[asterisk-bugs] [Asterisk 0010915]: Problems when doing an attended and unattended transfer with Thomson Phones

[noreply at bugs.digium.com]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=10915 
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Reported By:                ramonpeek
Assigned To:                
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Project:                    Asterisk
Issue ID:                   10915
Category:                   Channels/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.4.12.1  
SVN Branch (only for SVN checkouts, not tarball releases): N/A  
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
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Date Submitted:             10-08-2007 11:34 CDT
Last Modified:              10-08-2007 15:49 CDT
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Summary:                    Problems when doing an attended and unattended
transfer with Thomson Phones
Description: 
When doing an attended transfer (party A -> B -> C) the phones A & B start
ringing after they finished the active call, but there is no audio
(ghostcalls)

When doing an unattended transfer the phone B (transferrer) start ringing
immediatly after transfering the party A to C.
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---------------------------------------------------------------------- 
 mavince - 10-08-07 15:49  
---------------------------------------------------------------------- 
Loaded new rtp.c... usual compile warnings... tried attended transfer with
Polycom.... Asterisk crashed.. CLI trace below.. PSTN ->polycom(garrish)
->transfer to polycom (utano) ->crash

-- Executing [7323680452 at pri_from_als:1] Macro("SIP/172.16.4.4-b7d14e80",
"stdexten|0452|SIP/garrish") in new stack
    -- Executing [s at macro-stdexten:1]     -- Goto (macro-stdexten,s,4)
    -- Executing [s at macro-stdexten:4] Dial("SIP/172.16.4.4-b7d14e80",
"SIP/garrish|30") in new stack
 Extension Changed 0452 new state Ringing for Notify User utano
    -- Called garrish
 Extension Changed 0452 new state Ringing for Notify User vince
        -- SIP/garrish-08a5edb0 is ringing
        -- SIP/garrish-08a5edb0 answered SIP/172.16.4.4-b7d14e80
 Extension Changed 0452 new state InUse for Notify User utano
 Extension Changed 0452 new state InUse for Notify User vince
    -- Native bridging SIP/172.16.4.4-b7d14e80 and SIP/garrish-08a5edb0
        -- Started music on hold, class 'sip', on SIP/172.16.4.4-b7d14e80
        -- Stopped music on hold on SIP/172.16.4.4-b7d14e80
        -- Started music on hold, class 'sip', on SIP/172.16.4.4-b7d14e80
        -- Executing [0451 at mt_thorium:1] Macro("SIP/garrish-b7d11bb8",
"stdexten|0451|SIP/utano") in new stack
    -- Executing [s at macro-stdexten:1]     -- Goto (macro-stdexten,s,4)
    -- Executing [s at macro-stdexten:4] Dial("SIP/garrish-b7d11bb8",
"SIP/utano|30") in new stack
 Extension Changed 0451 new state Ringing for Notify User garrish
    -- Called utano
 Extension Changed 0451 new state Ringing for Notify User vince
        -- SIP/utano-08a535f8 answered SIP/garrish-b7d11bb8
 Extension Changed 0451 new state InUse for Notify User garrish
    -- Native bridging SIP/garrish-b7d11bb8 and SIP/utano-08a535f8
 Extension Changed 0451 new state InUse for Notify User vince
        -- Stopped music on hold on SIP/172.16.4.4-b7d14e80
thorium*CLI> 
Disconnected from Asterisk server
Executing last minute cleanups
Asterisk cleanly ending (0).
[root at thorium asterisk]# 

Issue History 
Date Modified   Username       Field                    Change               
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10-08-07 15:49  mavince        Note Added: 0071663                          
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