Summary: | ASTERISK-26048: Asterisk crashes with PJSIP Assertion "Invalid transport name" | ||
Reporter: | Erico Mattos (tdmnetwork) | Labels: | |
Date Opened: | 2016-05-20 15:08:12 | Date Closed: | 2016-05-20 15:14:55 |
Priority: | Critical | Regression? | |
Status: | Closed/Complete | Components: | pjproject/pjsip |
Versions: | 13.7.1 13.9.1 | Frequency of Occurrence | Constant |
Related Issues: | |||
Environment: | Kernel: 2.6.32-504.8.1.el6.x86_64 FreePBX 13.0.120 Before update: FreePBX Distro: 10.13.66-9 Asterisk 13.7.1 After update: FreePBX Distro: 10.13.66-12 Asterisk 13.9.1 | Attachments: | ( 0) TDM--error_after_call--summary.txt |
Description: | Relevant scenario:
- An incomming call is redirected to a group with an extension using transport TCP or TLS The following message showes on stderr and then crashes asterisk: {{asterisk: ../src/pjsip/sip_transport.c:299: pjsip_transport_get_type_from_name: Assertion `!"Invalid transport name"' failed.}} On the stdout (before the stderr message) using TCP: {{Executing [s@macro-dial:17] Dial("PJSIP/NET-6238770660-00000003", "PJSIP/401/sip:401@10.62.10.151:5061;transport=CP,20,trM(auto-blkvm)b(func-apply-sipheaders^s^1),") in new stack}} On the stdout (before the stderr message) using TLS: {{Executing [s@macro-dial:17] Dial("PJSIP/NET-6238770660-00000002", "SIP/401&PJSIP/402/sip:402@191.190.6.203:5061;transport=LS,20,trM(auto-blkvm)b(func-apply-sipheaders^s^1),") in new stack}} On both situation the first letter "T" is not shown. If I use UDP the error doesn't occurs since que transport is not appended. Full scenario (to understand the log attached): - Incomming call cames from a trunk on a SIP UDP gateway - Call is redirected to a IVR with a announcement - The option selected (4) is associated to a Ring Group (400) - The Ring group has two extensions, one offline and one online with TCP transport - Crashes when option 4 is selected | ||
Comments: | By: Asterisk Team (asteriskteam) 2016-05-20 15:08:12.847-0500 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) 2016-05-20 15:14:55.092-0500 This is not a bug in Asterisk. If PJPROJECT Is built with assertions enabled (which are for developers) then it will assert when this happens. Additionally the lack of 'T' is a bug in FreePBX. |