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...th1: 10.2.20.2/30 # First WAN connection eth2: 10.2.20.6/30 # Second WAN connection teql0: 10.2.20.10/30 # TEQL connection On the WAN router I route 10.2.18.0/24 to 10.2.20.10 dev teql0. On the LAN router I route anything from 10.2.18.0/24 to 10.2.20.9 dev teql0. Thanks, Daniel Daniel Bergqvist NetAtOnce AB 0470-72 99 00 www.netatonce.se daniel@netatonce.se

Hey! I'm having a small problem. I'm using Realtime to store SIP account information. Dialing works just fine, but when dialing a person already on the phone I don't get a busy tone. Eg, Phone 100 calls 200 and they chat with each other phone 150 calls 100, and gets a regular ringing tone what I would is for phone 150 to receive a busy tone since phone 100 is already speking with

...if I have a SIP gateway, something like an Audiocodes Mediant 1000. How do I prohibit the caller id presentation on that one? Regards, Kristian -- Kristian Larsson KLL-RIPE Network Engineer Net At Once [AS35706] email: kristian@netatonce.se irc: kll@ircnet phone: +46 470 592717 cell: +46 704 910401

Dear All : I need to link my HQ to some Remote Sites - I need a Router which supports VOIP , and VPN Also the Router Should has 3 FXS ports and 1 FXO ... The call should be routed from the Remote Site to the HQ through a VPN tunnel ( 3DES ) ... Any Advise ? Mohamed Farid ,, Notice: This e-mail (including attachments) is confidential and is intended solely for the addressee. Unless you are

Hi, I have a problem with setting outgoing caller id to "nothing" (secret) on our Wildcard TE405P connected to a swedish euroisdn line. Caller ID seems to work fine when connecting the same line to a Ericsson PBX - so something must be wrong in my settings, but I don't know what. I've tried: exten => _*70X.,1,Set(CALLERID(name)="") exten =>

Hello All, Ive been doing more and more research on trying to setup a cluster/load balancer for Asterisk. All the Asterisk boxes would be using a config that is the same between them all (via a DB), but we want one location to point the phones to, and from there that machine/device will send it to a Asterisk server so the call can be processed. I know you cant balance the whole call, ie: once the

Hello. I need help with limiting bandwidth. I have read every tutorial I''ve come over and I just can''t make anything work. Ok, here''s the scenario: I have a gateway, which has five network interfaces (eth0 -> eth4), eth0 is the ''external'' one and eth[1-4] are supposed to be limited to 128Kbit/s each. The interfaces eth[1-4] each have a C-class

Hi, we are currently considering different options for rolling out a large scale IP PBX to handle around 3,000 + concurrent calls. Can this be done with Asterisk? Has it been done before? I really would like an input on this. Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: