Outgoing is free to 18 countries, but (sigh) there are restrictions, but getting voipStunt (Germany) to respond to my questions as to what these restrictions are, seems to take an Act of God. $11/month for a home phone anyone can pick up and use, is really great. Outgoing calls are free – but to call non-free places, their rates are 1.5 times as much as Skype’s rates. They (voipStunt) require a $10 deposit which they take out the cost of the calls to NON free places and mobile phones, which are NOT free, OR if you exceed their UNKNOWN quota which they won’t tell you what it is.

Sometimes I have to “power cycle” the AIX card, but it works just great. Sometimes the circuits are busy when making international calls, but still it’s worth the price.

Now I want to be able to “assign” extension numbers to my PBX to route an extension to a Gizmo user. Is this possible, or have they locked us out from taking advantage of the SIP compatibility.

Gizmo is useless to me if I cannot use it with other SIP servers. If someone comes up with a hack that allows me to use it with other SIP servers, I would consider using it. I would like to talk to you offline, if you could send me private Email – I sent my address above in the earlier posting. Use that address, but please put SIP in the subject line so I can find your message among the 6000 spam messages per day that I’m getting into that address. Once I find your message, I’ll send you my real email address.

JS

Here are the Skype Rates in US dollars.

Here are the Gizmo Rates in US dollars.

At the time i’m writing this comment, I’m seeing, on the Skype side, 2.5 cents per minute for calls to the U.S.

And I’m seeing 1.8 cents per minute for calls to the U.S.

But Skype appears to be 2.2 cents per minute for calls to France.
And Gizmo appears to be 4.9 cents per minute for calls to France.

So if you’re calling France, you’re likely better off using Skype. If you’re calling the U.S., you’re likely better off using Gizmo.

Yesterday, I was sucessful in getting a clean “Build” which includes the “reSIProcate” .dylib, which I’m using for the SIP stack.

I already have the bi-directional audio streaming working, using the GSM codec I found on the net that’s 10 yrs old, and modified it to work and be compatible with Windows.

I already have it setup to use a number of crypto algs, and find that RC4 and AES appear to have good performance, but RC4 has less strain in my slow running G3 laptop I’m using to develop it on.

Am using X-Code 1.5 currently – want to be backward compatible with 10.3 if possibe.

I’m being funded by a company in Dubai, so the code belongs to them, but the reSIProcate license allows commercial use of their code.

If you want to contact me directly, Email me at crunch at webcrunchers dot com, and please put the word “SIP” in any place in the subject line to get past my very strict spam filters.

Since gizmo is not Open Source, I’m not very interested in it, except to use it for testing. But is it really running on native Mac OS? Including the Audio and Media? Does it use QT?

I hope to have time to play with it in about 3 weeks time, as I still have my nose to the grinder, grinding out Cocoa code.

Cheers
JD