Asterisk@Home Handbook Wiki Chapter 6

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Chapter 6 VOIP Service Providers


There are many service providers. Some provide proxy server that make it possible to connect to other members of that provider. Other providers offer both incoming and outgoing PSTN to VOIP termination. Here are a few common providers and how to make the work with Asterisk@Home.

Most providers will give you phone number and a password for that provider some will also give you a user name. If you get a real PSTN number from the provider it will be a normal 10 digit number (US providers). some providers give out shorter number that can only be used by other members of that provider.

The following site provides alot of useful information regarding
VOIP providers rates, connection types, and county availability
VOIP Charges


6.1 Free World Dialup (FWD)

Contact: http://www.freeworlddialup.com/
Service: proxy to other FWD users, Gateway to other providers
Protocol: SIP or IAX
Cost: free

You should have a phone number (123456) and a password (wibble). You also need to have your FWD account setup for IAX. This is achieved by visiting http://www.freeworlddialup.com, logging in and turning on IAX. This is done in the "Extra Features" area of your account page. It does take a little bit of time to be set up (10 mins or so), so do that first. Once you have turned it on and clicked 'Submit' enough times (I noticed I had to click Submit two or three times before it came up with 'Changes Successful', that may have just been a temporary glitch) you're ready to proceed below.

Once again, you need to be in AMP, the Asterisk Management Portal. Click on Setup at the top of the page, but this time click on Trunks on the left. Click on Add IAX2 Trunk

Outbound Caller ID should (but does not have to be) set to your FWD Number. This is what is displayed when you call someone else through FWD. They would normally just see your Extension (200).

Outgoing Settings

Trunk Name: fwd (This is just a descriptive name, and is what appears on the left of the screen)
PEER Details: (Change '123456' and 'wibble' to be your FWD Number and Password)


host=iax2.fwdnet.net
type=peer
username=123456
secret=wibble


Incoming Settings
USER Context: iaxfwd (Pay attention here. Don't change it. or it won't work)
USER Details (Nothing needs to be changed here, this can be pasted straight in)


allow=ulaw
auth=rsa

context=from-pstn
disallow=all
inkeys=freeworlddialup
type=user


Register String: should be set to yournumber:yourpassword@iax2.fwdnet.net, using our examples above, it would be 123456:wibble@iax2.fwdnet.net


Click 'Outbound Routing' from the menu, and then click 'Add Route'

Name your route something like 'fwd'
The dial prefix is, usually, 393 — That's 'FWD' on your phones pad.
Dial Patterns: 393|X.
Trunk Sequence: IAX2/fwd

Click 'Submit Changes'
You may have to move the trunk further up the priority list.

From the asterisk command line type the following to see if the new connection is registered.
iax2 show registry

Assuming you've got your username and password correct, you should now be able to dial '393612', Which will read out the time to you. IF you are feeling exceptionally brave, call '393613', which is a useful little echo tester - it'll just bounce back to you everything you say to it. You can then try '393514' which is FWD's 'Coffee Lounge' - I've never actually successfully had a conversation with anyone there, however, or '39355555', which calls a random volunteer, so you can actually speak to a live person!


6.2 Free World Dialup OUT (FWD)

Contact: http://www.fwdout.net/web/
Service: Gateway to other providers
Protocol: IAX
Cost: Share and Share Alike

FWDout is The Service Formerly Known as <name withheld>
You must read the documentation carefully and be aware that a poorly configured Asterisk@Home box can be used by other people on the fwdOUT network to make long distance calls that you may end up paying dearly for.
Create an account on http://www.fwdout.net/bell-cgi/signup.cgi

Once again, you need to be in AMP, the Asterisk Management Portal. Click on Setup at the top of the page, and click on Trunks on the left. Click on Add IAX2 Trunk

Outbound Caller ID should left blank


Outgoing Settings
Trunk Name: fwdOUT (This is just a descriptive name, and is what appears on the left of the screen)
PEER Details: (Change '123456' and 'wibble' to be your fwdOUT Number and Password)


username=123456
type=peer
secret=wibble
host=iax.fwdOUT.net


Incoming Settings
USER Context: iaxfwdOUT (Pay attention here. Don't change it. or it won't work)
USER Details (Nothing needs to be changed here, this can be pasted straight in)


type=user
inkeys=freeworlddialup
disallow=all
context=from-pstn
auth=rsa
allow=ulaw
allow=gsm


Register String: should be set to yournumber:yourpassword@iax2.fwdOUT.net, using our examples above, it would be 123456:wibble@iax2.fwdnet.net


Click 'Outbound Routing' from the menu, and then click 'Add Route'

Name your route something like 'fwdOUT'
The suggested Dial prefix for fwdOUT is 394, although this is optional

Dial Patterns: 394|X.
Trunk Sequence: IAX2/fwdOUT

Click 'Submit Changes'
You may have to move the trunk further up the priority list.

From the asterisk command line type the following to see if the new connection is registered.
iax2 show registry

If you have another provider for long distance place the fwdOUT before your providers Trunk so that outbound calls are routed through fwdOUT

fwdOUT will allow you to make long distance phone calls using other people's asterisk boxes, while allowing other people to route calls through your asterisk box. The idea is that you do not pay for calls in your local area, so you can let people route calls through your server, and other people do the same for you.

6.3 VoicePulse

Contact: http://connect.voicepulse.com/
Service: PSTN termination
Protocol: IAX
Cost: pay

Once again, you need to be in AMP, the Asterisk Management Portal. Click on Setup at the top of the page, but this time click on Trunks on the left. Click on Add IAX2 Trunk

Dial Prefix 9 if you're not already
Leave Default Trunk switched off (or make this the default if you want all your calls to use it)
Outbound Caller ID should (but doesn't have to be) set to your VoicePulse Number.

Outgoing Settings
Trunk Name: voicepulse-out-01 (This is just a descriptive name)
PEER Details: (Change <your username> and <your password> to be your VoicePulse Number and Password)


host=gwiaxt01.voicepulse.com
secret=<your password>
type=peer
username=<your username>


Incoming Settings

USER Context: voicepulse-in-01 (Pay attention here. Don't change it. or it won't work)
USER Details (Nothing needs to be changed here, this can be pasted straight in)


auth=rsa
context=from-pstn
inkeys=voicepulse01
type=user


Register String: should be set to <your username>:<your password>@gwiax-in-01.voicepulse.com example (bob:abc123 @gwiax-in-01.voicepulse.com)

That's it. Click on Submit Changes, and then on the big red 'You have made changes' bar and you're done.
For a test make a call (try 1-800-555-1212)

6.4 Sixtel

Contact: http://www.iax.cc/
Service: PSTN termination
Protocol: IAX
Cost: pay

Merged into Vitelity Communications LLC. You may need to refer to Section 6.20

Iax.cc, also known as sixTel is a small VOIP termination provider that offers very low rates for inbound and outbound calls. With rates at low as 1.43 cents per minute and a good number of local and toll free numbers to choose from, sixTel is a popular choice for home and small business users.

Once again, you need to be in AMP, the Asterisk Management Portal. Click on Setup at the top of the page, but this time click on Trunks on the left. Click on Add IAX2 Trunk

Use the following example to get you up and going:

Outbound caller ID: "Your Name" <1XXXXXXX>
Maximum Channels: 4

Trunk Name: sixTel

Peer Details:
allow=all
context=ext-did
host=iax2.sixtel.net
secret=myPassword
type=friend
username=myUserName

User Context: <<blank>>
User Details: <<blank>>

Registration String: myusername:mypassword@iax2.sixtel.net

On the DID tab, create a new DID
DID: 949XXXXXXX <- use sixtel DID number

Set this new DID to use your "Normal Incoming Calls Setting".

