[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:                     acknowledged
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:              11-16-2007 08:44 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 - 11-16-07 08:44  
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the jitter buffer problem causing the 1-way-audio in 10% of the cases is
caused by the adaptive settings. I changed sip.conf and misdn.conf to use
fixed buffer:

;JITTER BUFFER CONFIGURATION
jbenable = yes
jbforce = no
;jbimpl = adaptive <=== reason of the problem
jbimpl = fixed
;jbmaxsize=200
jblog = no

I still have the SIP transfer issue. On GXP2000 LINE2 get stuck in "HOLD"
position and there is no operator to recover it. Line2 is on hold forever
until it hangups. 

Issue History 
Date Modified   Username       Field                    Change               
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11-16-07 08:44  agx            Note Added: 0073806                          
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