| Summary: | ASTERISK-26848: Problem with call return when atxfer | ||
| Reporter: | Vladislav Krivoshchekov (superenemy) | Labels: | |
| Date Opened: | 2017-03-09 04:48:21.000-0600 | Date Closed: | 2020-01-14 11:13:51.000-0600 |
| Priority: | Major | Regression? | |
| Status: | Closed/Complete | Components: | Features |
| Versions: | 13.4.0 | Frequency of Occurrence | Constant |
| Related Issues: | |||
| Environment: | Linux version 2.6.32-5-686 (Debian 2.6.32-48squeeze4) | Attachments: | |
| 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) ะก 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@ac_2712_route_customer-0000010b;1) [Mar 7 17:41:46] VERBOSE[20054][C-00001969] app_dial.c: Local/100@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} | ||
| Comments: | By: Asterisk Team (asteriskteam) 2017-03-09 04:48:23.616-0600 Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution. A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report. Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process]. By: Joshua C. Colp (jcolp) 2017-03-09 04:56:21.806-0600 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 By: Asterisk Team (asteriskteam) 2017-03-24 12:00:02.525-0500 Suspended due to lack of activity. This issue will be automatically re-opened if the reporter posts a comment. If you are not the reporter and would like this re-opened please create a new issue instead. If the new issue is related to this one a link will be created during the triage process. Further information on issue tracker usage can be found in the Asterisk Issue Guidlines [1]. [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines | ||