Calling Web Services in Your Asterisk IVR

Web services in Asterisk dialplans

CRM or data integration aren’t things that you often associate with your inbound IVR.  That usually happens at the agent screen level. Sometimes, it’s handy to be able to pull data from an external data source and redirect the call based on the result. Continue reading Calling Web Services in Your Asterisk IVR

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

What To Do Before Someone Lets The Smoke Out of Your Server

The client was worried because one of the Asterisk servers had gone down without any notice at all. The overseer process on the other Asterisk server had noticed, and had taken over as the active server. The disruption was minimal. Agents were at work. But the client wanted to know what had happened, and what we could do to prevent the issue in the first place. After 30 minutes of poring over logs, digging around, and contacting the colo, we discovered that a tech had decided to swap out the power bar connected to that server. There was no notice, and not even a courtesy three-finger salute. Continue reading What To Do Before Someone Lets The Smoke Out of Your Server

Four Keys To Soft Phone Delight

A colleague once visited a client site, and found their server room to be a nightmare. Among other things, they had insisted their telco give them a PRI connection due to the improved reliability they thought they’d get. They had also decided to use a PRI to SIP connection in the server room to allow the flexibility of moving the telphony connection from one server to another if needed. When Justin got there, he found a SIP to PRI device on the telco side, and a PRI to SIP device connecting in to Asterisk. Continue reading Four Keys To Soft Phone Delight

Thousands helping Millions, Helping Your Federated Call Center Deployment

Thousands of volunteer experts have worked on Asterisk since 1999. This has made it into a powerhouse platform for telephony that you can use in your own system. Today it is a proven technology used by millions of users. Call centers and major PBX users are abandoning their legacy proprietary telephone switches to get the benefits offered by this Open-Source system. Continue reading Thousands helping Millions, Helping Your Federated Call Center Deployment

One Big Mistake You May Be Making With Your IVR

“The dialplan isn’t working!” Four words that increase your stress without telling you anything useful. In the last couple of weeks, though, I’ve heard it a couple of times. Both times it was the same cause, so I thought I’d tell you about it.

You can’t leave people hanging in an IVR (Interactive Voice Response). Silence is bad practice in writing dialplans, just like it’s bad in radio. But, it’s easy to miss some of the points where silence gets introduced. Continue reading One Big Mistake You May Be Making With Your IVR

SIP Phone Configuration the Easy Way

Phones Over SilhouetteFor large installations configuring all the VoIP phones can be a pain. Perhaps you have polycom SIP phones which I have seen in the past delay starting the web configuration interface until after the phone is ready to handle calls. This is great when the phone is in use as the user can use the phone sooner while not waiting on the processes providing the configuration interface, however when configuring a large number of phones it causes delays if manually configuring each phone. Even when the config interface comes up fast it’s still a labour intensive task to go through each phone manually configuring them. During this manual process mistakes can and will, especially given a large enough set of phone be made, some of which may not be initially obvious. For example, if the dtmf mode of the 47th phone configured today left at the default of RFC2833 but the system is using SIP INFO? That phone gets used for a while but then the user goes to use an IVR and has to file a trouble ticket to get it resolved after finding out dtmf isn’t functioning as expected.

The solution to this manual headache and time drain? Provisioning. This is where the phone system will have the configuration for each phone and the phones download their configuration so they are synced with the phone system options.

Provisioning is usually accomplished via a tftp where the phones configuration files reside. At boot the phone gets an IP via the local dhcp server and the dhcp response has an extra configuration setting, usually Option 66, to inform the phone where the tftp server is. From the tftp the phone will generally request a general model configuration file, this can handle firmware upgrades and common settings specific to a model or make of phones. After that file a request is usually made for a file with the MAC ID to differentiate the phones. This MAC ID file is specific to that individual phone and will contain the details of the SIP connections, extensions, and anything else just for that phone.

The Q-Suite supports provisioning a number of SIP phones from various manufactures. With easy support via templating to extend to new models as needed. The administrator only needs to collect and input the MAC ID’s and choose the proper template when configuring the extensions and the generation of the config files will be done for them. With DHCP Option 66 is set properly boot up the phones and they’ll be functional and making calls.

SIP With TCP Benefits Asterisk Call Centers

You need both reliability and performance in your VoIP (Voice over Internet Protocol) software to get the best out of it. Traditionally, performance has been emphasised in telephony software such as Asterisk and Asterisk-based call center software with reliability coming from the infrastructure you set up around it. Network and hardware performance are key components of a reliable voice connection. For this reason, call signalling in VoIP has been done via UDP (User Datagram Protocol) most of the time. Continue reading SIP With TCP Benefits Asterisk Call Centers

Save Money, Do More With Asterisk-based Call Center Software

You can no longer doubt that Asterisk is here to stay in the telecommunications world. Over the last decade, the rate of adoption has been skyrocketing. Call centers are no longer complaining about Asterisk-based solutions, or worrying about having to have an Asterisk expert on-hand. With Asterisk providing a platform for contact center ACD (Automatic Call Distribution) solutions to sit on, you can transition from a legacy system to one built on readily available and mature software. Continue reading Save Money, Do More With Asterisk-based Call Center Software