[asterisk-bugs] [Asterisk 0016862]: One-legged Transfer (INVITE / Replaces) not working anymore

Asterisk Bug Tracker noreply at bugs.digium.com
Fri Feb 19 09:34:20 CST 2010


A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=16862 
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Reported By:                pwalker
Assigned To:                
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Project:                    Asterisk
Issue ID:                   16862
Category:                   Channels/chan_sip/Transfers
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     new
Asterisk Version:           1.4.29 
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:             2010-02-18 07:57 CST
Last Modified:              2010-02-19 09:34 CST
====================================================================== 
Summary:                    One-legged Transfer (INVITE / Replaces) not working
anymore
Description: 
It seems that the "One legged Transfer" (INVITE with replaces: header) is
not working anymore in newer Versions of the 1.4 Branch.
The function has been working up to Asterisk 1.4.26.3, but doesn't
starting from 1.4.27.1 (including latest stable 1.4.29 and 1.4.30-rc2).
(With exactely the same configuration.)

Scenario:
- 10013 ("Test3") calls 10012  ("Test2")
- 10014 ("Test4") tries to pick up the call
Result:
- On phone 10012, the call is ended
- 10013 seems to answer / pick up the call, but there's no connection (no
audio)
- 10014 keeps playing the Ring-back Tone

====================================================================== 

---------------------------------------------------------------------- 
 (0118281) pwalker (reporter) - 2010-02-19 09:34
 https://issues.asterisk.org/view.php?id=16862#c118281 
---------------------------------------------------------------------- 
Further investigation with more debugging options enabled:
After performing the above Scenario and terminating all calls, there still
are two lock left (core show locks):
=======================================================================
=== Currently Held Locks ==============================================
=======================================================================
===
=== <file> <line num> <function> <lock name> <lock addr> (times locked)
===
=== Thread ID: -1211597936 (do_monitor           started at [17088]
chan_sip.c restart_monitor())
=== ---> Lock https://issues.asterisk.org/view.php?id=0 (chan_sip.c): MUTEX 9695
get_sip_pvt_byid_locked
&sip_pvt_ptr->lock 0x834a678 (1)
=== ---> Lock https://issues.asterisk.org/view.php?id=1 (chan_sip.c): MUTEX 9731
get_sip_pvt_byid_locked (channel
lock) 0x8349870 (1)
=== -------------------------------------------------------------------
===
=== Thread ID: -1224025200 (pbx_thread           started at [ 2641] pbx.c
ast_pbx_start())
=== ---> Waiting for Lock https://issues.asterisk.org/view.php?id=0 (channel.c):
MUTEX 1745 ast_waitfor_nandfds
(channel lock) 0x8349870 (1)
=== --- ---> Locked Here: chan_sip.c line 9731 (get_sip_pvt_byid_locked)
=== -------------------------------------------------------------------
===
======================================================================= 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-02-19 09:34 pwalker        Note Added: 0118281                          
======================================================================




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