[asterisk-users] Traffic Crossover

[Eric Wieling]

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.


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From: asterisk-users-bounces at lists.digium.com [asterisk-users-bounces at lists.digium.com] On Behalf Of Geoffrey Yeoh [pbyeoh at gelxis.co.uk]
Sent: Monday, April 15, 2013 7:28 PM
To: asterisk-users at lists.digium.com
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