[asterisk-bugs] [JIRA] (ASTERISK-26848) Problem with call return when atxfer

[Joshua Colp (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-26848?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=235636#comment-235636 ] 

Joshua Colp commented on ASTERISK-26848:
----------------------------------------

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





> Problem with call return when atxfer
> ------------------------------------
>
>                 Key: ASTERISK-26848
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-26848
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Features
>    Affects Versions: 13.4.0
>         Environment: Linux version 2.6.32-5-686 (Debian 2.6.32-48squeeze4)
>            Reporter: Vladislav Krivoshchekov
>
> Hello!
> We have a problem with the return of the call during the transfer.
> Example: subscriber A calls subscriber B and B makes a transfer to C. If the C-subscriber does not answer, B presses "*" to end the call to C and return to the conversation with A.
> 1) A calls B 
> 2) B - attended transfer to C
> 3) С no answer
> 4) B presses * to end the call
> 5) A listens to MOH, B listens to silent, call to C continues.
> asterisk -rx 'features show': 
> Builtin Feature           Default Current
> ---------------                -------    -------
> Pickup                        *8         *8     
> Blind Transfer              #          #      
> Attended Transfer                   *7     
> One Touch Monitor                        
> Disconnect Call            *           *      
> Park Call                                
> One Touch MixMonitor                     
> Dynamic Feature           Default Current
> ---------------           ------- -------
> preconfigured_call_forward no def  *57    
> Feature Groups:
> ---------------
> (none)
> logs:
> [Mar  7 17:41:45] DEBUG[20052][C-00001969] res_rtp_asterisk.c: Creating BEGIN DTMF Frame: 42 (*), at 10.9.117.82:59606
> [Mar  7 17:41:45] DTMF[20052][C-00001969] channel.c: DTMF begin '*' received on SIP/0080265-00006e67
> [Mar  7 17:41:45] DTMF[20052][C-00001969] channel.c: DTMF begin passthrough '*' on SIP/0080265-00006e67
> [Mar  7 17:41:45] DEBUG[20052][C-00001969] res_rtp_asterisk.c: Creating END DTMF Frame: 42 (*), at 10.9.117.82:59606
> [Mar  7 17:41:45] DTMF[20052][C-00001969] channel.c: DTMF end '*' received on SIP/0080265-00006e67, duration 120 ms
> [Mar  7 17:41:45] DTMF[20052][C-00001969] channel.c: DTMF end accepted with begin '*' on SIP/0080265-00006e67
> [Mar  7 17:41:45] DTMF[20052][C-00001969] channel.c: DTMF end passthrough '*' on SIP/0080265-00006e67
> [Mar  7 17:41:45] DEBUG[20052][C-00001969] bridge_channel.c: DTMF feature string on 0xb22cb234(SIP/0080265-00006e67) is now '*'
> [Mar  7 17:41:45] DTMF[20054][C-00001969] channel.c: DTMF end emulation of '*' queued on SIP/0096003-00006e69
> [Mar  7 17:41:46] DEBUG[20052][C-00001969] bridge_channel.c: DTMF feature string collection on 0xb22cb234(SIP/0080265-00006e67) timed out
> [Mar  7 17:41:46] DEBUG[20055][C-00001969] bridge_channel.c: Playing DTMF stream '*' out to 0xb23c39c4(Local/100 at ac_2712_route_customer-0000010b;1)
> [Mar  7 17:41:46] VERBOSE[20054][C-00001969] app_dial.c: Local/100 at ac_2712_route_customer-0000010b;2 requested media update control 20, passing it to SIP/0096003-00006e69
> [Mar  7 17:41:46] DEBUG[20054][C-00001969] res_rtp_asterisk.c: Setting the marker bit due to a source update



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