ShoreTel SIP trunks using Asterisk?

February 13th, 2012

Since we published the blog on the ShoreTel SIP implementation using Ingate, we have been deluged with requests for more detail.   Specifically, people want to know if there is a more economical  solution for bringing  SIP dial tone into the ShoreTel architecture when you are trying to add just a few local lines to a branch office.    Historically, given the feature depreciation that users might experience with a SIP trunk on ShoreTel, we had discouraged the adoption of SIP dial tone entirely!    In fact, if you already have a ShoreTel T1/PRI supporting your deployment  there is little economic benefit to be gained by replacing the PRI with SIP.   After all you already own the PRI gateway and now you have to go buy an Ingate box.  Might make sense for a new deployment, but it does not make since for an existing deployment or for adding just a few SIP dial peers at a remote office location.

Well then, what about, keeping the ShoreTel PRI but bringing the dial tone in over a SIP trunk?

The fact of the matter is, that many carriers do bring the PRI circuit in over a SIP trunk!   They then run the circuit through a Integrated Access Device (e.g. IAD) that converts the SIP trunk to PRI and the resulting hand off to the ShoreTel is a traditional TDM flavor of PRI.   This makes the carrier happy and the ShoreTel PRI continues to operate with no feature depreciation as everything from a ShoreTel perspective remains TDM!   The original investment in ShoreTel equipment is protected and you get some of the benefits of having a SIP trunk  like foreign DID’s and more fail over options.

This got  us wondering if we could do the same configuration by rolling our own?  We came up with two solutions.

First we created our own SIP Appliance using a low cost Linux box running CentOS and our favorite Asterisk dis tro, both of which are available as freeware.  For the SIP to PRI configuration we added a Digium T1 card bringing our total hardware cost to less than $900 for both the PRI hardware and the Linux computer!   Next we setup a  simple T1 Tie line between the ShoreTel iPBX and the SIP appliance.   We then created an IAX trunk to our SIP provider from the WAN side of the SIP appliance,  assigned our DID’s to the SIP dial peer and we were up an running in no time!  The configuration is a classic B2BUA and it works just like the carrier provided CISCO or Adtran IAD!

Next we set out to build a SIP solution that was a pure dial peer with no TDM hardware at all!    Lets assume you have a remote branch office and you would like to provide a local DID for that branch.   Using a SIP DID and less than $500 dollars worth of Linux computer hardware ( see our blog on the Shiva plug) you can easily make this happen.   We used the same appliance we created for the PRI conversion, this time without the Digium T1 card.   We specified the remote SG50 as the IP trunk source and terminated it on the SIP Appliance.  Again we created a SIP dial peer from the Appliance to the SIP dial tone provider and assigned our DID numbers.  It worked flawlessly as a simple, end point SIP dial tone solution providing local DID numbers to that remote branch office.

One of the challenges of ShoreTel is that the entire architecture needs to be described, from an IP topology, inside a private address space.   That is why you can’t just interconnect a ShoreTel SIP trunk group with an ISP.   When establishing your SIP Peer to your provider,  you are generally going to link with a public IP address.  Using our SIP appliance we were able to bring up a remote branch office with an end point that supported 4 SIP Peers and used a local DID for that branch.     (ShoreTel SIP trunks are licensed in packages of 5, while all SIP dial peers provide dual channels?) Again, the SIP trunks between the ShoreGear SG50 and the SIP appliance was created completely within the required private IP address space, yet the appliance interfaced with a public IP address to create the multichannel SIP dial peer.    The Asterisk configuration we created enabled us to configure on the LAN side of the firewall, yet specify a public IP address for the dial peer end point with no NAT issues to contend with.

At the end of the day, there are a variety of solutions for bringing SIP into your ShoreTel deployment!  Sometimes the best things in life are free!  We are no busy integrating the ViciDialer with ShoreTel to provide an enterprise class Outbound dialing solution for the contact center using freeware! More on this later……

6 responses to “ShoreTel SIP trunks using Asterisk?”

  1. DrVoIP says:

    ShoreTel phones are MGCP based and will not provide sip services.

  2. I am actually glad to read this blog posts which consists of tons of valuable information, thanks for providing these data.

  3. DrVoIP says:

    If you are a DrVoIP member, you will find the configuration file in your download section. Hit the AskDrVoIP option for follow on Tech Support! Thanks – Gandalf!

  4. Brian says:

    can you post your SIP.conf file..???

  5. DrVoIP says:

    Look for New video clip now in production! Or you could just purchase a couple of hours of tech support, after all that is how we pay the rent!

  6. Tim Miller Dyck says:

    Hello, thanks for this post showing this could be done (ShoreTel to Asterisk trunking). We are setting up a ShoreTel install based on ShoreTel 12.2 and FreePBX Distro 2.10. If you would be able to share your Asterisk SIP trunk settings and ShoreTel SIP trunk settings, that would be helpful in knowing what the right SIP configuration is to connect the two. Thanks! -Tim Miller Dyck, Ontario, Canada

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