Pay as you go on minute basis VoIP plans....

jonathanztaub · 2005-02-18T03:09:05+00:00

I'm new to this whole VoIP thing.I looked into it some months ago but hadn't taken any action yet.My situation is this:I don't talk a lot on the phone. Currently, the cheapest plan I foundwas BroadVoice in state calling which is $9.95 a month.My question is: Is there a plan which you pay as you go along on aminute basis?I've found the website LiveVoIP which seems pretty cheap (you must ofcourse by initialliy lots of minutes - but still cheap). There are someterms out there I don't quite understand: Asterisk, Sipura.Can please somebody explain to me exactly what these things are and arethey complicated/complex to install?Ideally (and it probably doesn't exist), I would like to have a phonewhich I can use to call other people and other people can call me whichuses VoIP technology.In order to minimize costs, I want to pay as I go by the minutes.Any insights will be most helpful.

Feb 18, 2005 25 Replies

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Wolfgang S. Rupprecht 21 years ago

That doesn't sound like fun. I guess any port in a storm...

Yes, that is another problem with Gafachi. It is pretty common though. I recall I had the same problem with broadvoice too. Gafachi and broadvoice need "dtmfmode=inband" to make touch-tones work. Eg. inband touch tones sent as actual tones down the audio path. I'm not sure why they don't provide sip or rtp command mappings for the touch tones, but they don't.

Teliax runs fine with my default setting of "dtmfmode=RFC2833". I've tested it with various auto-attendants and keypresses to navigate remote voice mail systems seem to work just fine. The one thing that does disturb me about Teliax is that when I plot the ping times to their machine it is very, very noisy with some of the pings 4x as long as the average. That makes me think they are a much smaller outfit and might easily get overloaded.

-wolfgang

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Miguel Cruz 21 years ago

Here's what I do - caller ID seems to be sent properly:

exten => _1NXXNXXXXXX,1,SetCallerID(321-654-0987) exten => _1NXXNXXXXXX,2,Dial,SIP/${EXTEN}@gafachi

miguel

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Kyler Laird 21 years ago

This has not been my experience. I was using my Gafachi account last week for some CallerID experimentation and had no problem setting the data.

(Right now I can't seem to make a call with that account.)

--kyler

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Wolfgang S. Rupprecht 21 years ago

Interesting! I wonder what I'm doing differently that it objects to. How are you setting the caller ID? Are you setting it to a 10-digit number or an 11-digit "1+" number?

This is what I'm doing:

extension.conf:

[to-pstn] exten => _1NXXXXXXXXX/6001,1,SetCallerID(WOLFGANG RUPPRECHT ,a) .... exten => _1NXXXXXXXXX,5,Dial(sip/${EXTEN}@gafachi-out)

sip.conf:

[gafachi-out] type=peer secret=mysecret username=myacctnumber fromuser=myacctnumber host=sip.gafachi.com dtmfmode=inband canreinvite=no

Now that I look at it, I wonder if the "fromuser" is interfering with things...

-wolfgang

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Wolfgang S. Rupprecht 21 years ago

Ok, I see how to get callerid to work with them. The trick was indeed to remove the "fromuser" in sip.conf. That caused the authentication to fail. To get that to work again one one needed to use the phony hostname myaccountnumber.sip.gafachi.com and their proxy would gleam the account number from the hostname. Weird, but clever! Thanks for the info that it was now doable.

One down, one to go. Getting reinvite working is important to cutting my local proxy out of the audio path. While the machine sitting on my desk makes is a reasonable enough sip-engine, moving the RTP packets themselves doesn't work out that well, especially when I'm recompiling the world.

-wolfgang

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