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

[Cao Minh Hiep (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-27857?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=243504#comment-243504 ] 

Cao Minh Hiep commented on ASTERISK-27857:
------------------------------------------

Hi Richard Mudgett
Thanks for your feedback.

Here is the information of the timing modules we have installed and in use:

On our Working Server2 which the issue occurs, we have:
"CLI> module show like res_timing
Module                         Description                              Use Count  Status      Support Level
res_timing_dahdi.so            DAHDI Timing Interface                   0          Running              core
res_timing_pthread.so          pthread Timing Interface                 0          Running          extended
res_timing_timerfd.so          Timerfd Timing Interface                 1          Running              core
3 modules loaded"

On our Working Server1 which the attended transfer works well aslo have the same above.

Please let us know if you need more information about the issue.
Thank you


> 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:
> {noformat}
> Action: Atxfer
> ActionID: 1
> Channel: SIP/100002-0000008b 
> Exten: 201
> {noformat}
> 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".
> {noformat}
> 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
> {noformat}
> 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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