When clicking on transfer in the user interface, there is a sound of DTMF signals in the users conversation. I assume this must be matched in the dialplan? What signales does it send? And how can I match this in my dialplan?

I am using asterisk 1.6.2.2, a pretty stripped down version, so no trixbox or other GUI.

Could you please refer to the specific bug?

Using trunk is not an option in production, but upgrading to 1.6.2.3 might be, if the issue is fixed.

Hi,
I have the Problem too. But it wasn't fixed after updating. Now I use asterisk 1.6.2.11 with FreePBX.

fop2.cfg

  1 [general]
  2 ; AMI definitions
  3 manager_host=localhost
  4 manager_port=5038
  5 manager_user=admin
  6 manager_secret=*****
  7 ;event_mask=call,agent
  8
  9 ; Daemon definitios
 10 ;listen_port      = 4445
 11 ;restrict_host    = www.asternic.org
 12 ;web_dir          = /var/www/html/operator/fop2
 13
 14 ; Global Config
 15 language           = en
 16 poll_interval      = 86400
 17 poll_voicemail     = 1
 18 monitor_ipaddress  = 0
 19
 20 ; Force blind transfer on asterisk 1.6
 21 blind_transfer     = 0
 22
 23 ; Force supervised transfer on asterisk 1.4
 24 ; requires the atxfer manager backport patch
 25 supervised_transfer = 1
 26
 27 ; Force delimiter for asterisk applications
 28 force_parameter_delimiter = ","
 29
 30 ; When adding or removing members to a queue, fop2 will default to
 31 ; AddQueueMember/RemoveQueueMember commands. If you set use_agentlogin
 32 ; to 1, together with the QueueChannel in a button definition set to
 33 ; an Agent number it will use AgentCallbackLogin and Agentlogoff instead.
 34 ;
 35 ; use_agentlogin = 0
 36
 37
 38 ; Master Password that overrides any individual one
 39 ;master_key = 5678
 40
 41 ; Filename to use when start monitoring, you can use ${UNIQUEID},
 42 ; ${ORIG_EXTENSION}, ${DEST_EXTENSION}
 43 ; and date formats %Y %m %d to construct the filename.
 44 ;
 45 ; Settings for modifying the recording filename
 46 ; Available variables are:
 47 ; ${UNIQUEID} = Unique Id of the call
 48 ; ${TIMESTAMP} = Unix Timestamp when the recording was initiated
 49 ; ${DEST_EXTENSION} = Target extenstion being monitored
 50 ; ${ORIG_EXTENSION} = Extension/User that started the recording (not
 51 ;                     the other leg)
 52 ; Date variables:
 53 ; %Y 4 digits year
 54 ; %y 2 digits year
 55 ; %m 2 digits month
 56 ; %d 2 digits day
 57 ; %h 2 digits hour
 58 ; %i 2 digits minute
 59 ; %s 2 digits seconds
 60
 61 monitor_filename=g${DEST_EXTENSION}-${UNIQUEID}
 62 monitor_format=wav
 63 monitor_mix=true
 64
 65 ; --- SAMPLE GROUPS ---
 66 ;group=queues:QUEUE/100,QUEUE/101
 67 ;group=deptA:SIP/100,SIP/101,SIP/102
 68 ; --- END SAMPLE ---
 69
 70 ; --- SAMPLE USER LIST ---
 71 ; format: user= EXTENSION : SECRET : PERMISSIONS : GROUPS
 72 ; You can enumerate several permissions and groups separated by comma
 73 ; available permissions:  'all', 'dial', 'hangup', 'meetme', 'pickup',
 74 ;                         'record', 'spy', 'transfer', 'whisper',
 75 ;                         'queuemanager', 'queueagent', 'phonebook'
 76 ;
 77 ;user=620:1234:all:queues
 78 ;user=621:1234:dial,transfer,pickup:deptA
 79 ;user=622:1234:all
 80 ;user=623:1234:meetme,pickup
 81 ;buttonfile=buttons.cfg
 82 ; ------ END SAMPLE ------
 83
 84 ; This line is NOT commented, it executes
 85 ; the autoconfig configuration for FreePBX
 86 #exec autoconfig-users-freepbx.sh

Attended transfer (atxfer) is run on asterisk too. Someone can help me?

I tried 1.6.2.3-rc2, 1.6.2.5, 1.6.2.9, 1.6.2.11 and trunk and no Version works. Could you tell me a working version or which asterisk configuration files could change?

Hi! I have the same problem with asterisk 13.8
When I call from panel to extension and try to transfer via 'Dial' window, I hear DTMF. But if I try to transfer via 'transfer' button, all works fine.

The issue happens only if call originated from me and I try to transfer by panel. If I transfer with phone (yealink) transfer works.
Is it asterisk bug?

PS. Sorry for bad English(

UPD. Commands in log looks the same. It means asterisk issue.

• > Action: Atxfer
• > Channel: SIP/10-00000148
• > Exten: 20#
• > Context: incom
• > Priority: 1

• > Action: Atxfer
• > Channel: SIP/10-00000149
• > Exten: 20#
• > Context: incom
• > Priority: 1