Introducing ISN: Free SIP Dialing From Any Asterisk Phone
Posted by admin | Posted in asterisk scripts | Posted on 01-07-2009
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Wouldn’t it be nice to pick up any telephone on your Asterisk system and place free SIP calls to anywhere in the world by dialing joe@sip.asterisk.com or any SIP URI? The problem, of course, is that most phones don’t include alphanumeric keyboards much less the @ symbol. Well, not to worry. A group of Asterisk gurus headed up by John Todd came up with a clever plan using DNS that lets you dial any SIP URI using the 10 numeric keys plus the asterisk key on any standard telephone keypad. Today, we’ll show you how to set up your Asterisk system to support ISN’s (aka ITAD Subscriber Numbers).
Overview. In laymen’s terms, the trick to ISN dialing is that we pass a number such as 1234*1061 to a DNS server that knows how to translate the numeric sequence into a SIP URI that looks like this: 1234@sip.pbxinaflash.com. In short, it takes the number after the asterisk and resolves it to a fully-qualified domain name which is preconfigured at freenum.org. And the result is inter-domain numeric SIP addressing using ordinary telephone instruments. For our recommended setup, you’ll actually dial ISN numbers like this: **1234*1061. The leading asterisks will tell FreePBX to treat this as an ISN dial string.1
Prerequisites. We’re assuming that you already have one of the FreePBX-enhanced Asterisk aggregations in place such as PBX in a Flash. If not, start there and then run the Orgasmatron Installer which provides all of the SIP URI functionality you’ll need for this project. If you’re not using PBX in a Flash, then review our tutorial on SIP URI’s which will walk you through getting this functionality set up on your FreePBX-enhanced Asterisk server.
Adjusting Your Phones to Support ISN Dialing. We’ll be using a somewhat different dial plan to make ISN calls so you’ll probably have to adjust the default dialplan on your actual phones or ATA to get this to work. If you can place ISN calls with a softphone but you get a fast busy when you dial the same number on your hardware-based phones, then it’s a dialplan problem. For Aastra phones, you can access the Aastra dialplan settings with a web browser. Just go to the IP address of the phone and login with admin:22222. Click on thePreferences option and you should see Local Dial Plan at the top of the page with an entry that looks like this: x+#|xx+*. Just change it to: x+#|xx+*|’*'xx+* and click the Save Settings button. No reboot of the phone is required. Notice that we’ve enclosed the asterisk in single quotes in the third option. That’s the trick to getting Aastra phones to recognize * as part of an actual dial string. If you’re using other phones, consult your user’s guide for tips on modifying your dialplan to accommodate an asterisk as the first character in the dial string.
Enabling Outbound ISN Dialing. There are a number of ways to get ISN outbound dialing to work with Asterisk. We’re going to show you a couple of methods. You can either set up a trunk and outbound route to handle the calls, or you can add an extension to your system which actual prompts for the ISN number when you dial that extension. There are also two ways to look up ISN numbers at freenum.org. The preferred method is using DNS queries with the new Asterisk ENUMLOOKUP function. An alternative method (which is especially useful with older versions of Asterisk that do not support ENUMLOOKUP) is to use FreeNUM’s external public resolver to map ISN dial strings to SIP URIs. WithPBX in a Flash and Asterisk 1.4.21.2 or later, both methods work.
Implementing the Trunk Method for ISN Dialing. With this option, you’ll be able to pick up any (properly configured) phone on your Asterisk system and dial **1234*1061 to complete a free ISN SIP call. To set this up, we’ll add a new trunk and outbound route in FreePBX. Then we’ll insert a dialplan script in extensions_custom.conf to finish up. Once you reload your Asterisk dialplan, you’ll be good to go.
Open FreePBX in a web browser, and choose Admin, Setup, Trunks, Add Trunk, Add Custom Trunk. Leave the General Settings blank for now. In the Dial Rules, insert X.*X. (be sure to include trailing period!) and, for the Custom Dial String, insert: local/$OUTNUM$@freenum. Click the Submit button to save your settings and reload the dialplan when prompted. Now add an Outbound Route calledOutFreeNUM. For the Dial Pattern, use **|X.*X. with the trailing period again. For the Trunk Sequence, choose local/$OUTNUM$@freenum. Save your entries and reload the dialplan once more. Finally, log into your server as root and edit extensions_custom.conf in /etc/asterisk. At the bottom of the file, insert the following code:
[freenum]
exten => _X.,1,Set(TIMEOUT(absolute)=10800)
exten => _X.,2,NoOp(Number to Call: ${EXTEN})
exten => _X.,3,Set(isnresult=${ENUMLOOKUP(${EXTEN},sip,,1,freenum.org)})
exten => _X.,4,GotoIf($["${isnresult}"=""]?6:5)
exten => _X.,5,Dial(SIP/${isnresult},40,r)
exten => _X.,6,Background(ss-noservice)
exten => _X.,7,Congestion
exten => _X.,8,Hangup
exten => h,1,Hangup
exten => i,1,Hangup
exten => T,1,Hangup
Make sure you eliminate the line-wrap on line 3 above. Then save the file and reload your dialplan: asterisk -rx “dialplan reload”. Now place a test call by dialing: **1234*1061. If the call doesn’t connect to Nerd Vittles’ demo site, check the Asterisk CLI and fix any reported errors.
