New York World Phone

I’m using WRT54GS + Openwrt + asterisk at home. It is good because we have all controls. But admin password will be lost from time to time, then I’ll have to reset it: http://blog.nyworldphone.com/2008/02/06/openwrt-on-wrt54gs-reset-the-root-password/. And, for some providers, I can’t make it work with my asterisk on wrt54gs.

Then I tried pbxes.org. It is still free and works perfect. I can do everything I’m doing right now with my asterisk.  It is a good option for personal use which cost nothing. Here I would like to explain some configurations that I did.

1. Receiving calls on Sip phone/device.

A) Internal between extensions.

• Extensions-> Add Extension
• Display name(e.g. 3000)
• Password(e.g. 3000)
• Outbound CID(e.g. 3000)
• Voicemail(optional, disabled by default)
• Submit

You could do the same thing above to add more extensions (e.g. 6000 etc).

When you configure it on your sip phone/device, usename is “pbxes_username”-exten, e.g. username-3000, password is your pbxes extension password and domain/sip proxy is pbxes.org. Now you can dial extensions from one to another, e.g. dial 6000.

B) Through a DID number

a. Add extensions as A).

b. Add trunks

• Trunks->Add Trunk->(e.g. Add SIP Trunk)
• Trunk Name (e.g. freedigits)
• username (e.g. freedigits number)
• password (e.g. freedigits password)
• Sip server(e.g. freedigits.net)
• Submit changes

 c. Add Inbound Routing

• Inbound Routing->Add Incoming Routing
• Trunk: (e.g. freedigits)
• Caller ID Number (optional, specify allowed incoming CID)
• Set Destination: mark Extension, and choose one from the list, e.g. 3000.
• Submit

Now when people call the freedigits DID, calls will be routed to the extension 3000.

2. Make calls

A) Add extensions as 1.A.

B) Add trunk as 1.B.b with your favorite voip provider, e.g. voipbuster.

C) Set outbound routing

• Outbound Routing->Add Route
• Route name: (e.g. sip out)
• Trunk sequence: (e.g. voipbuster)
• Set Destination (optional)
• Save changes

Now you can make calls usign your favorite vsp from your sip phone/device.

3. Call Forwarding/Receiving calls on PSTN phones (landline or cell phones).

A) Add trunks as 1.B.b.

B) Add ring groups

• Ring Groups->Add Ring Group
• Group number: (e.g. 1)
• extension list: landline or cell phone number plus #. e.g. 12126668888# (it may differ according to your provider)
• ring time, e.g. 20
• Destination if no answer (optional)
• Submit changes

C) Add inbound routing

Same as 1.B.c, except in “Set Destination” mark Ring Group(e.g. #1).

D) Add outbound routing as 2.C.

Now if people call your freedigits DID, the call will be forwarded to your regular phone using through your vsp.

4. Call back

Start with callthru by adding an inbound route with the number of your phone entered as Caller ID. Leave Trunk Name empty. If it doesn’t work check your call monitor for the right format of your Caller ID. When dialing the callthru destination you may always press the * key to redial and # to end digits. After the called party hangs up you will hear another dialtone for your next call. To hangup by yourself transfer the call to an invalid destination by dialing *2.

- alen

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This entry was posted on Wednesday, June 11th, 2008 at 8:39 pm and is filed under Asterisk, DID, Voip Services. You can follow any responses to this entry through the RSS 2.0 feed. You can leave a response, or trackback from your own site.