[asterisk-bugs] [Asterisk 0018339]: (Call Completion / SIP) Displaying Information After We Send A INVITE With URI From A NOTIFY(cc-ready)

Asterisk Bug Tracker noreply at bugs.digium.com
Fri Nov 19 10:46:18 CST 2010


The following issue has been RESOLVED. 
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https://issues.asterisk.org/view.php?id=18339 
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Reported By:                GeorgeKonopacki
Assigned To:                mmichelson
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Project:                    Asterisk
Issue ID:                   18339
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   tweak
Priority:                   normal
Status:                     resolved
Asterisk Version:           1.8.0 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
Resolution:                 fixed
Fixed in Version:           
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Date Submitted:             2010-11-19 09:01 CST
Last Modified:              2010-11-19 10:46 CST
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Summary:                    (Call Completion / SIP) Displaying Information After
We Send A INVITE With URI From A NOTIFY(cc-ready)
Description: 
This is NOT as Asterisk bug, but more of a weakness in the bliss call
completion specification.

We send an INVITE to the Asterisk server (Note: The URI was provided by a
NOTIFY (cc-ready) which we receive from the server for call-completion).

When we receive either a 180 Ringing, 200 OK or a 486 Busy Here, we
display on our phone screen the number/uri provided in the 'To' field
(which is the 32 digit uri for call-completion).

Because the Asterisk server knows that the URI provided in the INVITE is
related to call-completion, could we have a 'P-Asserted-Identity' header
added to 180 Ringing / 200 OK/ 486 Busy Here, so we can display something
meaningful to the end user.

This would be fantastic if you could!

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---------------------------------------------------------------------- 
 (0128989) mmichelson (administrator) - 2010-11-19 10:46
 https://issues.asterisk.org/view.php?id=18339#c128989 
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This is entirely doable already. Your best bet is going to be to make use
of the CONNECTEDLINE function in the dialplan. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-11-19 10:46 mmichelson     Note Added: 0128989                          
2010-11-19 10:46 mmichelson     Status                   new => resolved     
2010-11-19 10:46 mmichelson     Resolution               open => fixed       
2010-11-19 10:46 mmichelson     Assigned To               => mmichelson      
======================================================================




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