 'technique', on channel 'SIP/siptrunk.ovh.co.uk-0000003c'
[2017-06-18 19:25:02.627] VERBOSE[18859][C-000066d0] netsock2.c: Using SIP RTP CoS mark 5
[2017-06-18 19:25:02.627] VERBOSE[18859][C-000066d0] app_queue.c: Called SIP/111
[2017-06-18 19:25:02.628] VERBOSE[18859][C-000066d0] netsock2.c: Using SIP RTP CoS mark 5
[2017-06-18 19:25:02.628] VERBOSE[18859][C-000066d0] app_queue.c: Called SIP/112
[2017-06-18 19:25:02.629] VERBOSE[18859][C-000066d0] app_queue.c: SIP/112-0000003e connected line has changed. Saving it until answer for SIP/siptrunk.ovh.co.uk-0000003c
[2017-06-18 19:25:02.629] VERBOSE[18859][C-000066d0] app_queue.c: SIP/111-0000003d connected line has changed. Saving it until answer for SIP/siptrunk.ovh.co.uk-0000003c
[2017-06-18 19:25:02.744] VERBOSE[18859][C-000066d0] app_queue.c: SIP/112-0000003e is busy
[2017-06-18 19:25:02.744] VERBOSE[18859][C-000066d0] app_queue.c: Nobody picked up in 0 ms
[2017-06-18 19:25:02.838] VERBOSE[18859][C-000066d0] app_queue.c: SIP/111-0000003d is ringing
[2017-06-18 19:25:03.916] VERBOSE[18859][C-000066d0] res_rtp_asterisk.c: 0x7f1940370550 -- Probation passed - setting RTP source address to 90.104.106.66:8000
[2017-06-18 19:25:03.935] VERBOSE[18859][C-000066d0] res_rtp_asterisk.c: 0x7f1940370550 -- Probation passed - setting RTP source address to 90.104.106.66:8000
[2017-06-18 19:25:03.943] VERBOSE[18859][C-000066d0] app_queue.c: SIP/111-0000003d connected line has changed. Saving it until answer for SIP/siptrunk.ovh.co.uk-0000003c
[2017-06-18 19:25:03.944] VERBOSE[18859][C-000066d0] app_queue.c: SIP/111-0000003d answered SIP/siptrunk.ovh.co.uk-0000003c
[2017-06-18 19:25:03.944] VERBOSE[18859][C-000066d0] res_musiconhold.c: Stopped music on hold on SIP/siptrunk.ovh.co.uk-0000003c
[2017-06-18 19:25:03.944] VERBOSE[18862][C-000066d0] bridge_channel.c: Channel SIP/111-0000003d joined 'simple_bridge' basic-bridge <992a486a-2860-40bf-80e1-3086566dd9dc>
[2017-06-18 19:25:03.945] VERBOSE[18859][C-000066d0] bridge_channel.c: Channel SIP/siptrunk.ovh.co.uk-0000003c joined 'simple_bridge' basic-bridge <992a486a-2860-40bf-80e1-3086566dd9dc>
[2017-06-18 19:25:03.956] VERBOSE[18862][C-000066d0] res_rtp_asterisk.c: 0x7f1940370550 -- Probation passed - setting RTP source address to 90.104.106.66:8000
[2017-06-18 19:25:03.976] VERBOSE[18862][C-000066d0] res_rtp_asterisk.c: 0x7f1940370550 -- Probation passed - setting RTP source address to 90.104.106.66:8000
[2017-06-18 19:25:09.503] VERBOSE[18859][C-000066d0] bridge_channel.c: Channel SIP/siptrunk.ovh.co.uk-0000003c left 'simple_bridge' basic-bridge <992a486a-2860-40bf-80e1-3086566dd9dc>
[2017-06-18 19:25:09.503] VERBOSE[18862][C-000066d0] bridge_channel.c: Channel SIP/111-0000003d left 'simple_bridge' basic-bridge <992a486a-2860-40bf-80e1-3086566dd9dc>
[2017-06-18 19:25:09.503] VERBOSE[18859][C-000066d0] pbx.c: Spawn extension (technique-fermer, 1, 1) exited non-zero on 'SIP/siptrunk.ovh.co.uk-0000003c'
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