similar to: What is the best way to resale termination/origination?

I've been losing patience with my current provider, a small company called Sellvoip. Their termination is good, and they are asterisk based, but they are understaffed and have no concept of customer service. So I'm shopping. I am interested in the opinions of others on the providers they work with. Here are my criteria, roughly in order a) Decent quality, low latency. In

Anyone have a good provider for International (US/Canada at least) 800 termination/origination? I have a customer that had us port one of their 800 numbers and apparently didn't realize that they had published that number in Canada as well. Our current origination/termination provider can't handle Canadian inbound calls to that number, so I need to find another provider that can.

I'm looking for sip origination and termination companies. Anyone know of reliable ones? I am interested in wholesalers also. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090417/644bcbad/attachment.htm

I thought XO was reselling Level 3s (old Genuity assets) network/voip just like Qwest ? -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of jk@bingoconsulting.com Sent: Wednesday, October 11, 2006 3:38 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Load balance Asterisk server,when it is a SIP

Hi, I have a strange problem, when I try to connect to les.net from our local asterisk server through Procurve router I seems to be connecting on any port above 1024 and when I reload sip the port is changing too ... So I never get 5060? Any ideas on what is going on and how to resolve it? ? Sincerely, Robert Augustyn 519-997-3106 ext:802 www.linqone.com ? ? -------------- next part

>Shoval Tomer wrote: >> As far as I know it's not legal to join bellster in Israel. >> >> It means that you're reselling the minutes you buy from the telco >> company. > >Wouldn't you need to be selling them to be reselling? > >Does that make DISA illegal, and VoIP connections between offices if you >dial out the other end? Well, a thing

Hi List, After quite a bit of struggle, it looks like I'm all ready to roll out prepaid cards on my small island. I now have a 4 E1s with a bit of spare capacity in order to accept incoming calls, and I can route Reunion Island mobile and fix through my own installations. For all other destinations, I need a carrier. I need good wholesale prices to Comoros, Mauritius, Madagascar, India,

None of the unlimited packages you get plan on you using them up. It's like a couple of years ago when the long distance carriers gave us the first phone call of the holiday for free, and people let the phone off hook for the entire week end. It's illegal for sure, and for a couple of reasons. a. you're not allow to resell anything someone else sold you and not pay taxes on it. b.

John, I spoke with Level(3) last week regarding SIP termination. They quoted $0.01/minute, with an 11 Million Minute / Month minimum. Ugh! -dg -------------------------------------------------------------- Darnell Gadberry President binaryMedia darnell AT binmedia DOT com ------------ Date: Fri, 19 Dec 2003 21:12:22 -0500 To: asterisk-users@lists.digium.com From: John Todd

All we have a deal on Canadian termination. Rate: $0.0039 US Dollars Billing: 1/1 Protocol: SIP or H323 Codec: G729 Terms: Prepaid Only. We have a real-time web interface where you can monitor or download your CDR's. Please e-mail me offlist if you are interested: jweisman@ibell.net Thanks, Jon -------------- next part -------------- An HTML attachment was scrubbed... URL:

Anyone investigated the new service offerings from Level(3) in the last few months? They claim to be using ENUM and SIP - see http://www.level3.net/2192.html for details. Any idea of their pricing model for mid-sized enterprise applications or call centers for origination/termination? More specifically, do they interoperate with Asterisk? Some providers insist on certain hardware that

Hey guys, I'm actively trying to get the "big" picture on how all this works and relates to each other. I've gone through some basic examples from the book and from the sample files just fine. Now, I've setup an account with a VOIP provider which does IAX termination (exgn.net) After getting an account and following their steps, I can make calls out using my IAX (cubix) and

I hope this is an appropriate question for the list.. I am looking for a VOIP termination provider who can offer the following: -Flat Rate DID's in lots of areas -GOOD customer service/support with quick response times -Toll Free DID's at a reasonable rate -Reliable/Redundant network and availability etc. So far I have tested 4 providers which I will not mention here. I have found two

From the bend me over news department. 2 March 2006 Mr. Leonard Katz Executive Director Broadcasting and Telecommunications Canadian Radio-television and Telecommunications Commission Ottawa, Ontario K1A 0N2 Dear Mr. Katz: Associated with Bell Canada Tariff Notice No. 6929 1. Attached for the Commission's approval are proposed revisions to Bell Canada's Access Services Tariff Item

I'm looking for IAX (preferred) or SIP termination in Israel, with DIDs. Can anyone give me a price, or better yet, point me to a resource for buying/selling minutes? Thanks, Mark

I'm confused about VoIP termination service. Would this mean that I could have an Asterisk server in my basement and receive/call PSTN phones via a VoIP termination provider? The "call" out from my box would go over the Internet to a termination provider which would in turn convert it to analog over the PSTN to someone's plain old phone? And vise-versa, a plain old phone caller

Mike, Martin O'Shield here from WindyCitySDR. >From: mike castleman <mlc@democracynow.org> >Subject: [Asterisk-Users] origination providers >To: Asterisk Users Mailing List ><asterisk-users@lists.digium.com> >Message-ID: ><20050524190016.GG11096@pinetree.mlcastle.net> >Content-Type: text/plain; charset="us-ascii" >hi folks, >Has

Hi all, We have put together a firmware for a range of inexpensive routers. It has been configured to provide optimum VoIP performance. We have internally tested it for number of months and it looks very good. You should be able to run it easily with 20+ phones on local network ( we still did not hit the upper limit ) assuming that you have bandwidth. Your VoIP will get prioritized over other

Hello, Quick Diagram: Telco-PRI -> Asterisk <- Norstar PRI -> Norstar PBX (DMS100) (TE405P) (DMS100) | | V Cisco 7960G (SIP) I'm trying to change the Origination Number on my outgoing PRI, and running into a weird problem. If I make a call from a SIP extension off asterisk using the following context:

I missed your original email. I might be willing to do this for you. How many clients are you talking? I'm already reselling web hosting and have Icecast running, just haven't gotten around to rolling it into my packages yet. ------------- Forwarded message follows ------------- Geoff I have sent this to one of my Tech service proivders Weather he gets back to you or not i suppose is