[asterisk-bugs] [JIRA] (ASTERISK-26848) Problem with call return when atxfer
Richard Mudgett (JIRA)
noreply at issues.asterisk.org
Thu Mar 9 11:21:10 CST 2017
[ https://issues.asterisk.org/jira/browse/ASTERISK-26848?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]
Richard Mudgett updated ASTERISK-26848:
---------------------------------------
Description:
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':
{noformat}
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)
{noformat}
logs:
{noformat}
[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
{noformat}
was:
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
> 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':
> {noformat}
> 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)
> {noformat}
> logs:
> {noformat}
> [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
> {noformat}
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