2008/10/2 Gregory Malsack [EMAIL PROTECTED]
Hi All,
Can anyone recommend a good VOIP provider in the Milwaukee/Chicago area? We
need flat rate billing per line/trunk, trunking, did's, and iax or G.729
compatibility.
Thanks,
Greg
No virus found in this outgoing message.
Checked by
Hi.
We recommend Fonet Global, they work with Asterisk many years ago and
provide sip termination, DIDs, etc.
At 03:39 p.m. 02/10/2008, Steve Totaro wrote:
2008/10/2 Gregory Malsack mailto:[EMAIL PROTECTED][EMAIL PROTECTED]
Hi All,
Can anyone recommend a good VOIP provider in the
Gregory Malsack wrote on 1/4/08 4:48 PM:
Does anyone know of a good VOIP dialtone provider in the northern
Chicago area. My client has tried Broadvoice and Mix and is having
problems with latency in the middle of the traceroute between him and
the provider.
I use Broadvoice and haven't had
Good luck with that one. Most unlimited providers have limits. (even
if they say unlimited)
/b
On Sep 19, 2007, at 12:32 AM, Jim Boykin wrote:
Can someone suggests a good and resonable cost voip provider with
business unlimited plan in USA and allows simultaneous outgoing
calling.
Am Mittwoch, den 19.09.2007, 11:02 +0530 schrieb Jim Boykin:
Can someone suggests a good and resonable cost voip provider with
business unlimited plan in USA and allows simultaneous outgoing
calling.
My experience with business unlimited is that they very well know which
customer uses more
How would we be able to determine the reasonable cost for an unlimited
plan for an unspecified business? If the business was General Electric,
I would bet they would consider $1M/month very reasonable for unlimited
service. A plan for a corner shop might be reasonable at $19.95/month,
typical for
- Original Message -
From: Jody Gugelhupf [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Monday, August 27, 2007 3:55 PM
Subject: [asterisk-users] voip provider settings problem, please help
hi ppl, i'm using asterisk 1.2 because i'm making use of voiceone, but
before i
hi Anselm :)
thx for your tip, though i have qualified turned on, anyhow here are my
complete sip.conf and
extensions.conf, thx for any help :)
sip.conf
[general]
allowoverlap = yes
realm = mydomain.tld
bindport = 5060
bindaddr = 0.0.0.0
srvlookup = yes
tos = lowdelay
disallow = all
allow =
Am Montag, den 27.08.2007, 08:55 -0400 schrieb Jody Gugelhupf:
hi ppl, i'm using asterisk 1.2 because i'm making use of voiceone, but before
i was using asterisk
1.4 and had the same problem, it concerns an italian voip/sip provider called
eutelia/skypho, my
problem is the following one:
Plainvoip has a very good A-Z and I have
found they are fairly inexpensive.
They also offer TollFree orig and some
local dids.
www.plainvoip.com
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Crazy Boy
Sent: Friday, May 26, 2006 9:21 AM
To:
Thanks for the information, I will surely look into it!
Nitin
On 5/10/06, Kerry Garrison [EMAIL PROTECTED] wrote:
Have you looked at CBeyond? I like their T1 SIPConnect product.
From: [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED]] On Behalf Of Nitin GuptaSent: Wednesday, May 10, 2006 7:04 PM
Have you looked at CBeyond? I like their T1 SIPConnect
product.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nitin
GuptaSent: Wednesday, May 10, 2006 7:04 PMTo: Asterisk
Users Mailing List - Non-Commercial DiscussionSubject:
[Asterisk-Users] VOIP provider
Title: Message
Hi,
feel free to contact me off-list, we can have a test if you
want.
[EMAIL PROTECTED]
-Message d'origine-De:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Mark
AdamsEnvoyé: samedi 28 janvier 2006 15:50À:
On Jan 28, 2006, at 6:50 AM, Mark Adams wrote:
x-tad-smallerHi Everyone,/x-tad-smallerx-tad-smallerI know this may be off subject but I am not sure who to ask. I am currently looking for voip termination that is closest to replicating U.S. pots service. I run I.V.R. systems and I want to point
Dear trixter
Our software AstBill is now in use/beeing implemented by many smaal service providers and a few very large. It is Open Source.