Implementing the Extension Method for ISN Dialing. With this option, you’ll be able to pick up any phone on your Asterisk system and dial FREE (3733) to place an ISN call. You’ll be prompted to enter the number using the following format: 1234*1061. Note that there are no leading asterisks with this method. Instead of using ENUMLOOKUP to find the ISN number, we’ll use FreeNUM’s external public resolver to do the ISN translation into a SIP URI.
Log into your Asterisk server as root and edit extensions_custom.conf in /etc/asterisk. At the bottom of the file, insert the following context:
[custom-freenum]
exten => s,1,Answer
exten => s,2,Wait(2)
exten => s,3,Background(pls-entr-num-uwish2-call)
exten => s,4,Read(NUM2CALL,beep,30)
exten => s,5,GotoIf($["foo${NUM2CALL}" = "foo"]?10)
exten => s,6,Set(TIMEOUT(absolute)=10800)
exten => s,7,Background(pls-hold-while-try)
exten => s,8,Dial(SIP/${NUM2CALL}@public.freenum.org,30,m)
exten => s,9,Congestion
exten => s,10,Hangup
exten => h,1,Hangup
exten => i,1,Hangup
exten => T,1,Hangup
Now move to the top of the file and insert the following line in the [from-internal-custom] context:
exten => 3733,1,Goto(custom-freenum,s,1)
Save the changes you’ve made to the file and then edit (or create, if necessary) sip_custom.conf and insert the following line:
promiscredir=yes
Save the file and then restart Asterisk: amportal restart. Now place a test call by dialing 3733. When prompted for the ISN number, enter 1234*1061 and press # to avoid the timeout delay. Be aware that on non-FreePBX systems, this code would go in sip.conf; however, that file gets overwritten with any FreePBX reload. Hence the reason that we’ve placed the code in sip_custom.conf.
Creating a SIP URI for Your Asterisk Server. Before you can receive any inbound calls with ISN dialing, you’ll need at least one SIP URI for your Asterisk server. The format of a SIP URI is much like an email address: somename@yourdomain.dyndns.org or somenumber@yourdomain.dyndns.org. Step 1 is to register a fully-qualified domain name (FQDN) for your Asterisk server. Step 2 is to actually set up the SIP URI’s on your server.
If you already have a registered domain, then we recommend you create a sip subdomain: sip.yourname.org. Then point that subdomain to the IP address of your Asterisk server. If your Asterisk server has a dynamic IP address, then register a subdomain with a service such as dyndns.org and point that domain at your Asterisk server. We’ve previously covered how to install software on your Asterisk server to make sure your FQDN always resolves to the correct dynamic IP address. Here’s the link for DNS-O-Matic.
Once you have FQDN covered, you’re ready to set up a SIP URI. With Orgasmastron builds of PBX in a Flash, the work already has been done for you. You should already have a SIP URI of mothership@yourFQDN. For everyone else, the drill involves moving a copy of the [from-sip-external] context intoextensions_override_freepbx.conf in /etc/asterisk so that it can be edited without risking an overwrite from FreePBX. To find out the location of the [from-sip-external] context, issue the following commands while logged into your server as root:
cd /etc/asterisk
grep from-sip-external *
The result will look something like this:
extensions.conf:[from-sip-external]
extensions_override_freepbx.conf:[from-sip-external]
sip_general_additional.conf:context=from-sip-external
If the middle line is there, the context already has been copied over. Otherwise, list out the file showing [from-sip-external] which varies depending upon your version of FreePBX: cat extensions.conf. Now cut-and-paste the entire [from-sip-external] context into extensions_override_freepbx.conf. Then edit the override file and add an entry for each SIP URI you wish to create. The entries should be inserted just below the exten => s,1… line. Here are some samples:
exten => 16781234567,1,Goto(from-trunk,${DID},1)
This entry would let you control the routing of 16781234567 by creating a new incoming route in FreePBX with a DID entry of 16781234567. Then you can point the SIP URI to any FreePBX resource, e.g. an extension, ring group, IVR.
exten => e164,1,Goto(from-trunk,e164,1)
This entry would route e164@yourFQDN to the Inbound Route created for a DID number entry of e164.
