Does anyone know the MJ SIP to SIP syntax.

[Mr_Dasher]

Hi,

Does anyone know the syntax for making VOIP to VOIP calls.

It's something like "

"your phone number"@proxy.talk4free.com

or

"USER-ID"@talk4free.com

Thanks for your help.

D

[Mr_Dasher]

I must be missing something here. Sorry :0)

I think it's called SIP to SIP dialing.

What I want to do is to do is have a friend who lives in Europe and using a European VOIP provider to be able to call me directly using my MJ VOIP address? What info would he have to enter?

Would it be something like:

[email protected]

[email protected]

"phone_number"@talk4free.com

Thanks,

D

[Mr_Dasher]

Sorry Domingo,

I'm just being stupid here.

When I want to phone him via SIP only its really simple:

"phone_number"@"his_sip_server":5060 (it could be his user name because it's one and the same)

If he was going to phone my MJ account what would his stream look like?

He's tried:

"MJ_phone_number"@216.234.69.8:5070
"MJ_User_ID"@216.234.69.8:5070

But they don't work.

Thanks again,

D

[Mr_Dasher]

Thanks for the info,

Bare with me here. :0)

When I follow the method you suggest it's really just dialing a standard phone number through his provider. I maybe wrong but I thought I should be able to dial a number not as if it's a phone number but really as if it's an Internet address.

For example when I enter his contact info I don't include my server but I include his.

"His_phone_number"@"the_address_of the server_where_I_can_find_this_phone_number":5060

Soooo..... it looks like this:

[email protected]:5060 (all his info not mine)

This completely avoids using any phone system and keeps it all on the Internet. I think this allows me to call any VOIP phone anywhere in the world without paying long distance.

It works when I do it but it doesn't work when he tries it calling my MJ.

I'm looking for the MJ address info that he can try i.e.

"my_phone_number"@"the_address_of the server_where_he_can_find_my_phone_number":5070

I hope I'm making sense.

Thanks,

D

[crackerjack]

magicjack does not allow any sip peering, for that matter, neither does Vonage. It is a conscious decision on the part of both these providers to behave in this fashion. Many other providers do allow this functionality. Any traffic carried by the MJ infrastructure must ultimately involve PSTN.

[Mr_Dasher]

Thanks Crackerjack.

I thought it would be too good to be true.

D

[dishem]

If you have Asterisk and a publicly routable IP address directly on it or with correct ports forwarded, he could just dial sip:<your_extension>@<your_ip>