Finally, on your "Outbound Routes" tab, you will need to add the sixTel trunk to one of your outbound trunks.

Once you save your settings, click on the red bar at the top of the screen, wait a few seconds, and you should be able to send and receive calls through Iax.cc/sixTel.
For a test make a call (try 1-800-555-1212)


6.5 VoipJet

Contact: http://www.voipjet.com/
Service: PSTN termination
Protocol: IAX
Cost: trial/pay

VoipJet allows you to create a free trial account to test your system. Once you have things working and have made a call you can buy extra credit. Once you have registered, for free, and received your free credit to do some basic testing, VoipJet provides instructions on how to configure Asterisk to use your new account.

In the Asterix@home management portal, Click on: Setup at the top, Trunks on the left, add IAX2 Trunk in the middle.
General Settings, leave this section blank.
Outgoing Dial Rules, leave this section blank.
Outgoing Settings, edit this section:
Trunk Name: 7374@voipjet (7374 is the userID given to you, you can get it from the VoipJet page which helps with the settings)
PEER Details:
auth=md5
context=ext-did
host=64.34.45.100
notransfer=yes
username=7374
secret=ecbf7ffca7631227 (this is the MD5 hash that is given to you, on the help page)
type=peer

Incoming settings
User Context: voipjet
User Details:
auth=md5
context=ext-did
host=64.34.45.100
notransfer=yes
username=7374
secret=ecbf7ffca7631227 (this is the MD5 hash that is given to you, on the help page)
type=peer

Registration, leave this section blank

Submit and click the red save bar at the top.


6.6 MyNetfone - AUSTRALIA

Contact: http://www.mynetfone.com.au//
Service: PSTN termination
Protocol: SIP
Cost: pay

MyNetfone provides free user to user calls and A$0.10 (~US$0.065) untimed calls to Australian landline numbers (+6113 +612, +613, +617 +619). Check their plans here: http://www.mynetfone.com.au/plans/

Outgoing settings are:
allow=alaw&ulaw&g729
authname=0911XXXX
canreinvite=no
disallow=all
dtmfmode=rfc2833
fromuser=0911XXXX
host=sip.myfone.com.au
insecure=very
pedantic=no
qualify=yes
secret=<your password>
type=peer
username=0911XXXX


They also provide indial numbers to people with an Australian address (yes that's still regulated in Australia).
Incoming settings are:
You don't need this to make outgoing calls only


Registration:
You don't need this to make outgoing calls only

Submit and click the red save bar at the top.


6.7 TelaSIP

Contact: http://www.telasip.com/
Service: PTSN Termination
Protocol: SIP
Cost:pay

TelaSIP trunk configuration:
Oubound caller ID: "J Smith" <5212314214> (substitute with your name and DID)
Maximum channels: 2
Dialing rules: (substituting your local area code for 404 below)
1|NXXNXXXXXX
NXXNXXXXXX
404+NXXXXXX
Outgoing Settings:
Trunk Name: telasip-gw
Peer details (using your own account name/password):
type=peer
host=gw4.telasip.com
qualify=yes
insecure=very
context=from-pstn
username=<username>
secret=<secret>
Registration: youraccountname:yourpassword@telasip-gw

Configure outbound routing
Add route: outgoing
Dial patterns:
1NXXNXXXXXX
NXXNXXXXXX
NXXXXXX
Trunk sequence: 0=SIP/telasip-gw



6.8 Exgn LLC

Contact: http://www.exgn.net
Service: PTSN Termination
Protocol: SIP, IAX
Cost: Pay

Merged into Vitelity Communications LLC. You may need to refer to Section 6.20

Exgn is a VOIP service provider that uses Asterisk themselves and is very Asterisk friendly. Supports both SIP and IAX protocols. Provides failover forwarding and voicemail in case your Asterisk server is offline. Their user portal is one of the best seen in this industry which allows instant activation for DIDs in hundreds of ratecenters across the country along with a pool of toll free numbers to choose from. As of March 5, 2006 they also offer e911 service for their DIDs in the USA with Canadian e911 service coming soon.

These instructions are for AMP (Asterisk Management Portal). Click on 'Setup' at the top of the page, then click on 'Trunks' on the left, Then click 'Add IAX2 Trunk'.

Enter the following information in the appropriate fields:

Outbound Caller ID: "Your-Name" <NPANXXNXXX>
Maximum Channels: 2

Trunk Name: exgn

Peer Details:
allow=all
context=ext-did
host=iax.exgn.net
secret=your-password
type=friend
username=your-username

User Context: <blank>
User Details: <blank>

Registration String: your-username:your-password@iax.exgn.net

This DID should be set to use your "Normal Incoming Calls Setting"

Lastly, on your "Outbound Routes" tab, you will need to add the exgn trunk to one of your outbound trunks.

Now you should be able to make and receive calls!

If you cannot, please feel free to email support@exgn.net or open a trouble ticket within our user portal/control panel.


6.9 Gizmo Project / SIPphone

Contributed by: Casey

Contact: http://www.gizmoproject.com
Service: DID, PTSN Termination, Gateway to other providers and universities
Protocol: SIP
Cost: Gateway is free. DID and PSTN termination are pay.

Its free to setup an account on the Gizmo Project. It's free to use as a gateway to other providers.

Gizmo In provides DID service (a standard telephone phone number others can call from traditional phones that rings in to your Asterisk@Home box) . And Gizmo Out offers PSTN termination to most countries.

The following setting work with both the Gizmo IN / DID service, and the Gizmo Out long-distance service. I try to use the minimal settings to get things working, then add the bells and whistles later. Here are the basic settings I've used to setup A@H with Gizmo Project/SIPphone with Asterisk@Home 2.8.

In the settings below 17470000000 should be replaced with your Gizmo or SIPphone number. And Password111 should be replaced with your Gizmo or SIPphone password.

In Asterisk@Home, add a new SIP trunk. Remove any pre-filled text from the fields, then only add:

Trunk name: proxy01.sipphone.com

Peer Details:
allow=ulaw&alaw&gsm&ilbc
canreinvite=no
context=from-pstn
disallow=all
dtmfmode=rfc2833
fromdomain=proxy01.sipphone.com
fromuser=17470000000
host=proxy01.sipphone.com
insecure=very
secret=Password111
type=peer
username=17470000000


Register String: 17470000000:Password111@proxy01.sipphone.com

Submit the changes, then click the red link at the top of the page to apply the changes. That's it. If you're interested in how/why I set the items I did, read on...

First, I had read that beginning with recent versions of Asterisk, the User Context/Details (Incoming Settings) have been depreciated. Instead, it had been combined with the Peer Details (Outgoing Settings). I've eliminated the User Context/Details completely from my configuration, and it continues to work.

I found that if I didn't specify insecure=very and allow=ulaw that DID would not work. Instead, the incoming caller would be greeted with Gizmo / SIPphone's "the person you are calling has not setup voice mail" message. FYI allow=ilbc will also work. The context=from-pstn makes the incoming DID calls get handled according to the settings in the AMP Incoming Calls tab.

If you don't have Gizmo Out (long-distance) minutes, you don't need the fromdomain=proxy01.sipphone.com and the fromuser=17470000000 settings. You'll be limited to the gateway features, and toll-free calls, without those settings.

FYI Gizmo Project / SIPphone officially announced support for Asterisk (which, coincidentally, would include Asterisk@Home) on May 23, 2006. The original instructions, above, were posted March 6, 2006. We're on it like Blue Bonnet!


6.10 Iristel

Contact: http://www.iristel.ca
Service: DID, PTSN Termination
Protocol: SIP
Cost: $15.95/month CAD for one DID with unlimited local termination.

