[Asterisk-dev] goiax expanded with free us domestic calling

[Mark Aiken]

One thing that really puzzles me is why people keep inventing new call
control protocols now that SIP has proven itself and has been adopted by all
the major carriers and 3GPP. Skype was a hit because it solved one the major
weakness of SIP - NAT traversal. That problem is quickly going away and
Skype's value is now its user base as a P2P network. What does IAX bring to
the table? Why bother with Yet Another Call Control Protocol.
 Mark

 On 10/2/05, Matthew Simpson <matthew at txlink.net> wrote:
>
> >> I launched www.goiax.com <http://www.goiax.com> last week, which is
> intended to promote the
> >>use
> >> of IAX as a free and open source alternative to products like skype.
> >> There is no charge for the service. Right now I have free outbound >>to
> >> united states toll-free and us domestic numbers working.
>
> >While I appreciate your efforts to evangelicize Asterisk (not that I am
> >anybody in particular), please... this post has *nothing* to do with
> >Asterisk
> >development at all. Why did you post here?
>
> Uh... did you miss the last paragraph that talks about adding some IAX
> scalability features to * and asking for developer input? :boggle:
>
> Just in case you did, here it is again:
>
> "I intend to also modify asterisk to allow some QoS checking to avoid
> the problems IaxTel had with scalability. I would like some developer
> input on this. My idea right now is to add an app to asterisk that
> listens on a port for a status packet that will be sent from a softphone
> or IAX-compatible device. Asterisk would then reply with a packet that
> would contain current CPU load, current mem load, and number of channels
> up at that point. I would also like to have the asterisk app know the
> status of other IAX servers on that "network" and be able to reply with
> the IP address of the "best" available server at that time."
>
> yours,
> Matthew
> _______________________________________________
> Asterisk-Dev mailing list
> Asterisk-Dev at 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
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-dev/attachments/20051002/ad96ac5f/attachment.htm