[asterisk-users] Dial to PJSIP Channel with Typo PJSIP// Causes Asterisk Shutdown

2015-03-26 Thread Trey Hilyard
I found an issue with how PJSIP handles a typo in the Dial application. If
the Channel is mistakenly typed with two slashes (i.e Dial(PJSIP//...),
the Dial applications fails (obviously), but it also kills the server.

I put some code in my pbx_config to check for that string and not let the
dialplan reload, but it seems like there should be a better way to handle
in in the PJSIP stack or Dial app so that it doesn't take the server down
if it gets through.

I am not a developer, but I was hoping maybe someone who monitors this
mailing list might feel like taking this on as a bug fix.I haven't tried
with any other channel drivers, so it may cross to others.
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Dial to PJSIP Channel with Typo PJSIP// Causes Asterisk Shutdown

2015-03-26 Thread Matthew Jordan
On Thu, Mar 26, 2015 at 9:28 AM, Trey Hilyard kct...@gmail.com wrote:
 I found an issue with how PJSIP handles a typo in the Dial application. If
 the Channel is mistakenly typed with two slashes (i.e Dial(PJSIP//...),
 the Dial applications fails (obviously), but it also kills the server.

 I put some code in my pbx_config to check for that string and not let the
 dialplan reload, but it seems like there should be a better way to handle in
 in the PJSIP stack or Dial app so that it doesn't take the server down if it
 gets through.

 I am not a developer, but I was hoping maybe someone who monitors this
 mailing list might feel like taking this on as a bug fix.I haven't tried
 with any other channel drivers, so it may cross to others.


Please open an issue on the issue tracker:

https://issues.asterisk.org/jira

A backtrace from the crash will be needed as well. Instructions on
generating a backtrace can be found on the wiki here:

https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace

-- 
Matthew Jordan
Digium, Inc. | Director of Technology
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com  http://asterisk.org

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users