When you first sign up with Iristel, select the "I will use my own SIP gateway" option. When your account is activated, they will e-mail you a PDF document with sample Asterisk configuration. Use that document for reference. It has on it your assigned DID, you user ID number, and your password to access the service. To setup Iristel's service using the AMP GUI, follow these instructions:

  1. Click on Setup in the menu at the top of the page.
  2. Click on Trunks in the menu on the left of the page.
  3. Click on the Add SIP Trunk link.
  4. Under Outbound Caller ID, enter the following: "You Name" <11231234567> (replacing Your Name with your desired caller ID name, and 11231234567 with your assigned DID. Make sure this DID is in international dialing format (re: Include the leading one!). Also ensure that you do enclose your name in quotes, as this is the format Asterisk is expecting.
  5. Under Dialing Rules, enter the following: 1+NXXNXXXXXX This will add a leading one to locally dialed outgoing calls on this trunk. Locally dialed calls must be dialed in international dialing format or Iristel's SIP proxy will reject the call.
  6. Under Trunk Name enter: irisbax.iristel.net
  7. Under Peer Details enter:
callerid=1416xxxxxxx
dtmfmode=rfc2833
host=irisbax.iristel.net
insecure=very
secret=1111
type=peer
username=40932998

Leave the Incoming settings boxes blank. Under Register String, enter: 11231234567:<password>:<userID>@irisbax.iristel.net/11231234567 (replacing 11231234567 with your assigned DID, <password> with your password (excluding the angle brackets) and <userID> with your assigned user ID number (without the angle brackets) This string is provided to you in the setup PDF, you may copy and paste it here).

Your Iristel trunk is now ready to send and receive calls. Simply setup an outbound route to match your local area code, or all long distance calls if you wish. Just make sure that any call being sent to the Iristel SIP proxy is in international dialing format.


6.11 Voxee

Contributed by: Casey

Contact: http://www.voxee.com
Service: PTSN Termination
Protocol: IAX or SIP
Cost: Pay

Voxee provides outbound call termination to the PSTN. At the time of this handbook entry, Voxee's rate for the U.S. was 1.1-cents per minute, with 6-second (1/10-minute) billing. Here is the basic IAX configuration to get you started:

Add a new IAX trunk to Asterisk. Delete any pre-filled Peer Details information, and delete any pre-filled User Details information. Then add only the following (replace Username111 with the username Voxee assigned to you, and Password111 with your Voxee password):

Trunk Name: Voxee

Peer Details:
allow=alaw&ulaw&gsm&ilbc
canreinvite=no
disallow=all
host=66.246.246.52
secret=Password111
type=peer
username=Username111

The config above includes all the non-royalty codecs supported by Voxee. They also support g729. Feel free to remove any codec from the allow= line that you don't want to use (or add g729 if you have a license for it).

6.12 Gafachi

Contact: http://gafachi.com/
Service: PSTN termination
Protocol: SIP
Cost: pay

Get your GAFACHI_USER+GAFACHI_SECRET from the gafachi page, they are different from your login!

Once again, you need to be in AMP, the Asterisk Management Portal. Click on Setup up the top, but this time click on Trunks on the left. Click on Add SIP Trunk. Empty out the values.

Outbound Caller ID should be set without the country code. E.g 212XXXXXXX instead of 1212XXXXXXX


Outgoing Settings
Trunk Name: gafachi
PEER Details:

allow=ulaw
canreinvite=no
context=from-pstn
dtmfmode=rfc2833
fromuser=GAFACHI_USER
host=GAFACHI_USER.sip.gafachi.com
secret=GAFACHI_SECRET
type=friend
user= GAFACHI_USER
username=GAFACHI_USER


Incoming Settings

Leave blank (Took time before I got that far)

Register String: GAFACHI_USER:GAFACHI_SECRET@GAFACHI_USER.sip.gafachi.com


Note for Gafachi inbound 800 users.
I was fighting the issue of incoming calls being rejected for days until I happened to set the Incoming User Context to a blank field.
This wiped out the incoming settings that I had been tweaking. It also fixed my incoming call problem!
I am happily running with only the trunk name and registration string populated.



6.13 Acanac

Contact: http://www.acanac.ca/ or http://www.acanac.com/
Service: DID, PSTN termination
Protocol: SIP
Cost: pay
Asterisk@Home Ver. Tested: 2.7

You will need your username (your phone number) and your password from Acanac

1. Click on Setup in the menu at the top of the page.
2. Click on Trunks in the menu on the left of the page.
3. Click on the Add SIP Trunk link.
4. Under Outbound Caller ID, enter the following: "You Name" <1231234567>
(replacing Your Name with your desired caller ID name, and 1231234567 with your assigned DID. Ensure that you do enclose your name in quotes, as this is the format Asterisk is expecting.)
5. Under Trunk Name enter: acanac
6. Under Peer Details enter:

NOTE: The IP provided here is for East server 1, there are many servers so choose the correct IP.

callerid=<your acanac phone number>
dtmfmode=inband
host=66.49.255.38
insecure=very
secret=<your acanac password>
type=peer
username=<your acanac phone number>


1. Under User Context enter: <Your acanac phone number>
2. Under User Details enter:

callerid=<your acanac phone number>
context=from-pstn
host=66.49.255.38
insecure=very
secret=<your acanac password>
type=user
username=<your acanac phone number>


1. Under Register String, enter: <your acanac phone number>:<your acanacpassword>@66.49.255.38/<your acanac phone number>
2. Click "Submit Settings


At this time you can receive calls and send calls; however, you still need to set your Inbound Routing with your acanac phone number as your "DID Number"


6.14 Stanaphone

Contact: http://www.stanaphone.com/
Service: DID, PSTN termination
Protocol: SIP
Cost: free(inbound), pay(outbound)

You will need your username (your phone number) and your password from Stanaphone. Use the information provided in the SIP Settings section of the Account Information page.

1. Click on Setup in the menu at the top of the page.
2. Click on Trunks in the menu on the left of the page.
3. Click on the Add SIP Trunk link.
4. Under Outbound Caller ID, enter the following: "Your Name" <1231234567>
(replacing Your Name with your desired caller ID name, and 1231234567 with your assigned DID. Ensure that you do enclose your name in quotes, as this is the format Asterisk is expecting)
5. Under Maximum Channels enter: 2
6. Under Dial Rules enter: 1+NXXNXXXXXX
(or whatever other dial rules would be appropriate for your locale)
7. Under Trunk Name enter: stanaphone-out (or whatever you want to call this trunk)
8. Under Peer Details enter:
canreinvite=no
dtmfmode=rfc2833
fromdomain=sip.stanaphone.com
fromuser=<your stanaphone username — NOTE: Not your account login>
host=sip.stanaphone.com
insecure=very
nat=yes (if you are behind a router which you probably are)
qualify=yes
secret=<your stanaphone password>
type=friend
username=<your stanaphone username>

9. Under User Context enter: <your stanaphone username>
10. Under User Details enter:
auth=md5,plaintext
canreinvite=no
context=from-pstn
fromuser=<your stanaphone username>
host=sip.stanaphone.com
insecure=very
nat=yes
qualify=yes
secret=<your stanaphone password>
type=peer

11. Under Register String, enter:
<your stana username>:<your stana password>@sip.stanaphone.com/<your stana username>
example: 08123456:randomletterpasswd@sip.stanaphone.com/08123456
12. Click "Submit Settings

At this time you can receive calls and send calls; however, you still need to set your Inbound Routing with your stanaphone phone number as your "DID Number". You can then route this as any other route (most likely to the "Use Incoming Calls Setting")

6.15 VBuzzer

Contact: http://www.vbuzzer.com/
Service: DID, PSTN termination
Protocol: SIP
Cost: free(inbound), free(local outbound), pay(outbound)
Asterisk@Home Ver. Tested: 2.7

You will need your username, your phone number, and your password from VBuzzer. Note that I had to install thier software and connect for the first time in order to activate the DID. After such time, the software was unnecessary.

