IP external paging with Asterisk
June 26th, 2008 | Published in Uncategorized | 6 Comments
In the spirit of exploring the capabilities of Asterisk beyond its typical use as an IP PBX, external paging is a great feature to increase the value of your Asterisk system by having it assume the functionality typically reserved for a distinctly separate system and additional investment. Most people configure Asterisk systems for paging via phones but when dedicated paging is required, and when the need to page must take priority over a voice call, then external speakers are the way to go. This is often the case for hospitals and schools.
Just the other day I had a meeting with a car service organization who needs a new IP PBX system. Unfortunately they had recently made a large investment in a brand new paging system and had to postpone their IP PBX investment. When I learned that this 50 seat organization spent $30K on a dedicated paging system I whacked the guy on the back of the head for stupidity and explained that they could have purchased a brand new IP PBX + Paging system for not much more.
So some of you might ask “what external paging speakers work with Asterisk?”. Well I’ve only found one company that builds a completely open IP/SIP paging speaker and that’s CyberData. They have a whole line of paging speakers including ceiling speakers, wall speakers, voice boxes, paging amplifiers, and more…. all of which come in POE and non-POE options. Although the hardware is a little more expensive than analog paging speakers (most range from $250-$350), the fact that they can be tied into an existing investment in Asterisk and that they operate over the same ethernet wiring already in place makes the TCO very compelling.
Other than CyberData, does anyone know of or have used other SIP paging speakers?
August 18th, 2008at 8:53 pm(#)
Hi, please send more info, prices and distribution of system Part# 010965
May 31st, 2009at 3:42 am(#)
[...] IP external paging with Asterisk OS VoIP Open Source VoIP Posted by root 2 hours 13 minutes ago (http://www.os-voip.com) June 26th 2008 published in uncategorized 1 comment they have a whole line of paging speakers including ceiling speakers wall speakers voice boxes powered by wordpress using the monochrome author theme by graph paper press Discuss | Bury | News | IP external paging with Asterisk OS VoIP Open Source VoIP [...]
April 6th, 2010at 9:20 am(#)
Impressive piece of information, let me elaborate more on VoIP. Voice over Internet Protocol has been around since many years. But due to lack of sufficient and affordable bandwidth it was not possible to carry carrier grade voice over Internet Protocol. But since the arrival of low cost internet bandwidth and new speech codecs such as G.729, G.723 which utilizes very low payload to carry carrier class voice it has recently been possible to leverage the true benefits of VoIP. G.723 codec utilizes only 6 Kbps (Kilo Bytes/sec) which is capable of maintaining a constant stream of data between peers and deliver carrier grade voice quality. Lets put this way if you have 8 Mbps internet connection, by using G.723 codec you can run upto 100 telephone lines with crystal clear and carrier grade voice quality. I am also a user of VoIP and have setup a small PBX at home. Since I have discovered VoIP I have never used traditional PSTN service.
Dear readers, if you have not yet tried VoIP I suggest that you try VoIP technology and I bet you will never want to use the traditional PSTN phone service ever again. VoIP has far more superior features to offer which traditional PSTN sadly cannot offer.
Also It has recently been possile to carry Video alongwith VoIP by using low payload video codecs. I cannot resist to tell you that by using T.38 passthrough and disabling VAD VoIP can carry FAX transmission, but beaware FAX T.38 passthrough will only work when using wide band protocols such as G.711, a-Law and u-Law.
By using ATA (Analog Telephone Adapter) which converts VoIP signals into traditional PSTN you can also using Dial-up modems to connect to various dialup services. I wont go in to the details what VoIP can offer, to cut my story short VoIP is a must to have product for every business and individual.
How VoIP Works
When we make a VoIP call, a communication channel is established between caller and called party over IP (Internet Protocol) which runs on top of computer data networks. A telephony conversation that takes place over VoIP are converted into binary data packets streams in real time and transmitted over data network, when these data packets arrive at the destination these are again converted into standard telephony conversation. This whole process of voice conversion into data, transmission and data conversion into back voice conversation takes place within less than few milliseconds. That is how a VoIP is call is transmitted over data networks. I hope that now you understand basics of how a VoIP call takes place.
What are speech codec’s and what role codec plays in VoIP?
Speech codec play a vital role in VoIP and codec determines the quality and cost of the call. Let me explain you what exactly VoIP codec’s are and how they work. You may have heard about data compression, or probably you have heard about air compressor which compresses a volume of air in enclosed container, VoIP codec’s are no different than a air compressor. Speech codec’s compresses voice into data packets and decompresses it upon arrival at destination. Some VoIP codec’s can compress huge amount of voice while maintaining QoS which means use this type of codec will cost less because it will consume just a fraction of data network. Some codec’s are just not capable of encoding huge amount of voice they simply consume huge amount of data networks bandwidth hence the cost goes up.
Following is a list of VoIP codec’s along with how much data network bandwidth they consume.
* AMR Codec
* BroadVoice Codec 16Kbps narrowband, and 32Kbps wideband
* GIPS Family – 13.3 Kbps and up
* GSM – 13 Kbps (full rate), 20ms frame size
* iLBC – 15Kbps,20ms frame size: 13.3 Kbps, 30ms frame size
* ITU G.711 – 64 Kbps, sample-based Also known as alaw/ulaw
* ITU G.722 – 48/56/64 Kbps ADPCM 7Khz audio bandwidth
* ITU G.722.1 – 24/32 Kbps 7Khz audio bandwidth (based on Polycom’s SIREN codec)
* ITU G.722.1C – 32 Kbps, a Polycom extension, 14Khz audio bandwidth
* ITU G.722.2 – 6.6Kbps to 23.85Kbps. Also known as AMR-WB. CELP 7Khz audio bandwidth
* ITU G.723.1 – 5.3/6.3 Kbps, 30ms frame size
* ITU G.726 – 16/24/32/40 Kbps
* ITU G.728 – 16 Kbps
* ITU G.729 – 8 Kbps, 10ms frame size
* Speex – 2.15 to 44.2 Kbps
* LPC10 – 2.5 Kbps
* DoD CELP – 4.8 Kbps
Switch to VoIP Today and you will never want to use traditional PSTN ever again.
Thanks
-Imran
July 14th, 2010at 12:23 pm(#)
I think this post was actually a great beginning to a potential series of blog posts about this topic. A lot of people act like they know what they’re preaching about when it comes to this area and really, very few people actually get it. You seem to understand it though, so I think you ought to take it and run. Thanks a lot!
August 21st, 2010at 1:11 pm(#)
Bose seems to make the best wall speakers on the market today”‘;
October 6th, 2010at 12:09 pm(#)
wall speakers that are manufactured by Bose are superb*`;