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

[Vladislav Krivoshchekov (JIRA)]

Vladislav Krivoshchekov created ASTERISK-26848:
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             Summary: 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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