How to configure Asterisk for VoicePulse Connect

Who is VoicePulse?

VoicePulse is a broadband phone company that offers connectivity via SIP and IAX to the PSTN. The most appealing thing about VoicePulse is how easy it is to connect your Asterisk server to VoicePulse. It allows you to truly start playing with Asterisk without any special hardware in your computer. They offer phone numbers in several hundred cities in several dozen US states.

Purchasing an account

The first step is to purchase an account with VoicePulse's connect service. Go to their website at http://connect.voicepulse.com and click the signup link located on the toolbar. This service costs 2.40 cents a minute for outbound calls to the US48 and $11/month for an inbound phone number. So I recommend that you purchase a $10 account. That will provide you enough money to purchase a phone number and do some outbound calling. It's the smallest amount of their service that you can purchase. It will allow you to play around with the service and see if you like it. If you like the service, then charge up your account with more money.

Logging back in to your VoicePulse account

This section is to save you from the stupid mistake that so many of us have made. There are TWO seperate login pages for VoicePulse. The one on the main menu of http://www.voicepulse.com is NOT for the connect service. You must return to http://connect.voicepulse.com to log back in and manage your account.

Configuring outbound dialing

After signing up with the service you'll receive an email with some instructions. The email will contain a couple of important pieces of information that you'll want to place in your Asterisk configuration.

Example (data is fake, don't try to actually use it!):
 Host: gw5.voicepulse.com
 Login: Abc1dEF2GH
 Password: zYX9VUt8sr
 Context: VPWS

First thing to do is edit the /etc/asterisk/extensions.conf and place the line from the email that below the [default] line:
 exten => _1NXXNXXXXXX,1,Dial,IAX2/Abc1dEF2GH@voicepulse/${EXTEN}

This line instructs Asterisk to route outbound calls to VoicePulse. The next thing to do is to edit the /etc/asterisk/iax.conf and add the information about how to connect to VoicePulse. At the bottom of the file paste the information from the email. It should look something like this:

  [voicepulse] 
 context = foo
 secret=zYX9VUt8sr
 auth=md5
 type=friend
 host=gw5.voicepulse.com

At this point you should be able to start Asterisk up, connect with your SIP or IAX phone, and place an outbound call to your favorite land line friend. DIAX is a good IAX based phone and X-Lite is a good SIP based phone.

To provide your own CallerID number on outbound calls, add the following prior to the Dial line. (Renumber the priority as appropriate)

 exten => _1NXXNXXXXXX,1,SetCallerID("Me" <2165551212>)

The name portion ("Me") will not apply when contacting PSTN lines, but the phone number will be matched to the number database for the local telephone company. Remember that providing a 313-xxx-xxxx identification to a phone in the 216 area code will likely just result in "Michigan Call" instead of a person's name. Providing a number that is local to the destination will result in a proper name lookup.

Configuring inbound dialing (optional)

If you're interested in receiving inbound calls from VoicePulse on your Asterisk box, then complete the following section. The first thing you will need to do is purchase a phone number. Log in to the VoicePulse connect interface at http://connect.voicepulse.com. Click on the phone numbers tab. Select the area code and prefix that you would like your phone number. Pick your area code so that your friends can call you locally. Or you could purchase a phone number in another city where your friends and family live, etc. Remember that there is a charge of like $7.95/month for this phone number. So you'll have to have at least that much credit on your account.

Important: The inbound lines are billed the first of every month, and VoicePulse does not do prorated billing for inbound numbers. If you sign up for a number on the 30th of the month, you will get billed $7.95 and then another $7.95 a day or two later.

After you've purchased the phone number through the web interface, you will receive an email with the information on how to register your Asterisk to VoicePulse so that it receives the inbound calls. The email will contain a line that says something like:

 register => Abc1dEF2GH:zYX9VUt8sr@gw5.voicepulse.com

Remember to replace the example login and password with the one that you received in your introductory email. You will want to place this register statement under the general section of the /etc/asterisk/iax.conf. I typically place it just under the port statement. This will cause Asterisk to register with VoicePulse when it is started up.

Don't go placing that inbound call just yet! You'll need to define an extension for your inbound number. Let's say the phone number that you're assigned is 212-555-1212 just for the sake of this example. Then you will want to edit the /etc/asterisk/extensions.conf file and add a section like this:

 [voicepulse]
 exten => 2125551212,1,Goto(mycompany,s,1)

This line instructs Asterisk to route the call to the start of the mycompany context. You'll want to edit mycompany to be whatever the context name is for your Asterisk install. At this point you should be able to place an inbound call, enter your extension number from the menu, and talk away.

Note that if you do not have a dedicated context set up to handle the inbound Voicepulse number, as shown above, but instead allow the inbound calls to go to a default context, such as:

   incoming
   exten => s,1,answer...

for some reason the "s" extension will not work with Voicepulse. To use a default handler like this you must use "_.", such as:

   incoming
   exten => _.,1,answer...

