[Asterisk-Dev] sip gateway provider recommendations?

[Rusty Dekema]

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 at sbcglobal.net> wrote:
>
> Howdy,
> I am seeking a sip provider to originate calls to the pstn network, and be
> 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
>
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