SIP and IAX trunks and providers | Forum profile

Sorry be to such a pain. I found my answer here. http://pushkarbhatkoti.wordpress.com/2009/01/10/cm e-sip-trunking-... I was getting information that was telling me that if endpoints were configured using SCCP that they could not use an SIP trunk. But I found that to be false. -Jeff

I can get numbers on Grand Bahama, Nassau Freeport and the Abaco's Send me a PM if you are interested.

I just signed up for the free DID number from IP Comms and got it working pretty easy. The sign up process was pretty easy, and i just had to activate my account and got my number in about two hours. They have it setup to point to an ip address only right now. So no sip registeration for the free account. You have to sign up for that. The tech support was pretty good especially ...

try setting reinvite=no as it looks like the reinvite was what was unacceptable..

Should work fine. Need more details. Does the provider see your calls?

Check out http://www.flowroute.com They charge 0.0195 per min for toll free service. Their numbers are $1.39 per month but there is a $9.95 setup fee for each number. Seems a bit pricey if you need a lot of numbers but the per minute rates are good.

Have you tried: dtmfmode=rfc2833 rfc2833compensate=yes ???? I used voice pulse for a while (only recently switched out) and have has zero issues with their service AND you shouldnt have these issues with your version of asterisk.

The simplest way to do this without violating the TOS with your SS7 provider (I assume it's Illuminet) is to use a SS7/ENUM gateway. Most ENUM servers are based on BIND and can cache. You can then cache on the enum side. You would have to do this: A Link ====== ENUM ======= F Link ======> Class 5 Switch A Link ====== ENUM ======= F Link ======> Class 5 Switch That would...

You have to add both of your SIP accounts into the Trunks section; this will get your setup to communicate with both accounts. Once you have them communicating (click on the "PBX" pulldown menu then "PBX Status" to see if they are registering) then you can move onto creating the inbound route. From there under "Inbound Call Control" you need to set up a route. The DID is the ...

I have a vBuzzer account where I pay $25 a year for a 416 (toronto) DID and I get free unlimited calls to toronto (416) area...free incoming as well.

Sorry be to such a pain. I found my answer here. http://pushkarbhatkoti.wordpress.com/2009/01/10/cm e-sip-trunking-... I was getting information that was telling me that if endpoints were configured using SCCP that they could not use an SIP trunk. But I found that to be false. -Jeff