search for: telecoms

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 connect a PC with 2

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

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello I am running 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

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 |

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

I don't completely understand this.. Lemme try it out.. [default] exten => 1112223333,1,Macro(happy-did) [macro-happy-did] exten => s,1,Goto(${MACRO_EXTEN},1) exten => _XXXXXXXXXX,1,NoOp(Normal "s" exten stuff here) 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: ....

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

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've done this, but I get: Sep

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

I created some scripts to logrotate. I am having a problem. After I do it, 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

Hello Everyone! In 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

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

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

Hi, in december last year I posted the following problem: <QUOTE> When I dial the number of our client, located in another town, I get a connection to the asterisk server, I can talk to my client or listen to his mailbox. If someone in the town of this client calls him, he gets the ISDN error "service not available". With some more debugging we saw what happens with these

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

OK. I'm getting out the fireproof suit because it's coming and my hackles have been raised by a number of comments on the list of late. Disclaimer: No disrespect intended to the individuals of any *specific* thread. I'm a little frustrated over energy wasted on pedantic top/bottom posting crap rather than understanding the technology and industry best-practices which have been

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

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

Message: 16 Date: Thu, 19 May 2005 00:16:34 -0600 Michael, Do both! As for Sip Termination: ----------------------- Contact Kristi Eggers @ Txlink.net for month to month Originating/Termination VoIP Toll Free or Local USA DID #s. Yes they do both Sip and IAX. You must have seperate accounts for either Sip or IAX and fund your account with a minimum of $100. This is what I did. Once I get