Outbound dialing in the call center has undergone a revolutionary change in the past decade. In October of 2004, the Supreme Court of the United States allowed a ruling from a lower court to stand that enabled the FTC Do Not Call regulations. The widespread registration of home phones, along with restrictions on dialing cellphones (and their increasing share of the number of phones outstanding), signaled a massive shift in the way outbound contact centers would operate. Automatic or predictive dialing was not killed off then, but it has been in critical condition since. Continue reading “Warm Leads and Outbound Dialing in Today’s Environment”
Upgrading from TDM-Based PBX
Upgrading from a TDM-based PBX to a pure VoIP solution will reduce costs and allow for more flexibility, but these changes will also require changes to your business processes before you’ll be able to maximize the benefits of the new platform.
The biggest change will be the ability to use a soft phone instead of a desktop phone. Call centre phones are often big and bulky and require expensive headsets. Soft-phones are a program that runs on the computer. This allows for any standard PC, USB, or Bluetooth headset to be used. Softphones mean that the agent can easily work on a laptop from anywhere, even at home.
As the contact centre solution isn’t tied to phone calls, agents are able to handle other types of customer interactions on the same platform. This can include Twitter and e-mail support. The mixing and matching of “call” types allow the agent’s performance to be monitored with one system.
With the removal of the physical phone lines, calls can easily be routed to different sites worldwide. This allows for global companies to offer 24/7 service without having employees work graveyard shifts. With multiple installations at different colocation sites, disaster recovery is easy and in many cases, seamless.
These new features will require staff to be trained and processes to change in order for the benefits to be fully realized.
Housing a Contact Center ACD on a Hosted Platform
As mentioned in a prior post about softphones, cutting the initial startup costs of a contact center can be a crucial step in getting your operation off the ground. While some may find it favourable to buy, maintain, and store their own server hardware in a nearby location or even on site, this might not always be possible due to the relatively enormous costs of purchasing and housing said hardware. This is where hosted platforms come into play. Continue reading “Housing a Contact Center ACD on a Hosted Platform”
Using Agent Skill Priority in Skills-Based Routing
Skill priority is a topic that has been covered before in this blog, in the case where skills are being used and we want to ensure that calls are routed evenly. However, we have found that some are unclear on the benefits of assigning differing levels of skill to agents.
Continue reading “Using Agent Skill Priority in Skills-Based Routing”
Softphones for Contact Center ACD
In the ever-evolving realm of contact centers, minimizing startup and operating costs should be high on the list of items that a center should want to accomplish. There are numerous way to go about this, such as buying used office chairs instead of new ones, but if you are looking to cut costs without sacrificing functionality, using softphones instead of physical phones can be a good start. Continue reading “Softphones for Contact Center ACD”
VoIP Essentials: Codecs
When it comes to the contact center industry, there is a plethora of terms, jargon and concepts that can be new, foreign and confusing. Voice over IP, or more commonly, VoIP, adds even more to our shared vernacular.
One very important concept that permeates our new digital landscape is codec. It is important to understand what a codec is and why different codecs are used in different applications.
A codec can simply be thought of as a tool that converts audio and or video from one digital format to another digital format. When a signal is captured from a camera or mic is it transformed with a codec into a digital form, then stored and transmitted to our screens and audio devices. For example, a very popular codec is the MPEG family of codecs: MPEG-2 is used to transmit audio and video in digital cable boxes and satellite receivers and MPEG-3 is the codec that is best know as MP3.
With VoIP there are a few popular codecs that are broadly used and while they all serve the same purpose, they can be vastly different. VoIP has special constraints that digital broadcasting and digital audio do not have to constrained by. VoIP has to transmit the audio back and forth to both ends of the call with in milliseconds and it has to do that using a minimal amount of bandwidth. This means that we need to keep in mind:
- how much time and CPU resources it takes to run the audio through the codec ?
- how well does the codec compress the audio?
- how is the codec is affected by network latency?
When picking a codec there is trade offs with each one, as some codecs such as G711 use very little compression and sound excellent but they require more bandwidth to transmit so it is more susceptible to audio problems when the network is saturated or if you’re on a WIFI network. G729 uses a compression process that is highly effective but that comes at a cost. Most people would not be able to notice the difference a call encoded with G729 vs G711 but G729’s compression process is patented and any phone using it must have the proper license to use it. Because of this, it’s impossible to find legal softphones that can do the G729 for free. Being able to use G729 can make the all the difference when the Internet connect is subpar because of the advanced compression. I’ve been able to have conversations using G729 over weak WIFI signals when the same soft phone with G711 has been unusable.
