Any time something happens "around 20 mins", check your session-timers in 
sip.conf and on your other SIP devices.

If you specify your peer as a hostname, make sure you do NOT have multiple A 
records for that hostname.


________________________________
From: [email protected] 
[[email protected]] On Behalf Of Geoffrey Yeoh 
[[email protected]]
Sent: Monday, April 15, 2013 7:28 PM
To: [email protected]
Subject: [asterisk-users] Traffic Crossover

Hi all,

I am having this problems for a while and could not figure out the cause of 
this.

I have FreePBX version of Asterisk (1.8.11-cert) routing calls to 10 different 
FreePBX servers (same version of Asterisk) depending on the destination 
numbers.  The incoming calls into the main Asterisk server with 4 x  Sangoma 
A102 E1 card are coming through a SS7 link from an ISUP interface of a telecom 
grade Qualcomm gateway switch. There is also a SIP B2BUA server and 
PortaBilling gateway at the very end-point before the call is passed on to the 
end destination.

Issue #1
All calls routed by the main server to the different FreePBX kind of go weird 
after roughly 20 minutes.  The calls will get disconnected from the telecom 
gateway switch but the calls is still showing as connected on the main Asterisk 
servers and end-point Asterisk servers.

Issue #2
The second issue occurs around the same time roughly after 20 minutes.  Some 
current calls to one end-point Asterisk server will start getting audio from 
another call from another separate end-point Asterisk server. I’ve tried 
disabling RTP from each point of the connection but still no joy.

I hope somebody could give some pointers or direction on how to troubleshoot 
this.

Best Regards,
Geoffrey


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