Asterisk setup success 5

OACYS Technology

2/25/2004

We are using 12 phones in the office with one spare. All are SNOM 200 model phones (more about this later) that we purchased from ABP Technology (http://www.abptech.com/). We have a total of seven co lines, all analog (plus two DS3 for our Internet connection :) right now, that we connect using an Audiocodes MP-108FXO gateway (Also from ABP). The first four lines are part of a PacBell hunt group, while the next two are reserved for outgoing calls. The last line is a private incoming line to our Network Operations Center.

We have the MP108 configured to use the first six lines for outgoing calls, but to give priority to the last two, keeping our incoming lines clear. Incoming calls cause our receptionist phones to ring for 30 seconds, then play a message ("all our staff are busy...") and put the caller into an Asterisk Queue. We are eagerly awaiting the upcoming patch to announce queue position while in the queue. At any time callers can dial the extensions for Sales (1000), Customer Service (1001), Customer Service (1002) or Technical support (1003) all of which ring at several different phones. They can also dial direct extensions for any of our salespersons.

Some of our direct dial extensions are available to customers, others can only be used during a transfer from a receptionist. This protects our CEO and Senior Tech Support people from annoyances. Similarly, not all extensions will go to voicemail... Customer Service and Tech Support do not - they play a message to call back later or send an email.

The MP-108:

We were recommended this product by ABP and have been quite happy with it. The documentation is pretty weak, so I spent a couple of hours learning how the features work. The 4.0 build that came on the hardware is useless, so I immediately upgraded to 4.2 which solved all my problems and added some features I hadn't known I needed.

The 4.2 build supports polarity reversal, silence detection and current drop for disconnect detection. We use current drop here, which has the annoying tendency to take a couple of seconds to notice that someone has disconnected. I don't know whether this is an Asterisk thing, a telco thing, or a MP-108 thing, but I don't remember having any trouble with Digium's FXO card set to kewlstart.

Thanks to Clif Jones for a sample INI file and some advice in getting the MP-108 to work.

The SNOM 200 phones:

Audio quality and latency have not been an issue, which is to say that I've received no complaints. Connecting them to Asterisk is no trouble at all, though I'm having some problems getting them to stay registered (I think this is my fault, but I haven't had the time to debug it). I find the CallerID display to be very useful. And, once we got used to it, the conference calling feature became one of our favorite features.

The handsets are on the light side and the handset cords are just a little too short for large desks, but this is a problem that can happen with any phone, and they do use standard RJ-12 connectors. The headset and Speakerphone features are extremely high quality and well thought out - in general - and work as well as one could hope. You can't (quickly - and by this I mean in less than a second while the phone is ringing) change between headset and handset until AFTER you answer the call, but this is not a problem in most situations.

Our only major problem with the phones right now is with their surprise conference call feature. Here's how it works: If you have one call on the phone, you can press the transfer button to transfer, and hang up the handset to hang up. Great, just like a normal phone. Now the catch: If there is a call on hold and you want to hang up on the person you are talking to... DON'T HANG UP THE HANDSET. Doing so will conference the live call with the call on hold. OUCH! Similarly, don't use the transfer button, because this conferences the call on hold into the live call.

So far, I don't know of any way to do an unattended transfer while you have someone on hold. There are workarounds for most of this, but the sheer inconsistency of it is driving our office manager up the wall. Now, I may still have some settings wrong, but I think I've been pretty thorough. ABP is being very helpful in getting this figured out.

My solution: Asterisk has the ability to intercept the # key and use it as a transfer button. We've instructed our staff to a) never hang up by putting down the handset, and b) never use the SNOM's built-in transfer button. Instead, they press ESC to hang up and use the # key to initiate a transfer.

ABP also tried to sell us a Power-over-Ethernet device that would provide power to the SNOM phones during a blackout (in conjunction with a UPS of course). This is a problem that affects all VoIP phones, but our wiring is not PoE friendly. So, we went with the external power supplies for an extra charge. When we recommend/resell SNOM phones to our customers, we intend to sell the PoE system as well, wiring permitting.

All-in-all the office staff here is settling down to the new phones and there have been no show-stopping issues so far.

