[asterisk-bugs] [Asterisk 0018981]: Call transfer Avaya - Asterisk ooh323 one way audio
Asterisk Bug Tracker
noreply at bugs.digium.com
Wed Mar 16 15:52:21 CDT 2011
A NOTE has been added to this issue.
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https://issues.asterisk.org/view.php?id=18981
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Reported By: red_emotion
Assigned To: may213
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Project: Asterisk
Issue ID: 18981
Category: Addons/chan_ooh323
Reproducibility: always
Severity: major
Priority: normal
Status: assigned
Asterisk Version: 1.4.40
JIRA:
Regression: No
Reviewboard Link:
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
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Date Submitted: 2011-03-16 04:11 CDT
Last Modified: 2011-03-16 15:52 CDT
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Summary: Call transfer Avaya - Asterisk ooh323 one way audio
Description:
I have an Avaya IP Office 403 working in H.323. I managed to connect it to
my asterisk server through library ooh323 ... even though I was forced to
set as default codec ULAW (which is quite heavy), though the quality is
acceptable.
My problem born when I do a call transfer to another extension of AVAYA:
A = internal of avaya (H.323) n.1
B = internal of avaya (H.323) n.2
C = internal of asterisk (sip)
Scenario:
A calls B, B runs the call of A to C.
B and C (Avaya and Asterisk) are communicating properly, but when they
turn the call B of A to C, C hears the voice of A, but A does not hear the
voice of C.
I read on some forums that might be some problem ooh323, but I do not know
where to put hands ...
I hope I explained well ...
If you have any suggestions let me know
Thanks
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(0132967) may213 (manager) - 2011-03-16 15:52
https://issues.asterisk.org/view.php?id=18981#c132967
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Hi,
can you upgrade your asterisk from 1.4 to latest released 1.8?
1.4 include older version of ooh323 channel driver which have number of
bugs.
Issue History
Date Modified Username Field Change
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2011-03-16 15:52 may213 Note Added: 0132967
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