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

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

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(

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?

Just for starters: https://issues.asterisk.org/view.php?id=16816

Best regards,

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