So i'm looking for a good provider for DIDs and 800# from the US and CA,
who offer online signup and ordering. The provisioning should be less
than 12 hours, preferably instantly.
BinFone Telecom has instant provisioning and DIDs/800s for New York,
Washington, D.C., Baltimore,
I every one, looking for suggestions, or even better yet, a how to
guide.
I have read most of the wiki's on both, so I know this is completely
possible
I have Asterisk and SER configured on the same server x6
yes, 6 servers.
If I can get this to work on one, we are golden!
Number one
Give me an idea of your application.
I personally created a really cool asterisk system for the US Army where
the phones would ring on silent for about an hour before people would pick up
for a conference call.
I Know this config like the back of my head.
Keep in mind, Asterisk has many
Lousy pricing. If you reply/advertise on-list, at least be competitive.
-Original Message-
From: Nathan E. Pralle [mailto:[EMAIL PROTECTED]
Sent: Sunday, July 24, 2005 10:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] DID + 800
Correct -- and, solicited or not, any commercial email message (that is,
any message soliciting business), must include unsubscribe instructions,
which the one in question did not.
-Original Message-
From: Steve Totaro [mailto:[EMAIL PROTECTED]
Sent: Sunday, July 24, 2005 8:45 PM
Only softphone I ever downloaded (and promptly deleted) was xten. And
no, I never register with this email address.
-Original Message-
From: C F [mailto:[EMAIL PROTECTED]
Sent: Sunday, July 24, 2005 5:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Dude, just delivering you reality. DIDs are like an ISP IP address, has to
be hosted from carrier.
Just the way life is
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To
PS, I never gave a price,,, just reality!
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unless you are willing to drop $650,000.00 for DID distribution
segment.
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I simply answered a question dealing with a project that I my self worked
on for a year. I am not trying to profit. Only assist.
On this list, we are all in this together!
Brad
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you must be using Windows.
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Ive just gotten off a skype conference call and it
pisses me off that the quality of skype is higher than my asterisk calls.
Is there such a thing as a super high bandwidth codec?
In a situation that you have the bandwidth to share is there
something that I can use for important
Apart from Packet 8, is there anyone else who offers
unlimited calls in both the USA
and Australia?
Im starting to get the shits with their call quality and looking at
alternatives.
I currently use the Packet 8 Asia unlimited service for $49
a month and its a great price point but
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Monday, 25 July 2005 1:42 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Did anyone else get spammed by GIZMO?
I simply answered a question dealing
Can anyone tell me if the acc_flag is only supported under the options
request?
one of my telco pipes does not support the options field.
Yes, I know, non RFC.
Brad
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It has nothing to do with bandwidth.
It has everything to do with your routing gear!
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On Sunday 24 July 2005 22:36, Jay Milk scribbled:
Lousy pricing. If you reply/advertise on-list, at least be competitive.
Lousy attitude. If you reply on-list, at least be courteous.
As far as pricing, listed prices are not always what we might quote for
quantity.
But this isn't an
I personally have performed 'real time test" of plugging a Cisco 7940 phone
behind a Netgear router and had is sound like "butt-crack".
and then plugged it in (within 2 min) behind a Cisco router, and it was
Crystal Clear!
ROUTING GEAR IS ALWAYS THE KEY TO VOICE QUALITY ON VOIP.
That and a
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Nathan E. Pralle
Sent: Monday, 25 July 2005 2:14 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] DID + 800 Providers
On Sunday 24 July 2005 22:36, Jay
On Sun, Jul 24, 2005 at 11:45:07PM -0400, [EMAIL PROTECTED] wrote:
you must be using Windows.
Maybe he does. Maybe you do. I don't. Your message still lack any
reference to the ones you respond to.
--
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il |
How do you figure?
How does skype sounds so damm good on the
same network/machine? I think you might be wrong.
Cheers,
Dean
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
Sent: Monday, 25 July 2005 12:11
AM
To:
Title: Message
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]Sent: Sunday, July 24, 2005 9:11
PMTo: asterisk-users@lists.digium.comSubject: Re:
[Asterisk-Users] super high bandwidth codec
It has nothing to do
I'll give you an example. My staff does a lot of interviews with new
hires and salesmen. I do not personally sit in on the whole thing
(they can really drag out!), but I would like to be able to listen in
to make notes as needed while doing other work.
I had another idea whereas I could dial
dbruce wrote:
If you use the polycom provided config files, the default ring_answer class
is 4 and the auto_answer class is 3. So for your RANR alertinfo entry,
change the class to 3 and it will work as you expect. ie:
alertInfo voIpProt.SIP.alertInfo.3.value=RANR
Kristian Kielhofner wrote:
Billy Dunn wrote:
I have a bunch of Polycom Soundpoint 600 phones and they are working
great. The only thing I can't seem to get them to do is to
ring-answer without the ring.
This is what I have in my sip.cfg file on the boot server:
alertInfo
Last month I saw something funny which I can't reproduce anymore:
A 0500 number in .au is a service phone number and are forwarded
on exchange level to a real phonenumber. So if A calls B it gets
forwarded to C. Very simple.
Now the funny thing, on the phone of C, I saw both A and B as the
No ring is at the phone level,
only from what I know
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-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Edwin Groothuis
Sent: Monday, 25 July 2005 3:04 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Caller ID, Called ID and Forwarded ID
Last month I saw something funny which I
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