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Make Your Call Center Technology Work For You

Your call center is constantly having to deal with new challenges. Your client decides things now have to be done this way or that. Regulations change and now you need to record your data differently. Or not record it at all. A new business opportunity springs out of nowhere, and you have to respond quickly to capitalize. When you’re handling change every day, it’s important to have a flexible call center system. Our Q-Suite software is such a system.

Continue reading “Make Your Call Center Technology Work For You”

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Running the Beachfront Call Center

In 2004, we had a client with a call center in Northern New Brunswick.  For a dozen seats, he required thousands of dollars in telephony equipment, including the Pika board required to wire in the multiple incoming telephone channels, CTI server and a server to manage the leads and agent interaction. A few years later, and after a downturn in the economy, he was able to repurpose the equipment. He moved it to his basement, kept a few call center seats there, and used DSL to connect to a SIP provider. If he were to start today, he wouldn’t need the telephony card, the servers, and the wiring. He could start in his basement, using the Cloud,  and only move to an outside office when his growth demanded. Continue reading “Running the Beachfront Call Center”

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Feature Highlight: Conference Rooms

The common conference room, a necessity in today’s world with employees working from home, travelling, and distant customers. No product with PBX functionalities should be without one.

Lets take a look at the options for the conference rooms provided by the Q-Suite.
Q-Suite-Conference-Room-Admin-Page

The options seen are:

  • Name – a Friendly name. If the room is used for a purpose, as our example is, then it’s best to name it after that. This will be the name seen elsewhere within the system.
  • Extension – the extension used for directly dialing it from an extension.
  • PBX Server – This is hidden if your system has a single PBX server. In the case where there is multiple this can be used to distribute load or in the case of a geo-diverse system ensure the conference is on the server closest to the majority of participants to ensure quality and limit bandwidth usage.
  • PIN – a numeric pin to secure the conference with.
  • Disable first member message – If checked this will not playback a message stating the first member is the only one in the conference.
  • Announce user count on join – If checked this will announce to the user how many users are already in the conference they are joining.
  • Play music on hold for single member – If checked this will play music on hold when the conference has a single user. Otherwise the user will only hear silence.
  • Music On Hold Class – what music on hold to play when it’s a single user conference and the above is checked.
  • Suppress enter/leave sound – If checked a user joining or leaving the conference will not trigger the tones to be played to all conference members.
  • Announce names on enter/leave – If checked each user joining will be prompted to record their name before they join. This recording will be used as the join or leave

With all of these options you can configure the conference rooms you needed. However to fully leverage conference rooms, or any single feature, the ability to combine and interlink them to suit your needs is required. Let’s take the example of a daily standup which needs a conference room for a few remote employees. It’s always starts at 8am and lasts about 30 minutes. One way to ensure outside callers do not use it outside of this hours is a pin but you can also mix it with the Routing Rules so the DID or IVR/Auto Attendant option is not configured to point directly to the conference room but to a routing rule first. This routing rule will only route callers there at 8am until 8:30, although I’d recommend 15 mins earlier at least to allow early callers in and maybe a bit longer depending if the meeting has a strict end time or not.  The routing rule can then send them to an auto attendant or elsewhere in the system if it’s outside of the hours — add multiple rules to allow different routing if it’s before to note they are to early, or if they are late to send them to a recording of the conference to hear what they missed.

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Schedule the Sale

Callbacks can be a great way to give that last push to get your sale. Callbacks can also drain the performance (and profitability) of your call center. For such an important tool, they can be woefully misunderstood. Your call center software likely has a number of settings surrounding callbacks. Make sure you understand what your agents are doing with their callbacks.

There are two primary types of scheduled callbacks: Continue reading “Schedule the Sale”

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One Way to Stop Overloading Your Telephony Server

Too much traffic can bring down your call center

There is a subset of your staff doing most of the work. This is the well-known Pareto Principle, where 80% of results are achieved by 20% of causes. 20% of your employees are doing 80% of the work. 20% of your clients are responsible for 80% of your profits. Understanding how this works in your cloud-based call center can help you be more efficient. Having 20% of your telephony servers handling 80% of the calls can be a recipe for disaster.

You may have one number that comes in on one trunk, and use smart IVR routing to get calls to the right spot. That’s pretty common. Your SIP provider may only allow one IP to communicate with it. That’s also pretty common. If you just point it to the first of many telephony servers, though, that server is going to be doing a lot of work. One strategy is to have agents distributed across multiple servers to spread things out. Another is to have multiple trunks. None of these solutions is ideal for heavy usage cases. On commodity or Cloud hardware, you will reach the capacity of a server, and be stuck. It’s worse if you have occasional bursts of activity over one trunk or another.

Load balancing is very important under heavy call volumes. For telephony, this is usually accomplished by having a load-balancing SIP Proxy in front of your telephony servers. Handling the media (voice, usually) is the hard part of a Voice over IP (VoIP) call. Signalling is fairly lightweight. Telling the server a call is coming in, accepting it, saying “Yes, I’m still here” is really just some text being passed back and forth. Taking the audio, encoding it, breaking it into packets and sending it off, possibly recording it, is the hard part.

One interesting fact about most VoIP traffic, such as SIP, is the signalling and media can happen on different servers. In the case where only one server is allowed to connect to the provider, this almost always means the signalling. The media can, and often does, connect to a different server.

On inbound, a SIP proxy handles the easy part. It can also decide which of the available servers will take the next call, and arrange the details between your server and your service provider. This way, there’s not one single server in a multi-server call center that’s struggling with 80% of the call volume.

For outbound, the usual solution is to have your trunk proxied, and the outbound load distributed evenly. This usually means spreading your agents out so the outbound call volume doesn’t overwhelm the server. Again, your SIP proxy looks like the trunk provider to each of the servers using the proxy. The call gets dialed, then the media is processed as normal.

In either case, whether inbound or outbound, you can avoid having the Pareto Principle cause disruption. The better you do with call distribution, the fewer complaints you’ll have with call problems.