I love to work with you on this and if any features are missing we be happy to implement it.
Are Casilla --
http://astartelecom.com - Independent VOIP
On Tue, 2005-10-25 at 08:21 +0100, Are wrote:
Dear trixter
Our software AstBill is now in use/beeing implemented by many smaal
service providers and a few very large. It is Open Source.
I love to work with you on this and if any features are missing we be
happy to implement it.
I didnt
We don't have a complete package quite yet. I think we have most of
what you will need but we do not have support at present yet to accept
customers payments. We can do that easily via 3rd party sofware but we
can't do it ourselves yet. Anyway, www.aleph-com.net/astpp is the link.
Darren
Bret,
See my recent post:
http://lists.digium.com/pipermail/asterisk-users/2005-October/130542.html
I'll send you an email off list with the features and future roadmap.
Alistair Cunningham,
Integrics Ltd,
+44 (0)7870 699 479
http://integrics.com/
trixter aka Bret McDanel wrote:
I am
Hey ho,
We have something like that (tailored for huge installations), contact
me off list for more info.
zoa.
trixter aka Bret McDanel wrote:
I am tasked with evaluating ready made solutions for a voip provider.
Does anyone have any recommendations for software, specifically the
We have a turn-key solution available that does exactly what you are asking
for. You can reach someone for more information at 415.442.4010.
TKS
Paul
[EMAIL PROTECTED]
trixter aka Bret McDanel wrote:
I am tasked with evaluating ready made solutions for a voip provider.
Does anyone have
Doesn't www.sipgate.co.uk do that? After all, they provide free NCFA
numbers to the asking.
-Original Message-
From: trixter http://www.0xdecafbad.com
[mailto:[EMAIL PROTECTED]
Sent: Thursday, June 02, 2005 9:26 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users]
On Fri, 2005-06-03 at 01:17 -0500, Jay Milk wrote:
Doesn't www.sipgate.co.uk do that? After all, they provide free NCFA
numbers to the asking.
You misunderstand I am asking for termination *to* NCFA. I want to be
able to call them, as my signature indicates I already have a NCFA for
inbound
You misunderstand I am asking for termination *to* NCFA.
Can you also terminate through Sipgate? They say: United Kingdom, 1.19 p/min
I figure if they can provide origination for NCFA numbers, they can
also terminate to them... your +44 870 number is a NCFA one, no?
--Luki
-
From: trixter http://www.0xdecafbad.com
[mailto:[EMAIL PROTECTED]
Sent: Friday, June 03, 2005 1:33 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] voip provider request
On Fri, 2005-06-03 at 01:17 -0500, Jay Milk wrote:
Doesn't www.sipgate.co.uk
On Fri, 2005-06-03 at 00:10 -0700, Luki wrote:
You misunderstand I am asking for termination *to* NCFA.
Can you also terminate through Sipgate? They say: United Kingdom, 1.19 p/min
I figure if they can provide origination for NCFA numbers, they can
also terminate to them... your +44 870
On Fri, 2005-06-03 at 03:33 -0500, Jay Milk wrote:
Naw, I understood you full well. You'd think if they provide
origination to NCFA numbers, they'd provide termination to them as well,
wouldn't you? As far as their website is concerned, there are only two
UK rates, and no disclaimers that
: [Asterisk-Users] voip provider request
On Fri, 2005-06-03 at 03:33 -0500, Jay Milk wrote:
Naw, I understood you full well. You'd think if they provide
origination to NCFA numbers, they'd provide termination to them as
well, wouldn't you? As far as their website is concerned
On Fri, 2005-06-03 at 10:14 -0500, Jay Milk wrote:
Ahhh... Sneaky. Because of the special billing agreements on NCFA
numbers, there's bound to be a lower limit to how these calls are
priced. I doubt BT gives sipgate (or any other VOIP provider) a
signigicant discount on these calls. If you
On Fri, Jun 03, 2005 at 10:14:58AM -0500, Jay Milk wrote:
Ahhh... Sneaky. Because of the special billing agreements on NCFA
numbers, there's bound to be a lower limit to how these calls are
priced. I doubt BT gives sipgate (or any other VOIP provider) a
signigicant discount on these calls.
