[asterisk-bugs] [Asterisk 0019036]: fromuser not respected during OPTIONS message (qualify)

Asterisk Bug Tracker noreply at bugs.digium.com
Tue Apr 5 20:48:11 CDT 2011


A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=19036 
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Reported By:                jkister
Assigned To:                
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Project:                    Asterisk
Issue ID:                   19036
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   trivial
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.8.2.3 
JIRA:                        
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:             2011-03-29 13:43 CDT
Last Modified:              2011-04-05 20:48 CDT
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Summary:                    fromuser not respected during OPTIONS message
(qualify)
Description: 
register = 2155551941:123456 at 10.0.138.226/2155551941~600

[peer](!)
type=peer
context=inbound
qualify=yes
qualifyfreq=300
insecure=port,invite
nat=yes
outgoinglimit=4
incominglimit=4

[mypeer](peer)
host=10.0.138.226
defaultuser=2155551941
fromuser=2155551941
md5secret=023f30a320a5781e8ffd1af9888012af
incominglimit=10

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 (0133420) jkister (reporter) - 2011-04-05 20:48
 https://issues.asterisk.org/view.php?id=19036#c133420 
---------------------------------------------------------------------- 
i dont see how that's any different from what i've provided, but sure.


pbx1*CLI> sip set debug peer mypeer
SIP Debugging Enabled for IP: 10.0.138.226
pbx1*CLI> sip qualify peer mypeer
Reliably Transmitting (NAT) to 10.0.138.226:5060:
OPTIONS sip:10.0.138.226 SIP/2.0
Via: SIP/2.0/UDP 10.0.1.3:5060;branch=z9hG4bK25e0a3b6;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk at 10.0.1.3>;tag=as55e1bb2c
To: <sip:10.0.138.226>
Contact: <sip:asterisk at 10.0.1.3:5060>
Call-ID: 016369855b5cf1fd11dad4921666ca14 at 10.0.1.3:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.2.3
Date: Wed, 06 Apr 2011 01:44:45 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces
Content-Length: 0


---

<--- SIP read from UDP:10.0.138.226:5060 --->
SIP/2.0 403 From: URI not recognized
Via: SIP/2.0/UDP
10.0.1.3:5060;received=10.0.1.3;branch=z9hG4bK25e0a3b6;rport=5060
From: "asterisk" <sip:asterisk at 10.0.1.3>;tag=as55e1bb2c
To: <sip:10.0.138.226>;tag=metaswitch+1+0+e02fc5f2
Call-ID: 016369855b5cf1fd11dad4921666ca14 at 10.0.1.3:5060
CSeq: 102 OPTIONS
Server: DC-SIP/2.0
Organization: 
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog
'016369855b5cf1fd11dad4921666ca14 at 10.0.1.3:5060' Method: OPTIONS
pbx1*CLI> 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2011-04-05 20:48 jkister        Note Added: 0133420                          
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