[asterisk-bugs] [Asterisk 0015876]: Interoperability with Exchange 2007 UM
Asterisk Bug Tracker
noreply at bugs.digium.com
Thu Sep 10 18:43:01 CDT 2009
A NOTE has been added to this issue.
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https://issues.asterisk.org/view.php?id=15876
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Reported By: rsw686
Assigned To:
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Project: Asterisk
Issue ID: 15876
Category: Channels/chan_sip/Interoperability
Reproducibility: always
Severity: major
Priority: normal
Status: new
Asterisk Version: 1.6.1.5
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: 2009-09-10 16:38 CDT
Last Modified: 2009-09-10 18:43 CDT
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Summary: Interoperability with Exchange 2007 UM
Description:
Starting with versions 1.6.0.14 and 1.6.1.5 Asterisk fails to connect with
Exchange 2007 UM. Both versions 1.6.0.13 and 1.6.1.4 which were released on
18-Aug-2009 work.
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(0110546) rsw686 (reporter) - 2009-09-10 18:43
https://issues.asterisk.org/view.php?id=15876#c110546
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I wanted to provide some logs from the Asterisk console. On version 1.6.1.4
when a call goes to Exchange I receive the following output.
== Using SIP RTP CoS mark 5
-- Executing [3300 at default:1] Answer("SIP/8678-094863c0", "") in new
stack
-- Executing [3300 at default:2] Dial("SIP/8678-094863c0",
"SIP/SIP_VM/3300") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP_VM/3300
-- Got SIP response 302 "Moved Temporarily" back from 10.9.1.13
-- Now forwarding SIP/8678-094863c0 to
'SIP/3300::::TCP at 10.9.1.13:5065' (thanks to SIP/SIP_VM-094f1cd0)
== Using SIP RTP CoS mark 5
-- SIP/10.9.1.13:5065-094f6f58 is ringing
-- SIP/10.9.1.13:5065-094f6f58 answered SIP/8678-094863c0
-- Packet2Packet bridging SIP/8678-094863c0 and
SIP/10.9.1.13:5065-094f6f58
asterisk*CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
8678/8678 10.9.5.107 D 32971
Unmonitored
SIP_VM 10.9.1.13 5060 OK (1 ms)
On version 1.6.1.5 when trying to call Exchange I see
== Using SIP RTP CoS mark 5
-- Executing [3300 at default:1] Answer("SIP/8678-09978b88", "") in new
stack
-- Executing [3300 at default:2] Dial("SIP/8678-09978b88",
"SIP/SIP_VM/3300") in new stack
== Using SIP RTP CoS mark 5
[Sep 10 19:37:53] WARNING[12237]: app_dial.c:1528 dial_exec_full: Unable
to create channel of type 'SIP' (cause 20 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [3300 at default:3] Busy("SIP/8678-09978b88", "") in new
stack
asterisk*CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
8678/8678 10.9.5.107 D 32971
Unmonitored
SIP_VM 10.9.1.13 5060
UNREACHABLE
This is coming from the same machine. All I have done is stop asterisk,
run make install on the other version, and then started asterisk.
Issue History
Date Modified Username Field Change
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2009-09-10 18:43 rsw686 Note Added: 0110546
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