[asterisk-bugs] [JIRA] (ASTERISK-27857) Attended Transfer: Attended transfer has failed if using AMI terminal to send.

[Benjamin Keith Ford (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-27857?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=243441#comment-243441 ] 

Benjamin Keith Ford commented on ASTERISK-27857:
------------------------------------------------

[~hiepcm], I can successfully make an attended transfer following this scenario. The steps I took are:
# Make an external call to phone A (with "t" in Dial options)
# Pick up phone A
# Do the attended transfer AMI action
# Phone B will ring - answer it
# Hang up phone A
# The outside phone is now bridged with phone B
Are these the exact steps you took? Did phone B ever ring when doing the Atxfer action via AMI?

> Attended Transfer: Attended transfer has failed if using AMI terminal to send.
> ------------------------------------------------------------------------------
>
>                 Key: ASTERISK-27857
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-27857
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Core/Bridging
>    Affects Versions: 13.21.0
>         Environment: Asterisk v13.21.0, AMI Terminal, Telephone.
>            Reporter: Cao Minh Hiep
>            Assignee: Unassigned
>         Attachments: attended transfer failed with AMI_13.21.0.txt, attended transfer failed with AMI.txt
>
>
> We have made an attended transfer with the following scenario:
> 1.Two phones(A and B) in one work-group.
> 2. Make a call to one of them(phone A) from an outside phone.
> 3. On AMI interface(used Tera Term terminal) to make an attended transfer from phone A to other(phone B).
> We could not make an attended transfer from A phone to B phone.
> It outputs a beep sound also.
> When we tried to investigate the causing of this problem.
> We found the difference logs between bug log and a normal log as below:
> =>*2 201 (*2: attended transfer, 201: Extention)
> We do that by the following AMI command:
> Action: Atxfer
> ActionID: 1
> Channel: SIP/100002-0000008b 
> Exten: 201
> We found the logs of "Channel Local/20 at a_context_01-0000006e"
> instead of "Channel Local/201 at a_context_01-0000006e".
> We also found there are different process ID in progress of  "*2 201"
> It's same process ID(30450) with "*2 20" and It turns to 3584 ID with 1.
> Note: In the normal attended transfer log we found the same process ID for "*2201".
> "DTMF[30450][C-0000002d]: channel.c:3972 __ast_read: DTMF end '*' received on SIP/100002-0000008b, duration 0 ms
> DTMF[30450][C-0000002d]: channel.c:3999 __ast_read: DTMF begin emulation of '*' with duration 100 queued on SIP/100002-0000008b
> DTMF[30450][C-0000002d]: channel.c:4092 __ast_read: DTMF end emulation of '*' queued on SIP/100002-0000008b
> DTMF[30450][C-0000002d]: channel.c:3972 __ast_read: DTMF end '2' received on SIP/100002-0000008b, duration 0 ms
> DTMF[30450][C-0000002d]: channel.c:3999 __ast_read: DTMF begin emulation of '2' with duration 100 queued on SIP/100002-0000008b
> DTMF[30450][C-0000002d]: channel.c:4092 __ast_read: DTMF end emulation of '2' queued on SIP/100002-0000008b
>     -- <SIP/100002-0000008b> Playing 'pbx-transfer.gsm' (language 'ja')
> DTMF[30450][C-0000002d]: channel.c:3972 __ast_read: DTMF end '2' received on SIP/100002-0000008b, duration 0 ms
> DTMF[30450][C-0000002d]: channel.c:4031 __ast_read: DTMF end accepted without begin '2' on SIP/100002-0000008b
> DTMF[30450][C-0000002d]: channel.c:4042 __ast_read: DTMF end passthrough '2' on SIP/100002-0000008b
> DTMF[30450][C-0000002d]: channel.c:3972 __ast_read: DTMF end '0' received on SIP/100002-0000008b, duration 0 ms
> [May 16 11:57:37] DTMF[30450][C-0000002d]: channel.c:4031 __ast_read: DTMF end accepted without begin '0' on SIP/100002-0000008b
> DTMF[30450][C-0000002d]: channel.c:4042 __ast_read: DTMF end passthrough '0' on SIP/100002-0000008b
> DTMF[3584][C-0000002d]: channel.c:3972 __ast_read: DTMF end '1' received on SIP/100002-0000008b, duration 0 ms
> DTMF[3584][C-0000002d]: channel.c:4031 __ast_read: DTMF end accepted without begin '1' on SIP/100002-0000008b
> DTMF[3584][C-0000002d]: channel.c:4042 __ast_read: DTMF end passthrough '1' on SIP/100002-0000008b
> "
> Please have a look at attached test logs file.
> And could you please show us the causing of the problem and fixed patch for it?



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