1. Click on Setup in the menu at the top of the page.
2. Click on Trunks in the menu on the left of the page.
3. Click on the Add SIP Trunk link.
4. Leave "Outbound Caller ID" blank
5. Under Trunk Name enter: vbuzzer
6. Under Peer Details enter:

allow=ulaw&gsm
authname=<your username>
canreinvite=no
context=from-pstn
disallow=all
dtmf=rfc2833
dtmfmode=rfc2833
fromdomain=vbuzzer.com
fromuser=<your username>
hidecallerid=yes
host=vbuzzer.com
insecure=very
nat=no
port=80
qualify=yes
secret=<your password>
type=peer
user=<your username>
useragent=VBuzzer/1.1.0.9
username=<your username>


1. Under User Context enter: <Your vbuzzer phone number>
2. Under User Details enter:

authname=<your password>
canreinvite=no
context=from-pstn
dtmfmode=inband
fromdomain=vbuzzer.com
fromuser=<your password>
host=vbuzzer.com
insecure=very
nat=yes
port=80
secret=<your password>
type=user
user=<your password>
useragent=vbuzzer/1.1.1.0
username=<your password>



1. Under Register String, enter: <your username>:<your password>:<your username>@vbuzzer/<your phone number>
2. Click "Submit Settings

NOTE: For the registration Asterisk has a bug where if you put vbuzzer.com:80, it will continue to try and register on 5060. you must put the context in the register string.. (in this case vbuzzer)
NOTE 2: All your phone number entries should have the leading 1 on the number
NOTE 3: You will need to work your dialplans a bit as to dial out you must have the leading 1 or 011.

At this time you can receive calls and send calls; however, you still need to set your Inbound Routing with your acanac phone number as your "DID Number"


6.16 Broadvoice

Contact: http://www.broadvoice.com/
Service: DID, PSTN termination
Protocol: SIP
Cost: pay
Asterisk@Home Ver. Tested: 2.7

You will need to determine which of the Broadvoice SIP servers is closest to your location and then set it in your hosts file as sip.broadvoice.com.

1. Click on Setup in the menu at the top of the page.
2. Click on Trunks in the menu on the left of the page.
3. Click on the Add SIP Trunk link.
4. Leave "outbound caller ID" blank
5. Under Trunk Name enter: sip.broadvoice.com
6. Under Peer Details enter:

authname=<your phone number>
canreinvite=no
context=from-pstn
dtmf=inband
dtmfmode=inband
fromdomain=sip.broadvoice.com
fromuser=<your phone number>
host=sip.broadvoice.com
insecure=very
qualify=yes
secret=<your password>
type=peer
user=phone
username=<your phone number>


1. Under User Context enter: <your phone number>
2. Under User Details enter:

authname=<your phone number>
canreinvite=no
context=from-pstn
dtmf=inband
dtmfmode=inband
fromdomain=sip.broadvoice.com
fromuser=<your phone number>
host=sip.broadvoice.com
insecure=very
secret=<your password>
type=user
user=phone
username=<your phone number>



1. Under Register String, enter: <your phone number>@sip.broadvoice.com:<your password>:<your phone number>@sip.broadvoice.com
2. Click "Submit Settings

At this time you can receive calls and send calls; however, you still need to set your Inbound Routing with your broadvoice phone number as your "DID Number"

6.17 WENGO

Contact: http://www.wengo.fr, http://www.wengo.com
Service: DID, PSTN termination
Protocol: SIP
Cost: free(wengo only), pay(DID, outbound)
Asterisk@Home Ver. Tested: 2.5, 2.8

1. Click on Setup in the menu at the top of the page.
2. Click on Trunks in the menu on the left of the page.
3. Click on the Add SIP Trunk link.
4. Leave "outbound caller ID" blank
5. Under Trunk Name enter: wengo
6. Under Peer Details enter:

allow=g729&alaw&ulaw&ilbc
canreinvite=no
context=from-pstn
disallow=all
fromdomain=voip.wengo.fr
fromuser=<your username>
host=voip.wengo.fr
insecure=very
nat=yes
promiscredir=yes
qualify=yes
secret=<your SIP password>
type=peer
username=<your username>


To retrieve your SIP password:
- on wengo.com: connect you to your account center my wengo, select my account settings, then ATABox
- on wengo.fr: connect you using espace perso, mon profil, mes coordonnées, then paramétres de votre compte, then wenbox

7. Under User Context enter: <leave empty>
8. Under User Details enter: <leave empty>

9. Under Register String, enter: <your username>:<your SIP password>@voip.wengo.fr/<you wengo number>
10. Click "Submit Settings"

At this time you can receive calls and send calls; however, you still need to set your Inbound Routing with your wengo phone number as your "DID Number"

6.18 QuantumVoice

Contact: http://www.QuantumVoice.com
Service: DID, PSTN termination
Protocol: SIP / Plans for IAX
Cost: Various Residental/Business Plans
Asterisk@Home Ver. Tested: 2.8

1. Click on Setup in the menu at the top of the page.
2. Click on Trunks in the menu on the left of the page.
3. Click on the Add SIP Trunk link.
4. Leave "outbound caller ID" blank
5. Under Trunk Name enter: QuantumVoice
6. Under Peer Details enter:

allow=g729&gsm&ulaw&alaw
canreinvite=yes
context=from-pstn
disallow=all
host=sipdr.quantumvoice-sip.com
insecure=very
nat=yes
secret=Password
type=friend
username=Normally telephone number

7. Under User Context enter: <leave empty>
8. Under User Details enter: <leave empty>

9. Under Register String, enter: <your username>:<your SIP password>@sipdr.quantumvoice-sip.com/<you QuantumVoice number>
10. Click "Submit Settings"

At this time you can receive calls and send calls; however, you still need to set your Inbound Routing with your QuantumVoice phone number as your "DID Number"

6.19 TalkLITE.NET

Contact: http://www.talklite.net
Service: PSTN termination
Protocol: SIP / IAX
Cost: Starts at $0.01 for US48
Asterisk@Home Ver. Tested: 2.8
Trixbox v.1.0

1. Click on Setup in the menu at the top of the page.
2. Click on Trunks in the menu on the left of the page.
3. Click on the Add SIP Trunk link.
4. Leave "outbound caller ID" blank
5. Under Trunk Name enter: talklite
6. Under Dial Rules:
1360+NXXXXXX <----- This will allow you to dial 7 Digit numbers for the 360 areacode.
Change the areacode to match your own areacode to dial 7
Digits from your home phone.
1+NXXNXXXXXX <----- This will allow you to call just AreaCode+Number

7. Under Peer Details enter:

allow=ulaw
canreinvite=yes
disallow=all
host=sip,talklite.net
nat=yes
qualify=yes
secret=Password From E-Mail
type=friend
username=Username From E-Mail

8. Under User Context enter: <leave empty>
9. Under User Details enter: <leave empty>
10. Under Register String, enter: <your username>:<your SIP password>@sip.talklite.net
11. Click "Submit Settings"

Under Outbound Routes
1. Route Name: talklite
2. Dial Patterns
X.

3.Trunk Sequence SIP/talklite or IAX/talklite
4. Click "Submit Settings"


6.20 Vitelity Communications LLC (merger between Exgn LLC and Sixtel)

Contact: http://www.vitelity.net
Service: PTSN Termination
Protocol: SIP, IAX
Cost: Pay

Vitelity Communications announced its official launch on 07/18/2006, combining two of the industries leading VoIP providers, Exgn LLC (www.exgn.net) & Sixtel Communications (www.iax.cc). Vitelity began migrating EXGN customers shortly after this mid-July announcement.
Return to the Asterisk@Home Wiki Handbook Table of Contents


Chapter 6 VOIP Service Providers


There are many service providers. Some provide proxy server that make it possible to connect to other members of that provider. Other providers offer both incoming and outgoing PSTN to VOIP termination. Here are a few common providers and how to make the work with Asterisk@Home.

