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List:       asterisk-dev
Subject:    Re: [Asterisk-Dev] sip gateway provider recommendations?
From:       Rusty Dekema <rdekema () gmail ! com>
Date:       2005-11-28 15:22:31
Message-ID: 68171c120511280722j6758c280vcdd9eca93fd10770 () mail ! gmail ! com
[Download RAW message or body]

Confusingly enough, if you want to place a call from your Asterisk machine
to a telephone on the public-switched telephone network using a 3rd party
phone company to provide the connection, this is called VoIP Termination,
not Origination. VoIP Origination is when a 3rd party phone company assigns
you a telephone number and sends calls that come in to that number to your
Asterisk box (or whatever).

With that aside, you might check any one of the following companies:

http://connect.voicepulse.com
http://www.broadvoice.com
http://www.teliax.com
http://www.voipjet.com
http://www.gafachi.com

Many of these will let you get started for $10, and one will even give you =
a
free account preloaded with $0.25 for testing purposes. ($0.25 lasts a fair
while when USA termination is $0.01 or so per minute.)

Finally, the asterisk-dev mailing list is not the right place to ask this
kind of question. The asterisk-dev list is intended for discussions by
people who are working on developing the Asterisk software itself. The
mailing list you are looking for is asterisk-users. You will be much more
likely to get answers to operational kinds of questions there (and probably
will get flamed less :)).

Regards,
Rusty



On 11/25/05, John Brookes <johnbrookes@sbcglobal.net> wrote:
>
> Howdy,
> I am seeking a sip provider to originate calls to the pstn network, and b=
e
> able to scale up to multiple lines.
> (I remember I saw a provider with $25 upfront pay-as-you-go, but lost
> their name...)
>
> Anyone
> Thanks in advance,
> John B
>
> _______________________________________________
> Asterisk-Dev mailing list
> Asterisk-Dev@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-dev
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-dev
>
>

[Attachment #3 (text/html)]

Confusingly enough, if you want to place a call from your Asterisk
machine to a telephone on the public-switched telephone network using a
3rd party phone company to provide the connection, this is called VoIP
Termination, not Origination. VoIP Origination is when a 3rd party
phone company assigns you a telephone number and sends calls that come
in to that number to your Asterisk box (or whatever). <br>
<br>
With that aside, you might check any one of the following companies:<br>
<br>
<a href="http://connect.voicepulse.com">http://connect.voicepulse.com</a><br>
<a href="http://www.broadvoice.com">http://www.broadvoice.com</a><br>
<a href="http://www.teliax.com">http://www.teliax.com</a><br>
<a href="http://www.voipjet.com">http://www.voipjet.com</a><br>
<a href="http://www.gafachi.com">http://www.gafachi.com</a><br>
<br>
Many of these will let you get started for $10, and one will even give
you a free account preloaded with $0.25 for testing purposes. ($0.25
lasts a fair while when USA termination is $0.01 or so per minute.) <br>
<br>
Finally, the asterisk-dev mailing list is not the right place to ask
this kind of question. The asterisk-dev list is intended for
discussions by people who are working on developing the Asterisk
software itself. The mailing list you are looking for is
asterisk-users. You will be much more likely to get answers to
operational kinds of questions there (and probably will get flamed less
> )). <br>
<br>
Regards,<br>
Rusty<br>
<br>
<br><br><div><span class="gmail_quote">On 11/25/05, <b class="gmail_sendername">John \
Brookes</b> &lt;<a href="mailto:johnbrookes@sbcglobal.net">johnbrookes@sbcglobal.net</a>&gt; \
wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, \
204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">







<div><font face="Arial" size="2">Howdy,</font></div>
<div><font face="Arial" size="2">I am seeking a sip provider to originate calls to 
the pstn network, and be able to scale up to multiple lines.</font></div>
<div><font face="Arial" size="2">(I remember I saw a provider with $25 upfront 
pay-as-you-go, but lost their name...)</font></div>
<div><font face="Arial" size="2"></font>&nbsp;</div>
<div><font face="Arial" size="2">Anyone</font></div>
<div><font face="Arial" size="2">Thanks in advance,</font></div>
<div><font face="Arial" size="2">John B</font></div>

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