[asterisk-bugs] [Asterisk 0017805]: SIP REFER auth fails, RTP timeout ignored, and other discrepancies between e/ingress calls

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=17805 
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Reported By:                kkm
Assigned To:                
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Project:                    Asterisk
Issue ID:                   17805
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     acknowledged
Asterisk Version:           SVN 
JIRA:                       SWP-2015 
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!): 281083 
Request Review:              
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Date Submitted:             2010-08-06 01:22 CDT
Last Modified:              2010-08-06 18:01 CDT
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Summary:                    SIP REFER auth fails, RTP timeout ignored, and other
discrepancies between e/ingress calls
Description: 
The following per-peer parameters in sip.conf are ignored for incoming
calls, but applied correctly to outgoing calls:

auth= :: if incoming call is transferred by Transfer() dialplan
application, user name and secret not used when constructing the digest in
reply to the 407 response.

rtptimeout= :: overrides global value of the same for outgoing calls but
not for incoming calls.
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 (0125673) kkm (reporter) - 2010-08-06 18:01
 https://issues.asterisk.org/view.php?id=17805#c125673 
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I am sorry, that's my bad -- I really chose a confusing separator symbol.
No IPv6 around here... 

Issue History 
Date Modified    Username       Field                    Change               
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2010-08-06 18:01 kkm            Note Added: 0125673                          
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