[asterisk-bugs] [JIRA] (ASTERISK-26048) Asterisk crashes with PJSIP Assertion "Invalid transport name"

Asterisk Team (JIRA) noreply at issues.asterisk.org
Fri May 20 15:08:56 CDT 2016


    [ https://issues.asterisk.org/jira/browse/ASTERISK-26048?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=230718#comment-230718 ] 

Asterisk Team commented on ASTERISK-26048:
------------------------------------------

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].

> Asterisk crashes with PJSIP Assertion "Invalid transport name"
> --------------------------------------------------------------
>
>                 Key: ASTERISK-26048
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-26048
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: pjproject/pjsip
>    Affects Versions: 13.7.1, 13.9.1
>         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
>            Reporter: Erico Mattos
>            Severity: Critical
>
> 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 at macro-dial:17] Dial("PJSIP/NET-6238770660-00000003", "PJSIP/401/sip:401 at 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 at macro-dial:17] Dial("PJSIP/NET-6238770660-00000002", "SIP/401&PJSIP/402/sip:402 at 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



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