Most providers will give you phone number and a password for that provider some will also give you a user name. If you get a real PSTN number from the provider it will be a normal 10 digit number (US providers). some providers give out shorter number that can only be used by other members of that provider.

The following site provides alot of useful information regarding
VOIP providers rates, connection types, and county availability
VOIP Charges


6.1 Free World Dialup (FWD)

Contact: http://www.freeworlddialup.com/
Service: proxy to other FWD users, Gateway to other providers
Protocol: SIP or IAX
Cost: free

You should have a phone number (123456) and a password (wibble). You also need to have your FWD account setup for IAX. This is achieved by visiting http://www.freeworlddialup.com, logging in and turning on IAX. This is done in the "Extra Features" area of your account page. It does take a little bit of time to be set up (10 mins or so), so do that first. Once you have turned it on and clicked 'Submit' enough times (I noticed I had to click Submit two or three times before it came up with 'Changes Successful', that may have just been a temporary glitch) you're ready to proceed below.

Once again, you need to be in AMP, the Asterisk Management Portal. Click on Setup at the top of the page, but this time click on Trunks on the left. Click on Add IAX2 Trunk

Outbound Caller ID should (but does not have to be) set to your FWD Number. This is what is displayed when you call someone else through FWD. They would normally just see your Extension (200).

Outgoing Settings

Trunk Name: fwd (This is just a descriptive name, and is what appears on the left of the screen)
PEER Details: (Change '123456' and 'wibble' to be your FWD Number and Password)


host=iax2.fwdnet.net
type=peer
username=123456
secret=wibble


Incoming Settings
USER Context: iaxfwd (Pay attention here. Don't change it. or it won't work)
USER Details (Nothing needs to be changed here, this can be pasted straight in)


allow=ulaw
auth=rsa

context=from-pstn
disallow=all
inkeys=freeworlddialup
type=user


Register String: should be set to yournumber:yourpassword@iax2.fwdnet.net, using our examples above, it would be 123456:wibble@iax2.fwdnet.net


Click 'Outbound Routing' from the menu, and then click 'Add Route'

Name your route something like 'fwd'
The dial prefix is, usually, 393 — That's 'FWD' on your phones pad.
Dial Patterns: 393|X.
Trunk Sequence: IAX2/fwd

Click 'Submit Changes'
You may have to move the trunk further up the priority list.

From the asterisk command line type the following to see if the new connection is registered.
iax2 show registry

Assuming you've got your username and password correct, you should now be able to dial '393612', Which will read out the time to you. IF you are feeling exceptionally brave, call '393613', which is a useful little echo tester - it'll just bounce back to you everything you say to it. You can then try '393514' which is FWD's 'Coffee Lounge' - I've never actually successfully had a conversation with anyone there, however, or '39355555', which calls a random volunteer, so you can actually speak to a live person!


6.2 Free World Dialup OUT (FWD)

Contact: http://www.fwdout.net/web/
Service: Gateway to other providers
Protocol: IAX
Cost: Share and Share Alike

FWDout is The Service Formerly Known as <name withheld>
You must read the documentation carefully and be aware that a poorly configured Asterisk@Home box can be used by other people on the fwdOUT network to make long distance calls that you may end up paying dearly for.
Create an account on http://www.fwdout.net/bell-cgi/signup.cgi

Once again, you need to be in AMP, the Asterisk Management Portal. Click on Setup at the top of the page, and click on Trunks on the left. Click on Add IAX2 Trunk

Outbound Caller ID should left blank


Outgoing Settings
Trunk Name: fwdOUT (This is just a descriptive name, and is what appears on the left of the screen)
PEER Details: (Change '123456' and 'wibble' to be your fwdOUT Number and Password)


username=123456
type=peer
secret=wibble
host=iax.fwdOUT.net


Incoming Settings
USER Context: iaxfwdOUT (Pay attention here. Don't change it. or it won't work)
USER Details (Nothing needs to be changed here, this can be pasted straight in)


type=user
inkeys=freeworlddialup
disallow=all
context=from-pstn
auth=rsa
allow=ulaw
allow=gsm


Register String: should be set to yournumber:yourpassword@iax2.fwdOUT.net, using our examples above, it would be 123456:wibble@iax2.fwdnet.net


Click 'Outbound Routing' from the menu, and then click 'Add Route'

Name your route something like 'fwdOUT'
The suggested Dial prefix for fwdOUT is 394, although this is optional

Dial Patterns: 394|X.
Trunk Sequence: IAX2/fwdOUT

Click 'Submit Changes'
You may have to move the trunk further up the priority list.

From the asterisk command line type the following to see if the new connection is registered.
iax2 show registry

If you have another provider for long distance place the fwdOUT before your providers Trunk so that outbound calls are routed through fwdOUT

fwdOUT will allow you to make long distance phone calls using other people's asterisk boxes, while allowing other people to route calls through your asterisk box. The idea is that you do not pay for calls in your local area, so you can let people route calls through your server, and other people do the same for you.

6.3 VoicePulse

Contact: http://connect.voicepulse.com/
Service: PSTN termination
Protocol: IAX
Cost: pay

Once again, you need to be in AMP, the Asterisk Management Portal. Click on Setup at the top of the page, but this time click on Trunks on the left. Click on Add IAX2 Trunk

Dial Prefix 9 if you're not already
Leave Default Trunk switched off (or make this the default if you want all your calls to use it)
Outbound Caller ID should (but doesn't have to be) set to your VoicePulse Number.

Outgoing Settings
Trunk Name: voicepulse-out-01 (This is just a descriptive name)
PEER Details: (Change <your username> and <your password> to be your VoicePulse Number and Password)


host=gwiaxt01.voicepulse.com
secret=<your password>
type=peer
username=<your username>


Incoming Settings

USER Context: voicepulse-in-01 (Pay attention here. Don't change it. or it won't work)
USER Details (Nothing needs to be changed here, this can be pasted straight in)


auth=rsa
context=from-pstn
inkeys=voicepulse01
type=user


Register String: should be set to <your username>:<your password>@gwiax-in-01.voicepulse.com example (bob:abc123 @gwiax-in-01.voicepulse.com)

That's it. Click on Submit Changes, and then on the big red 'You have made changes' bar and you're done.
For a test make a call (try 1-800-555-1212)

6.4 Sixtel

Contact: http://www.iax.cc/
Service: PSTN termination
Protocol: IAX
Cost: pay

Merged into Vitelity Communications LLC. You may need to refer to Section 6.20

Iax.cc, also known as sixTel is a small VOIP termination provider that offers very low rates for inbound and outbound calls. With rates at low as 1.43 cents per minute and a good number of local and toll free numbers to choose from, sixTel is a popular choice for home and small business users.

Once again, you need to be in AMP, the Asterisk Management Portal. Click on Setup at the top of the page, but this time click on Trunks on the left. Click on Add IAX2 Trunk

Use the following example to get you up and going:

Outbound caller ID: "Your Name" <1XXXXXXX>
Maximum Channels: 4

Trunk Name: sixTel

Peer Details:
allow=all
context=ext-did
host=iax2.sixtel.net
secret=myPassword
type=friend
username=myUserName

User Context: <<blank>>
User Details: <<blank>>

Registration String: myusername:mypassword@iax2.sixtel.net

On the DID tab, create a new DID
DID: 949XXXXXXX <- use sixtel DID number

Set this new DID to use your "Normal Incoming Calls Setting".

Finally, on your "Outbound Routes" tab, you will need to add the sixTel trunk to one of your outbound trunks.

