similar to: SIP 800 Origination/Termination - International

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

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:

Hi, Does anyone know of Canadian VoIP termination providers? I have Canadian customers and would like to provide Canadian dial in and dial out (canadian callerid). Thanks!

Somebody know if they are down? Let me know, Julius C. Barber ventas@gringotel.com www.GringoTel.com Tel. USA: 1-408-705-1189 GringoTel - ahorre en sus llamadas internacionales. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060509/924605b6/attachment.htm

Hi, We have been selling * systems for a while and always have used other companies for origination and termination and let the client pay directly. Since we do not have enough traffic to justify building our own infrastructure we would like to start reselling someone else's service. Any ideas? It would have to include easy way for billing. Thank you. robert -------------- next part

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

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

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

Hey I just setup service with IPComms and they are telling me to setup such as this: iax.conf: [IPCommsNet] type=user host=69.15.xxx.xx context=voicepulse-in ;(changed by me) nat=yes dtmfmode=rfc2833 disallow=all allow=ulaw allow=alaw allow=gsm When I'm calling once of my numbers it's giving me this though: Oct 5 12:11:06 NOTICE[49584]: chan_iax2.c:5476 socket_read: Rejected connect

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

Dear Asterisk Community, Does your company provide inbound 800# origination? If so, please read this message and e-mail us a quote for monthly co-lo hosting of our asterisk server and per-minute inbound 800# origination. The Prostate Cancer Research and Education Foundation (PC-REF) is a non-profit organization dedicated to helping prostate cancer sufferers and their loved ones. We have

hi folks, Has anyone found a good (and, ideally, cheap -- we don't really want any per-minute charges) origination provider which can handle a moderate number of simultaneous incoming calls (to the same, single DID)? Many of the providers I've tried contacting either won't call me back, or want me to sign an NDA just to get a rate quote, or some other bullshit. Most of the

Hi all, I have here several IAX2 Softphones(IDEFISK, DIAX and an own develop based on iaxclient.lib). I have follow dialrules in my std-test extension: [std-test] exten => *601,1,Answer exten => *601,n,Dial(IAX2/pbxnetwork/xxxxxx,30,m) exten => *601,n,Hangup exten => *602,1,Answer exten => *602,n,Dial(IAX2/pbxnetwork/xxxxxx,30) exten => *602,n,Hangup No I have a problem when

A long time ago (Asterisk 0.x, 1.0.x) my experience is that there were alot of interoperability issues, a common troubleshooting issue was to make sure all endpoints where using the latest version of Asterisk. I have not seen these issues in a while. However I've been working with a customer of mine and this ITSP called IP Communications (IPComms.net) well turns out we have had constant

Hi - I am looking for suggestions for ITSPs for faxing with asterisk 1.8. We are based in the US, so would need an ITSP with US DIDs. #1) We would like to use Fax For Asterisk with asterisk 1.8.4 in order to receive faxes via T.38. Sending faxes is not a requirement. Does anyone have a working asterisk 1.8.4 configuration and ITSP provider that they can recommend? We have been trying T.38

Hello guys I am a new ubuntu user and i am trying to install play65 application without success. I have installed the wine but nothing has been achieved. Anyone who can help me on this?

I am looking for a good provider for IAX2/800 termination. I am currently using FreeWorldTel and wanted to use NuFone but it seems that both of them don't provide customer service. FreeWorld has terrible voice quality and NuFone never answers their phone or responds to messages. Thanks, Linn

Can anyone recommend (offlist) a good IAX or SIP based 800 provider? Intention is for high volume calling card traffic from the US Virgin Islands and Puerto Rico. Thanks, j

Just on the off chance that Canadian Asterisk users might be interested in a place to discuss topics specific to the "great white north" (sources, services, telcos, etc.), I created the asterisk-canada mailing list: http://lists.syonex.com/mailman/listinfo/asterisk-canada or asterisk-canada-subscribe@lists.syonex.com Cheers! John

when I run console with ruby 1.9.1 and rails 2.3.2, and trying to do something like User.first.name.encoding I''m getting #<Encoding:ASCII-8BIT>, though I''ve set "encoding: utf8" in database.yml any suggestions? --~--~---------~--~----~------------~-------~--~----~ You received this message because you are subscribed to the Google Groups "Ruby on Rails: