similar to: How to configure Asterisk as a VoIP(SIP) to PSTN Gateway?

Hello, I'm very new with * and I would really appreciate some help to implement a SIP to PSTN Gateway. My current scenario includes an * box with a TE405P board. I have a 1.5Mb connection to the outside world (using a router with firewall capabilities) and channel banks that allow me to connect the T1s coming out of the TE405 board to the PSTN network (carrier). I need to configure * to

Hello, I need help in creating a simple PSTN Gateway. This is the scenario: -I have one client sending me VoIP traffic (they don't have asterisk, so IAX is out of the picture for me) and I need to validate that traffic (only accept calls coming from his IP). After that I would terminate the calls to the PSTN network and keep logs for billing purposes. -I have a TE405P board and

Hello, We use all SCSI PCI card hardware RAIDs on all 4 of our production Asterisk servers. They all have Digium quad T1 cards and they all have from 2 to 4 T1s hooked up to them. We have had no noticable problems with dropped calls/poor quality. What are you looking to do with this system? what kind of traffic will be going through these 4 T1s? MATT--- -----Original Message----- From: Deon

Hello, I'm running * on a very basic configuration. I have a Wildcard TE405P with the first T1 connected to a PRI line and the remaining three to Adtran TA750 channel banks with FXS modules. I successfully configured everything to work with a couple of Swissvoice IP10S handsets (MGCP) and analog extensions connected to the channel banks. The problem I'm having is that when I pick up any

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

Hi all, I'm having some problems getting TDMoE setup for the 1st time. I have a TE405P installed in the main server with an ethernet cross-connection to the secondary machine. (Yes, I know about IAX2 but I want to use TDMoE to simulate using T1s.) I'm using -HEAD from yesterday. On the main machine /etc/zaptel.conf: loadzone = us defaultzone=us

We're having some difficulty connecting a T1 fax board card to asterisk, with E&M Wink. We've successfully connected it to other E&M Wink T1s, but have been unable to get asterisk to hear it's dialing. Asterisk can dial down to it with no problem, and it hears all the digits, but whenever it sends digits down to asterisk, * always only "gets" the first one. This is

Did anybody know, Is it possible to establish a ISDN DIAL up Connection and Analog Dial up Connection (V90) trough asterisk with Digium TE405? Thanks a lot for help. Nico Giefing -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060328/3b657058/attachment.htm

Hi, I would like to realize a voip testbed that should simulate the scenario in which two companies have an asterisk PBX connected through a PRI-ISDN to the Telco operator. I have no experience of T1/E1 connection but I think that the above could be relized with 3 asterisk boxes equipped with Digium TE405P cards. One of the box will represent the Telco, the other two, the two companies PBX. I

worked came to one ring only now. Thank you very much. If I use TE410 or TE405 instead of X100P. do it make that first ring disappear? Shakil -----Original Message----- From: tony@softins.clara.co.uk [mailto:tony@softins.clara.co.uk] Sent: Tuesday, April 20, 2004 12:27 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Re: Auto Answering PSTN --> Asterisk using X100PCard In

Hello, We just had some new T1s turned up today to replace others that our contract has run out on and we are now getting more echo on the new T1 lines than we had on the old ones. The T1 type is 24-channel, D4/AMI SF Robbed-bit(the same as the T1s they replaced) The problem is that we are getting echo on about 10% of the calls in and out placed on these new T1s compared to less than 1% with

Hello, Several people asked to get a hold of the stats I used to determine that the Sangoma T1/E1 boards performed better in our real-world tests than the Digium boards. We've decided to post our results after confirming them over the past month of operations. Here is a link to the last two reviews I wrote about Sangoma with Asterisk:

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, 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

Greetings, We are currently testing a Sipura SPA-3000 as a gateway from our Asterisk system to a PSTN line for 911 access. We have a number of locations and want to place an SPA-3000 in each, connected to a PSTN line that will provide the correct ANI/ALI information to 911 for each location. It all works great, except for a reasonably significant (4 seconds) delay between when the SPA-3000

Hello, I'm looking at building a couple new PRI Gateway boxes using TE405P cards, and was wondering if anyone has had any experiences (good or bad) with the Intel SE7210TP1_E motherboards from Intel. General Technics builds some really nice (and cost effective) 1U servers based on the board: Server: http://www.gtweb.net/gt637.html Specs:

When dtmfmode is set to inband for SIP, and i originate a call from sip out to the PSTN, I can hear the DTMF digit twice in the audio stream. Once very briefly and once for normal duration. Our Theory: While Asterisk is parsing the DTMF, for a fraction of a second, while the end user generated DTMF is being detected, the DTMF is passed inband. Once the DTMF is detected Asterisk silences it

Hi all, I am handed a project to setup *. The requirement is that it can handle 8 T1s. Half of the calls coming into the system will be routed to SIP extensions (with transcoding). The machine we have in our disposal is a new dual Xeon 3.2gHz server with 2g of ram and an dual 1000mb nic. Voice will be coming in from the PSTN (through 2 quad digium cards) in g711ulaw, and most of the time will

You can even set it to zero. Mine works well when in zero. The line pick up immediately :> -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Chris Stenton Sent: Thursday, February 09, 2006 6:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SPA-3000 VOIP-PSTN

My Sangoma Experience in Asterisk: Followup 2005-05-06 original review can be found at: http://astguiclient.sourceforge.net/Sangoma_experience.txt It's now been a month since I finished my last round of tests with the Sangoma A104u board in Asterisk. I have had a lot of conversations with the guys at Sangoma about how my system runs so that they could set up their test systems to try to