[asterisk-bugs] [Asterisk 0018367]: [patch] Missing P-Asserted-Identity

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=18367 
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Reported By:                GeorgeKonopacki
Assigned To:                rmudgett
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Project:                    Asterisk
Issue ID:                   18367
Category:                  
Channels/chan_sip/CallCompletionSupplementaryServices
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     assigned
Target Version:             1.8.5
Asterisk Version:           SVN 
JIRA:                       SWP-2649 
Regression:                 No 
Reviewboard Link:           https://reviewboard.asterisk.org/r/1199/ 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2010-11-24 07:19 CST
Last Modified:              2011-05-06 11:19 CDT
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Summary:                    [patch] Missing P-Asserted-Identity
Description: 
Phone A is monitoring Phone B.

Phone B becomes available so the Asterisk server sends a NOTIFY(cc-ready)
to Phone A.

Phone A calls Phone B (using the URI provided by the NOTIFY(cc-ready)).

Phone B receives P-Asserted-Identity in its INVITE message – GOOD

Phone A does NOT receive P-Asserted-Identity in any of its messages –
BAD

This means the Phone A is displaying the 32 digit URI. Phone B displays
the information provided by the P-Asserted-Identity.


SIP.CONF

sendrpid = yes
sendrpid = pai

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 (0134577) svnbot (reporter) - 2011-05-06 11:19
 https://issues.asterisk.org/view.php?id=18367#c134577 
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Repository: asterisk
Revision: 317670

U   branches/1.8/channels/chan_sip.c

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r317670 | rmudgett | 2011-05-06 11:19:19 -0500 (Fri, 06 May 2011) | 22
lines

Fix SIP connected line updates.

This patch fixes a couple SIP connected line update problems:

1) The connected line needs to be updated when the initial INVITE is sent
if there is a peer callerid configured.  Previously, the connected line
information did not get reported until the call was connected so SIP could
not report connected line information in ringing or progress messages.

2) The connected line should not be updated on initial connect if there is
no connected line information.  Previously, all it did was wipe out any
default preset CONNECTEDLINE information set by the dialplan with empty
strings.

(closes issue https://issues.asterisk.org/view.php?id=18367)
Reported by: GeorgeKonopacki
Patches:
      issue18367_v1.8.patch uploaded by rmudgett (license 664)
Tested by: rmudgett

Review: https://reviewboard.asterisk.org/r/1199/

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http://svn.digium.com/view/asterisk?view=rev&revision=317670 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2011-05-06 11:19 svnbot         Checkin                                      
2011-05-06 11:19 svnbot         Note Added: 0134577                          
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