[asterisk-bugs] [Asterisk 0016491]: Asterisk sip.conf realtime register, contact problem

Asterisk Bug Tracker noreply at bugs.digium.com
Mon Dec 21 19:02:23 CST 2009


A NOTE has been added to this issue. 
====================================================================== 
https://issues.asterisk.org/view.php?id=16491 
====================================================================== 
Reported By:                jamicque
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   16491
Category:                   Resources/res_realtime
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     new
Asterisk Version:           1.6.0.20- 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
====================================================================== 
Date Submitted:             2009-12-21 09:59 CST
Last Modified:              2009-12-21 19:02 CST
====================================================================== 
Summary:                    Asterisk sip.conf realtime register, contact problem
Description: 
When I have a config of sip_ps in realtime I've noticed the regestration
problem in sip.conf. 
The config in DB looks like: 
sip.conf	general	register	jamicque:xxx at sip.xxx.xxx:5060/jamicque

than the  register send by asterisk to provider looks like (please look at
the contact header):

REGISTER 13 headers, 0 lines
Reliably Transmitting (no NAT) to 213.218.116.66:5060:
REGISTER sip:sip.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP 10.0.0.199:5060;branch=z9hG4bK40e31a75;rport
Max-Forwards: 70
From: <sip:jamicque at sip.xxx.xxx>;tag=as48167c5e
To: <sip:jamicque at sip.xxx.xxx>
Call-ID: 0a812221399f1a927fb46039266a4136 at 127.0.1.1
CSeq: 104 REGISTER
User-Agent: xxx
Authorization: Digest username="jamicque", realm="sip.xxx.xxx",
algorithm=MD5, uri="sip:sip.freeconet.pl",
nonce="4b2f9aed189e6af29c9b5b4307ff625414486c2b",
response="00f2b29e3a240406fa47bcbdff9e10b6"
Expires: 60
Contact: <sip:s at 10.0.0.199>
Event: registration
Content-Length: 0

When I remove the port information from my reltime registration:
sip.conf	general	register	jamicque:xxx at sip.xxx.xxx/jamicque

the contact in register is ok.




====================================================================== 

---------------------------------------------------------------------- 
 (0115621) wdoekes (reporter) - 2009-12-21 19:02
 https://issues.asterisk.org/view.php?id=16491#c115621 
---------------------------------------------------------------------- 
It works fine over here with asterisk 1.6.1.11 and realtime_static and
res_odbc.

If I append "/abc" after the host/port, I get """Contact:
<sip:abc at 10.20.30.40>""" like expected.


You wouldn't happen to have your var_val column length set to 30
characters, would you? 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-12-21 19:02 wdoekes        Note Added: 0115621                          
======================================================================




More information about the asterisk-bugs mailing list