On Fri, 3 Jun 2005, trixter http://www.0xdecafbad.com wrote:
Anyone else know of any providers that allows you to call a UK NCFA (+44
870) for $0.05 USD or less per minute and is BYOD?
The closest I've seen is http://www.iax.cc/ (sixTel) who charges just
over 5c/min.
Direct from their web
Asterisk wrote:
Hi all,
Is there a VOIP provider that can deliver local Rio de Janeiro numbers?
I am looking for a normal Rio number for my Asterisk box.
I'm using a RJ DID, right now http://www.libretel.com with IAX DID (they
offer SP also).
Have not tried much on it, noticed DTMF can be a
Arruda
Sent: Sunday, May 15, 2005 10:22 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Voip Provider in Brazil
Asterisk wrote:
Hi all,
Is there a VOIP provider that can deliver local Rio de Janeiro numbers?
I am looking for a normal Rio number for my
I have no idea on rates, services or quality, I just happened to read
their name and store it away in my head. maxvoip.com.br
Unfortunately I do not think they are BYOD, but they might be.
On Sun, 2005-05-15 at 10:55 -0300, Asterisk wrote:
Uhmm it is great that I can get it but it is a little
No, I'm not ignorant of how this works. You'll notice I put it
appears bad when I posted my results. Yes, it's not a perfect way to
show problems -- but taken with a grain of salt it's not half bad.
Especially when sampled over a longer period of time, and if the
original poster can correlate
On Friday 01 April 2005 04:28, Joseph Gutowski wrote:
Ok, since I guess no one else wanted to bite -- I will.
I installed PingPlotter, switched to UDP just to be the same as you,
and ran it against sip.broadvoice.com. Absolutley no problems, no
packet loss at all.
Ran it with all of the
Bob Goddard wrote:
The apparent packet loss you are seeing may be just fine tuning
of the routers in question.
This is the conclusion I came to as well; however, with the way
PingPlotter works the router is not sending ICMP unreachables but rather
ICMP TTL expired responses. In any case, the
The apparent packet loss you are seeing may be just fine tuning
of the routers in question.
This is the conclusion I came to as well; however, with the way
PingPlotter works the router is not sending ICMP unreachables but rather
ICMP TTL expired responses. In any case, the routers in
Rich Adamson wrote:
In other words, as the ttl value is increased and additional icmps
are sent, you might see what you believe is congestion, but you still
don't have any clue as to whether hop #2, #5, or #10 actually was
involved with that congestion.
Sure. But there is a way around this.
The
No, I'm not ignorant of how this works. You'll notice I put it
appears bad when I posted my results. Yes, it's not a perfect way to
show problems -- but taken with a grain of salt it's not half bad.
Especially when sampled over a longer period of time, and if the
original poster can correlate the
Johnathan Corgan wrote:
First off, I have Sprint Broadband Direct internet service, a fixed
wireless setup with a 2-5 Mbps downlink and a terrible 128 kbps uplink.
So I know I'm in for trouble anyway.
The broadvoice edge router (63.251.209.126, their lax site) is another
11 hops away. One hop
Discussion
Subject: Re: [Asterisk-Users] VoIP Provider problems
Johnathan Corgan wrote:
First off, I have Sprint Broadband Direct internet service, a fixed
wireless setup with a 2-5 Mbps downlink and a terrible 128 kbps
uplink.
So I know I'm in for trouble anyway.
The broadvoice edge router
On Thursday 31 March 2005 15:59, Kellner, Peter wrote:
Ping runs as a low priority service so it is not realistic to measure
response time using ping.
Try tracepath. It's not using port 7 and can be used by normal users.
--
Steve Szmidt
They that would give up essential liberty for
Ok, since I guess no one else wanted to bite -- I will.