Once you save your settings, click on the red bar at the top of the screen, wait a few seconds, and you should be able to send and receive calls through Iax.cc/sixTel.
For a test make a call (try 1-800-555-1212)


6.5 VoipJet

Contact: http://www.voipjet.com/
Service: PSTN termination
Protocol: IAX
Cost: trial/pay

VoipJet allows you to create a free trial account to test your system. Once you have things working and have made a call you can buy extra credit. Once you have registered, for free, and received your free credit to do some basic testing, VoipJet provides instructions on how to configure Asterisk to use your new account.

In the Asterix@home management portal, Click on: Setup at the top, Trunks on the left, add IAX2 Trunk in the middle.
General Settings, leave this section blank.
Outgoing Dial Rules, leave this section blank.
Outgoing Settings, edit this section:
Trunk Name: 7374@voipjet (7374 is the userID given to you, you can get it from the VoipJet page which helps with the settings)
PEER Details:
auth=md5
context=ext-did
host=64.34.45.100
notransfer=yes
username=7374
secret=ecbf7ffca7631227 (this is the MD5 hash that is given to you, on the help page)
type=peer

Incoming settings
User Context: voipjet
User Details:
auth=md5
context=ext-did
host=64.34.45.100
notransfer=yes
username=7374
secret=ecbf7ffca7631227 (this is the MD5 hash that is given to you, on the help page)
type=peer

Registration, leave this section blank

Submit and click the red save bar at the top.


6.6 MyNetfone - AUSTRALIA

Contact: http://www.mynetfone.com.au//
Service: PSTN termination
Protocol: SIP
Cost: pay

MyNetfone provides free user to user calls and A$0.10 (~US$0.065) untimed calls to Australian landline numbers (+6113 +612, +613, +617 +619). Check their plans here: http://www.mynetfone.com.au/plans/

Outgoing settings are:
allow=alaw&ulaw&g729
authname=0911XXXX
canreinvite=no
disallow=all
dtmfmode=rfc2833
fromuser=0911XXXX
host=sip.myfone.com.au
insecure=very
pedantic=no
qualify=yes
secret=<your password>
type=peer
username=0911XXXX


They also provide indial numbers to people with an Australian address (yes that's still regulated in Australia).
Incoming settings are:
You don't need this to make outgoing calls only


Registration:
You don't need this to make outgoing calls only

Submit and click the red save bar at the top.


6.7 TelaSIP

Contact: http://www.telasip.com/
Service: PTSN Termination
Protocol: SIP
Cost:pay

TelaSIP trunk configuration:
Oubound caller ID: "J Smith" <5212314214> (substitute with your name and DID)
Maximum channels: 2
Dialing rules: (substituting your local area code for 404 below)
1|NXXNXXXXXX
NXXNXXXXXX
404+NXXXXXX
Outgoing Settings:
Trunk Name: telasip-gw
Peer details (using your own account name/password):
type=peer
host=gw4.telasip.com
qualify=yes
insecure=very
context=from-pstn
username=<username>
secret=<secret>
Registration: youraccountname:yourpassword@telasip-gw

Configure outbound routing
Add route: outgoing
Dial patterns:
1NXXNXXXXXX
NXXNXXXXXX
NXXXXXX
Trunk sequence: 0=SIP/telasip-gw



6.8 Exgn LLC

Contact: http://www.exgn.net
Service: PTSN Termination
Protocol: SIP, IAX
Cost: Pay

Merged into Vitelity Communications LLC. You may need to refer to Section 6.20

Exgn is a VOIP service provider that uses Asterisk themselves and is very Asterisk friendly. Supports both SIP and IAX protocols. Provides failover forwarding and voicemail in case your Asterisk server is offline. Their user portal is one of the best seen in this industry which allows instant activation for DIDs in hundreds of ratecenters across the country along with a pool of toll free numbers to choose from. As of March 5, 2006 they also offer e911 service for their DIDs in the USA with Canadian e911 service coming soon.

These instructions are for AMP (Asterisk Management Portal). Click on 'Setup' at the top of the page, then click on 'Trunks' on the left, Then click 'Add IAX2 Trunk'.

Enter the following information in the appropriate fields:

Outbound Caller ID: "Your-Name" <NPANXXNXXX>
Maximum Channels: 2

Trunk Name: exgn

Peer Details:
allow=all
context=ext-did
host=iax.exgn.net
secret=your-password
type=friend
username=your-username

User Context: <blank>
User Details: <blank>

Registration String: your-username:your-password@iax.exgn.net

This DID should be set to use your "Normal Incoming Calls Setting"

Lastly, on your "Outbound Routes" tab, you will need to add the exgn trunk to one of your outbound trunks.

Now you should be able to make and receive calls!

If you cannot, please feel free to email support@exgn.net or open a trouble ticket within our user portal/control panel.


6.9 Gizmo Project / SIPphone

Contributed by: Casey

Contact: http://www.gizmoproject.com
Service: DID, PTSN Termination, Gateway to other providers and universities
Protocol: SIP
Cost: Gateway is free. DID and PSTN termination are pay.

Its free to setup an account on the Gizmo Project. It's free to use as a gateway to other providers.

Gizmo In provides DID service (a standard telephone phone number others can call from traditional phones that rings in to your Asterisk@Home box) . And Gizmo Out offers PSTN termination to most countries.

The following setting work with both the Gizmo IN / DID service, and the Gizmo Out long-distance service. I try to use the minimal settings to get things working, then add the bells and whistles later. Here are the basic settings I've used to setup A@H with Gizmo Project/SIPphone with Asterisk@Home 2.8.

In the settings below 17470000000 should be replaced with your Gizmo or SIPphone number. And Password111 should be replaced with your Gizmo or SIPphone password.

In Asterisk@Home, add a new SIP trunk. Remove any pre-filled text from the fields, then only add:

Trunk name: proxy01.sipphone.com

Peer Details:
allow=ulaw&alaw&gsm&ilbc
canreinvite=no
context=from-pstn
disallow=all
dtmfmode=rfc2833
fromdomain=proxy01.sipphone.com
fromuser=17470000000
host=proxy01.sipphone.com
insecure=very
secret=Password111
type=peer
username=17470000000


Register String: 17470000000:Password111@proxy01.sipphone.com

Submit the changes, then click the red link at the top of the page to apply the changes. That's it. If you're interested in how/why I set the items I did, read on...

First, I had read that beginning with recent versions of Asterisk, the User Context/Details (Incoming Settings) have been depreciated. Instead, it had been combined with the Peer Details (Outgoing Settings). I've eliminated the User Context/Details completely from my configuration, and it continues to work.

I found that if I didn't specify insecure=very and allow=ulaw that DID would not work. Instead, the incoming caller would be greeted with Gizmo / SIPphone's "the person you are calling has not setup voice mail" message. FYI allow=ilbc will also work. The context=from-pstn makes the incoming DID calls get handled according to the settings in the AMP Incoming Calls tab.

If you don't have Gizmo Out (long-distance) minutes, you don't need the fromdomain=proxy01.sipphone.com and the fromuser=17470000000 settings. You'll be limited to the gateway features, and toll-free calls, without those settings.

FYI Gizmo Project / SIPphone officially announced support for Asterisk (which, coincidentally, would include Asterisk@Home) on May 23, 2006. The original instructions, above, were posted March 6, 2006. We're on it like Blue Bonnet!


6.10 Iristel

Contact: http://www.iristel.ca
Service: DID, PTSN Termination
Protocol: SIP
Cost: $15.95/month CAD for one DID with unlimited local termination.

