[asterisk-bugs] [JIRA] (ASTERISK-24642) 13 branch broken, can't make SIP calls
Matt Jordan (JIRA)
noreply at issues.asterisk.org
Wed Dec 24 10:49:34 CST 2014
[ https://issues.asterisk.org/jira/browse/ASTERISK-24642?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=224218#comment-224218 ]
Matt Jordan commented on ASTERISK-24642:
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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?
> 13 branch broken, can't make SIP calls
> --------------------------------------
>
> Key: ASTERISK-24642
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-24642
> Project: Asterisk
> Issue Type: Bug
> Security Level: None
> Environment: -r429719 works, -r429739 is broken
> Reporter: Malcolm Davenport
>
> 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}
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