[asterisk-bugs] [Asterisk 0018185]: Blind transfer failure, A calls B, B transfers to C

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=18185 
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Reported By:                kwemheuer
Assigned To:                
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Project:                    Asterisk
Issue ID:                   18185
Category:                   Channels/chan_sip/Transfers
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     new
Asterisk Version:           1.8.0 
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:             2010-10-22 04:43 CDT
Last Modified:              2010-10-22 07:32 CDT
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Summary:                    Blind transfer failure, A calls B, B transfers to C
Description: 
The summary says it: Transfers are not working in one direction. In
detail:

I set up an asterisk system (version 1.8.0). While using a SIP only
environment I discovered a problem using blind transfer. The phones are
SNOM or Aastra and are using the SIP REFER Method.

For debugging purposes I stripped down the environment to a simple
dialplan with only three extensions. There are three phones (Snom)
registered to asterisk. I use release 1.8.0 from yesterday.

There are three extensions: 100, 150, and 180. There are three
SIP-Devices: phone1, phone2, and phone3. (See extensions.conf, sip.conf)

User on phone1 dials 150 (phone2). User on phone2 transfers the call to
phone3 (using blind transfer, SIP REFER Method). The calls fails (See debug
output in file failed.log)

The following scenario is working: User on phone1 dials 150 (phone2). User
on phone1 transfers the call to phone3 (using blind transfer, SIP REFER
Method). See debug output in file working.log.

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---------------------------------------------------------------------- 
 (0128310) pabelanger (manager) - 2010-10-22 07:32
 https://issues.asterisk.org/view.php?id=18185#c128310 
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We'll also need a SIP debug to see what is happening (see below).

---
We require a complete debug log to help triage the issue.

This document will provide instructions on how to collect debugging logs
from an Asterisk machine for the purpose of helping bug marshals
troubleshoot an issue:

http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt


Issue History 
Date Modified    Username       Field                    Change               
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2010-10-22 07:32 pabelanger     Note Added: 0128310                          
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