Problems with transfer and transfer to Vmail

6string

Thanks for your response, here is what I have, Pressing transfer to voicemail
I put the XXX in the remote address

127.0.0.1 <- Event: PeerStatus
127.0.0.1 <- Privilege: system,all
127.0.0.1 <- Peer: SIP/10
127.0.0.1 <- PeerStatus: Registered
127.0.0.1 <- Server: 0

XXX.64.90.134 <= <msg data="7|tovoicemail|3|70de7017deaa2b01190cf4fcd1c733ce" />

127.0.0.1 -> Action: Redirect
127.0.0.1 -> Channel: Local/FMPR-18@from-internal-feb6
127.0.0.1 -> Exten: *12
127.0.0.1 -> Context: default
127.0.0.1 -> Priority: 1
127.0.0.1 -> Async: True

127.0.0.1 <- Response: Error
127.0.0.1 <- Message: Channel does not exist: Local/FMPR-18@from-internal-feb6
127.0.0.1 <- Server: 0

Response: Error
Message: Channel does not exist: Local/FMPR-18@from-internal-feb6
Server: 0
XXX.64.90.134 <= <msg data="1|ping||" />
XXX.64.90.134 => { 'btn': '0', 'cmd': 'pong', 'data': '0', 'slot': '' }

127.0.0.1 <- Event: Registry
127.0.0.1 <- Privilege: system,all
127.0.0.1 <- ChannelDriver: SIP
127.0.0.1 <- Domain: inbound17.vitelity.net
127.0.0.1 <- Status: Registered
127.0.0.1 <- Server: 0
127.0.0.1 -> Action: Redirect
127.0.0.1 -> Channel: Local/FMPR-18@from-internal-feb6
127.0.0.1 -> Exten: *12
127.0.0.1 -> Context: default
127.0.0.1 -> Priority: 1
127.0.0.1 -> Async: True

127.0.0.1 <- Response: Error
127.0.0.1 <- Message: Channel does not exist: Local/FMPR-18@from-internal-feb6
127.0.0.1 <- Server: 0

Response: Error
Message: Channel does not exist: Local/FMPR-18@from-internal-feb6
Server: 0

XXX.64.90.134 <= <msg data="1|ping||" />
XXX.64.90.134 => { 'btn': '0', 'cmd': 'pong', 'data': '0', 'slot': '' }

nicolas

Ok. You are the 3rd person reporting this issue. However I was unable to reproduce the problem on my test system. I notice you are using followme, and that is a potential factor as it uses Local channels. However in my system I can transfer calls even if I have follow me enabled.

I must be able to reproduce the issue in order to work on a fix. Can you tell me exactly what versions are you running of everything? FreePbx version, Asterisk version. And how is the call being made and your extensions setup, including followme?

Do you call directly from one extension to the other? Do you call a did number, from there to an ivr and then your extension? Do you use ringgroups? Perhaps a combination of ring group and follow me?

Best regards,

6string

OK, thanks for the response
I am using Asterisk 1.4.24
Freepbx 2.5.1.5
Elastix 1.5.2-2.3
The calls on the previous trace go like this
A sip DID from Vitelity to an IVR
I dialed ext 18 from the IVR then answered the phone on ext 18
The call shows up with caller ID on the FOP2 panel
I click on the destination Ext 12 then click the Transfer to VMail

I have also tried ext to ext calls with the same result
Also I am not using ring groups

From reading your response I disabled the follow me settings on a few exts and tested
That appears to be the problem, the transfer and transfer to vmail works once I disabled the follow me
I will test further to see if I can find any other information

Thanks

6string

Just a little more information I found
The follow me on the destination exts makes no difference
However if follow me is defined (and has an additional ext or number defined) on the ext that is using FOP2 then it will fail
on my system when trying to transfer or transfer to vmail

This will solve most of my problem,
Thanks for all your help !!

6string

Sorry to be a pest
I have the transfer and transfer to a station working
But now I cannot transfer to any Queue or to Park
It just hangs up
Here is the trace of a Sip DID being transferred to Queue 50
XXX.XXX.80.146 <= <msg data="1|atxfer|23|f3d868c14c77785f89883320076e82ae" />

127.0.0.1 -> Action: Redirect
127.0.0.1 -> Channel: SIP/dannylarsen-b7d27d80
127.0.0.1 -> Exten: 50
127.0.0.1 -> Context:
127.0.0.1 -> Priority: 1
127.0.0.1 -> Async: True

127.0.0.1 <- Response: Success
127.0.0.1 <- Message: Redirect successful
127.0.0.1 <- Server: 0

