similar to: auto authentication for voip client

Hi All, I have one asterisks server and 3 client (i'm using voip sip client for my handphone). I've configured sip.conf and extension.conf with 3 user different. And nothing wrong with them, i could make them to make a call too. what i want to ask is, i was try to use 1 user (ex:1001) in 2 different client. example: client 1 (1001) make a call to client 2 (1002) --> ok then in client

Hi all, I want to make initial VoIP authentication process from asterisk server to be based on EAP-SIM authentication of Freeradius server (so it will be not necessary to insert account datas in the asterisk database). Is there any way of doing that from Freeradius and Asterisk? Or at least, is there any way to sync the EAP-SIM data on Freeradius to asterisk server? thank you -------------- next

Does anyone know a good (and stable) voip gateway product with 4 ports (2 fxo and 2 fxs), with the following requirements: * being able to connect analog phones to the FXS ports, and communicate over SIP with an REGISTRAR/PROXY server (SER in our case). * being able to connect the FXO port to local office PSTN network, and dial to that office pstn number and getting an internal dialtone, or

Peoplefone AG offers Voice over IP(VoIP) services with exceptional rates. Peoplefone is a certified partner of Siemens<http://www.siemens.ch/index.jsp?sdc_p=c175fi1012637lmno1012637psuz1&sdc_sid=1113876080&>and AVM/FRITZ!Box <http://www.fritz-shop.ch/> . Due to our rapid growth, we are currently seeking for: VOIP SPECIALIST Place of work: Zurich *Requirements:* -

Peoplefone AG offers Voice over IP(VoIP) services with exceptional rates. Peoplefone is a certified partner of Siemens<http://www.siemens.ch/index.jsp?sdc_p=c175fi1012637lmno1012637psuz1&sdc_sid=1113876080&>and AVM/FRITZ!Box <http://www.fritz-shop.ch/>. Due to our rapid growth, for our new Polish subsidiary we are currently seeking for: VOIP SPECIALIST Place of work:

Hi, I'm a new user of Aterisk, and I have to configure a VoIP Gateway. I have an Alcatel PBX with an E1 card, connected, for the moment, to a local carrier. I would like work with a french VoIP provider, but, for this, I need to use a VoIP Gateway for connect my E1. Thus, I want to build my own voip gateway. It very simple, I want to connect my PBX to the gateway (E1 link) for both call

We are planning to install a voip system based on asterisk for 2000-3000 retail locations and up to 6000-8000 sip accounts/users. Instead of setting up a new, centralized PSTN gateway, we are intend to use a CISCO gateway/router of an existing CISCO voip solution in the headquarter and we must able to call all CISCO based voip phones in the headquarter running together with a CCM. SIP-Phones

Have you looked at his website, www.netenable.co.uk ? Looks like he pays bills the same way as he answers followups ;-) g -----Original Message----- From: asterisk-users-bounces@lists.digium.com on behalf of Peter Bowyer Sent: Sun 03-Dec-06 8:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] VoIP GSM Gateways Not very good at answering followups to

I'm setting up a demo for two asterisk machines. One will be a central Asterisk server which will handle everything already in VoIP (office-like functions plus agents functionality). The second Asterisk box will be used strictly as a VoIP gateway to the first server. The gateway server will have 4 T1s connected to it and what I was thinking on doing was the following: in

Hello, Long time subscriber/reader of this list - thank you for all the great ideas. Scenario: We currently provide a hosted ACD system using Mitel phones (speaking the Minet protocol) to an NCI based server solution. The logic behind this choice was the emulation of key system features etc... Many of our clients have asked for basic call queue functionality: - Agents having the ability to

I'm looking for recommendations for UK-based VoIP-PSTN gateways. They should ideally offer: - IAX connection - Multiple simultaneous calls on a single account - Lower call rates than BT Business - Auto-top up or invoicing in arrears I can find several that offer one of these facilities, but none that offer all. Thanks! -- David Gurr Congruity Ltd. Hemel Hempstead, UK

Hi people, is there any good document on setup asterisk using only its voip functions , i mean without e1/t1 hardware, i have a cisco 5400 gateway and a cisco 7000 working as gatekeeper, can asterisk register with this GK?, im thinking in using cisco voip phones Any pointer will be apreciated, and sorry for my bad english Miguel

Folks, I am a newbie to the VOIP world and have a question (might as well sound silly to some). I would like to set up a PC-to-Phone call from my desktop to a regular PSTN number. Does the Asterisk PBX itself act as a VOIP-PSTN gateway or do I have to subscribe to a VOIP provider for this? Are there any free IP-PSTN gateways which I can subscribe to? Do I need any specific hardware to set up such

Hi all. I am currently thinking about monitoring a network link (bandwidth, latency, packet drop). I have a centos server which is connected to providers network through a switch. How to test the link? Should I: - use ICMP to ping gateway and collect staistics of time/packet loss/etc. - use software like here: http://www.cyberciti.biz/faq/linux-unix-windows-monitor-outgoing-bandwidth-latency/

hello, I would like to realize a VoIP Gateway, with some extra-features. The aim is to get the phone number of the caller, to make research in our database, and to put him automatically through the good employee. The company is equipped with a VoIP network : software : - PSTNgw - Ohphone - OpenGatekeeper hardware : - Quicknet phonejack - gateway : Voicetroniw OpenLine4 Is it possible to

Hello everyone. This is an other question from a relatively newbie. I'd like to provide auto callback ability for my *. From my mobile I want to be able to call a number on the * and have it call me back on my mobile. I know how to generate a .call file from a script and I know how to call a script from the dialplan (in order to get the .call file generated). I also found the scripts on

Hi, I'm trying to set up a voip gateway between a classic pbx and ip network with asterisk. phones -- pbx -- * -- ip network I would like a prefix ( 0 ) for the classic calls and another prefix ( 1 ) for voip calls. The problem is that pbx can talk with asterisk only with S0 synchro (like a terminal) and succeeded not to make call with prefix in this mode. I also try to consider asterisk

I bought a MV-372 for 2 SIM cards as the one channel model seems to work well (see http://www.voip-info.org/wiki/view/Setup+MV-370+GSM+Gateway+with+Asterisk). The setup is such: --------- Inet --> VoIP provider ---> POTS | | (iax2, NAT) | asterisk (on abox with iptables fw) | (SIP, LAN) |----------> SNOM190 phones | ----------> SIP-GSM-module ---> SIM

<http://www.vistaprint.com/vp/gateway.aspx?S=5176697856> Anyone on the list good with Linksys PAP2NA configuration, I am looking to take my ata's and emulate the operation of a pots phone line as close as I can get. One thing I need to change is the fast busy tone I get when someone hangs up on the call. We are also looking for a Voip/ Asterisk Consultant to set up hardware for a

Hello, I'm researching the possibility of using VOIP (SIP) with an existing PABX system. Ideally, the setup would be to dial an outside line through the PABX (that would actually link to the the VOIP gateway). At this point I would prefer not to purchase a hardware-based VOIP gateway. I would prefer to use a software-based gateway for research & testing purposes. Could anyone please