[asterisk-bugs] [Asterisk 0011169]: unable to perform attended transfer of incoming call from mISDN

[noreply at bugs.digium.com]

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 06:33 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 06:33  
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working on it, i did a few test and seems ok after i:
- upgraded 1.4.13 -> 1.4.14
- switched in sip.conf "directrtpsetup" from Yes to No

Give me 1 hour of more testing, i'm just waiting the people with that i
usually did those testing to be back 

Issue History 
Date Modified   Username       Field                    Change               
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11-20-07 06:33  agx            Note Added: 0074051                          
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