Stuck on SIPStuck on SIP

The next-gen IP network can take early lessons now, and let's hope providers continue to find ways to reduce costs and close gaps that cause delays or cost money.

In my last update: SIP Trunks: A Year Gone By, I mentioned we would revisit our service and reevaluate moving ahead and exploring new SIP Trunk services. The contract expired and we informed the provider in writing not to "auto renew." To our surprise, when we went to a new service we reduced costs again and added features we couldn't squeeze out of our former service.

About six months prior, we established service with Jive Communications using their PBX and integrating dial plans with our PBX. We also set up a single SIP extension using a Panasonic KX-UT670 desk phone at our IT partner's office, and this phone maintains service using Jive's hosted PBX. We have 3-digit dialing between each, and failovers to cell phones.

I've mentioned in other posts about the need to integrate the hosted PBX with premise PBXs, and we have done this primarily to accomplish energy savings (reduce energy consumption), strengthen our solution (no more local hard drive crash potential) and reduce rack space. One side effect was that our message waiting lamps went dead once we built the dial plan using the hosted PBX to take voice mail messages and provide automated attendant functions.

We issued the first port request to move our toll free fax service to the hosted PBX, and this took 7 days to complete. Our inbound faxes now route directly to the hosted service and we receive inbound fax via email. The key advantage is we no longer need to have two fax machines in a hunt group to guard against a machine running out of paper; we also now don't have to worry about modem hangs or typical reboots of the fax machine to clear errors, or electrical-related issues with fax machines.

Our second port request to move our DID numbers to the hosted SIP trunk service took 11 days to complete. Besides costs being reduced, we gained a better solution to route calls to our cell phones.

The inbound DID numbers route to the hosted PBX dial plan and then concurrently ring through to the premise PBX and to each cell phone. This frees up a channel or concurrent call session for each inbound call by not having the premise PBX route the calls. There is a one-ring delay before calls start to ring on the cell phones and this is a slight improvement over the two-ring delay we have when using the premise PBX to route concurrent calls.

Now that we need fewer concurrent calls because routing is accomplished through the cloud, we use fewer SIP trunks (SIP trunks and concurrent call sessions are one and the same). Fewer SIP trunks subscribed equates to a reduction of approximately 50% in cost over our former SIP trunk solution.

Some our desk phones are used for back office functions and testing, and so we must block Caller-ID from these phones. With this provider we entered in 0000000000 to display the extension number digits for those desk phones. We'd discovered that using outbound CLIR (Caller-ID restriction feature) disabled all outbound dialing, so for any CLID modifications or manipulations, it's a good thing to test before implementation.

The $15 port charges per DID was worth the cost because of the number of SIP trunks we no longer needed for routing concurrent calls to both the PBX and to cell phones.

Because we are striving to reduce our energy footprint, I targeted the voice processing system (VPS) for retirement. In our office, we use Microsoft hosted Exchange for company email. Having synched our mail to our iPhones and desktop computers using Exchange is a huge improvement over Gmail. We implemented hosted Exchange with the end goal of eliminating our voice processing system. Now, our voice mail messages are delivered to both our desktop computers and iPhones.

The other cool feature we gained is the ability to use CounterPath's BRIA for iPhones and iPads. We can always connect to either the hosted PBX or office PBX from our laptops when we're in the field or if we are mobilized for support during inclement weather or emergency situations.

Our former provider requires 30 days' written notice of service termination. The issue we could not address was giving notice to the provider, since our new provider stated that disconnecting service before port completion would likely cause loss of the DID numbers we wanted to retain. So we delayed giving notice until the port requests were completed, tested and verified. The former provider only billed for billable usage on the final bill that arrived in just 3 days after submitting the disconnection of services.

While abandoning premise voice processing systems for hosted or hybrid configurations like ours isn't for everyone, it's important to look at what's really required. Some companies must have voice mail transfer directly to another user's mailbox; so far we haven't even started exploring a solution, since we don't use it.

Overall, the process isn't ideal but it's certainly easier than dealing with most Telcos. I can't say I miss message waiting lights or the days of haggling with the Telcos to find ways to reduce costs or improve services. SIP trunks at least for an SMB such as my company have proven over time to be better, faster and cheaper. The risks are worth the issues and thus far the issues are so insignificant that I can't imagine going back to copper.

The next-generation IP network can take early lessons now, and let's hope that providers continue to find ways to reduce costs and close gaps that cause delays or cost money. My biggest personal joy is calling on our new provider (Jive Communications) and picking their brain with, "Can you do...?" Thus far, they always responded, "Yes, we're working on that request already," or, "Let's look at this." Pretty cool words coming from a "telephony provider.

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