"DID Number" for Asterisk

[choanhemnhe]

Need DID number for 2 phone numbers on one Asterisk server.
I setup 2 FPL numbers on one Asterisk server. Every number must be forward to different IVR. To do this I need to fill in "DID number" for each trunk. I tried to fill in username (my phone number); FPL login name (email address), but both do not work!

anyone know this, please help!

Thanks a lot,

[FONGO_steve]

Need DID number for 2 phone numbers on one Asterisk server.
I setup 2 FPL numbers on one Asterisk server. Every number must be forward to different IVR. To do this I need to fill in "DID number" for each trunk. I tried to fill in username (my phone number); FPL login name (email address), but both do not work!

anyone know this, please help!

Thanks a lot,

You need to buy our config file.
Are you using these DID's for business? Business use is strictly prohibited on freephoneline as we are a residential phone provider only!

[choanhemnhe]

Yes I did buy Conf files for both numbers. I use these number for home only. There are 2 family living in my house. The numbers must be forwarded to correct its owner or correct extension.

Please help!
thanks

[evilmonkey]

If you bought the config file, then login to freephoneline.ca and you will see a tab called SIP. Use the credential information that is there for your trunk settings. Use the following information under Outgoing PEER Settings:

Code: Select all

host=voip.freephoneline.ca
context=from-pstn
username=***your_fpl_phone_number***
secret=***Your password from the SIP Page***
type=peer
insecure=very
dtmfmode=inband

Use your FPL phone number for user context and the following for incoming USER Details:

Code: Select all

type=peer
fromdomain=voip.freephoneline.ca
fromuser=***phone number***
username=***phone number***
secret=***SIP password***
dtmfmode=inband

And the following for the registration string: ***fpl_phone_num***:***Sip password***@voip.freephoneline.ca/fpl_phone_num (so it looks like 1xxxXXXxxxx:pass@voip.freephoneline.ca/1xxxXXXxxxx)

Do that twice, once for each line, and you should be good.

[choanhemnhe]

Ok,
Thanks for all helps. Following evilmonkey instructions. I can create Inbound Trunk for my 2 numbers. However my case as follow:
I setup 2 number on the same trixbox server. I want when somebody call phone number #1 the phone up-stair will be ring and when they call phone number #2, down-stair phone will be ring. When setup Inbound trunk, DID number could not leaved blink. Somebody who know what DID number FPL pass to your Trixbox server or ATA box when your number be dialed. I tried to fill in "email address", "account number", "phone number". But both did not work.
thanks

[evilmonkey]

You need to set up two inbound routes. For the first one, fill in the DID of the number you want to ring the upstairs phone and set the destination as the extension that you set up upstairs. Do the same thing for downstairs.

The DID number is your FPL phone number starting with 1. For example, if your phone number is (647)123-4567, then your DID is 16471234567. Good luck!

Sorry, I don't check this forum every day, so it may take a bit of time for me to answer.

[soupalognon]

You need to set up two inbound routes. For the first one, fill in the DID of the number you want to ring the upstairs phone and set the destination as the extension that you set up upstairs. Do the same thing for downstairs.

The DID number is your FPL phone number starting with 1. For example, if your phone number is (647)123-4567, then your DID is 16471234567. Good luck!

Sorry, I don't check this forum every day, so it may take a bit of time for me to answer.

I just tried it and it work, make sure you put the 1 in front of your phone number like evilmonkey said.

Soupalognon.

[patx]

I followed this post and it worked out of the box.... but after my asterisk server crashed and i had to recompile the hole thing... I followed this post again but now i get "this account is not valid" message on my outbound calls... inbound is perfect...

I have check and recheck for bad typo but that is not the problem... my config is exactly the same as it was before the crash...

Asterisk CLI return:

-- Called 1418xxxxxxx-out/1819xxxxxxx
-- SIP/1418xxxxxxx-out-00000005 is making progress passing it to SIP/xxxxxxx-00000004
-- Got SIP response 603 "Declined" back from 208.65.240.142

208.65.240.142 is a freephoneline IP.

any idea ?!

[patx]

OK got it to work again...

I used the info on this post to fix the problem:

h**p://fonality.com/trixbox/node/46383

(I dunno if external linking is permitted on this board so change ** to tt)