FreePBX Offers SIP Service

Not only does FreePBX provide one of the most feature rich PBXs in the market, with a price that can’t be beat, it is has also been the key for thousands of businesses to escape the lock that traditional telephony providers have had on them for so many decades.

With SIP phone service so readily available, it has led to hundreds of SIP VoIP telephony providers and with that, a lot of confusion as to what providers to use and who is going to provide reasonable service and be ready to support a FreePBX/Asterisk based platform, or who is even going to continue to be around as many have gone out of business. Having worked with hundreds of customer systems, I can tell you that the reliability and technical knowledge amongst these providers varies across the board and choosing who to use or trusting what they are telling you when things go wrong can be a daunting task for the customer base.

With the affiliation of Bandwidth.com as the primary corporate sponsor of FreePBX, we decided to help our customer base by developing a service that would be mutually beneficial to FreePBX users looking for reliable SIP trunking services and for the FreePBX project to gain a revenue stream to further support its development efforts. Bandwidth.com has one of the best VoIP infrastructures in the US market with a nationwide network and multiple points of redundancy across the country. They are also a strong and financially healthy company, with three INC 500 awards from 2005-2008, one of only nine companies to win the award three years running. Of those nine privately held companies, Bandwidth.com was the 4th fastest growing overall and the only Telecom firm on the list.

Although Bandwidth.com has been supplying one of the best SIP trunking services in the industry, they have not offered their service using instant online activation nor is it available to systems that do not have static IP addresses. In order to address this we have created a new FreePBX branded SIP Trunking service delivered on Bandwidth.com’s network with the goal of seamless interoperability with FreePBX and available for quick and easy activation so you can be up and running in minutes. Our initial offerings will target the commonly requested Unlimited 2-Way SIP Trunk service complete with one or multiple DIDs on an account and E911 enabled service across the US.

Although our service will not be free in the same way that FreePBX is, we are striving to create an excellent service at a reasonable price and will evolve the capabilities as new customers come online and provide feedback for what they are looking for. One example of “reasonable” is the cost of DIDs (inbound phone numbers). We have found that many providers serving SOHO and small SMB businesses charge $5-$15 per additional DID per month! There is no good reason for that and as a result, we will price our DIDs for only $1 each per month, so purchase as many as you need!

Are you looking to obtain top quality SIP service that runs on one of the best VoIP network in the country, is designed to work with FreePBX and will directly support the further development of the project?

Cisco Unified CM 6.1 to Asterisk and FreePBX SIP Trunks (Powered by Bandwidth.com)

One of the systems I manage is an 875 Extension Cisco Unified Call Manager(UCM). At the moment the system uses SCAN trunks for long distance calling. These SCAN Trunks are provided by the state of Washington and interconnect via a four port FXO card. Callers use a PIN to make long distance calls. This is some seriously old school technology and as such has sound quality to match.

So, armed with a four port SIP trunk account from Bandwidth.com, I set forth to make a SIP trunk from the UCM to Bandwidth.com. Long story short… it does not look like I can. I can set things up that should work, but don’t. There is an active TAC open with Cisco and when(if) we make it work then I’ll be back in a new blog entry.

Instead, we have “Cisco Unified CM 6.1 to Asterisk as SIP Proxy for Long Distance service.”

What is difficult to impossible with UCM is trivial in Asterisk w/FreePBX.

For Bandwidth.com I made a new SIP Trunk with the name of “freepbx” and here are the PEER Details:

username=myusername
type=peer
sendrpid=yes
secret=mypassword
qualify=yes
insecure=very
host=trunk.freepbx.com
fromdomain=trunk.freepbx.com
context=from-trunk

The Register name was formated like this:

myusername:mypassword@trunk.freepbx.com/360746XXXX
——-
For the UCM, I made a SIP Trunk named “ucm” and here are the PEER Details:

type=peer
context=from-callmanager
host=10.XXX.XXX.XXX
disallow=all
allow=ulaw&alaw
nat=no
canreinvite=no
qualify=yes

and in extensions_custom.conf, courtesy of Philippe, I added ( to have these calls bypass most of the dialplan logic )

[from-callmanager] include => from-internal
exten => _1NXXNXXXXXX,1,Dial(${OUT_12}/${EXTEN});
exten => _1NXXNXXXXXX,n,hangup

NOTE: the the “OUT_12” reflects the “freepbx” trunk and the last thing to do is allow anonymous SIP as the calls are routed w/o authentication from the UCM ( for now.)

From CallManagerToSIP

Now on to UCM.

From CallManagerToSIP

Since we are using no authentication to send calls from UCM to Asterisk this part is somewhat straight forward. First step is to define a “SIP Trunk Security Profile Configuration” making sure the “Outgoing Transport Type” is UDP, which looks like:

From CallManagerToSIP

Then we create the SIP Trunk, noting that the “Calling Search Space” in this case is “Reserved Incoming Calls” which basically does not allow inbound calls to go anywhere:

From CallManagerToSIP

The last three items to set up are the “Route Group, Route List and Route Pattern.”

Here is the Route Group:

From CallManagerToSIP

Here is the Route List:

From CallManagerToSIP

And since we need a way to direct calls out this SIP Trunk to make long distance calls, here is the Route Pattern so users dial “891NXXXXXXXXX” to dial out:

Bare in mind, I only want calls going out this trunk ( not in) so if you wanted bi directional calls… You would adjust the “Calling Search Space” in the trunk to allow it.

Otherwise, that is the setup that is live for now. Once an Authenticated SIP Trunk can be sorted out on the UCM, I plan to go directly to Bandwidth.com and skip the proxy.