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

[Asterisk Bug Tracker]

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:              2011-06-02 07:40 CDT
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Summary:                    [patch] 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.  
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 (0135636) remiq (reporter) - 2011-06-02 07:40
 https://issues.asterisk.org/view.php?id=19337#c135636 
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I patched asterisk to include 'sip:' in the uri.  I confirmed attended
transfers are working now with my sip proxy.  Here is the BYE message:


Reliably Transmitting (no NAT) to 209.191.39.117:5060:
BYE sip:322-eng at 209.191.39.117:5060;adtnpxyid-1i2c6kcj=8cg9c2 SIP/2.0
Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK1c20e4ae
Route: <sip:209.191.39.117;lr>
Max-Forwards: 70
From: <sip:312 at 64.19.145.13;user=phone>;tag=as06bd6aa9
To: "Poly_test ENG"<sip:322-eng at 64.19.145.13>;tag=8F24444A-B00E699F
Call-ID: 7f8fe88-8db45e5d-55791ff2 at 10.0.15.101
CSeq: 102 BYE
User-Agent: Asterisk PBX SVN-branch-1.8-r321155M
Proxy-Authorization: Digest username="322-eng", realm="asterisk",
algorithm=MD5, uri="sip:64.19.145.13", nonce="",
response="f8f04d4adff65548b5cb6b17f791f768"
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: "Remi Quezada"<sip:176 at 64.19.145.13>;tag=as39b0a74b
To: "Poly_test
ENG"<sip:322-eng at 209.191.39.117:5060;adtnpxyid-1i2c6kcj=8cg9c2>;tag=7E86B3E1-C3C48076
Call-ID: 763fcb0214fe7c230c228aa15e1eda74 at 64.19.145.13:5060
CSeq: 105 NOTIFY
Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK195c1cac
Contact: <sip:322-eng at 209.191.39.117:5060;adtnpxyid-1i2c6kcj=8cg9c2>
Record-Route: <sip:209.191.39.117;lr>
Event: refer;id=2
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.3.1734
Accept-Language: en
Content-Length: 0


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

Issue History 
Date Modified    Username       Field                    Change               
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2011-06-02 07:40 remiq          Note Added: 0135636                          
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