[asterisk-bugs] [Asterisk 0019337]: Call shows on hold after attended transfer with a Polycom phone

Asterisk Bug Tracker noreply at bugs.digium.com
Wed May 25 10:42:46 CDT 2011


A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=19337 
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Reported By:                remiq
Assigned To:                
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Project:                    Asterisk
Issue ID:                   19337
Category:                   Channels/chan_sip/Transfers
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     acknowledged
Asterisk Version:           SVN 
JIRA:                       SWP-3492 
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): 1.8 
SVN Revision (number only!): 319938 
Request Review:              
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Date Submitted:             2011-05-20 10:33 CDT
Last Modified:              2011-05-25 10:42 CDT
====================================================================== 
Summary:                    Call shows on hold after attended transfer with a
Polycom phone
Description: 
When I do an attended transfer from a Polycom IP650 the call is
transferring successfully, but the call is not releasing properly on the
phone that is initiating the transfer.  Instead the call shows that it is
on hold.  
====================================================================== 

---------------------------------------------------------------------- 
 (0135384) remiq (reporter) - 2011-05-25 10:42
 https://issues.asterisk.org/view.php?id=19337#c135384 
---------------------------------------------------------------------- 
Here is the BYE from asterisk 1.6.2.9:

Reliably Transmitting (no NAT) to 209.191.39.117:5060:
BYE sip:322-eng at 209.191.39.117:5060;adtnpxyid-1i2c6kcj=8cg8ia SIP/2.0
Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK6e12c410;rport
Max-Forwards: 70
From: <sip:312 at 64.19.145.13;user=phone>;tag=as671e9719
To: "Poly_test ENG"<sip:322-eng at 64.19.145.13>;tag=23813484-136FC2B
Call-ID: 8ced1f50-60723397-d9a44366 at 10.0.15.101
CSeq: 102 BYE
User-Agent: Asterisk PBX 1.6.2.9
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


<--- SIP read from UDP:209.191.39.117:5060 --->
SIP/2.0 200 OK
From: <sip:312 at 64.19.145.13;user=phone>;tag=as671e9719
To: "Poly_test ENG"<sip:322-eng at 64.19.145.13>;tag=23813484-136FC2B
Call-ID: 8ced1f50-60723397-d9a44366 at 10.0.15.101
CSeq: 102 BYE
Via: SIP/2.0/UDP 64.19.145.13:5060;rport=5060;branch=z9hG4bK6e12c410
Contact: <sip:322-eng at 209.191.39.117:5060;adtnpxyid-1i2c6kcj=8cg8ia>
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.3.1734
Accept-Language: en
Content-Length: 0


<-------------> 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2011-05-25 10:42 remiq          Note Added: 0135384                          
======================================================================




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