Asterisk:

Wonderful, it does everything I want. A few of the things it does:
  • Voicemail
  • Private and public extensions
  • Directory Service
  • Call Queues
  • Music on Hold (donated by our local High School jazz band)
  • Works with Cisco, SNOM, Pingtel, Budgetone, X-Lite and more
  • Automated attendent and IVR
  • A simple app that I wrote to play a message based on our network status (we are an ISP)
  • Fully customizable ring groups (i.e. ring all phones after hours, or only the receptionist during hours)
  • Conference Calling and Three-way-calling (enhanced by the SNOM phone's conference call feature)
  • Automatic Failover from PSTN to Internet and vice-versa
  • Support for multiple VoIP dialtone providers for low-cost long-distance
  • Transfer to cell phone

VoicePulse:

We use VoicePulse for our outgoing long-distance, and so far have not had any complaints. We were using NuFone, but are turned off by their lack of a web interface for refilling our account and viewing CDR. Both NuFone and VoicePulse have recently had (reported on the list, but not personally confirmed) outages that did not affect us. In both cases, setup and ordering were quick and easy (more so with VoicePulse).

ABP Technology Partners:

After an initial testing period with Asterisk and X-Ten, we purchased a single SNOM phone from ABP (I don't recall how we found them). They were very helpful and were willing to sell us a single phone, which is always a pleasant surprise when dealing with distributors. The single SNOM phone we received worked well, so we went ahead and purchased an additional 12 phones and the MP-108 gateway on the recommendation of our salesperson. Our experience with them has been very good, and the (uncompleted) RMA process of one defective unit has gone smoothly so far.

-Ernest W. Lessenger-
OACYS Technology


OACYS TECHNOLOGY is a 23-year company, founded in 1982 to develop and deploy computer solutions. Based in a semi-rural community, the company has long been accustomed to operating independently and developing self-reliant solutions with minimal external dependencies. We have developed our own solutions to address and manage substantial competition and adversity, and we continue to do so in the course of our own daily operations. For our consulting clients we explain step-by-step what we have done, why we have done it, and how to do what we did (or would do or avoid) to resolve any of the many challenges facing today's independent ISP/WISPS who are interested in winning their own battles and wars.

http://www.oacys.com/



Go back to Asterisk hardware recommendations

OACYS Technology

2/25/2004

We are using 12 phones in the office with one spare. All are SNOM 200 model phones (more about this later) that we purchased from ABP Technology (http://www.abptech.com/). We have a total of seven co lines, all analog (plus two DS3 for our Internet connection :) right now, that we connect using an Audiocodes MP-108FXO gateway (Also from ABP). The first four lines are part of a PacBell hunt group, while the next two are reserved for outgoing calls. The last line is a private incoming line to our Network Operations Center.

We have the MP108 configured to use the first six lines for outgoing calls, but to give priority to the last two, keeping our incoming lines clear. Incoming calls cause our receptionist phones to ring for 30 seconds, then play a message ("all our staff are busy...") and put the caller into an Asterisk Queue. We are eagerly awaiting the upcoming patch to announce queue position while in the queue. At any time callers can dial the extensions for Sales (1000), Customer Service (1001), Customer Service (1002) or Technical support (1003) all of which ring at several different phones. They can also dial direct extensions for any of our salespersons.

Some of our direct dial extensions are available to customers, others can only be used during a transfer from a receptionist. This protects our CEO and Senior Tech Support people from annoyances. Similarly, not all extensions will go to voicemail... Customer Service and Tech Support do not - they play a message to call back later or send an email.

The MP-108:

We were recommended this product by ABP and have been quite happy with it. The documentation is pretty weak, so I spent a couple of hours learning how the features work. The 4.0 build that came on the hardware is useless, so I immediately upgraded to 4.2 which solved all my problems and added some features I hadn't known I needed.

The 4.2 build supports polarity reversal, silence detection and current drop for disconnect detection. We use current drop here, which has the annoying tendency to take a couple of seconds to notice that someone has disconnected. I don't know whether this is an Asterisk thing, a telco thing, or a MP-108 thing, but I don't remember having any trouble with Digium's FXO card set to kewlstart.

Thanks to Clif Jones for a sample INI file and some advice in getting the MP-108 to work.

The SNOM 200 phones:

Audio quality and latency have not been an issue, which is to say that I've received no complaints. Connecting them to Asterisk is no trouble at all, though I'm having some problems getting them to stay registered (I think this is my fault, but I haven't had the time to debug it). I find the CallerID display to be very useful. And, once we got used to it, the conference calling feature became one of our favorite features.

