[asterisk-bugs] [Asterisk 0016992]: incoming INVITE received no progress, just 200 OK, causing Sipra pstn to go off-hook
Asterisk Bug Tracker
noreply at bugs.digium.com
Wed Mar 10 10:33:36 CST 2010
The following issue requires your FEEDBACK.
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https://issues.asterisk.org/view.php?id=16992
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Reported By: jw-asterisk
Assigned To:
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Project: Asterisk
Issue ID: 16992
Category: Channels/chan_sip/General
Reproducibility: always
Severity: major
Priority: normal
Status: feedback
Asterisk Version: 1.6.2.5
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: 2010-03-08 22:12 CST
Last Modified: 2010-03-10 10:33 CST
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Summary: incoming INVITE received no progress, just 200 OK,
causing Sipra pstn to go off-hook
Description:
With a Sipura 3102 using asterisk as its proxy (pstn->voip). the Sipura
sends asterisk an INVITE, then asterisk responds with a "100 Trying" almost
immediately, but then as soon as asterisk starts dialing extensions to
handle the incoming call it sends "200 OK" to the Sipura. This causes the
Sipura to prematurely go off-hook.
Shouldn't asterisk be sending a "180 Ringing" to the Sipura until one of
the extensions answers the call at which point the channels would be
bridged?
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(0119221) lmadsen (administrator) - 2010-03-10 10:33
https://issues.asterisk.org/view.php?id=16992#c119221
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You'll need to provide the dialplan you're using along with the SIP trace
of the call. Both of those things are at a minimum required when filing SIP
issues.
This page may also be a useful read:
http://www.asterisk.org/developers/bug-guidelines
Item 6 under Opening An Issue In The Issue Tracker - A Checklist deals
with filing SIP related issues.
Thanks!
Issue History
Date Modified Username Field Change
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2010-03-10 10:33 lmadsen Note Added: 0119221
2010-03-10 10:33 lmadsen Status new => feedback
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