VOIP etc > Voice Over IP - VOIP > VOIP Tech Chat > [CheapVoipInc] Beta testers needed!

reply to nitzan

Re: [CheapVoipInc] Beta testers needed!

said by nitzan:

said by Arne Bolen:

Do you proxy the audio?

Yes. Shouldn't matter for A-Z calls as we're 8ms from our international carriers. For US calls it might but so far it seems like quality is good.

reply to nitzan

So far, I have made calls to London. Call quality was very good. Any way to specify the caller ID to display to the called party?
--
So Much trouble in the world...Bob Marley

reply to nitzan
All my calls to Hong Kong so far have gone to voicemail, but from what I can tell, the quality is really good on grey route. No crackling, echoes or weird volume levels.

Also, you seem to bill from when the call connects. Voip.ms bills from the moment you call. (Anyone else notice this?)

reply to mrcamp

reply to mrcamp
OK- Caller ID interface is now available via the web console. You can set it yourself or set it to blank if you want to send your own CID from a PBX. Note you can't do that from an adapter - it'll display your username instead.

reply to nitzan
Setting the Caller ID via your web console works fine. I have configured my Asterisk to not proxy media and the Caller ID is sent.

reply to nitzan
Hi,

- I put money yesterday but I have not received the credit in my account. I guess for this beta the process is done manually but this will be automated with the final release.
- When I bought credit everything shows as Future Nine Corportation, when I click in the link that says "Return to Furture Nine Corporation" it took me to future nine instead of back to CheapVOIPInc

- In the speed dial. The Destination and Name are in the same line, it would be nice if you show them in separate lines.
- In the speed dial. I clicked edit button and the page shows the msg "Once you have completed the form above, click on the CONTINUE button." There is no CONTINUE button

- Currently there i no way for me to know what is registered to use my account. I configued my OBI but I do not see any page where I can check that it is registered with your server. It would be nice to be able to see that information, may be in the SIP INFO or in other page.

Thanks

Google checkout isn't ready yet - it's still going to the F9 account. I added your credit manually and added beta/test credit to that. Thanks!

I'll take a look at the speed dial screen. Not sure about the registration status - there will be issues with that once we add more servers, so we'll see.

reply to verix

reply to nitzan
Hi nitzan,

Thanks you for adding the credit.

I tested using Standard Route first. I setup speed dial (21, 23 and 24) to numbers in Guadalajara Mexico and I tried calling several times and have problems to connect to #23 and 24#. After I dialed and I got no ring or busy tone I got the message "The number you dialed 23 has not received a response from the service provider"

I switched to Premium Route, and I tried again 23# and 24# and failure, the calls failed.

I then tried with the number in speed dial 21# and I was able to connect and talk without any problem at all, quality of sound was excellent. My mother told me that it was raining a lot in her city and I believe this is the cause of the problem, the rain caused problem in the local phone lines in Guadalajara and so this is the reason for the calls to 23# and 24# to fail.

In the web page the history for the call to 21# says that it was type standard instead of premium as I expected, and I believe the cost per minute was premium and not standard 2012-06-19 04:23:59 STANDARD 0.015

Thanks

reply to nitzan
Hi nitzan,

I have a question
Are you going to have an access number to use your service as a calling card ? so we can call from a cell phone for example.

Thanks

Still thinking about it, but yes will probably add some calling card numbers later on.

Speed dial issue: will take a look.

Call type: the calls were premium the call type field is just the type between standard/callback/DID call/etc. - it's not the route.

reply to nitzan
OK- for those of you with multiple PBX's/clients/etc. - you can now add additional SIP peers and setup IP authentication under the IP AUTH link inside the web console. Please give it a try and let me know how it works.

It seems to be working even when fed a FQDN instead of an IP for those that might benefit. Still a work in progress, but what nice progress. Those with dynamic addresses should really benefit, and maybe even those that have those lonnnnng IPv6 addresses in the future.

reply to nitzan
Hi nitzan,

I have tried a couple of 800/888 numbers and the call fails. I guess in this beta this is not working but it will work. Will the call to these numbers be free or will your company charge for the call ?

Thanks

reply to nitzan
Call quality better now - no echo.

reply to nitzan
Hi nitzan,

The www.cheapvoipinc.com web does not have a signout/logout timeout. I did login in the morning, I even put my machine to sleep for several hours, then tonight I checked the call history and other pages without requiring to login again. This is convenient while testing and with a private computers but it is best practive to have a timeout for the session. I guess you are planning to implement a timeout for the session.

Regards

reply to nitzan
On the signup form, interesting that Alaska and Hawaii are listed under country .

reply to nitzan
Once set up CID can't be changed or removed. All changes are just ignored.
CIDs set for peers are not honored, either.