[asterisk-bugs] [Asterisk 0018837]: [patch] Deadlock with attended transfer of SIP call

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=18837 
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Reported By:                alecdavis
Assigned To:                
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Project:                    Asterisk
Issue ID:                   18837
Category:                   Core/RTP
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     ready for testing
Asterisk Version:           1.8.2.3 
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:             2011-02-17 19:25 CST
Last Modified:              2011-02-18 05:00 CST
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Summary:                    [patch] Deadlock with attended transfer of SIP call
Description: 
3 SIP phones.

A calls B, and B answers on line 1.
B puts A on hold by selecting line2.
B calls C, and C answers.
B initiates transfer of line1 to line2, phone uses REFER.


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Relationships       ID      Summary
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has duplicate       0018468 SIP crash on transfer
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 (0132128) Irontec (reporter) - 2011-02-18 05:00
 https://issues.asterisk.org/view.php?id=18837#c132128 
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You're right but with this config. the xfered call hangs.

Asterisk sends 2 consecutive INVITEs (different CSeq):


U 10.10.0.165:5060 -> 10.10.0.142:5064
INVITE sip:300 at 10.10.0.142:5064 SIP/2.0.
Via: SIP/2.0/UDP 10.10.0.165:5060;branch=z9hG4bK748b128b.
Max-Forwards: 70.
From: "200" <sip:200 at 10.10.0.165>;tag=as08a90e13.
To: <sip:300 at 10.10.0.142:5064>;tag=2be4485acdb9d4a5i4.
Contact: <sip:200 at 10.10.0.165:5060>.
Call-ID: 19bb6dd729b44c10706c9aa868aa3902 at 10.10.0.165:5060.
CSeq: 105 INVITE.
User-Agent: Asterisk PBX 1.8.2.3.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH.
Supported: replaces, timer.
Remote-Party-ID: "200"
<sip:200 at 10.10.0.165>;party=calling;privacy=off;screen=no.
Content-Type: application/sdp.
Content-Length: 205.
.
v=0.
o=root 2032588221 2032588224 IN IP4 10.10.0.165.
s=Asterisk PBX 1.8.2.3.
c=IN IP4 10.10.0.165.
t=0 0.
m=audio 14354 RTP/AVP 8.
a=rtpmap:8 PCMA/8000.
a=silenceSupp:off - - - -.
a=ptime:20.
a=sendrecv.


U 10.10.0.165:5060 -> 10.10.0.142:5064
INVITE sip:300 at 10.10.0.142:5064 SIP/2.0.
Via: SIP/2.0/UDP 10.10.0.165:5060;branch=z9hG4bK1c16a556.
Max-Forwards: 70.
From: "200" <sip:200 at 10.10.0.165>;tag=as08a90e13.
To: <sip:300 at 10.10.0.142:5064>;tag=2be4485acdb9d4a5i4.
Contact: <sip:200 at 10.10.0.165:5060>.
Call-ID: 19bb6dd729b44c10706c9aa868aa3902 at 10.10.0.165:5060.
CSeq: 106 INVITE.
User-Agent: Asterisk PBX 1.8.2.3.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH.
Supported: replaces, timer.
Remote-Party-ID: "100"
<sip:100 at 10.10.0.165>;party=calling;privacy=off;screen=no.
Content-Type: application/sdp.
Content-Length: 205.
.
v=0.
o=root 2032588221 2032588225 IN IP4 10.10.0.167.
s=Asterisk PBX 1.8.2.3.
c=IN IP4 10.10.0.167.
t=0 0.
m=audio 16834 RTP/AVP 8.
a=rtpmap:8 PCMA/8000.
a=silenceSupp:off - - - -.
a=ptime:20.
a=sendrecv.


And the peer can't handle this:

U 10.10.0.142:5064 -> 10.10.0.165:5060
SIP/2.0 500 Internal Server Error.
To: <sip:300 at 10.10.0.142:5064>;tag=2be4485acdb9d4a5i4.
From: "200" <sip:200 at 10.10.0.165>;tag=as08a90e13.
Call-ID: 19bb6dd729b44c10706c9aa868aa3902 at 10.10.0.165:5060.
CSeq: 106 INVITE.
Via: SIP/2.0/UDP 10.10.0.165:5060;branch=z9hG4bK1c16a556.
Server: Cisco/SPA509G-7.4.3.
Content-Length: 0.
.


So, there are no locks but call hangs up. (Using Cisco Spa5XX)


Have you tried directmedia=update ?? (dtmfmode=info) 
That is the issue of 0018734


Thanks. 

Issue History 
Date Modified    Username       Field                    Change               
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2011-02-18 05:00 Irontec        Note Added: 0132128                          
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