The handsets are on the light side and the handset cords are just a little too short for large desks, but this is a problem that can happen with any phone, and they do use standard RJ-12 connectors. The headset and Speakerphone features are extremely high quality and well thought out - in general - and work as well as one could hope. You can't (quickly - and by this I mean in less than a second while the phone is ringing) change between headset and handset until AFTER you answer the call, but this is not a problem in most situations.

Our only major problem with the phones right now is with their surprise conference call feature. Here's how it works: If you have one call on the phone, you can press the transfer button to transfer, and hang up the handset to hang up. Great, just like a normal phone. Now the catch: If there is a call on hold and you want to hang up on the person you are talking to... DON'T HANG UP THE HANDSET. Doing so will conference the live call with the call on hold. OUCH! Similarly, don't use the transfer button, because this conferences the call on hold into the live call.

So far, I don't know of any way to do an unattended transfer while you have someone on hold. There are workarounds for most of this, but the sheer inconsistency of it is driving our office manager up the wall. Now, I may still have some settings wrong, but I think I've been pretty thorough. ABP is being very helpful in getting this figured out.

My solution: Asterisk has the ability to intercept the # key and use it as a transfer button. We've instructed our staff to a) never hang up by putting down the handset, and b) never use the SNOM's built-in transfer button. Instead, they press ESC to hang up and use the # key to initiate a transfer.

ABP also tried to sell us a Power-over-Ethernet device that would provide power to the SNOM phones during a blackout (in conjunction with a UPS of course). This is a problem that affects all VoIP phones, but our wiring is not PoE friendly. So, we went with the external power supplies for an extra charge. When we recommend/resell SNOM phones to our customers, we intend to sell the PoE system as well, wiring permitting.

All-in-all the office staff here is settling down to the new phones and there have been no show-stopping issues so far.

Asterisk:

Wonderful, it does everything I want. A few of the things it does:
  • Voicemail
  • Private and public extensions
  • Directory Service
  • Call Queues
  • Music on Hold (donated by our local High School jazz band)
  • Works with Cisco, SNOM, Pingtel, Budgetone, X-Lite and more
  • Automated attendent and IVR
  • A simple app that I wrote to play a message based on our network status (we are an ISP)
  • Fully customizable ring groups (i.e. ring all phones after hours, or only the receptionist during hours)
  • Conference Calling and Three-way-calling (enhanced by the SNOM phone's conference call feature)
  • Automatic Failover from PSTN to Internet and vice-versa
  • Support for multiple VoIP dialtone providers for low-cost long-distance
  • Transfer to cell phone

VoicePulse:

We use VoicePulse for our outgoing long-distance, and so far have not had any complaints. We were using NuFone, but are turned off by their lack of a web interface for refilling our account and viewing CDR. Both NuFone and VoicePulse have recently had (reported on the list, but not personally confirmed) outages that did not affect us. In both cases, setup and ordering were quick and easy (more so with VoicePulse).

ABP Technology Partners:

After an initial testing period with Asterisk and X-Ten, we purchased a single SNOM phone from ABP (I don't recall how we found them). They were very helpful and were willing to sell us a single phone, which is always a pleasant surprise when dealing with distributors. The single SNOM phone we received worked well, so we went ahead and purchased an additional 12 phones and the MP-108 gateway on the recommendation of our salesperson. Our experience with them has been very good, and the (uncompleted) RMA process of one defective unit has gone smoothly so far.

-Ernest W. Lessenger-
OACYS Technology


OACYS TECHNOLOGY is a 23-year company, founded in 1982 to develop and deploy computer solutions. Based in a semi-rural community, the company has long been accustomed to operating independently and developing self-reliant solutions with minimal external dependencies. We have developed our own solutions to address and manage substantial competition and adversity, and we continue to do so in the course of our own daily operations. For our consulting clients we explain step-by-step what we have done, why we have done it, and how to do what we did (or would do or avoid) to resolve any of the many challenges facing today's independent ISP/WISPS who are interested in winning their own battles and wars.

http://www.oacys.com/



Go back to Asterisk hardware recommendations

Created by: JustRumours, Last modification: Wed 25 of Feb, 2004 (20:04 UTC)
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