| Summary: | ASTERISK-26472: Asterisk crashes when invalid Dial string used with PJSIP | ||
| Reporter: | xrobau (xrobau) | Labels: | |
| Date Opened: | 2016-10-16 20:12:23 | Date Closed: | 2016-10-17 08:58:56 |
| Priority: | Major | Regression? | No |
| Status: | Closed/Complete | Components: | pjproject/pjsip |
| Versions: | 14.0.2 | Frequency of Occurrence | Constant |
| Related Issues: | |||
| Environment: | FreePBX Distro 7 (Equivalent to CentOS 7) | Attachments: | ( 0) ast14-pjsip-segv.txt |
| Description: | When dialling a bogus (invalid syntax) PJSIP channel, asterisk segv's.
{code} -- Executing [s@macro-dial-one:47] ExecIf("PJSIP/200-00000000", "0?Set(D_OPTIONS=trII)") in new stack -- Executing [s@macro-dial-one:48] Dial("PJSIP/200-00000000", "PJSIP/xrobau/*43,,TtrIb(func-apply-sipheaders^s^1)") in new stack freepbx*CLI> Disconnected from Asterisk server Asterisk cleanly ending (0). Executing last minute cleanups [root@freepbx ~]# {code} | ||
| Comments: | By: Asterisk Team (asteriskteam) 2016-10-16 20:12:24.494-0500 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) 2016-10-17 08:58:56.276-0500 This isn't a bug in Asterisk itself but in PJSIP. It's already been fixed https://trac.pjsip.org/repos/ticket/1946 and will be in the next release of PJSIP. | ||