[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:all-tabpanel ]

Joshua Colp updated ASTERISK-26848:
-----------------------------------

    Assignee: Vladislav Krivoshchekov
      Status: Waiting for Feedback  (was: Triage)

> 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
>            Assignee: 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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