Categories
Peripherals Products

The Algo 8180 SIP Audio Alerter

The 8180 SIP Audio Alerter is a loud ringing, paging and intercom device for use with SIP telephone systems. It has two main uses. The first is as a loud ringer for use in places such as warehouses. The second use is as a paging and intercom system. It can operate in both these modes at the same time by appearing as two different extensions on your phone system at the same time.

When registered with a SIP server, one endpoint will play an audio file from internal memory upon ring detection. The second endpoint will auto-answer for voice paging or full intercom (two way audio).

Equipped with a high efficiency integrated amplifier and tuned high quality loudspeaker, the 8180 is typically eight times louder than a telephone speaker. Several audio files are pre-loaded into the 8180 internal memory for ring sounds but users may also record or upload custom audio files, music, sound effects, or voice announcements.

Ambient noise monitoring

The advanced features of the 8180 include SoundSureTM technology which automatically adjusts loud ring and loudspeaker volume to compensate for background ambient noise. Ideal for variable noise environments (restaurants, workshops, classrooms, etc.), SoundSureTM ensures that ringing or paging is always heard but not unnecessarily loud.

Applications / Usage scenario

  • Loud Ringer in noisy or variable noise environments (classroom, restaurant, machine shop)
  • Voice Paging (warehouse, workshop)
  • Outdoor ringing or paging
  • Multi-cast wide area notification

Easy to Install

  • Network managed SIP endpoint
  • Configuration is possible using the feature buttons or web interface
  • PoE eliminates local power supply

Features

  • Voice Paging with talkback capability
  • High efficiency and high output wideband speaker
  • Multicast receive or broadcast capability
  • Outputs for external speaker, slave amplifier, or visual alerter
  • Dual purpose loud ringing and/or talkback voice paging
  • Significantly louder (eight to twenty times) than typical telephones
  • Low frequency tones outperform traditional shrill electronic ringers
  • SoundSureTM ambient noise compensation adjusts output for noise level
  • Multicasting capability for wide area notification

Customisable

  • Pre-loaded with several ring tones including bell, chime, gong, buzzer, warble, and dogs
  • Supports custom uploaded WAV files or recorded messages
  • Selectable/customisable alert tones or announcements

Please contact for pricing and avalibility

Categories
Case Studies Technical

Man In The Middle

Man In The Middle Server, What does he mean I hear you ask?

Well its the best term for it, as it describes what it is and does. Basicly we have an AsteriskServer sitting in between the incoming lines and the main  PBX. Idealy We would have liked to replace the existing PBX but in some cases this just isnt possible and we have to accept this.

So why may we need to do this, Well we have used this solution to allow customers to have follow the sun  support call centers.  Calls arrive to a dedicated DDI number that is for support in that country and depending on the time of day it is routed to the call center that is open in another part of the world over a voip network. This is done by Asterisk checking the called number of all calls and if a match is made the call is passed off to Asterisk to handle the call in its dialplan, all other unmatched calls are passed on to the main system.

We have also used this method of connection to play prompts to callers before the call is sent to the main system. Calls in this case can be identified by CallerID name or number. So for example calls from the BT operator, International payphones or even just a certain CallerID Number are played a specific message and then either forwarded on or passed on to a IVR (Interactive Voice Response menu) to be handled in a specific manner.

A another often used reason for this type of connection is when migrating from a legacy pbx to and Asterisk server. The line connections are the same and the call flow is controlled by the dialplan, Routing calls on teh Asterisk system to its dialplan and routing calls for extensions still on the old system to it.

Once all extensions are migrated to the new system the old system can be turned off and removed with no interuption to the users on the new system.

Categories
Case Studies

Asterisk install in a campus style school

In late 2005 We we approached to replace an aging Panasonic system for a British Public School.
The driving force for moving to voip was that the site was spread over a wide area and different buildings and to provide telephones to the remote buildings would prove too expensive.
The system was replaced with a central Asterisk server with nearly 80 extensions. The core LAN was upgraded to Netgear Layer 3 switches with Powerdsine POE midspans.
The remote buildings added extra complexity as one was on the other side of a public road.

To overcome this, this building was connected to the main site via a “Point to Point” Wifi link. The other building was closer and could be connected via a Fibre Link between the it and the main building.

Campus Site

The system was configured to use account codes in public areas, These handsets can only make emergency or internal calls unless a validated account code is entered.
The handsets used were a mix of Aastra 480i and 9133i because of build quality, reliability and BLF support.

The system has now been in place for over 2 and a half years now and in that time the only faults have been either ISDN failing or cable faults. This is a great demonstration of the reliabilty of Asterisk and Aastra handsets and voip in general.

UPDATE

We have since writing this post updated the system on new hardware as part of a refresh. They are now running on a newer version of Asterisk and have more buildings connected to the network.

Categories
Case Studies

Solicitors Group

We were approached to provide a Voip solution for a London Solicitors. They were moving from a Single office to two offices in different parts of the city.

At the Primary office we installed an Asterisk server that was connected to the ISDN30e link where all DDI numbers were delivered for both sites. This site also had its own VOip connection as well

At the main site the operators were based using the Asternic FlashOsPanel for displaying extension status for both sites.

A key feature of the system was to provide a flexible system for out of hours callers to contact the duty Solicitor. This was provided so that callers on calling out of hours  can leave a message for the next day or press an option to contact the Duty solicitor. The Duty solicitor can call into the system and change the contact number at will if for example they are busy with a client.

A SIP trunk connected both sites together over a 20Meg link that was used for both voice and data.

The new sites were not in the exchange area of the original phone number so this was ported to Gradwell.net so that it could be delivered via an IAX2 connection to the relevent server.

The handsets used were a mixture of Aastra 53i and 55i with an xml appliction to set DND status in the Flash ops panel and light the light on the handset.