Transfer anywhere not working...
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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: 1387901217.0858497***********: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: True127.0.0.1 <- Response: Success
127.0.0.1 <- Message: Atxfer successfully queued127.0.0.1 <- Event: VarSet
127.0.0.1 <- Privilege: dialplan,all
127.0.0.1 <- Channel: SIP/23264-0000015f
127.0.0.1 <- Variable: TRANSFER_CONTEXT
127.0.0.1 <- Value: outgoing
127.0.0.1 <- Uniqueid: 13****.352Flash 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 -
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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: True127.0.0.1 <- Response: Success
127.0.0.1 <- Message: Atxfer successfully queued127.0.0.1 <- 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: 14664.1130TIMER asterisk_ami_connect
** Asterisk Manager logged in localhost for 415 seconds
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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.
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