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Summary:ASTERISK-24642: 13 branch broken, can't make SIP calls
Reporter:Malcolm Davenport (mdavenport)Labels:
Date Opened:2014-12-24 10:32:36.000-0600Date Closed:2014-12-24 10:50:17.000-0600
Priority:MajorRegression?
Status:Closed/CompleteComponents:
Versions:Frequency of
Occurrence
Constant
Related
Issues:
Environment:-r429719 works, -r429739 is brokenAttachments:
Description:In working revision, calls are fine.
In broken revision, calls don't happen.  Asterisk receives INVITE w/ auth and returns Trying...and that's it.

Dialplan tested:
{noformat}
exten => 306,1,NoOp()
same => n,Background(pbx-invalid)
same => n,Hangup()
{noformat}
Comments:By: Matt Jordan (mjordan) 2014-12-24 10:47:44.780-0600

So it's clearly not broken everywhere, since the Test Suite is still running just fine.

Revision {{429739}} is the following:

{quote}
{noformat}
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r429739 | mmichelson | 2014-12-18 08:43:53 -0600 (Thu, 18 Dec 2014) | 17 lines

Ensure the correct value is returned for CHANNEL(pjsip, secure)

Prior to this patch, we were using the PJSIP dialog's secure flag
to determine if a secure transport was being used. Unfortunately,
the dialog's secure flag was only set if a SIPS URI were in use,
as required by RFC 3261 sections 12.1.1 and 12.1.2. What we're interested
in is not dialog security, but transport security. This code change
switches to a model where we use the dialog's target URI to determine
what transport would be used to communicate, and then check if that
transport is secure.

AST-1450 #close
Reported by John Bigelow

Review: https://reviewboard.asterisk.org/r/4277


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{noformat}
{quote}

How is a modification to the {{CHANNEL}} function going to break basic playback?

By: Malcolm Davenport (mdavenport) 2014-12-24 10:50:17.721-0600

Updated pjproject.  The world is now a happier place.

By: Rusty Newton (rnewton) 2014-12-24 12:06:31.138-0600

Yup I ran into this yesterday and an update for pjproject fixed it for me too.