| Summary: | ASTERISK-18244: When caller hangs up on AGI dial app the code does not complete execution | ||
| Reporter: | sam mitchell (lakersman2006) | Labels: | |
| Date Opened: | 2011-08-08 13:25:15 | Date Closed: | 2011-09-14 09:00:22 |
| Priority: | Major | Regression? | |
| Status: | Closed/Complete | Components: | |
| Versions: | 1.8.5.0 | Frequency of Occurrence | |
| Related Issues: | |||
| Environment: | Attachments: | ||
| Description: | I have a php AGI script that bridges a call using the dial() app and when the caller hangs up first the AGI script immediately exits without returning the DIALSTATUS, ANSWEREDTIME, etc. But if the callee hangs up first then the AGI code continues executing as normal and returns the correct DIALSTATUS ANSWERTIME, etc. | ||
| Comments: | By: sam mitchell (lakersman2006) 2011-08-09 16:40:07.971-0500 It appears that asterisk 1.6.2.20 also shows this same behavior. Asterisk 1.6.5.15 works correctly. Also, it appears v1.8.4.4 works fine as well. By: Paul Belanger (pabelanger) 2011-08-26 14:25:52.463-0500 We are going to need some log files to see what is happening. Since you also know this works in 1.8.4.4 but not 1.8.5.0, you should be able to look into the changelog and see which revision made the change. This will make it easier for somebody to fix. --- Thank you for taking the time to report this bug and helping to make Asterisk better. Unfortunately, we cannot work on this bug because your description did not include enough information. You may find it helpful to read the Asterisk Issue Guidelines http://www.asterisk.org/developers/bug-guidelines. We would be grateful if you would then provide a more complete description of the problem. At a minimum, we need: 1. the specific steps or actions you took that caused you to encounter the problem, 2. the behavior you expected, and 3. the behavior you actually encountered (in as much detail as possible). This likely includes output from the console with debug level logging, a SIP trace (if this is SIP related), and configuration information such as dialplan (e.g. extensions.conf) and channel configuration (e.g. sip.conf). Thanks! By: Leif Madsen (lmadsen) 2011-09-14 09:00:17.238-0500 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 | ||