Another VoIP codec that is popular is the GSM codec. It was made popular by the cell phone industry as the codec is not patented and it offered an acceptable balance between network utilization and voice quality. When possible I recommend staying away from GSM because if you do use it, it will sound just like your on a cell phone. In general, if you are making VoIP calls for business reasons, you’ll want to avoid this codec for this reason.
When I’m setting up a contact center or PBX using the Q-Suite platform I always make sure that the prefered codec is G711 on the server side, and I let our clients know that, unless there is high bandwidth costs or limited bandwidth between different locations. I make sure that I disable the other codecs like GSM,G726 and G729 if there are no licenses for that codec. When there is remote users that may be using questionable networks or a need to conserve network bandwidth, G729 is the codec of choice.
The Challenge of VoIP System Failures Not Addressed by Most High Availability Designs
Hardware or software can fail at anytime and induce a system failure. It is not possible to reduce such failures to nil. When VoIP based systems experience such failures, it results in the loss of on-going calls. High availability (HA) or redundant systems cannot address this unless they are capable of restoring an on-going call without either one of the end-points re-initiating the call. Most high availability system for Session Initiation Protocol (SIP) based VoIP calls and their redundancy setup, deploy an immediate replacement of the failed component/sub-system to allow continued use of the system. It is good enough for many situations but it might not be adequate for mission critical applications when the HA cannot not restore on-going calls.
Imagine a scenario where an outside caller initiates a call and when it hits the demarcation point of the contact center installation. This could be a premise based contact center or a Cloud set up offering virtual contact center services. When the call setup reaches the intended peer and conversation starts, it is possible that your system, either Cloud based or on-premise solutions, could experience a failure. Once the system detects a failure, its high availability and redundant setup will kick-in and the system will be ready for future calls but what happens to the on-going call? They just die. This is the normal operating mode of traditional high availability systems including most high availability solutions offered for Asterisk. This issue becomes more critical for large contact centers using automatic call distribution (ACD) with significant traffic at any given time.
With contact center ACD, the importance of going beyond the traditional high availability is extremely important. Having the capability to keep calls alive through call survival is critical. This will allow the user to continue the phone conversation without the need for re-initiating the call. It is a sophistication in offering redundancy that goes beyond recognizing the need to bring into action the replacement software and hardware components. It introduces intelligence required in preserving all the on-going calls essential for mission critical systems.
Ease of trying Q-Suite Call Center software
When choosing software for your contact center, one important aspect is the feature list. You will already have an idea of the functionality you need, and can see the products feature list on their marketing material. A demo or two can be seen, and then the decision to buy or not buy is made. Continue reading “Ease of trying Q-Suite Call Center software”
Contact Center ACD + Zendesk = An enhanced Customer Service platform
Contact center ACDs are still a highly desirable method of allowing customers to connect and interact with customer service representatives (CSRs). Web-based ticketing systems are also a convenient method to keep track of any types of communication that may occur between customers and CSRs. The blending of Indosoft’s Q-Suite with Zendesk makes for a perfect combination of a fully featured contact center with a cloud-based issue tracking system. Continue reading “Contact Center ACD + Zendesk = An enhanced Customer Service platform”
SIP Registration Timeout Settings for High Availability
In setting up a high available telephony system most worry about the back end and ensure it functions as they would expect and require. However one highly visible user issue I have seen is a misconfiguration of the connected SIP phones in regards to the registration timeouts. When these are very high on your SIP phone then it may not notice a service has moved (via IP/DNS/etc changes) due to a HA switchover and can potentially miss incoming calls until it does. Typically an outgoing call attempt will work or at the very least cause a registration attempt to the new server the service has moved to.
For example take a look at Aastra, their defaults in a few models I’ve seen are at a half hour for a failed registration.
If the failed registration timeout is half an hour and the phone attempts to re-register and fails your phone will show an error or unregistered for the next half hour. This can happen in the cases where the registration comes in as a box is failing or a failover happened and the configuration is being written/updated due to the switchover process. More reasonable set of values are shown in the following.
In this one I’ve lowered the registration failed retry and also the timeout retry timers. These will make the SIP phone resolve the registration issue quicker by retrying more often than the defaults. They could be lower depending on the situation.
One precaution before everyone sets these very low. These settings should be set appropriately when the SIP phone is off-site and there are protections, for example Fail2Ban, in place to block brute force attacks. In these cases where the SIP phone is on an app on a mobile device this failed registration timeout should be set high enough to not trigger a lockout of a valid device. If the devices are in-house or IPs can be whitelisted then the values can be lower without worry.