When you first sign up with Iristel, select the "I will use my own SIP gateway" option. When your account is activated, they will e-mail you a PDF document with sample Asterisk configuration. Use that document for reference. It has on it your assigned DID, you user ID number, and your password to access the service. To setup Iristel's service using the AMP GUI, follow these instructions:

  1. Click on Setup in the menu at the top of the page.
  2. Click on Trunks in the menu on the left of the page.
  3. Click on the Add SIP Trunk link.
  4. Under Outbound Caller ID, enter the following: "You Name" <11231234567> (replacing Your Name with your desired caller ID name, and 11231234567 with your assigned DID. Make sure this DID is in international dialing format (re: Include the leading one!). Also ensure that you do enclose your name in quotes, as this is the format Asterisk is expecting.
  5. Under Dialing Rules, enter the following: 1+NXXNXXXXXX This will add a leading one to locally dialed outgoing calls on this trunk. Locally dialed calls must be dialed in international dialing format or Iristel's SIP proxy will reject the call.
  6. Under Trunk Name enter: irisbax.iristel.net
  7. Under Peer Details enter:
callerid=1416xxxxxxx
dtmfmode=rfc2833
host=irisbax.iristel.net
insecure=very
secret=1111
type=peer
username=40932998

Leave the Incoming settings boxes blank. Under Register String, enter: 11231234567:<password>:<userID>@irisbax.iristel.net/11231234567 (replacing 11231234567 with your assigned DID, <password> with your password (excluding the angle brackets) and <userID> with your assigned user ID number (without the angle brackets) This string is provided to you in the setup PDF, you may copy and paste it here).

Your Iristel trunk is now ready to send and receive calls. Simply setup an outbound route to match your local area code, or all long distance calls if you wish. Just make sure that any call being sent to the Iristel SIP proxy is in international dialing format.


6.11 Voxee

Contributed by: Casey

Contact: http://www.voxee.com
Service: PTSN Termination
Protocol: IAX or SIP
Cost: Pay

Voxee provides outbound call termination to the PSTN. At the time of this handbook entry, Voxee's rate for the U.S. was 1.1-cents per minute, with 6-second (1/10-minute) billing. Here is the basic IAX configuration to get you started:

Add a new IAX trunk to Asterisk. Delete any pre-filled Peer Details information, and delete any pre-filled User Details information. Then add only the following (replace Username111 with the username Voxee assigned to you, and Password111 with your Voxee password):

Trunk Name: Voxee

Peer Details:
allow=alaw&ulaw&gsm&ilbc
canreinvite=no
disallow=all
host=66.246.246.52
secret=Password111
type=peer
username=Username111

The config above includes all the non-royalty codecs supported by Voxee. They also support g729. Feel free to remove any codec from the allow= line that you don't want to use (or add g729 if you have a license for it).

6.12 Gafachi

Contact: http://gafachi.com/
Service: PSTN termination
Protocol: SIP
Cost: pay

Get your GAFACHI_USER+GAFACHI_SECRET from the gafachi page, they are different from your login!

Once again, you need to be in AMP, the Asterisk Management Portal. Click on Setup up the top, but this time click on Trunks on the left. Click on Add SIP Trunk. Empty out the values.

Outbound Caller ID should be set without the country code. E.g 212XXXXXXX instead of 1212XXXXXXX


Outgoing Settings
Trunk Name: gafachi
PEER Details:

allow=ulaw
canreinvite=no
context=from-pstn
dtmfmode=rfc2833
fromuser=GAFACHI_USER
host=GAFACHI_USER.sip.gafachi.com
secret=GAFACHI_SECRET
type=friend
user= GAFACHI_USER
username=GAFACHI_USER


Incoming Settings

Leave blank (Took time before I got that far)

Register String: GAFACHI_USER:GAFACHI_SECRET@GAFACHI_USER.sip.gafachi.com


Note for Gafachi inbound 800 users.
I was fighting the issue of incoming calls being rejected for days until I happened to set the Incoming User Context to a blank field.
This wiped out the incoming settings that I had been tweaking. It also fixed my incoming call problem!
I am happily running with only the trunk name and registration string populated.



6.13 Acanac

Contact: http://www.acanac.ca/ or http://www.acanac.com/
Service: DID, PSTN termination
Protocol: SIP
Cost: pay
Asterisk@Home Ver. Tested: 2.7

You will need your username (your phone number) and your password from Acanac

1. Click on Setup in the menu at the top of the page.
2. Click on Trunks in the menu on the left of the page.
3. Click on the Add SIP Trunk link.
4. Under Outbound Caller ID, enter the following: "You Name" <1231234567>
(replacing Your Name with your desired caller ID name, and 1231234567 with your assigned DID. Ensure that you do enclose your name in quotes, as this is the format Asterisk is expecting.)
5. Under Trunk Name enter: acanac
6. Under Peer Details enter:

NOTE: The IP provided here is for East server 1, there are many servers so choose the correct IP.

callerid=<your acanac phone number>
dtmfmode=inband
host=66.49.255.38
insecure=very
secret=<your acanac password>
type=peer
username=<your acanac phone number>


1. Under User Context enter: <Your acanac phone number>
2. Under User Details enter:

callerid=<your acanac phone number>
context=from-pstn
host=66.49.255.38
insecure=very
secret=<your acanac password>
type=user
username=<your acanac phone number>


1. Under Register String, enter: <your acanac phone number>:<your acanacpassword>@66.49.255.38/<your acanac phone number>
2. Click "Submit Settings


At this time you can receive calls and send calls; however, you still need to set your Inbound Routing with your acanac phone number as your "DID Number"


6.14 Stanaphone

Contact: http://www.stanaphone.com/
Service: DID, PSTN termination
Protocol: SIP
Cost: free(inbound), pay(outbound)

You will need your username (your phone number) and your password from Stanaphone. Use the information provided in the SIP Settings section of the Account Information page.

1. Click on Setup in the menu at the top of the page.
2. Click on Trunks in the menu on the left of the page.
3. Click on the Add SIP Trunk link.
4. Under Outbound Caller ID, enter the following: "Your Name" <1231234567>
(replacing Your Name with your desired caller ID name, and 1231234567 with your assigned DID. Ensure that you do enclose your name in quotes, as this is the format Asterisk is expecting)
5. Under Maximum Channels enter: 2
6. Under Dial Rules enter: 1+NXXNXXXXXX
(or whatever other dial rules would be appropriate for your locale)
7. Under Trunk Name enter: stanaphone-out (or whatever you want to call this trunk)
8. Under Peer Details enter:
canreinvite=no
dtmfmode=rfc2833
fromdomain=sip.stanaphone.com
fromuser=<your stanaphone username — NOTE: Not your account login>
host=sip.stanaphone.com
insecure=very
nat=yes (if you are behind a router which you probably are)
qualify=yes
secret=<your stanaphone password>
type=friend
username=<your stanaphone username>

9. Under User Context enter: <your stanaphone username>
10. Under User Details enter:
auth=md5,plaintext
canreinvite=no
context=from-pstn
fromuser=<your stanaphone username>
host=sip.stanaphone.com
insecure=very
nat=yes
qualify=yes
secret=<your stanaphone password>
type=peer

11. Under Register String, enter:
<your stana username>:<your stana password>@sip.stanaphone.com/<your stana username>
example: 08123456:randomletterpasswd@sip.stanaphone.com/08123456
12. Click "Submit Settings

At this time you can receive calls and send calls; however, you still need to set your Inbound Routing with your stanaphone phone number as your "DID Number". You can then route this as any other route (most likely to the "Use Incoming Calls Setting")

6.15 VBuzzer

Contact: http://www.vbuzzer.com/
Service: DID, PSTN termination
Protocol: SIP
Cost: free(inbound), free(local outbound), pay(outbound)
Asterisk@Home Ver. Tested: 2.7

You will need your username, your phone number, and your password from VBuzzer. Note that I had to install thier software and connect for the first time in order to activate the DID. After such time, the software was unnecessary.

