search for: telecom

Hello everyone. I want to migrate an old PBX which uses the ?1-PRI from one Telecom to VoIP by transparently moving the numbers one by one. I mean that the numbers that the PBX handle must be transparently moved from one operator to another. The old connection to the PBX is ?1-PRI and we must preserve that because no one knows how to configure this PBX. So my idea is to conne...

Will M$ ever stop!!.. Whats the bet their telecoms products will use non-standard protocols.. I really wouldn't like to run a telecom system on Windoze in the first place.. Full Story.. http://news.zdnet.co.uk/communications/0,39020336,39117099,00.htm

...ning an * box with just 8 extensions connected to our old Alcatel BCN 5200 PABX. The requirement is that we now scale it up to handle about 300 lines and get rid of our old PABX. Is there a way of hooking up 300 phones to asterisk without going via the PABX. I am more of a network person than a telecomms one so i may not fully understand some technologies. I note that an E1 channel bank might be used with FXS interfaces but looks like I may need a whole bunch of these to get than many lines. Any help/ponters will be appreciated Many thanks Anthony - -- I always tell the truth, even when I...

Hello, I recently bought a Junghanns Octobri Card. I have some problems with this card to make outbound calls but I can receive calls. I have 3 lines to PSTN and 3 lines to my existing PBX FRANCE TELECOM <-- OctoBRI --> Asterisk 1.2.4-BRIstuffed-0.3.0-PRE-1h <-- OctoBRI --> PABX e-Generis <----> ISDN Phones | | SIP Phones France Telecom --> SIP Phones :...

Disabled echo canceller because of tone (tx) on channel 10 I understand that the PSTN companies use their own echo canceller's, send a tone across 2100hz, the problem we're having is people are complaining of echo on random calls. I'm assuming this may be the cause. Is their anyway to 'ignore' the disabling of EC? Or would be just be a manual code change.. Matt

...So when this is ran it will cut the cdr and the s will show the actual DID not the s correct? But then the NoOp would be something like: .... 1,NoOp(s,1,Answer) yada yada yada ... Correct? .o-------------------------------------------------------o. Brian Fertig Network Engineer Planet Telecom, Inc. Tampa, FL Office 813.901.5182x107 Office 813.864.3164 Direct 813.817.9961 Cellular 813.881.9762 Fax Web: www.planet-telecom.com email: brian@planet-telecom.com -------------------------->IM's<--------------------------- MSN: brian@planet-telecom.com AIM: ptelebrian Yahoo: ptele_bria...

Hello all, Voisonics is pleased to introduce easySysAdmin, an automated support/security platform, designed to save your engineer's time and prevent hacking attempts and telecom fraud. It comprises of an online service run by us, and a lightweight and easy-to-install client on your side. Specifically of interest to Asterisk users is the monitoring of SIP registrations, and automatic blocking of repeated failed attempts. This is done though a local application on your serv...

I've followed the recent thread on caller id with UK British Telecom networks (where the caller id data is delivered before the first ring). My understanding is that if I use a recent CVS head (e.g. CVS-HEAD-09/18/04-17:45:52) and a TDM400 with FXO modules, all I need to do is include the line: usecallerid=uk In my zapata.conf (in the [channels] section) I'v...

Okay, my last post for the night. So, after tweaking my iptables rules I placed a test call from my * -> Vonage softphone account, to my Vonage main account (which uses the Motorola device). Once connected I tried punching a few DTMF's to check the quality and found I had been transferred to a fax machine. ???? Did I find a secret entrance to some fax machine at Vonage, or

...t, I am sending kill -HUP to the process its not using the newly created messages file again. Could someone help me out with how I can rotate asterisk's log's without killing the process? ..o-------------------------------------------------------o. Brian Fertig NOC/Network Engineer Planet Telecom, Inc. Tampa, FL Office This email was scanned by: Mcafee GroupShield ---------------- CONFIDENTIAL DISCLAMER ---------------- All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other t...

