[asterisk-bugs] [Asterisk 0018227]: CONNECTEDLINE is not working at all if do not introduce a simple SIP call

Asterisk Bug Tracker noreply at bugs.digium.com
Thu Oct 28 15:47:39 CDT 2010


A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=18227 
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Reported By:                voipgate
Assigned To:                
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Project:                    Asterisk
Issue ID:                   18227
Category:                   Functions/func_callerid
Reproducibility:            have not tried
Severity:                   major
Priority:                   normal
Status:                     new
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:              
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Date Submitted:             2010-10-28 06:38 CDT
Last Modified:              2010-10-28 15:47 CDT
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Summary:                    CONNECTEDLINE is not working at all if do not
introduce a simple SIP call
Description: 
I am testing CONNECTEDLINE on asterisk 1.8.0. My phones AASTRA and Polycom
support CALLERID updates on outbound calls.

What I try to achieve is to rewrite the name on the display of the phone
on outbound calls. 

Here is what I tried and which is not working. I used tshark to confirm
that asterisk is not sending any packets with a P-Asserted header.

Set("SIP/mixdev1_601-00000091", "CONNECTEDLINE(num)=3133731337") 
Set("SIP/mixdev1_601-00000091", "CONNECTEDLINE(name)=HUMBALA TEST")
Dial("SIP/mixdev1_601-00000091", "SIP/VOIPmixdev1-4/20333000") 

Here is another example that did not work either.
Set("SIP/mixdev1_601-00000091", "CONNECTEDLINE(num)=3133731337") 
Set("SIP/mixdev1_601-00000091", "CONNECTEDLINE(name)=HUMBALA TEST")
Echo()


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---------------------------------------------------------------------- 
 (0128450) voipgate (reporter) - 2010-10-28 15:47
 https://issues.asterisk.org/view.php?id=18227#c128450 
---------------------------------------------------------------------- 
Not every call ends in the DIAL application. Asterisk can do much more like
Meetme. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-10-28 15:47 voipgate       Note Added: 0128450                          
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