[asterisk-bugs] [LibPRI 0013936]: qsigchannelmapping incorrectly defaults to 'logical'

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=13936 
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Reported By:                alecdavis
Assigned To:                mattf
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Project:                    LibPRI
Issue ID:                   13936
Category:                   General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     assigned
Asterisk Version:           SVN 
SVN Branch (only for SVN checkouts, not tarball releases):  trunk 
SVN Revision (number only!): 155671 
Disclaimer on File?:        N/A 
Request Review:              
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Date Submitted:             2008-11-19 18:01 CST
Last Modified:              2008-11-24 14:21 CST
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Summary:                    qsigchannelmapping incorrectly defaults to 'logical'
Description: 
Default intention of http://bugs.digium.com/view.php?id=13454 was to act as
old behaviour '= physical'.

Calls coming in from channels > 16 incorrectly get incremented by 1.

result:
No Audio in either direction for calls coming in on channels > 16.

Calls coming in from channels 31 incorrectly get incremented to channel
32, which reports the following

WARNING[6184]: chan_dahdi.c:10958 pri_dchannel: Ring requested on
unconfigured channel 0/32 span 1

my workaround until fixed.
in file chan_dahdi.conf if switchtype=qsig
ensure that qsigchannelmapping=physical to act as old code
or qsigchannelmapping=logical if you require it.

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---------------------------------------------------------------------- 
 (0095406) alecdavis (reporter) - 2008-11-24 14:21
 http://bugs.digium.com/view.php?id=13936#c95406 
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Tested and functioning correctly with respect to defaulting to 'physical'
instead of logical.

Now to get call path replacement to work.

Should 2 Asterisk boxes back to back both setup as Qsig, one signalling
set as 'pri_net' and the other set to 'pri_cpe' be able to do automatic PR
as long as call is on same span in and out. We have tested this against a
QSIG pri network, provided by Jtec equipment, but not yet back to back
asterisk boxes.

example using SIP, but equally could be fixed analog lines from a PABX.

8500-8599 on Asterisk1                         8600-8699 on Asterisk2
SIP phone1 '8512' <=> Asterisk1 <==QSIG PRI==> Asterisk2 <=> Sip phone2
'8622'
SIP phone1 '8522'

Assuming 8512 calls 8622 on Asterisk2, 8622 uses 'Dial/dahdi/G0/8522' to
immediately transfer back to 8522 on Asterisk1.

On our Jtec we are seeing both trunk channels stay up. I understood that
asterisk would be monitoring the calls, and when both legs are on the same
span the call would have been handed back to the appropriate Asterisk box
to handle. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2008-11-24 14:21 alecdavis      Note Added: 0095406                          
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