[asterisk-bugs] [Asterisk 0016218]: Asterisk does not connect the call to internal extension

[Asterisk Bug Tracker]

The following issue has been UPDATED. 
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https://issues.asterisk.org/view.php?id=16218 
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Reported By:                mrmrmrmr
Assigned To:                
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Project:                    Asterisk
Issue ID:                   16218
Category:                   Channels/chan_sip/General
Reproducibility:            sometimes
Severity:                   major
Priority:                   normal
Status:                     closed
Asterisk Version:           SVN 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
Resolution:                 no change required
Fixed in Version:           
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Date Submitted:             2009-11-10 13:47 CST
Last Modified:              2011-01-28 11:16 CST
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Summary:                    Asterisk does not connect the call to internal
extension
Description: 
The following issue occurs in most of my calls received from an external
SIP proxy. So it is highly reproducable.
Very rarely, the call reaches the internal extension. 

Let me first describe the structure of my Asterisk Proxy:

It is running on a CentOS server with 2.6.30.4 kernel.
The server has 2 interfaces ; ppp0 accesses the external SIP proxy over
internet and has real IP address. br0 is the LAN interface which my SIP
clients are connected. Internal SIP clients are Linksys SPA3000 and Linksys
PAP2.
There is Shorewall firewall running on the server, but there is no NAT
either for the external call legs or internal call legs.

In the problematic call scenario, call is coming from an external SIP
proxy to my Asterisk server. The incoming rule sends the call to a ring
group (600) which should forward the call to 2 internal extensions (995 and
990).

Sometimes, the call is not connected to the internal clients. They don't
receive any SIP INVITE.
When I check the logs on Asterisk console I see "AGI Script
dialparties.agi completed, returning -1"
But I don't have any clue why dialparties.agi script returns "-1"
I am attaching the agi debug output to this issue.
I will also attach my existing sip configuration.

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Issue History 
Date Modified    Username       Field                    Change               
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2011-01-28 11:16 lmadsen        Status                   new => closed       
2011-01-28 11:16 lmadsen        Resolution               reopened => no change
required
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