Response: Success
Message: Redirect successful
Server: 0

127.0.0.1 <- Event: Unlink
127.0.0.1 <- Privilege: call,all
127.0.0.1 <- Channel1: SIP/dannylarsen-b7d27d80
127.0.0.1 <- Channel2: SIP/10-b7b01d78
127.0.0.1 <- Uniqueid1: 1267564614.458
127.0.0.1 <- Uniqueid2: 1267564615.461
127.0.0.1 <- CallerID1: 8012053201
127.0.0.1 <- CallerID2: 10
127.0.0.1 <- Server: 0

127.0.0.1 <- Event: ExtensionStatus
127.0.0.1 <- Privilege: call,all
127.0.0.1 <- Exten: 10
127.0.0.1 <- Context: ext-local
127.0.0.1 <- Status: 0
127.0.0.1 <- Server: 0

127.0.0.1 <- Event: Hangup
127.0.0.1 <- Privilege: call,all
127.0.0.1 <- Channel: SIP/10-b7b01d78
127.0.0.1 <- Uniqueid: 1267564615.461
127.0.0.1 <- Cause: 16
127.0.0.1 <- Cause-txt: Normal Clearing
127.0.0.1 <- Server: 0

127.0.0.1 <- Event: QueueMemberStatus
127.0.0.1 <- Privilege: agent,all
127.0.0.1 <- Queue: 50
127.0.0.1 <- Location: SIP/10
127.0.0.1 <- MemberName: SIP/10
127.0.0.1 <- Membership: dynamic
127.0.0.1 <- Penalty: 0
127.0.0.1 <- CallsTaken: 3
127.0.0.1 <- LastCall: 1267561947
127.0.0.1 <- Status: 1
127.0.0.1 <- Paused: 0
127.0.0.1 <- Server: 0

127.0.0.1 <- Event: Newexten
127.0.0.1 <- Privilege: call,all
127.0.0.1 <- Channel: SIP/dannylarsen-b7d27d80
127.0.0.1 <- Context: macro-dial
127.0.0.1 <- Extension: h
127.0.0.1 <- Priority: 1
127.0.0.1 <- Application: Macro
127.0.0.1 <- AppData: hangupcall
127.0.0.1 <- Uniqueid: 1267564614.458
127.0.0.1 <- Server: 0

127.0.0.1 <- Event: Newexten
127.0.0.1 <- Privilege: call,all
127.0.0.1 <- Channel: SIP/dannylarsen-b7d27d80
127.0.0.1 <- Context: macro-hangupcall
127.0.0.1 <- Extension: s
127.0.0.1 <- Priority: 1
127.0.0.1 <- Application: ResetCDR
127.0.0.1 <- AppData: vw
127.0.0.1 <- Uniqueid: 1267564614.458
127.0.0.1 <- Server: 0

127.0.0.1 <- Event: Newexten
127.0.0.1 <- Privilege: call,all
127.0.0.1 <- Channel: SIP/dannylarsen-b7d27d80
127.0.0.1 <- Context: macro-hangupcall
127.0.0.1 <- Extension: s
127.0.0.1 <- Priority: 2
127.0.0.1 <- Application: NoCDR
127.0.0.1 <- AppData:
127.0.0.1 <- Uniqueid: 1267564614.458
127.0.0.1 <- Server: 0

127.0.0.1 <- Event: Newexten
127.0.0.1 <- Privilege: call,all
127.0.0.1 <- Channel: SIP/dannylarsen-b7d27d80
127.0.0.1 <- Context: macro-hangupcall
127.0.0.1 <- Extension: s
127.0.0.1 <- Priority: 3
127.0.0.1 <- Application: GotoIf
127.0.0.1 <- AppData: 1?skiprg
127.0.0.1 <- Uniqueid: 1267564614.458
127.0.0.1 <- Server: 0

127.0.0.1 <- Event: Newexten
127.0.0.1 <- Privilege: call,all
127.0.0.1 <- Channel: SIP/dannylarsen-b7d27d80
127.0.0.1 <- Context: macro-hangupcall
127.0.0.1 <- Extension: s
127.0.0.1 <- Priority: 6
127.0.0.1 <- Application: GotoIf
127.0.0.1 <- AppData: 0?skipblkvm
127.0.0.1 <- Uniqueid: 1267564614.458
127.0.0.1 <- Server: 0

127.0.0.1 <- Event: Newexten
127.0.0.1 <- Privilege: call,all
127.0.0.1 <- Channel: SIP/dannylarsen-b7d27d80
127.0.0.1 <- Context: macro-hangupcall
127.0.0.1 <- Extension: s
127.0.0.1 <- Priority: 7
127.0.0.1 <- Application: NoOp
127.0.0.1 <- AppData: Cleaning Up Block VM Flag: BLKVM/10/SIP/dannylarsen-b7d27d80
127.0.0.1 <- Uniqueid: 1267564614.458
127.0.0.1 <- Server: 0