Choosing the codec to use with VoicePulse

If you would like a higher quality call from VoicePulse than GSM provides, then choose the codec you want with the following lines in the /etc/asterisk/iax.conf:

 disallow=all
 allow=ulaw
 allow=alaw

This example removed everything but G.711 ulaw and G.711 alaw. VoicePulse supports the following codecs:

• GSM
• G.711ulaw
• G.711alaw
• ADPCM
• ILBC
• SPEEX

For a list of the abbreviations used in the iax.conf file for the codecs click on Asterisk config iax.conf.

CallerID Name Lookup

Voicepulse does perform a database dip in order to present CallerID Name information. However, this is only done when you register with the connect01.voicepulse.com (and 02 and 03) SIP gateways. The SFO and NYC gateways do not do this.

Conclusion

If you follow this documents instructions on how to configure Asterisk and VoicePulse and think something was left out, then please signup for an account on this WIKI and edit the document. Your notes and tips are very important.

The example config did not work for me

I think partly because VoicePulse has changed their hardware configuration.  
This is an example of the configuration I am using that is working as of 8/2/2004.

Voicepulse Connect Flexrate for Local and Longdistance

Modified version of Voicepulse Flexrate to accommodate 7 and 10 digit dialing.

globals
VOICEPULSE_API_KEY=your voicepulse api key
VOICEPULSE_API_PREFIX=VOICEPULSE_

VOICEPULSE_GATEWAY_OUT_A=voicepulse02 ; your iax/sip.conf entries
VOICEPULSE_GATEWAY_OUT_B=voicepulse01

outbound context

       ; Local 7 Digit Numbers
       exten => _NXXXXXX,1,Set(CALLERID(all)="Callerid Name" <calleridnumber>)
       exten => _NXXXXXX,2,Set(OTHER_PROVIDERS_FLAT_RATE=0.02); 
       exten => _NXXXXXX,3,Macro(voicepulseflexrate,${VOICEPULSE_API_KEY},1NXX${EXTEN}) ;put  your areacode before ${EXTEN}
       exten => _NXXXXXX,4,Verbose(The rate is ${VOICEPULSE_FLEXRATE})
       exten => _NXXXXXX,5,GotoIf($${VOICEPULSE_FLEXRATE} > ${OTHER_PROVIDERS_FLAT_RATE}?outbound|${EXTEN}|700
       exten => _NXXXXXX,6,Dial(IAX2/${VOICEPULSE_GATEWAY_OUT_A}/1NXX${EXTEN});desired area code prefix
       exten => _NXXXXXX,7,GotoIf($${DIALSTATUS}=CHANUNAVAIL?${EXTEN}|500)
       exten => _NXXXXXX,500,Dial(IAX2/${VOICEPULSE_GATEWAY_OUT_B}/1NXX${EXTEN})

       exten => _NXXXXXX,700,Dial(IAX2/username@voip-providerA/1NXX${EXTEN}) ; if they're cheaper
       exten => _NXXXXXX,701,Dial(IAX2/${VOICEPULSE_GATEWAY_OUT_A}/1NXX${EXTEN}); failover voicepulse 1
       exten => _NXXXXXX,702,Dial(IAX2/${VOICEPULSE_GATEWAY_OUT_B}/1NXX${EXTEN}); failover voicepulse 2
       exten => _NXXXXXX,703,Dial(IAX2/username@voip-providerD/1NXX${EXTEN}) ; another failover 



       ; US Numbers
       exten => _1NXXNXXXXXX,1,Set(CALLERID(all)="Your Name" <your number>)
       exten => _1NXXNXXXXXX,2,Set(OTHER_PROVIDERS_FLAT_RATE=0.011)
       exten => _1NXXNXXXXXX,3,Macro(voicepulseflexrate,${VOICEPULSE_API_KEY},${EXTEN})
       exten => _1NXXNXXXXXX,4,Verbose(The rate is ${VOICEPULSE_FLEXRATE})
       exten => _1NXXNXXXXXX,5,GotoIf($${VOICEPULSE_FLEXRATE} > ${OTHER_PROVIDERS_FLAT_RATE}?outbound|${EXTEN}|800
       exten => _1NXXNXXXXXX,6,Dial(IAX2/${VOICEPULSE_GATEWAY_OUT_A}/${EXTEN})
       exten => _1NXXNXXXXXX,7,GotoIf($${DIALSTATUS}=CHANUNAVAIL?${EXTEN}|600)
       exten => _1NXXNXXXXXX,600,Dial(IAX2/${VOICEPULSE_GATEWAY_OUT_B}/${EXTEN})

       exten => _1NXXNXXXXXX,800,Dial(IAX2/username@voip-providerA/${EXTEN})
       exten => _1NXXNXXXXXX,801,Dial(IAX2/${VOICEPULSE_GATEWAY_OUT_A}/${EXTEN})
       exten => _1NXXNXXXXXX,802,Dial(IAX2/${VOICEPULSE_GATEWAY_OUT_B}/${EXTEN})
       exten => _1NXXNXXXXXX,803,Dial(IAX2/username@voip-providerD/${EXTEN})

See also

• Asterisk IAX channels
• IAX
• Asterisk config iax.conf
• Asterisk config extensions.conf