[asterisk-bugs] [Asterisk 0016585]: [patch] Asterisk does not fully support SIP connections to Internet Telephony Service Providers

Asterisk Bug Tracker noreply at bugs.digium.com
Tue Jan 19 06:01:28 CST 2010


A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=16585 
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Reported By:                Nick_Lewis
Assigned To:                
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Project:                    Asterisk
Issue ID:                   16585
Category:                   Channels/chan_sip/NewFeature
Reproducibility:            always
Severity:                   feature
Priority:                   normal
Status:                     ready for testing
Asterisk Version:           SVN 
JIRA:                       SWP-706 
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:             2010-01-12 05:37 CST
Last Modified:              2010-01-19 06:01 CST
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Summary:                    [patch] Asterisk does not fully support SIP
connections to Internet Telephony Service Providers
Description: 
Asterisk does not fully support SIP connections to ITSPs. For example
chan_sip.c does not support DID calls that have the extension defined only
in the to-header. It also does not support peer matching on the register
contact (callbackexten).

It is generally thought that these features should be captured in a new
peer type such as type=service  
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 (0116880) Nick_Lewis (reporter) - 2010-01-19 06:01
 https://issues.asterisk.org/view.php?id=16585#c116880 
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oej - since you have a veto on this I guess it is your way or the highway
as far as getting stuff into trunk. Please be specific regarding what
system design work and documentation you would like to see for type=service
and I will see whether I can find some time to do it 

Issue History 
Date Modified    Username       Field                    Change               
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2010-01-19 06:01 Nick_Lewis     Note Added: 0116880                          
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