[asterisk-bugs] [JIRA] (ASTERISK-27857) Attended Transfer: Attended transfer has failed if using AMI terminal to send.
Joshua Colp (JIRA)
noreply at issues.asterisk.org
Wed May 16 00:48:55 CDT 2018
[ https://issues.asterisk.org/jira/browse/ASTERISK-27857?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=243418#comment-243418 ]
Joshua Colp commented on ASTERISK-27857:
----------------------------------------
It appears the bug you have submitted is against a rather old version of a supported branch of Asterisk. There have been many issues fixed between the version you are using and the current version of your branch. Please test with the latest version in your Asterisk branch and report whether the issue persists.
Please see the Asterisk Versions [1] wiki page for info on which versions of Asterisk are supported.
[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions
> 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: . I did not set the category correctly.
> Affects Versions: 13.12.0
> Environment: Asterisk v13.0.0, AMI Terminal, Telephone.
> Reporter: Cao Minh Hiep
> Attachments: 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?
--
This message was sent by Atlassian JIRA
(v6.2#6252)
More information about the asterisk-bugs
mailing list