[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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call routing based on CID
Hi,
I've been trying to route incoming calls based on CID to a trunk but the
calls are not getting though. I am trying to use a wild card prefix based
on countries so I can point the call to the appropriate trunk.
I am running Asterisk 1.8 with FreePBX.
Here is a sample of my configuration in extentions_custom.conf
exten = _00336123412xx/44XX.,1,Set(RINGTIME=90,g)
exten = _00336123412xx/44XX.,n,Answer(10) exten =
_00336123412xx/44XX.,n,Dial(SIP/trunk01/${EXTEN},${RINGTIME})
exten = _00336123412xx/44XX.,n,Hangup()
Any kind suggestions is appreciated. Thanks.
Best Regards,
Geoffrey Yeoh
--
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[asterisk-users] Call routing based on CID
Thanks Eric. That works.
--
Try: exten = _00336123412xx/_44XX.,1,Set(RINGTIME=90,g)
Notice the _ on your callerid pattern
-Original Message-
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Geoffrey
Yeoh
Sent: Thursday, October 11, 2012 1:15 PM
To: asterisk-users at lists.digium.com
Subject: Re: [asterisk-users] Call routing based on CID
Hi,
I've been trying to route incoming calls based on CID to a trunk but the
calls are not getting though. I am trying to use a wild card prefix based
on countries so I can point the call to the appropriate trunk.
I am running Asterisk 1.8 with FreePBX.
Here is a sample of my configuration in extentions_custom.conf
exten = _00336123412xx/44XX.,1,Set(RINGTIME=90,g)
exten = _00336123412xx/44XX.,n,Answer(10) exten =
_00336123412xx/44XX.,n,Dial(SIP/trunk01/${EXTEN},${RINGTIME})
exten = _00336123412xx/44XX.,n,Hangup()
Any kind suggestions is appreciated. Thanks.
Best Regards,
Geoffrey Yeoh
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Complex Dialplan Help Needed
Hello all, I have a project which requires me to rout calls from ten blocks of sequential numbers i.e. 02081000100 - 02081000200 (each block - 100 numbers) coming in from a telco gateway via Dahdi-SS7 to 10 specific numbers outside the box through two to three SIP trunks (trunk 2 and 3 will be spare capacity/redundant for trunk 1). CLI is crucial here as I need to forward the CLI of the numbers from the blocks of numbers from the SS7 gateway, not the CLI of the originating caller. The Asterisk is behind a firewall with NAT setup. The traffic is one way only. Calls going to the switch goes to Asterisk, Asterisk accepts the call, looks at the CLI from the line (not the caller), routs the call to its assigned outside number through the primary trunk. If primary trunk is unavailable, trixbox will then rout the call to the spare trunks on the list. Hope anyone who has setup this before could give me some good tips on how to set this up. Geoffrey -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
