Website: SIPphone.com

Service launched in 2004 by founder of Lindows and MP3.com. Offers many unique and interesting features including a do-it-yourself conference bridge, and lots of diagnostic tools. These people seem to be on a mission to destroy the traditional phone system.

SIPphone supports open standards aggressively. They sell PSTN calling by the minute, claiming that most flat rate plans are a poor deal for the consumer. SIPPhone has even sued competitors that sell locked ATA devices without proper disclosure to the consumer.

SIPPhone offers three free softphones ( GizmoProject, X-Lite, and PhoneGaim ), and has a free-calling program All Calls Free under which registered users can call the landline and mobile phones of other Gizmo customers for free.

They have tons of Phone DIDs in different regions of USA, UK and France.

Setting it up on your asterisk
on your sip.conf
register => YOURNUMBER:YOURPASSWORD@proxy01.sipphone.com/EXTENSION_TO_RING

YOURNUMBER - includes the 1 (eg. 17475551212, NOT 7475551212 like others sip providers
YOURPASSWORD - is your password
EXTENSION_TO_RING - the extension on your asterisk box, or if you have some built in ones for your context

then at the end of your sip.conf
[sipphone]
host=proxy01.sipphone.com
context=stana
type=peer
disallow=all
allow=ulaw
canreinvite=no


You'd then put the appropriate line in your extensions.conf
I use FWD to call my SIPPHone account, and use these lines:
;-- SipPhone.com
exten => _1747NXXXXXX,1,SetCallerID(14165551212); <- real looking number (fake or real but real looking)
exten => _1747NXXXXXX,2,SetCIDName(YourName); <-- What ever name
exten => _1747NXXXXXX,3,Dial(IAX2/${FWDNUMBER}:${FWDPASSWORD}@iax2.fwdnet.net/**${EXTEN:1})

idealy the SetCallerID should be your own SIPPhone phone, but any phone number will work.

Dialing from FWD requires you to set your callerid and CDIName (maybe) or else you'll dial from your extension which will be invalid,and won't let you get through to the SIPPHone user.

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Now setting buying your DID from SIPPhone is VERY economical and since you can now connect it to your asterisk it's a fantastic way to give you a phone number in the US, or UK. (now if only they'd do it in canada!)

I got it to work using these settings:
In sip.conf

[general]
register => 17473861234:my_secret@proxy01.sipphone.com/17473861234

[sipphone]
type=friend
secret=mypassword
username=17473861234
fromuser=17473861234
fromdomain=proxy01.sipphone.com
host=proxy01.sipphone.com
context=default
disallow=all
allow=ulaw

In extensions.conf

exten => _1747XXXXXXX,1,Dial(SIP/${EXTEN}@sipphone)
exten => _1747XXXXXXX,2,Hangup

Adding a dial in (DID) number via Les.net

In addition to buying a dialin number from sipphone directly you can also get them from other vendors and then route the traffic to sip phone. For example you can by a DID at les.net for one dollar (canadian) a month with quite most reasonable per minute. I routed that to Gizmo Project (siphone) account easily. Well, easy after I figured out what syntax to use in the les.net UI. Create a peer with type URI and use the format: 17476XXXXXX@proxy01.sipphone.com for the Peer Address; then associate that with your DID by clicking on the your # in the "Your DIDs" page. Works great, real cheap!

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See also:

• Vendor Hosted Forums for SIPphone
• Asterisk settings Gizmo
• sip.conf
• extensions.conf
• GizmoProject