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

Thu May 17 23:27:55 CDT 2018

    [ https://issues.asterisk.org/jira/browse/ASTERISK-27857?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=243444#comment-243444 ] 

Cao Minh Hiep edited comment on ASTERISK-27857 at 5/17/18 11:27 PM:
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Hi Benjamin Keith Ford
Thanks for your feedback.

Our scenario is the following (quite same to you).
1. Make an external call to phone A (phone A and phone B are in the same word-group) 
2. Pick up phone A -> Answer it.
3. Do the attended transfer AMI action from Tera Term Terminal
4. Phone B did not ring - I heard a beep sound at phone A
5. Hang up phone A
6. The outside phone was not bridged with phone B.

We also successfully with the attended transfer AMI action on another test environment.
Note: We built asterisk and run it in the different servers.
Here is our test environment:
・Asterisk Build Server
　　- Virtual machine on a Cloud service:
　　　　CentOS Linux release 7.3.1611 (Core)
　　-Disable BUILD_NATIVE by menu-select.

・Executive Server: (Attended transfer works well on this server)
　　- Virtual machine on an other Cloud service:
　　　　CentOS Linux release 7.3.1611 (Core)
・Non-Executive Server: (The issue occurs on this server)
　　- VMWare Esxi 6.5
　　　　CentOS Linux release 7.3.1611 (Core)
OS are same for all. Basically we used packages that has installed in CentOS7.3 ISO image.

We also can successfully make an attended transfer on real desktop phone without AMI action like this:
1. Make an external call to phone A (phone A and phone B are in the same word-group) 
2. Pick up phone A -> Answer it.
3. Press *2201 on phone A for doing the attended transfer to phone B(201)
4. Phone B will ring - Answer it.
5. Hang up phone A.
6. The outside phone is now bridged with phone B -Answer it.
7. Hang up phone B and external phone.

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

was (Author: hiepcm):
Hi Benjamin Keith Ford
Thanks for your feedback.

Our scenario is the following (quite same to you).
1. Make an external call to phone A (phone A and phone B are in the same word-group) 
2. Pick up phone A -> Answer it.
3. Do the attended transfer AMI action from Tera Term Terminal
4. Phone B did not ring - I heard a beep sound at phone A
5. Hang up phone A
6. The outside phone was not bridged with phone B.

On our test environments, We also sometimes successfully with the attended transfer AMI action.
Note: We built asterisk and run it in the different servers.

We also can successfully make an attended transfer on real desktop phone without AMI action like this:
1. Make an external call to phone A (phone A and phone B are in the same word-group) 
2. Pick up phone A -> Answer it.
3. Press *2201 on phone A for doing the attended transfer to phone B(201)
4. Phone B will ring - Answer it.
5. Hang up phone A.
6. The outside phone is now bridged with phone B -Answer it.
7. Hang up phone B and external phone.

Please let us know if you need more information about 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:
> 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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