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Summary:ASTERISK-11422: CID name from sip.conf when no CID name is supplied
Reporter:pj (pj)Labels:
Date Opened:2008-02-12 05:02:05.000-0600Date Closed:2008-02-12 09:09:51.000-0600
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/General
Versions:Frequency of
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Description:when no caller id name is supplied in INVITE from peer, caller id name from sip.conf is ignored

it's very similar bug, that was already reported and fixed:
Fix CID name when no CID name is supplied (bug ASTERISK-2746)

****** ADDITIONAL INFORMATION ******

it can be easy reproductible, so I'm not sending full debug, only relevant part:
when following come in invite, caller id name is used from sip.conf to overwrite supplied "p"
From: "p"<sip:324@xxx.xxx.xx>;tag=674ca20e

when following come in invite, caller id name is ignored, callerid(all) function shows only number.
From: <sip:324@xxx.xxx.xx>;tag=8c0ccd4f


my sip.conf peer konfiguration:
type=peer
host=dynamic
nat=yes
canreinvite=no
callcounter=yes
busylevel=1
secret=xxxx
callerid="test324" <324>
Comments:By: pj (pj) 2008-02-12 05:42:21.000-0600

some options from my [general] section:
match_auth_username=yes
alwaysauthreject=yes
allowsubscribe=no
By: Digium Subversion (svnbot) 2008-02-12 09:07:32.000-0600

Repository: asterisk
Revision: 103385

U   branches/1.4/channels/chan_sip.c

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r103385 | file | 2008-02-12 09:07:31 -0600 (Tue, 12 Feb 2008) | 4 lines

Even if no CallerID name or number has been provided by the remote party still use the configured sip.conf ones.
(closes issue ASTERISK-11422)
Reported by: pj

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http://svn.digium.com/view/asterisk?view=rev&revision=103385
By: Digium Subversion (svnbot) 2008-02-12 09:09:51.000-0600

Repository: asterisk
Revision: 103386

_U  trunk/
U   trunk/channels/chan_sip.c

------------------------------------------------------------------------
r103386 | file | 2008-02-12 09:09:50 -0600 (Tue, 12 Feb 2008) | 12 lines

Merged revisions 103385 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r103385 | file | 2008-02-12 11:09:24 -0400 (Tue, 12 Feb 2008) | 4 lines

Even if no CallerID name or number has been provided by the remote party still use the configured sip.conf ones.
(closes issue ASTERISK-11422)
Reported by: pj

........

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http://svn.digium.com/view/asterisk?view=rev&revision=103386