I installed PingPlotter, switched to UDP just to be the same as you,
and ran it against sip.broadvoice.com. Absolutley no problems, no
packet loss at all.
Ran it with all of the published proxy addresses, again no problems.
I then used the
Joseph Gutowski wrote:
I installed PingPlotter, switched to UDP just to be the same as you,
and ran it against sip.broadvoice.com. Absolutley no problems, no
packet loss at all.
Well, that's good to hear.
I then used the 63.251.209.126 that you posted, and it was awful (at
least it appears awful).
We recently configure an asterisk server to use with an VoIP provider
to make calls to a PSTN. We use (voipjet, nufone, diamond)
We feel that we haven't got the quality that we hope. Sometimes our
calls gets mute, or we feel communication cuts on our phone calls.
We have got an QOS
It is not that simple. But you can begin by doing a traceroute to the
many providers at different times of the day. This will see the route
changes and time delays between hops to get to VoIP Providers gateways.
The best tool I've found for monitoring connections, routes, congestion,
is called
Give me a try! www.shelltel.com And don't use G711 for your calls.
invest in the G729 codec. you'll find your calls will start working
better. I'm a G729 shop.
Thanks
Michael D. Schelin
626-814-2454
Max W Blackmer Jr wrote:
We recently configure an asterisk server to use with an VoIP
Robert Terzi wrote:
The best tool I've found for monitoring connections, routes, congestion,
is called PingPlotter. http://pingplotter.com/ It's a shareware
visual traceroute. It continually graphs the traceroute style
responses. There is a scrollable timeline to view how things change.
You
On Tue, 2005-03-29 at 12:36 +0200, Ismael Gil wrote:
Hello all,
We recently configure an asterisk server to use with an VoIP provider
to make calls to a PSTN. We use (voipjet, nufone, diamond)
If you find the same problem with multiple ITSP's, then it may not be
them that is at fault.
James Rothenberger wrote:
I am testing a call flow in which an inbound SIP call (to the Asterisk
from a PSTN connection from a SIP VoIP provider) is not answered
(nobody there and no voicemail) and the call is terminated on the PSTN
side. After the SIP CANCEL is sent to the Asterisk from the
Hi,
* Erik Lagerway wrote/schrieb:
There is a provider in the US - www.AddaLine.com, who just launched a
SIP service with some great rates for North America
I have been using their service for months and I am extremely happy with
the
service.
looks like Germany is again laggin
Iconnecthere seems to have better rates...
-Original Message-
From: Martin Dommermuth [EMAIL PROTECTED]
Date: Thu, 12 Jun 2003 19:48:43 +0200 (MEST)
To: [EMAIL PROTECTED] [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] VoIP Provider
Hi,
* Erik Lagerway wrote/schrieb
Martin == Martin Dommermuth [EMAIL PROTECTED] writes:
Martin looks like Germany is again laggin behind all others in the
Martin communication field. Or I asked at the wrong place. There
Martin might not be to many people from Germany in this list.
One possibility is Pulver's LibrTel at
On 12 Jun 2003, James H. Cloos Jr. wrote:
One possibility is Pulver's Libr=C3=A9Tel at http://www.libretel.com.
Whenever I try any of their access numbers (at least the ones around me,
in the DC area), I get a recording The number you have reached is not in
service. This does not inspire great
(BHi
(B
(BWe can found a couple of ITSP at Jasomi networks's PR.
(B
(Bhttp://www.jasomi.com/pr_deployment.html
(B
(BDoes anyone try it?
(B
(Bmack
(B
(BOn Thu, 12 Jun 2003 19:48:43 +0200 (MEST)
(BMartin Dommermuth [EMAIL PROTECTED] wrote:
(B
(BHi,
(B
(B* Erik Lagerway
On Mon, 9 Jun 2003, Gary wrote:
Just a quick look at their rates show they just might be into rip
off's..
Australia0.06 (0-61-0)
Australia-Cellular 0.31 (0-61-7)
Australia-Cellular 0.31 (0-61-8)
Australia-Cellular
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