[asterisk-bugs] [Asterisk 0017999]: Problrm with attendant transfer
Asterisk Bug Tracker
noreply at bugs.digium.com
Thu Sep 16 05:52:18 CDT 2010
The following issue has been SUBMITTED.
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https://issues.asterisk.org/view.php?id=17999
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Reported By: iskatel
Assigned To:
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Project: Asterisk
Issue ID: 17999
Category: Core/PBX
Reproducibility: always
Severity: major
Priority: normal
Status: new
Asterisk Version: 1.6.2.13
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: 2010-09-16 05:52 CDT
Last Modified: 2010-09-16 05:52 CDT
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Summary: Problrm with attendant transfer
Description:
Hello!
There are such two situation during attendant transfer usage
Situation https://issues.asterisk.org/view.php?id=1
1) A (8123364000) calls B (0011*102). B answers.
2) B using DTMF dial *2 (code in features.conf for attendant transfer).
3) A hears MOH. B dial number C (3364021)
4) C ringing. A hears MOH.
5) B hangup. A still hears MOH. C ringing.
5) A hangup. C still ringing until "atxfernoanswertimeout" expires.
Problem: When A and B hangup C still ringing.
Situation https://issues.asterisk.org/view.php?id=2
1) A (8123364000) calls B (0011*102). B answers.
2) B using DTMF dial *2 (code in features.conf for attendant transfer).
3) A hears MOH. B dial number C (3364021)
4) C ringing. A hears MOH.
5) B hangup. A still hears MOH. C ringing.
6) "atxfernoanswertimeout" expires. After this asterisk tries callback "B"
but do it using such form "SIP/0011*102" and generates INVITE with
RURI=sip:@"dest_ip" i.e. without any number in RURI.
Because of this SIP remote device cannot handle call.
Problem: There is no number in RURI when try callback to B.
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Issue History
Date Modified Username Field Change
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2010-09-16 05:52 iskatel New Issue
2010-09-16 05:52 iskatel Asterisk Version => 1.6.2.13
2010-09-16 05:52 iskatel Regression => No
2010-09-16 05:52 iskatel SVN Branch (only for SVN checkouts, not tarball
releases) => N/A
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