[asterisk-bugs] [Asterisk 0012439]: Asterisk does not match trunk.

[noreply at bugs.digium.com]

The following issue has been CLOSED 
====================================================================== 
http://bugs.digium.com/view.php?id=12439 
====================================================================== 
Reported By:                bgitonga
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   12439
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     closed
Asterisk Version:           1.4.19 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
Resolution:                 open
Fixed in Version:           
====================================================================== 
Date Submitted:             04-14-2008 12:45 CDT
Last Modified:              05-13-2008 07:12 CDT
====================================================================== 
Summary:                    Asterisk does not match trunk.
Description: 
I’m reopening bug 12417 as the patch in 12362 only fixes the crash. Now
asterisk does not crash but does not get the trunk from real time. I get
the following error.
No such host: trunk1. 
Using the same configuration and database on previous releases of branch
1.4.

Below is my original post. This time (after applying patch in bug 12362)
asterisk does not crash but returns the error “No such host: trunk1”
------------------------------------------------------------------------------------------------------------------------------
I’m using asterisk realtime with Postgres and UnixODBC.
When you have the trunk defines in realtime, and dial a number, asterisk
crashes, but when you have the trunk defines in sip.conf, it works ok.


For example dialing using the following defined in sip.conf will dial out
ok but the database entry does not work in realtime and causes asterisk to
crash. This works in asterisk-1.4.18 but not asterisk-1.4.19

Exten=> _1NXNNXXXXXX,1,Dial(SIP/${EXTEN}@trunk1|60)


//This Works
[trunk1]
type=friend
username=username
secret=secret
callerid="Name" <NUM>
host=Provider IP
canreinvite=yes
context=context
dtmfmode=rfc2833
disallow=all
allow=ulaw



//This Does not work
INSERT INTO sip_buddies(name,callerid,canreinvite,context,dtmfmode,host,
,type,disallow,allow,username,secret)
VALUES
('trunk1','Name <NUM>','yes','context','rfc2833',’Provider
IP','friend','all','ulaw','username','sectret');


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

---------------------------------------------------------------------- 
 jmls - 05-13-08 07:12  
---------------------------------------------------------------------- 
at user request 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
05-13-08 07:12  jmls           Status                   new => closed       
05-13-08 07:12  jmls           Note Added: 0086771                          
======================================================================