| Summary: | ASTERISK-19442: unaccepted attend transfer hangup caller | ||||
| Reporter: | Alexander Cruz (chandero) | Labels: | |||
| Date Opened: | 2012-02-28 14:28:31.000-0600 | Date Closed: | 2012-04-24 14:42:26 | ||
| Priority: | Major | Regression? | |||
| Status: | Closed/Complete | Components: | |||
| Versions: | 1.8.9.2 | Frequency of Occurrence | Constant | ||
| Related Issues: |
| ||||
| Environment: | Asterisk 1.8.9.2 on CentOS 6 kernel Linux localhost.localdomain 2.6.32-71.29.1.el6.x86_64 #1 SMP Mon Jun 27 19:49:27 BST 2011 x86_64 x86_64 x86_64 GNU/Linux | Attachments: | ( 0) issue_19442_log | ||
| Description: | A call B, B transfer call to C, C answer to B and not accept the call hangingup, immediatly A is hangup. | ||||
| Comments: | By: David Woolley (davidw) 2012-02-29 05:32:47.940-0600 What technology? If SIP, SIP or features transfer? I think you are going to need to provide the technology driver debug output, as well. See https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information By: Alexander Cruz (chandero) 2012-02-29 15:55:21.900-0600 via feature atxfer => *2 By: Matt Jordan (mjordan) 2012-03-06 15:34:59.450-0600 We require a complete debug log to help triage the issue. This document will provide instructions on how to collect debugging logs from an Asterisk machine for the purpose of helping bug marshals troubleshoot an issue: https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information By: Vladimir Astafiev (vldmr) 2012-03-15 22:17:44.516-0500 I have some issue. Affected all versions after 1.8.8.2 By: Matt Jordan (mjordan) 2012-03-23 12:34:57.519-0500 It is incredibly difficult to tell what is going on in this log file, since you have IAX2, local channels, and SIP all interacting with each other. Let's try this: 1) Remove the MixMonitor from your IAX2 channel, its not helping matters here 2) Provide your extensions.conf, sip.conf, and iax.conf. Specify which parties and technologies are taking place in the attended transfer, who does the transfer, who is the transfer target, and what the expected result should be. 3) Provide a new DEBUG log with SIP DEBUG turned on (if a SIP channel is involved in the transfer) that illustrates the problem Thanks Matt By: Matt Jordan (mjordan) 2012-04-24 14:42:16.470-0500 Suspended due to lack of activity. Please request a bug marshal in #asterisk-bugs on the IRC network irc.freenode.net to reopen the issue should you have the additional information requested. Further information can be found at http://www.asterisk.org/developers/bug-guidelines | ||||