There is a constant demand from our contact center software customers for greater granularity in the details of the call center reports. Reports are based on data collected and retained by the system and there are numerous reporting possibilities on Agent, Queue and ACD performance. Here are some of the generic contact center reports based on Indosoft call center software, Q-Suite. Continue reading “Q-Suite has in-built CDR to capture cradle to grave data for calls”
Using Eyebeam with G729 codec on Asterisk based systems
Asterisk is modular and has so many incredible built-in applications. Modern day call centers require a lot of functionality and, Asterisk provides a contact center platform that is truly versatile, at tremendous cost savings. Live listening in Asterisk is an exciting feature for call center managers. Live listening is not new but Asterisk makes it much more accessible with its chan_spy. G729 is a popular codec (you can purchase G729 license from Digium) which reduces bandwidth consumption to around 45 Kbps per channel. When bandwidth costs are at a premium G729 can provide a 50% savings. Agent phones using G729 can minimize transcoding and reduce system load on Asterisk. Eyebeam, the licensed full featured version of X-lite, is a popular soft-phone with inbuilt G729 codec. Continue reading “Using Eyebeam with G729 codec on Asterisk based systems”
X-lite default setting hangs-up the phone if you are silent
Last updated: August 10, 2016
We have clients using our call center software Q-Suite with a variety of IP-phones and softphones. One of the popular softphones is X-lite and its licensed version eyeBeam. There are times when it takes considerable investigation of a problem to identify the cause. Recently we came across one which warrants a mention.We were investigating complaints about calls being dropped. Generally with VoIP and SIP, this raises all sorts of alarm and as always, we started a detailed investigation to determine cause of the connection drop between the softphone and the our contact center platform. Many of our customers are hosted multi-tenant contact center service providers and the usual suspect is the infrastructure. In this case, the debug log did not point to anything suspicious until we noticed a pattern of agents being silent before the the softphone sending a ‘BYE’. This was happening without any agent action. The SIP user-agent (X-lite) was hanging up. It did not take very long to determine the culprit within the X-lite settings. You have the screen-shot with the red arrow pointing to the checkbox.
X-lite seems to by default, check the ‘Call Inactivity’ check box and inflict considerable headache any to contact center technical staff who relies on default settings. We noticed it when the call center agents were waiting on one end listening to the interaction between their customer and an automated third party verification IVR. The more sophisticated the call center software suite, the more you will have to button down all the setting carefully. Silence suppression is an option not supported by Asterisk.
Other softphones have similar functionality. For instance, Zoiper phones appear to have a “Keep alive time-out” setting that defaults to on. Recent versions allow you to control that yourself under the “Advanced” tab of the Account options. Be sure to check out your particular brand of softphone for the same kind of settings.
Other softphones have similar functionality. For instance, Zoiper phones appear to have a “Keep alive time-out” setting that defaults to on. Recent versions allow you to control that yourself under the “Advanced” tab of the Account options. Be sure to check out your particular brand of softphone for the same kind of settings.
Basic network topology for Call Center setup
Efficient network setup can enhance the quality of your VoIP experience. Even if you are averse to VoIP terminations for routing long distance traffic, within your call center, you can take advantage of VoIP. Here is a simple but efficient topology for Asterisk based call center setup.
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VoIP – Bandwidth, Codec and Voice Quality
VoIP (Voice over Internet Protocol) has established itself as a dominant force in telecommunications. SIP (Session Initiation Protocol) is the application layer for creating and controlling the calls and the voice is governed by RTP (Real Time Transport). When we are setting up a contact center or converting to VoIP, we want to know how to take advantage of the VoIP without getting into the pit falls of the technology.
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Configuring large contact centers with Asterisk
The precise extent of Asterisk usage for PBX and call centers may be unknown but one can safely say that it is omnipresent. The clear give away is when you call an office and hear the default prompts that come with standard Asterisk installation when someone’s extension goes to voice-mail.
