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Summary:ASTERISK-14246: wrong sip port number display
Reporter:pj (pj)Labels:
Date Opened:2009-06-01 10:52:22Date Closed:2009-06-02 08:13:37
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/General
Versions:Frequency of
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Description:"sip show settings" output shows rtp port number used in last sip call, instead of sip signaling port. This port number is changing after every sip call and it's from range specified in rtp.conf.
I'm using canreinvite=no, so asterisk doing rtp proxy for sip2sip call.


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

sip show settings

Global Settings:
----------------
 UDP SIP Port:           17020  <<-----
 UDP Bindaddress:        0.0.0.0
 TCP SIP Port:           5060
 TCP Bindaddress:        0.0.0.0
 TLS SIP Port:           Disabled
Comments:By: Digium Subversion (svnbot) 2009-06-02 08:13:00

Repository: asterisk
Revision: 198762

U   trunk/main/rtp_engine.c

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r198762 | file | 2009-06-02 08:13:00 -0500 (Tue, 02 Jun 2009) | 5 lines

Fix a bug where we were passing in address information that should remain unmodified to a function that may modify it.

(closes issue ASTERISK-14246)
Reported by: pj

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http://svn.digium.com/view/asterisk?view=rev&revision=198762
By: Digium Subversion (svnbot) 2009-06-02 08:13:37

Repository: asterisk
Revision: 198763

_U  branches/1.6.2/

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r198763 | file | 2009-06-02 08:13:37 -0500 (Tue, 02 Jun 2009) | 11 lines

Blocked revisions 198762 via svnmerge

........
 r198762 | file | 2009-06-02 10:12:59 -0300 (Tue, 02 Jun 2009) | 5 lines
 
 Fix a bug where we were passing in address information that should remain unmodified to a function that may modify it.
 
 (closes issue ASTERISK-14246)
 Reported by: pj
........

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http://svn.digium.com/view/asterisk?view=rev&revision=198763