[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:                     acknowledged
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 08:22 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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Relationships       ID      Summary
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related to          0013034 [patch] 183 response although progressi...
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 (0107291) klaus3000 (reporter) - 2009-07-01 08:22
 https://issues.asterisk.org/view.php?id=15420#c107291 
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sorry - I do not read all the new bug reports.

Looks like in my testing I never had that issue - in my test cases
PROGRESS was always sent.

The problem is that the B channel is not opened until dialing=0. Thus,
when ANSWER is received, dialing should be set to 0.

Probably in pri_dchannel() in section "case: PRI_EVENT_ANSWER" there
should be 

  pri->pvts[chanpos]->dialing = 0;

somewhere. 

Issue History 
Date Modified    Username       Field                    Change               
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2009-07-01 08:22 klaus3000      Note Added: 0107291                          
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