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Considerations while setting up contact centers using Asterisk
Asterisk opens up a lot of options while setting up contact centers. Last ten years have heralded a revolution in the contact center technology platform. During this period VoIP technology has dramatically altered the landscape of of telecommunications. Asterisk being an hybrid PBX with inbuilt support for VoIP and TDM, is the force behind the change in the contact center technology platform. Let us consider some of the options that are available for Asterisk based call centers. Continue reading “Considerations while setting up contact centers using Asterisk”
Practical limits on predictive dialing algorithm
Last updated: August 12, 2016
Predictive dialing is used by outbound contact centers to improve the talk-time of their agents. A good predictive dialer algorithm will pace its dialing based on the hit-rate and the number of agents waiting to receive calls. Hit rate is the percentage of live connects versus call attempts. The dial rate of a predictive dialer is limited by the drop rate allowed in the country of operation. For instance, the FTC in the United States limits the drop rate to 3%. Drop rate is the ratio of calls dropped to the total calls answered by a person at the other end. Any call that goes off-hook (picked up by a live person) needs to be connected to an agent within 2 seconds, or it’s counted as a drop.
Predictive dialing is used by outbound contact centers to improve the talk-time of their agents. A good predictive dialer algorithm will pace its dialing based on the hit-rate and the number of agents waiting to receive calls. Hit rate is the percentage of live connects versus call attempts. The dial rate of a predictive dialer is limited by the drop rate allowed in the country of operation. For instance, the FTC in the United States limits the drop rate to 3%. Drop rate is the ratio of calls dropped to the total calls answered by a person at the other end. Any call that goes off-hook (picked up by a live person) needs to be connected to an agent within 2 seconds, or it’s counted as a drop.
Predictive dialing is more complicated than call progress analysis. The biggest factor in call progress detection is the no-answer duration. Typically for United States or Canada, it is at least 18 seconds. That’s roughly three rings on a phone. A good call progress algorithm estimates if the call will be answered or if it will be a no-answer. With digital signaling available for both TDM and VoIP, all other types of call terminations, such as busy, bad number, and disconnected, are known instantly. A predictive dialer algorithm has to be smarter than statistical call progress analysis. It has to keep the average wait time of agents between calls to a minimum without exceeding the drop-rate set by regulatory authorities.
There are also other challenges for predictive dialers. When the outbound dialing market gets saturated, it results in poor hit-rate. The dialer cannot just increase its dial rate to compensate for the poor hit rate. There are unwritten rules in the telecommunication industry limiting the number of call setup per second. Too many call attempts will result in telecom carriers throttling call setups by signaling network congestion. In scenarios where the dialer is already dialing at the rate of allowable call setups per second, there is little value for any statistical call progress analysis on calls in progress.
There is also a trend in using warm leads to increase hit-rate. Here there are fewer leads, so the predictive dialer needs to be very efficient. With warm leads, agents are talking longer and the dialer needs to work well for the smaller number of agents waiting for live connects.
With years of production experience, Q-Suite has a predictive dialer that is finely tuned to pace well and perform effectively. We introduced our call center solution for Asterisk in 2003 and have been improving it ever since.
Cost of Setting up a 100 Seat Call Center on Asterisk Platform
I am generally tasked with the Web DEMO of Indosoft call center software Q-Suite for Asterisk. A Web DEMO is usually very interactive where the viewer has full control of my laptop and we can do almost everything (not run the predictive dialer, of course). All the Inbound IVR, Queue, Agent nail-up and an incredible amount of contact center functionality, all are with a few clicks of the button. At the end of the demo we will invariably enter the topic of cost of setting up a call center platform. Continue reading “Cost of Setting up a 100 Seat Call Center on Asterisk Platform”
Skype for Asterisk, another wonderful development for Asterisk users
In 2003, when Indosoft started migrating its call center software to Asterisk, we knew it was the right move because we had switched over to Linux in 1999 and seen its benefits. We were not quite sure of the magnitude of change unleashed by Asterisk and its impact on the PBX market. After all, we were selling contact center solutions and were looking to get more control of the costs associated with development and support. When we look at Asterisk, it is from this narrow perspective of call center requirements even though we realize that Asterisk serves a very large community with a broader range of requirements.
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