[asterisk-bugs] [Asterisk 0015420]: [patch] No audio on calls from asterisk sip phones to nortel set until dtmf from sip phone

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=15420 
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Reported By:                scottbmilne
Assigned To:                
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Project:                    Asterisk
Issue ID:                   15420
Category:                   General
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     new
Asterisk Version:           1.4.25.1 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2009-06-29 13:58 CDT
Last Modified:              2009-07-01 05:46 CDT
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Summary:                    [patch] No audio on calls from asterisk sip phones
to nortel set until dtmf from sip phone
Description: 
When placing a call from an Asterick SIP phone (X-Lite) to Nortel phone set
(M3903), no voice will pass until any key is pressed on the SIP phone. SIP
to SIP calls - normal. SIP to external phone - normal. Nortel to SIP -
normal. All permutations of calls pass audio immediately upon answer except
an sip to nortel.

Asterisk is behind Nortel Option 11c via PRI
Telco -> PRI -> Nortel Option 11c -> PRI -> Asterisk

Very new to Asterisk and VOIP and don't know where to start...Any
suggestions would be GREATLY appreciated.

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 (0107280) alecdavis (reporter) - 2009-07-01 05:46
 https://issues.asterisk.org/view.php?id=15420#c107280 
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I've just been browsing release notes for 1.4.25,
http://www.asterisk.org/node/48596

" Delay signalling progress until a PRI channel really signals progress.
- Closes issue https://issues.asterisk.org/view.php?id=13034. Reported, patched,
and tested by klaus3000."

This above mentioned fix, is causing this issue.
Klaus mentioned in his later notes
https://issues.asterisk.org/view.php?id=13034#102009, that he believed,
"dialing=0" should also be in the event PRI_EVENT_RINGING, but it never
made it.

I also agree with klaus, and think it should also be there, but I'm
covering all basis. But I don't have access for a few days, only web
access. 

klaus3000: are you reading this, or can someone bring it to his attention. 

Issue History 
Date Modified    Username       Field                    Change               
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2009-07-01 05:46 alecdavis      Note Added: 0107280                          
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