exten => 18431234567,1,Goto(custom-windyhouse,s,1)
This entry would route incoming calls to 18431234567@yourFQDN to s,1 in a custom context called [custom-windyhouse] in extensions_custom.conf.
exten => 17065439876,1,Dial(SIP/17066313456@sip.otherdomain.com)
This entry would route incoming calls to 17065439876@yourFQDN to another SIP URI.
exten => 12021234567,1,Dial(local/12029876543@from-internal)
This entry would route incoming calls to 12021234567@yourFQDN to a cellphone at 12029876543 using your Asterisk dialplan to choose an appropriate trunk for the call.
exten => 18883331212,1,Dial(SIP/skype_joe@proxy01.sipphone.com)
This entry would route incoming calls to 18883331212@yourFQDN to a Skype user named joe using the free Gizmo5 gateway.
Once you’ve made all desired SIP URI entries, save the override file and reload your Asterisk dialplan.
Using the PBX in a Flash ITAD Number. So you’re probably asking, “What’s in this for me?” Well, a couple of things actually. First, if you’re a PBX in a Flash user, we want you to join our free calling network. We already have reserved the 1061 ITAD number for our group. Just cut-and-paste the form below, fill in the blanks, and email it to us. We’ll set up an ISN number for your server (one per customer, please) so that others can contact you without spending a dime. The other option is to obtain your own ITAD number for your organization and set it up on your own server. We’ll get to that in a minute.
If you want to join our club (and we really don’t mind if you’re not using PBX in a Flash), then cut-and-paste the form below into your email and fill it out. And here’s the email link. Once we receive your request, we’ll set up an ISN number for you that matches your existing phone number. So, if your phone number is 16781234567, your new ISN number will be 16781234567*1061. Please include your international codes with your phone number. Before we activate your ISN number, we’ll place a test call to your SIP URI to verify it’s working. Please be sure it is before applying.
Name:
Mailing Address:
Phone Number:
SIP URI for Your Server: _____________@_____________________________
ISN Number (leave blank):
Publish Entry in Directory? Yes or No (choose one)
Obtaining Your Own ITAD Number. We know there are lots of you that prefer to do things yourself. And that’s perfectly fine. We’re going to quickly show you how. But, if you want to be included in the PBX in a Flash directory, please send us the form above with your own ISN contact number once you get things working.
To get your own ITAD number, visit this link and follow the instructions for requesting your own number. It’s easy, but detail matters so do it right the first time! Within a few days, you’ll get your shiny new number. And, in a few more days, freenum.org will notify you that your account has been established.
Setting Up An ISN Account at FreeNum.org. Once you receive your login credentials from FreeNUM, log in to your account. Leave the DNS Wildcard setting the way it is. All you have to do is insert your fully-qualified domain name in the FQDN placeholder. For example, if your FQDN were sip.big.edu, then the last part of the DNS entry should look like this:
sip:\\1@sip.big.edu!” .
Save your entry and wait an hour. Then test it by dialing your new ISN number or, after logging into your server as root, use a command like the following. Turn your SIP URI around from 6781234567*1061 so that it looks like this:
dig @freenum.org NAPTR 7.6.5.4.3.2.1.8.7.6.1061.freenum.org.
Aretta Introduces Free NetPBX. In an industry first, Aretta Communications is rolling out a free Asterisk hosted solution known as NetPBX Free Edition. The only cost is for the minutes you use, and the free hosted service will support one inbound or outbound call at a time. Everything including the SIP trunking is preconfigured so the system is literally plug-and-play. We’ll provide a more in-depth review once we’ve had some time to play.
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New Vitelity Special. Vitelity has generously offered a new discount for PBX in a Flash users. You now can get an almost half-price DID and 60 free minutes from our special Viatalk sign-up link. If you’re seeking the best flexibility in choosing an area code and phone number plus the lowest entry level pricing plus high quality calls, then Vitelity is the hands-down winner. Vitelity provides Tier A DID inbound service in over 3,000 rate centers throughout the US and Canada. And, when you use our special link to sign up, the PBX in a Flash project gets a few shekels down the road while you get an incredible signup deal as well. The going rate for Vitelity’s DID service is $7.95 a month which includes up to 4,000 incoming minutes on two simultaneous channels with terminations priced at 1.45¢ per minute. Not any more! For PBX in a Flash users, here’s a deal you can’t (and shouldn’t) refuse! Sign up now, and you can purchase a Tier A DID with unlimited incoming calls for just $3.99 a month and you get a free hour of outbound calling to test out their call quality. To check availability of local numbers and tiers of service from Vitelity, click here. Do not use this link to order your DIDs, or you won’t get the special pricing! After the free hour of outbound calling, Vitelity’s rate is just 1.44¢ per minute for outbound calls in the U.S. There is a $35 prepay when you sign up. This covers future usage and any balance is fully refundable if you decide to discontinue service with Vitelity.
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