[asterisk-bugs] [Asterisk 0017886]: Sip Reason header "Call completed elsewhere" is not passed through local channel in queues

[Asterisk Bug Tracker]

The following issue has been UPDATED. 
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https://issues.asterisk.org/view.php?id=17886 
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Reported By:                francesco_r
Assigned To:                
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Project:                    Asterisk
Issue ID:                   17886
Category:                   Applications/app_queue
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     acknowledged
Asterisk Version:           1.6.2.11 
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:             2010-08-19 08:16 CDT
Last Modified:              2010-08-24 13:43 CDT
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Summary:                    Sip Reason header "Call completed elsewhere" is not
passed through local channel in queues
Description: 
If i put in a queue two members with local channels, the sip reason header
is not passed to the under layer sip channel, so if some one answer the
queue with ringall strategy, the other phones lists the call as missed. In
other words the SIP "Call completed elsewhere" is not trasmitted through
local channels.

If i set a queue like this:

299 has 0 calls (max unlimited) in 'ringall' strategy (0s holdtime, 0s
talktime), W:0, C:1, A:0, SL:100.0% within 60s
   Members:
      SIP/203 (Not in use) has taken no calls yet
      SIP/202 (Not in use) has taken 1 calls (last was 437 secs ago)
   No Callers

all works as expected. Enabling the sip debug i can see:

Reliably Transmitting (NAT) to 192.168.1.213:5060:
CANCEL sip:203 at 192.168.1.213:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK70d70222;rport
Max-Forwards: 70
From: "201" <sip:201 at 192.168.1.111>;tag=as405f3769
To: <sip:203 at 192.168.1.213:5060;transport=udp>
Call-ID: 53923e6f0bb1583f320936cd7bc17346 at 192.168.1.111
CSeq: 102 CANCEL
User-Agent: Asterisk PBX 1.6.2.11
Reason: SIP;cause=200;text="Call completed elsewhere"
Content-Length: 0

If i set the same queue like this:

299 has 0 calls (max unlimited) in 'ringall' strategy (0s holdtime, 0s
talktime), W:0, C:0, A:0, SL:0.0% within 60s
   Members:
      Local/203 at from-queue/n (Not in use) has taken no calls yet
      Local/202 at from-queue/n (Not in use) has taken no calls yet
   No Callers

the sip cancel is without Reason:

Reliably Transmitting (NAT) to 192.168.1.213:5060:
CANCEL sip:203 at 192.168.1.213:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK5e0e1565;rport
Max-Forwards: 70
From: "201" <sip:201 at 192.168.1.111>;tag=as2330a1d9
To: <sip:203 at 192.168.1.213:5060;transport=udp>
Call-ID: 3f9298175c425a3d5b4a517d6360e1c4 at 192.168.1.111
CSeq: 102 CANCEL
User-Agent: Asterisk PBX 1.6.2.11
Content-Length: 0

I have attached two log of a call coming from queue 299 with two members
202 and 203 exposing the different behaviours.
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Issue History 
Date Modified    Username       Field                    Change               
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2010-08-24 13:43 lmadsen        Description Updated                          
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