Transfer anywhere not working...

10 years ago

bahunt
24 Dec 2013

I have the latest version with Centos using Asterisk. I am unable to transfer to another ext. or external number when I try the number dials and nothing happens. The dial feature doesn't work either any help would be greatly appreciated.
127.0.0.1 -> Action: Ping
127.0.0.1 <- Response: Success
127.0.0.1 <- Ping: Pong
127.0.0.1 <- Timestamp: 1387901217.085849
7***********:1203 <= <msg data="1_11|dragatxfer|12|9e2e95717bce335f679554518595fe70" />
boton por canal SIP/23******** esta blessed
127.0.0.1 -> Action: Atxfer
127.0.0.1 -> Channel: SIP/2*****3264-0000015f
127.0.0.1 -> Exten: 2345
127.0.0.1 -> Context: outgoing
127.0.0.1 -> Priority: 1
127.0.0.1 -> Async: True
127.0.0.1 <- Response: Success
127.0.0.1 <- Message: Atxfer successfully queued
127.0.0.1 <- Event: VarSet
127.0.0.1 <- Privilege: dialplan,all
127.0.0.1 <- Channel: SIP/2******3264-0000015f
127.0.0.1 <- Variable: TRANSFER_CONTEXT
127.0.0.1 <- Value: outgoing
127.0.0.1 <- Uniqueid: 13**********.352
Flash clients connected: 1
-----------------------------------------------------------------
Client ***********:****, user: 2**********@GENERAL, type: websockets
-----------------------------------------------------------------
127.0.0.1 <- Event: Registry
127.0.0.1 <- Privilege: system,all
127.0.0.1 <- ChannelType: SIP
127.0.0.1 <- Username: ****
127.0.0.1 <- Domain: *****
127.0.0.1
admin
26 Dec 2013 Administrator

Does 2345@outgoing exists in your dialplan? It seems to me that you have an incorrect configuration in your button contexts, or similar to that.
asterisk -rx "dialplan show 2345@outgoing"
Best regards,
bahunt
30 Dec 2013

2345 does exist in my dial plan. The number is dialed when I try to transfer but nothing happens....
admin
30 Dec 2013 Administrator

Then you are missing the tT options on your original dialplan dial command. Or you do not have the atxfer feature enabled in /etc/asterisk/features.conf
bahunt
30 Dec 2013

I added tT to the extension and made sure atxfer was enabled and still get the same response...
ATXFER using native atxfer in AMI for SIP/2-00007
127.0.0.1 -> Action: Atxfer
127.0.0.1 -> Channel: SIP/2-00007
127.0.0.1 -> Exten: 23
127.0.0.1 -> Context: C21LD
127.0.0.1 -> Priority: 1
127.0.0.1 -> Async: True
127.0.0.1 <- Response: Success
127.0.0.1 <- Message: Atxfer successfully queued
127.0.0.1 <- Event: VarSet
127.0.0.1 <- Privilege: dialplan,all
127.0.0.1 <- SequenceNumber: 2670
127.0.0.1 <- File: pbx.c
127.0.0.1 <- Line: 11404
127.0.0.1 <- Func: pbx_builtin_setvar_helper
127.0.0.1 <- Channel: SIP/2-0007
127.0.0.1 <- Variable: TRANSFER_CONTEXT
127.0.0.1 <- Value: C21LD
127.0.0.1 <- Uniqueid: 14664.1130
TIMER asterisk_ami_connect
** Asterisk Manager logged in localhost for 415 seconds
admin
30 Dec 2013 Administrator

FOP2 is sending the correct command (it seems, as I do not know your dialplan). If the atxfer is not working, you will have to look at your asterisk config. Your problem it is *not* fop2 related.
Best regards,
admin
30 Dec 2013 Administrator

Confirmed via live help: The dialplan was NOT including the tT options in the Dial command, making the attendant transfer not to work.
So, for other users that might have issues, please check and re check many times that you are using the correct dial options and that the atxfer feature is enabled also.
This was NOT a fop2 issue, but an asterisk misconfiguration.
bahunt
30 Dec 2013

thanks for the great support!