[asterisk-bugs] [Asterisk 0011169]: unable to perform attended transfer of incoming call from mISDN
noreply at bugs.digium.com
noreply at bugs.digium.com
Tue Nov 20 07:28:49 CST 2007
A NOTE has been added to this issue.
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http://bugs.digium.com/view.php?id=11169
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Reported By: agx
Assigned To:
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Project: Asterisk
Issue ID: 11169
Category: Channels/General
Reproducibility: always
Severity: minor
Priority: normal
Status: feedback
Asterisk Version: 1.4.13
SVN Branch (only for SVN checkouts, not tarball releases): 1.4
SVN Revision (number only!): 88862
Disclaimer on File?: N/A
Request Review:
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Date Submitted: 11-06-2007 08:16 CST
Last Modified: 11-20-2007 07:28 CST
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Summary: unable to perform attended transfer of incoming call
from mISDN
Description:
After talked with Blitzrage about bug 11085 on #asterisk-bugs we agreed on
opening a new bug.
Its always reproducible when there is an incoming call from mISDN. If
instead i generate the call from a SIP or IAX2 phone it does not happen.
The flow involves all GXP2000 phones with firmware 1.1.2.27, also tried
1.1.1.14.
The bug involes asterisk 1.4.13 also tried with svn-1.4.
mISDN used is 1.1.6 also tried to downgrade to 1.1.5.
The flow is like this:
1. SIP/12 answer call from mISDN on gxp's Line1
2. SIP/12 pick Line2 and call SIP/11 (mISDN get MOH)
3. SIP/11 answer SIP/12 and say its ok to talk with misdn people
4. SIP/12 pick line1 (SIP/12 get MOH) and say that the other people is
free
5. SIP/12 hit TRANSFER button and press LINE2 and here happen the
problem:
- 90% of the time SIP/12 get a message onto the screen "TRANSFER
CANCELLED" and is unable to repick LINE2
- 10% of the time the call is transferred but is one-way-audio
I'll attacch full debug output.
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agx - 11-20-07 07:28
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triple-checked it with different phones (snom, gxp2000, idefisk, etc.)
its solved now
:)
Issue History
Date Modified Username Field Change
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11-20-07 07:28 agx Note Added: 0074059
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