127.0.0.1 <- Event: Newexten
127.0.0.1 <- Privilege: call,all
127.0.0.1 <- Channel: SIP/dannylarsen-b7d27d80
127.0.0.1 <- Context: macro-hangupcall
127.0.0.1 <- Extension: s
127.0.0.1 <- Priority: 8
127.0.0.1 <- Application: DBdel
127.0.0.1 <- AppData: BLKVM/10/SIP/dannylarsen-b7d27d80
127.0.0.1 <- Uniqueid: 1267564614.458
127.0.0.1 <- Server: 0

127.0.0.1 <- Event: Newexten
127.0.0.1 <- Privilege: call,all
127.0.0.1 <- Channel: SIP/dannylarsen-b7d27d80
127.0.0.1 <- Context: macro-hangupcall
127.0.0.1 <- Extension: s
127.0.0.1 <- Priority: 9
127.0.0.1 <- Application: GotoIf
127.0.0.1 <- AppData: 1?theend
127.0.0.1 <- Uniqueid: 1267564614.458
127.0.0.1 <- Server: 0

127.0.0.1 <- Event: Newexten
127.0.0.1 <- Privilege: call,all
127.0.0.1 <- Channel: SIP/dannylarsen-b7d27d80
127.0.0.1 <- Context: macro-hangupcall
127.0.0.1 <- Extension: s
127.0.0.1 <- Priority: 11
127.0.0.1 <- Application: Hangup
127.0.0.1 <- AppData:
127.0.0.1 <- Uniqueid: 1267564614.458
127.0.0.1 <- Server: 0

127.0.0.1 <- Event: Hangup
127.0.0.1 <- Privilege: call,all
127.0.0.1 <- Channel: SIP/dannylarsen-b7d27d80
127.0.0.1 <- Uniqueid: 1267564614.458
127.0.0.1 <- Cause: 16
127.0.0.1 <- Cause-txt: Normal Clearing
127.0.0.1 <- Server: 0

nicolas

XXX.XXX.80.146 <= <msg data="1|atxfer|23|f3d868c14c77785f89883320076e82ae" />

127.0.0.1 -> Action: Redirect
127.0.0.1 -> Channel: SIP/dannylarsen-b7d27d80
127.0.0.1 -> Exten: 50
127.0.0.1 ->** Context:**
127.0.0.1 -> Priority: 1
127.0.0.1 -> Async: True

Context is missing. Do you use manual buttons configuration or automatic? How does your queue button config looks like?

6string

I am using the automatic config
Here is my Fop2.cfg
[general]
; AMI definitions
manager_host=127.0.0.1
manager_port=5038
manager_user=fop2
manager_secret=fop2secret
;event_mask=call,agent

; Daemon definitios
; listen_port = 4445
;restrict_host = www.asternic.org
;web_dir = /var/www/html/operator/fop2

; Global Config
language = en
poll_interval = 86400
poll_voicemail = 1
;monitor_ipaddress = 0

; Force blind transfer on asterisk 1.6
; blind_transfer = 1

; Force supervised transfer on asterisk 1.4
; requires the atxfer manager backport patch
; supervised_transfer = 0

; Force delimiter for asterisk applications
; force_parameter_delimiter = ","

; When adding or removing members to a queue, fop2 will default to
; AddQueueMember/RemoveQueueMember commands. If you set use_agentlogin
; to 1, together with the QueueChannel in a button definition set to
; an Agent number it will use AgentCallbackLogin and Agentlogoff instead.
;
use_agentlogin = 0

; Filename to use when start monitoring, you can use ${UNIQUEID},
; ${ORIG_EXTENSION}, ${DEST_EXTENSION}
; and date formats %Y %m %d to construct the filename.

; Master Password that overrides any individual one
master_key = 6string

; Settings for modifying the recording filename
; Available variables are:
; ${UNIQUEID} = Unique Id of the call
; ${TIMESTAMP} = Unix Timestamp when the recording was initiated
; ${DEST_EXTENSION} = Target extenstion being monitored
; ${ORIG_EXTENSION} = Extension/User that started the recording (not
; the other leg)
; Date variables:
; %Y 4 digits year
; %y 2 digits year
; %m 2 digits month
; %d 2 digits day
; %h 2 digits hour
; %i 2 digits minute
; %s 2 digits seconds

monitor_filename=g${DEST_EXTENSION}-${UNIQUEID}
monitor_format=wav
monitor_mix=true

nicolas

Hi,

Run the autoconfig-buttons-freepbx.sh script at the linux cli and look for the queue button configuration and paste it here..

jmituzas

Hi,

If you use asterisk 1.4 unpatched, you must set supervised_transfer=0 and leave it that way.

Will that work for asterisk 1.6 as well ?

and place within fop2.cfg file, correct?

nicolas

supervised_transfer=0 is for disabling the use of the atxfer manager command in Asterisk 1.4

blind_transfer=1 is for forcing standard redirects (and disabling atxfer) on asterisk 1.6.2

FOP2 Community Forum

Problems with transfer and transfer to Vmail