similar to: What are the various models of DID providers

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

Hello Everyone, We are currently having talks with various service providers, and trying to determine what the best way is to interconnect in order to have access to the PSTN network. As you know there are two ways of doing this: Traditional PRI: Have trunks grouped into a transport layer such as OC3/12. With DIDs attached to the group. As you many know, this approach would also require a POP

Almost all DID providers refer to XO and L3. Are they the only sources for DIDs in USA and everybody has to go to them? Could someone explain how exactly it works? Thanks -- Zeeshan A Zakaria -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070531/170b14cb/attachment.htm

Hi, Does anyone know anything about the following? In a hosted environment where several area DIDs are provisioned on a single server, how do most carriers establish the origination DID, number. Asterisk allows us to modify the CallerID, name, number and DNID channel variables before dialling out via SIP. Most carriers will allow us to spoof a callerID when placing a call, and pass it forward.

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 presently working on an office move and evaluation of telecommunications services needed at the new location. I'm presently wrastling with an issue related to portability and geography between landline carriers. Presently certain people within the organization are hopelessly in love with our 909-822-xxxx number(provided by pacbell/att). As that number is presently provisioned it

Sadly it's a limited to PRIs(at least that's what I've been told) You just can't send that much call data with a good old non-PRI T1. The up side is you get one extra voice channel to use as compared to a PRI. It's strange that the carrier doesn't have PRI cabability, I've run into many more carriers that can't do non-PRI. I've even had one that couldn't

Hello everyone, I'm hoping someone can help me with this. I have a business customer in the U.S. (Michigan, AT&T Territory). I need to get 4 trunks into an asterisk Box. My intention is to use an Eicon Diva Server card with 2 BRI Circuits. The reason for this is that the business needs DID's on the trunks (20 of them). A full or fractional PRI is overboard for them, as they will

Is there any good free / accurate online resources with detailed country numbering plans? Failing that let's get something running ourselves. I was also thinking maybe people present could contribute some information on this list for now. The countries I am after are below. To start this off I will provide the information for Australia +61 and New Zealand +64. NZ Cellular: area code 21

Just curious... Is there anyone out there who has given this outfit money and actually received any service from them?

I want to collocate an * box somewhere, where better than where voicepulse chose to put their servers? They probably did their homework and selected someplace where good handoff to the pstn can be found, right/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050520/9f5975b8/attachment.htm

> > There is a big confusion here about Stir Shaken. It is NOT a provider > issue. Un fact, all providers are whasing their hands and modifying their > swihtches to pass-through the Signature. They cannot sign the call because > then the become the responsible party for the call before the FCC, and > liable for any illegal call. Every owner of a PBX that sends calls to the >

Hello Everyone, Unlike going through DIDx, DIDLogic etc.., we have an option of getting DIDs directly from local telco Bell Canada. Currently our SIP Trunk provider assigned a DID to us, and as you know, they just redirect requests it to our PBX. However, when dealing directly with a telco, what equipment will we need? Basically giving us the same capability as a DID provider. If someone can

With asterisk 1.8.7.0 has been running ok for months. Now, this morning, it's crashing. I can restart it, but it crashes after 10+ minutes. It dies like this -- Executing [s at macro-stdexten:2] Dial("SIP/teliax-00000019", "SIP/176,18,rtT") in new stack == Using UDPTL TOS bits 184 == Using UDPTL CoS mark 5 == Using SIP RTP TOS bits 184 == Using SIP RTP

Hello, this is Jesse with Webformix. We are managing an Asterisk installation for residential VOIP service, and we are having a problem where all inbound calls to DIDs which are assigned to us by our wholesaler but not yet assigned to a downstream customer get caught in a routing loop. ** For example: we get assigned DID block 1230-1239 and only 1230-1233 are given to our customers, then our

Check out the proposed prices when this is approved. BELL CANADA REPORT ON THE ECONOMIC EVALUATION FOR THE TARIFF REVISION OF Bell Canada's Access Services Tariff Item 315 ? Zero-Dialed Emergency Call Routing Service (0-ECRS) *2 March 2006 TABLE OF CONTENTS Page 1.0 GENERAL 3 1.1 Purpose of the Study 3 2.0 SERVICE DESCRIPTION 3 2.1 Service Characteristics 3 2.2

I'm pretty new to this whole Asterisk system & VoIP thing, but being a programmer by trade the complexity didn't scare me off (at least not yet)... I have setup an Asterisk system for my home & home office. My wife & I run two separate businesses from home, and we have a general family home phone line as well. The cost of all these lines with analog carriers was getting

[crossposted to isp-clec and asterisk-users] As part of several larger projects, the question of rate importation from termination carriers has come up. If a firm has four different LD partners, as an example, then it is obvious that the firm needs to determine at the origination of a call where that call would be best sent on a cost basis. I'll ignore things like quality of service and

Sorry for the off-topic, but perhaps this will be of interest to other asterisk based ITSPs. We are starting service in a rural area where the ILEC has the rural "monopoly". From what we have read in the FCC docs this does NOT exempt them from number portability, but what does it take for us to qualify to receive their numbers? To date we simply have a few voice trunks to them,

On Mon, 13 Jul 2020 15:44:12 -0400, Matthew Fredrickson wrote: > > On Mon, Jul 13, 2020 at 2:34 PM Saint Michael <venefax at gmail.com> wrote: > >> > >> There is a big confusion here about Stir Shaken. It is NOT a provider issue. Un fact, all providers are whasing their hands and modifying their swihtches to pass-through the Signature. They cannot sign the call