...n planning for my new asterisk environment, I would like to have some nice presentation graphics for "breaking down" the system to management here where I work. I like to use Visio for these kinds of things, and was wondering if anyone had a good source for Visio stencils that fit into a telecom plan. There was a set from Cisco, and it'll probably be 'good enough' for what I need, but I thought I would ask the community. Thanks everyone! - Don

Ciao, I have an Asterisk 1.2.9.1 box with a beronet dualBRI (install-misdn-queue) on a Debian distro. I'm experiencing problem with some Telecom Italia lines: some people cannot choose menu selections or extensions after hearing intro message. Is there anybody who knows if there is a particular parameter to set inside misdn.conf (or maybe some other configuration files) ? TIA Giorgio Incantalupo

Hi, For those who heard of the recent France Telecom IPR disclosure against Opus, here's our response: https://hacks.mozilla.org/2013/02/defending-opus/ Cheers, Jean-Marc

Hi all, We've installed Asterisk on a rural development project and we're testing a prepaid phone service. As far as now we're having terrific service results but there's a problem with the calls billing at our local telecom. For instance, a farmer buys a 1 dollar phone card and use it to dial a USA number, the call should lasts for 60 seconds. Asterisk is doing a great job finishing the call exactly at 60 seconds. The problem is that the telecom company billing system adds a two second delay for each call, so the bill...

...so the call fails. And this is not a single failure, it happens every time. </QUOTE> We were finally able to track down this problem and solve it. I'm not sure whether it is only related to ISDN, but maybe some of you find this explanation helpful. This description deals with the German Telecom, as far as I know Colt Telecom handles things the same way. There are two ways to transfer a telephone number, either digit by digit or as a whole block. Mobile phone company e.g. always transfer number as a block. The German Telecom is able to switch from digit-by-digit (which is the default...

Hi, Someone reported me that from a PBX on which someone gained fraudulent access, he could observe hundreds of calls to the same destination number. For curiosity's sake, I'm wondering why would this happen (dialing the same number over and over) ? Some special numbers generate here and there revenues for callees (and not for callers). Beside sharing interests with the callee that get

...rated over energy wasted on pedantic top/bottom posting crap rather than understanding the technology and industry best-practices which have been built upon for years. I'm not against change - far from it. I'm against throwing out good work and history done by an entire industry to make telecom one of the most complex and yet stable computing environments (class 4/5 entrants from Nortel, Lucent, etc.) We should learn from and extend best practices where they do not address circumstances which weren't available 20 years ago (or more) but not to ignore proven practices simply becaus...

Hi everyone! Here is my configuration and messages taken from Asterisk startup. The E1 PRI trunk is connected to our national telecom company here in Croatia. When I call from outside over this trunk to my company I get 'error in connection' respnse. In the same moment I can't see anything in Asterisk, nothing that will tell me that the call reached Asterisk. I think there is a problem with PRI synchronization or PRI...

I currently have a single server with a few SIP and IAX upstreams for origination and termination with IAX clients. I am adding a second server that will have a much higher capacity and will be handling a larger call volume. However, this second server is not going to be geographically near the first. It will largely share the same upstreams. I would like for this to be an integrated system

...r the Digium DS3 PCI Channelized Voice card is EXCELLENT! :):) I might, at best, add a SER server to feed my Asterisk, but beyond that, the above is E X A C T L Y what I am doing! :) Sincerely, SoftwareRadioGuy >From: Michael Welter <mike@introspect.com> >Subject: [Asterisk-Users] Telecom SIP termination vs. >DS3 >To: Asterisk Users Mailing List - Non-Commercial >Discussion <asterisk-users@lists.digium.com>, >Commercial and Business-Oriented Asterisk Discussion ><asterisk-biz@lists.digium.com> >Message-ID: <428C2F42.9050704@introspect.com> >...