1. Click on Setup in the menu at the top of the page.
2. Click on Trunks in the menu on the left of the page.
3. Click on the Add SIP Trunk link.
4. Leave "Outbound Caller ID" blank
5. Under Trunk Name enter: vbuzzer
6. Under Peer Details enter:

allow=ulaw&gsm
authname=<your username>
canreinvite=no
context=from-pstn
disallow=all
dtmf=rfc2833
dtmfmode=rfc2833
fromdomain=vbuzzer.com
fromuser=<your username>
hidecallerid=yes
host=vbuzzer.com
insecure=very
nat=no
port=80
qualify=yes
secret=<your password>
type=peer
user=<your username>
useragent=VBuzzer/1.1.0.9
username=<your username>


1. Under User Context enter: <Your vbuzzer phone number>
2. Under User Details enter:

authname=<your password>
canreinvite=no
context=from-pstn
dtmfmode=inband
fromdomain=vbuzzer.com
fromuser=<your password>
host=vbuzzer.com
insecure=very
nat=yes
port=80
secret=<your password>
type=user
user=<your password>
useragent=vbuzzer/1.1.1.0
username=<your password>



1. Under Register String, enter: <your username>:<your password>:<your username>@vbuzzer/<your phone number>
2. Click "Submit Settings

NOTE: For the registration Asterisk has a bug where if you put vbuzzer.com:80, it will continue to try and register on 5060. you must put the context in the register string.. (in this case vbuzzer)
NOTE 2: All your phone number entries should have the leading 1 on the number
NOTE 3: You will need to work your dialplans a bit as to dial out you must have the leading 1 or 011.

At this time you can receive calls and send calls; however, you still need to set your Inbound Routing with your acanac phone number as your "DID Number"


6.16 Broadvoice

Contact: http://www.broadvoice.com/
Service: DID, PSTN termination
Protocol: SIP
Cost: pay
Asterisk@Home Ver. Tested: 2.7

You will need to determine which of the Broadvoice SIP servers is closest to your location and then set it in your hosts file as sip.broadvoice.com.

1. Click on Setup in the menu at the top of the page.
2. Click on Trunks in the menu on the left of the page.
3. Click on the Add SIP Trunk link.
4. Leave "outbound caller ID" blank
5. Under Trunk Name enter: sip.broadvoice.com
6. Under Peer Details enter:

authname=<your phone number>
canreinvite=no
context=from-pstn
dtmf=inband
dtmfmode=inband
fromdomain=sip.broadvoice.com
fromuser=<your phone number>
host=sip.broadvoice.com
insecure=very
qualify=yes
secret=<your password>
type=peer
user=phone
username=<your phone number>


1. Under User Context enter: <your phone number>
2. Under User Details enter:

authname=<your phone number>
canreinvite=no
context=from-pstn
dtmf=inband
dtmfmode=inband
fromdomain=sip.broadvoice.com
fromuser=<your phone number>
host=sip.broadvoice.com
insecure=very
secret=<your password>
type=user
user=phone
username=<your phone number>



1. Under Register String, enter: <your phone number>@sip.broadvoice.com:<your password>:<your phone number>@sip.broadvoice.com
2. Click "Submit Settings

At this time you can receive calls and send calls; however, you still need to set your Inbound Routing with your broadvoice phone number as your "DID Number"

6.17 WENGO

Contact: http://www.wengo.fr, http://www.wengo.com
Service: DID, PSTN termination
Protocol: SIP
Cost: free(wengo only), pay(DID, outbound)
Asterisk@Home Ver. Tested: 2.5, 2.8

1. Click on Setup in the menu at the top of the page.
2. Click on Trunks in the menu on the left of the page.
3. Click on the Add SIP Trunk link.
4. Leave "outbound caller ID" blank
5. Under Trunk Name enter: wengo
6. Under Peer Details enter:

allow=g729&alaw&ulaw&ilbc
canreinvite=no
context=from-pstn
disallow=all
fromdomain=voip.wengo.fr
fromuser=<your username>
host=voip.wengo.fr
insecure=very
nat=yes
promiscredir=yes
qualify=yes
secret=<your SIP password>
type=peer
username=<your username>


To retrieve your SIP password:
- on wengo.com: connect you to your account center my wengo, select my account settings, then ATABox
- on wengo.fr: connect you using espace perso, mon profil, mes coordonnées, then paramétres de votre compte, then wenbox

7. Under User Context enter: <leave empty>
8. Under User Details enter: <leave empty>

9. Under Register String, enter: <your username>:<your SIP password>@voip.wengo.fr/<you wengo number>
10. Click "Submit Settings"

At this time you can receive calls and send calls; however, you still need to set your Inbound Routing with your wengo phone number as your "DID Number"

6.18 QuantumVoice

Contact: http://www.QuantumVoice.com
Service: DID, PSTN termination
Protocol: SIP / Plans for IAX
Cost: Various Residental/Business Plans
Asterisk@Home Ver. Tested: 2.8

1. Click on Setup in the menu at the top of the page.
2. Click on Trunks in the menu on the left of the page.
3. Click on the Add SIP Trunk link.
4. Leave "outbound caller ID" blank
5. Under Trunk Name enter: QuantumVoice
6. Under Peer Details enter:

allow=g729&gsm&ulaw&alaw
canreinvite=yes
context=from-pstn
disallow=all
host=sipdr.quantumvoice-sip.com
insecure=very
nat=yes
secret=Password
type=friend
username=Normally telephone number

7. Under User Context enter: <leave empty>
8. Under User Details enter: <leave empty>

9. Under Register String, enter: <your username>:<your SIP password>@sipdr.quantumvoice-sip.com/<you QuantumVoice number>
10. Click "Submit Settings"

At this time you can receive calls and send calls; however, you still need to set your Inbound Routing with your QuantumVoice phone number as your "DID Number"

6.19 TalkLITE.NET

Contact: http://www.talklite.net
Service: PSTN termination
Protocol: SIP / IAX
Cost: Starts at $0.01 for US48
Asterisk@Home Ver. Tested: 2.8
Trixbox v.1.0

1. Click on Setup in the menu at the top of the page.
2. Click on Trunks in the menu on the left of the page.
3. Click on the Add SIP Trunk link.
4. Leave "outbound caller ID" blank
5. Under Trunk Name enter: talklite
6. Under Dial Rules:
1360+NXXXXXX <----- This will allow you to dial 7 Digit numbers for the 360 areacode.
Change the areacode to match your own areacode to dial 7
Digits from your home phone.
1+NXXNXXXXXX <----- This will allow you to call just AreaCode+Number

7. Under Peer Details enter:

allow=ulaw
canreinvite=yes
disallow=all
host=sip,talklite.net
nat=yes
qualify=yes
secret=Password From E-Mail
type=friend
username=Username From E-Mail

8. Under User Context enter: <leave empty>
9. Under User Details enter: <leave empty>
10. Under Register String, enter: <your username>:<your SIP password>@sip.talklite.net
11. Click "Submit Settings"

Under Outbound Routes
1. Route Name: talklite
2. Dial Patterns
X.

3.Trunk Sequence SIP/talklite or IAX/talklite
4. Click "Submit Settings"


6.20 Vitelity Communications LLC (merger between Exgn LLC and Sixtel)

Contact: http://www.vitelity.net
Service: PTSN Termination
Protocol: SIP, IAX
Cost: Pay

Vitelity Communications announced its official launch on 07/18/2006, combining two of the industries leading VoIP providers, Exgn LLC (www.exgn.net) & Sixtel Communications (www.iax.cc). Vitelity began migrating EXGN customers shortly after this mid-July announcement.
Created by: GinelLipan, Last modification: Sat 31 of Dec, 2011 (02:38 UTC) by admin
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