similar to: Call limits and VoIP providers

Displaying 20 results from an estimated 20000 matches similar to: "Call limits and VoIP providers"

2003 Jul 18
8
"Best" VoIP provider for Asterisk?
Hello! I would like to get connected with a VoIP provider for home. At some point, I'm sure I will be connecting to it via an Asterisk box, but for now, I will be using whatever hardware they provide. What recomendations do you in the Asterisk community have for a reliable VoIP service that will hopefully interoperate with Asterisk? A company that is actually willing to work with an
2004 May 08
5
1800 Provider
Hi list, I'm interested in receiving incoming call to my Asterisk PBX thru an 1800 number. Anybody knows a provider with best minute rate? I heard that that Nufone can provide this service for around 3cents/min for calls made within 48 continental states. Any provider that can give better rate, even with additional limitation such as much few states that a call can originate? How do
2005 Jan 23
1
VoIP Providers and Backbone Servers
Hello All, Well, my explorations in to the world of VoIP is proving fruitful and in the near future I am hoping to have my small VoIP online service up and running ready to help promote the industry and hopefully gain a few customers in the process. Additionally, I will soon have my IAX and SIP softphone ready that will handle video, audio, and text communications. I am looking for quality and
2011 Mar 18
1
remove old files
Hello I use the dovecot 2.0.10 server and wonder how to clean up correctly the old mails (especially for spam folder). I use maildir storage, and wonder if deleting files with an rm and file command combinaison would be good or would corrupt indexes files/whatever. for exemple, the spam folder is in ~/.mail/.INBOX.spam/ Thus, can I "cd ~/.mail/.INBOX.spam/ && rm 'files older
2004 Jul 07
8
VoIP hackers gut Caller ID
The Register is carrying a article written by Kevin Poulsen of Securtiy Focus, calling asterisk "..the most powerful tool for manipulating and accessing CPN data.." > http://www.theregister.co.uk/2004/07/07/hackers_gut_voip/ I hope NuFone doesn't drop asterisk-set-able callerid's after this article; i've been wanting that feature from voicepluse for a long time.
2004 Jan 24
13
Has Nufone gone belly-up
Folks, I've ordered a new account from Nufone last month. Transferred money to Nufone through their paypal account. I had communication with Nufone sales up until two weeks back. Since then there were no replies to my emails. I am afraid with this kind of unresponsiveness how one would run a reliable service with this company. Have no bad feeling with Jeremy as the author of widely used h323
2004 Jun 15
1
(sans objet)
Dear users I have a problem with the dr function: "dimension reduction". I give you my example, and i'll be pleased to read your comments. #let be X a matrix 50*100: library(dr); X<- matrix(rnorm(50*100,5,1),50,100); #and let be Y a vector response: Y<- sample(0:1,50,replace=T); #I choose (for the expérience, but in reality i don't have it) a few variables #which are
2005 Oct 09
1
enter a survey design in survey2.9
Hi dears, I expect that Mr Thomas Lumley will read this message. I have data from a complexe stratified survey. The population is divide in 12 regions and a region consist to and urban area and rural one. there to region just with urbain area. stratification variable is a combinaison of region and area type (urban/rural) In rural area, subdivision are sample with probabilties proporionnal to
2007 May 11
4
Dealing with 2 SIP providers
Hi, I have a question of using 2 SIP providers. Let's say I have provider A and provider B, and I would like my calls to go to A, and then B if A wasn`t available Something like this would work: exten => 1234,1,Dial(SIP/providerA) exten => 1234,2,Dial(providerB) exten => 1234,3,Hangup But what if I want to put in a delay? If I put 30 seconds on each of them, I'll wait a
2006 Apr 21
2
confused about iax and voip providers termination
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
2006 Nov 20
1
Reliable European SIP/IAX Providers?
I know that the wiki has an extensive list of European VoIP providers out there....but there's so many that it's kind of hard to sort through. So I was wondering if anyone could recommend some reliable SIP/IAX termination providers in Europe? Something like VoicePulse Connect, NuFone, Vitelity, or Junction Networks based out of Europe. I really don't trust a US VoIP company for
2007 Dec 17
3
VoIP service providers/PSTN termination points
Hello ppl, Am looking at some PSTN termination providers in US. If this question has been repeated, please point me to the correct link, as I've tried searching the archives but have been unsuccesful so far. I have come across quite a few companies which provide the same, such as : Iconnecthere <http://www.iconnecthere.com> Vonage <http://www.vonage.com> Teliax
2005 Aug 25
4
VoIP providers -- California, U.S.
Hi, Just wondering if people could suggest a good VoIP provider that can service the San Francisco Bay Area and the Los Angeles area. I've tried race.com (recommended to me) but they're kind of hard to get ahold of. Any other suggestions? This is for a business, so reliability is key. I did see the recent thread about this, and while I saw a few mentioned, I didn't see anything
2006 Apr 17
4
Looking for a good VoIP Provider in the UK-
Any recommendations for a VoIP provider in the UK? I have a few guys in a field office in the UK with SIP phones and a VPN tunnel back to a working Asterisk setup in the US. The Asterisk setup has an IAX trunk with TelaSIP/VoipXpress with local DiD's for US offices, so they can call vendors, customers etc in the US at local rates. I'd like to get the same thing for the UK, so that UK
2006 May 23
13
Now that Nufone is dead...
Now that Nufone is dead, what are other providers of 800 numbers that work with Asterisk? -- Carlos Chavez Director de Tecnolog?a Telecomunicaciones Abiertas de M?xico S.A. de C.V. Tel: +52-55-91169161 Ext 2001
2003 Aug 20
13
VoIP dialtone?
Hi all, While pondering my choices for local dial tone service via a bunch of POTS lines or a T1, I began to wonder if perhaps there is another way. Are there VoIP dialtone providers? That is, could I use only my internet connection for voice calls and not have a separate T1/POTS bank for that? I guess I am imagining a company that gateways between the PTSN and the internet backbone.
2006 May 15
1
VOIP adapters to connect PSTN lines to SIP phones
Hi, I have a question on VoIP adapters. As far as I understand, those adapters are usually used to connect DSL/Cable access to a normal phone (Internet to Adapter, then to PSTN phones). I want to know if you can use those adapters to do the opposite: connect a few lines (1-4 let`s say) to the adapters, then deliver via SIP to an Asterisk box. (I know I could use a TDM400 and Asterisk, but I
2003 Nov 12
2
Canadian VoIP termination?
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!
2007 Oct 08
2
inbound call voip providers
Hello: I want to have a local telephone number that, when the people calls this number (via mobile or normal PSTN), the voip provider stablishes a SIP session to my asterisk box. It is possible? If yes... What providers have this service in Europe? It is difficult to configure and get things working ok? Will my asterisk box see the mobile or normal PSTN phone# that is calling the number
2005 Mar 03
0
New user - problem getting dtmf tones through VOIP providers?
Just setting up Asterisk. I'd like to be able to dial out through VOIP providers and have customers type in a code in response to a prompt. So far, I've been able to set things up to make the call and play the prompt. However, my problem now is the DTMF tones; they don't register when I call get_data. When I make a call in person (from the DIAX softphone, through a VOIP provider,