similar to: Rate file formats: a standard?

Pardon my ignorance, but I was hoping someone could clear up something for me. - For a few POTS lines, digium has a single port card for that, or a T1 card to a channel bank. - For 10 or more lines, digium has a T1 or E1 card for that too based on PRI channels - For 100's to 1000's of lines, I suspect a soft-switch is in order??? A traditional phone company will sell: - POTS lines for

If you are a Canadian VoIP provider or CLEC, help make a difference by joining the CAVP: www.cavp.ca >OMFG, >I thought April Fools day was over. This is hard to believe. If true it >tells me that Bell has not changed at all. They are still trying to >manipulate and take advantage of the parts of the market they have absolute >control over. The CRTC was right to continue to

This is an idea from FreeSWAN, which was implemented in the recently released version 1.0. Basically the idea is that FreeSWAN sites automatically encrypt traffic between them when possible, without having to set up the link ahead of time. How this works is: The sites publish some info in DNS. FreeSWAN gets some traffic destined for that site. - looks up the info in DNS - if the info is there:

Hi, Inspired by a recent rant about one particular provider, I am getting very curious about something I've never mastered. I'd like someone to explain this here or at least post a link or two that can educate me and probably countless others who have no knowledge in this area. I'm sure there are several of you reading this that know all about the subject. What are the various

Greetings folks, I'm currently dealing with a company to let me set Caller-ID-Name on outbound calls. So far pretty happy with their services. The basic service works like this: * CLEC sets Point Code to point to this company * CLEC has to sign LOA saying they give me permission to set the Caller-ID-Name through this company. * I go into web interface and set name. However, the CLEC is

Using a SIP connection with a CLEC, the downstream (received) audio is perfect when the mute button is activated on the phone. However, when there is upstream audio (i.e., talking or even breathing into the microphone), the downstream audio is cut off. It's kinda like having a half-duplex audio connection. When I divert outgoing calls to another provider, these calls are fine.

I have the following configuration: CLEC -> T-1 -> Asterisk -> Adtran Channel Bank -> (analog) -> Nortel Don't complain that it's ugly. I've already done plenty of that. The CLEC manages their Adtran remotely and needs to be able to continue to do so. I assume they use FDL to do the management. We are using the Zaptel DACS/DACSRBS to cross connect some of the

OK.. I'm up to my eyes in LD BS! I can't for the life of me understand how any carrier, either VoIP or traditional service provider can make heads or tails of how to hand off an * based call to an LD provider. Every provider I talk to, says I have to have a traditional T1 put in to their respective networks. I don't want to do this. I want a LD provider that can take a IP, SIP, IAX

I'm in the process of turning up a PRI in one of my markets and have run into a problem I have never seen before. I am unable to place a local outgoing call. Long Distance over the same PRI works fine. When I attempt to place a local call using the PRI I see Asterisk attempt to dial, and am greeted with a busy signal. This signal appears to originate on the telco's switch. I have had

I've been working with Asterisk for a little while now, and have been looking recently at my next career opportunity. It seems from searching the various job sites that the predominant VOIP technology is not the applications-based open source approach we took, but Cisco, with a really heavy emphasis on the networking (ie network engineer) aspect. If you do a job search for (VOIP or

It seems like it might be nice to have a mailing list to talk about (and to) voip providers for Asterisk users. It would be a good place to share info about config, pricing news, customer service, local numbers, transient outages, etc. Providers would be encouraged to contribute sales info. Users would be able to help each other out with technical and non-technical issues. Seems good for

Hi all, I am trying to setup a ZAP interface to do MF signaling for a handoff to a 911 tandem. The signaling I need to perform on the T1 is this: 9-1-1 Tandem: Wink CLEC end office: KP (Keypulse) NPA ST (Start) 9-1-1 Tandem: Wink CLEC end office: KP I (Info Digit) NXX XXXX ST As I'm not as familiar with the Zaptel configurabliity, I'm not really sure how to do this. Do I dial twice

http://www.boingboing.net/2006/06/05/play_zork_by_phone.html Let me preface this idea with one comment: I don't have the time to do this - I don't even have time to eat these days. But someone out there has the cycles to do this... and it would be very cool. OK, so now Zork is attached to Asterisk, but using the less-than-clear Festival engine. There are beta tests of the LumenVox

Hallo, I would like to use to read.xls function from the gdata package to read data from Microsoft Excel files but I experienced a problem: For example I used the following code: testfile<-read.xls("/home/.../wsjecon0603.xls", #file path header=F, dec=",", na.strings="n.a.", skip=5, sheet=2,

I want to test asterisk with its DID functions on a smaller scale without the expense of a PRI. I am in the US, and want the most reliable/stable, and feature rich solution that I can have. What ISDN card would be reccommended? My clec switch operator says he can/will send DID info to my BRI's if I wish. Can Asterisk, or one of its addons support this? What BRI solution would be

My CLEC just called and asked if we will support the "MI2" protocol on our proposed T1 circuit. I think this is for CallerID name. Will the T100P support this? Thanks, Mike

Central business location has a PRI with a CLEC. Remote offices access the PRI for all voice traffic via VoIP. How does one get the telco to report the address of a remote office to the 911 call center when the call is made from that respective location?

What is the deal with "CSR's"? TWTelecom is telling me that I can't port a number to their service without a Customer Service Record. Apparently this is easy with Verizon, and not so easy with some other companies. Basically I'm at a brick wall with a couple of ports because TWTelecom is telling me I HAVE to get a CSR and certain other providers (Time Warner Cable for

We are an asterisk user... small time system 50-100 users or so. But, we have an opportunity to get into a big time telecom activity. It would have 2000 to 30,000 user lines per city, and we would like to have those brought back to a central location for control and because transport can be more economical than remote site rentals, maintenance and personnel. We could take the local lines into

Hi all, Newbie quest here. I have 3 PSTN lines (2430-2432) setup by the CLEC in a hunt group coming in to a TDM04B and 3 Grandstream gxp-2000's (say a,b and c) and Asteriskathome installed. I want all phones (a,b,c) to be able to take calls from the hunt group. Does this mean I must make 9 SIP extensions on * ? 3 for each IP phone? and group them in trtiplets (ring groups?)