Asterisk PBX

Friday, December 31, 2010

Asterisk embedded systems

History
Backlinks...AsteriskAsterisk tips and tricksAsterisk hardware recommend...Asterisk setup minimumAsterisk Install CDROMRolePlayingTwoPhonesTwoRooms
There is a lot of interest in running Asterisk on "embedded" plattforms like the Linksys NSLU2. 2010-11-01 Nicherons: ipPbx08 & SPoGSM now support BLF (Busy Lamp Field) to show on a phone who is talking on other phones 2010-07-17 Nicherons: ipPbx08 & SPoGSM are the latest development of embedded Asterisk ip PBX phone systems 2010-06-03 - DM Link releases their first IP PBX. 2010-04-15 http://code.google.com/p/omapterisk/.Omapterisk Asterisk port for OMAP ,that could run in Beagleboard or Hawkboard. 2009-12-21-http://www.plugpbx.org/ Project Starts. Run Asterisk and FreePBX on a SD card on the SheevaPlug Embedded Arm computer. Flash and go! Open source! 2009-12-09-IP01 has the power to run 32 concurrent calls and size of an ATA at $135. 2009-08-14-Xenia Labs Sucessfully test drives the Positron Telecom embedded Asterisk PCI card. 2009-06-16-Uni-Ta New firmware of embedded Asterisk system S800 based on Blackfin is available. 2009-06-10-FIVN releases simple Asterisk GUI version 1.6 for embedded systems. 2009-01-14-IP BRI Atcom's latest PBX with four BRI TE (line) interfaces. Powerful Blackfin BF537 600Mhz DSP provide all the power needed for a production PBX. 2009-01-14-BR4-APPLIANCE (IPBRI) , Astfin powered quad BRI Asterisk appliance. 2008-12-30 uCpbx releases PR1-Appliance based on Blackfin / Astfin. More 2008-10-28 OpenVox IX100 IPPBX kit, the award winning product, is designed for small and embedded asterisk systems and it's the first one that combines analog mini pci card with ISDN bri card in a little box. More 2008-05-21 The Asterisk Orgasmatron: A $199 Turnkey PBX Install in Under 15 Minutes Part I Part II 2008-05-08- ATCOMrealeases first 8 ports asterisk embedded IPPBX : IP08 2008-04-15-Soundwin releases modular 2/4 ports Asterisk embedded IPPBX: IPX200/400 on Blackfin. BRI Appliance (April 2008), New improved version of embedded Asterisk on a Blackfin processor with a Quad BRI interface. All hardware designs open. 2008-02-28 - Bicom Systems launches officeBOX, solid state, fan less IP-PBX appliance Older News http://www.plugpbx.org Download pre-built SD card images to run FreePBX/Asterisk based on Debian Squeeze via the ARM SheevaPlug embedded computer. Ease of use core focus and building a collaborative community. Come run FreePBX on a 3 watt computer with plenty of power that's entirely open and easy to hack. $99 USD device. http://code.google.com/p/omapterisk/.Prebuilt Root FS for Omapterisk -Asterisk port for OMAP ,that could run in Beagleboard or Hawkboard. Uni-Ta embedded Asterisk IP PBX S800 with commercialized GUI, based on Blackfin. 1~8 analog ports. VoIPtel: On CeBIT'08 we launched VoIPtelGUI for Atcom's IP04, now we are responsible for the VoIPtel CE and SE Firmware for their entire line of PBX's. Don't forget to visit our Forum and Blog. PBX in a Flash - We've tried to collect in one place everything you will need to create your own PBX in a Flash system in under an hour! ATCOM:Asterisk based on Blackfin chipset, uClinux: IP04 IP08 AstBoxLinux small linux with "Bristuffed" Asterisk, Zaptel, Libpri, zaphfc and development tools for CF, USB sticks and Flash disks Astfin Asterisk for Blackfin based devices, uClinux, support for open source hardware (fxo, fxs, bri, pri) AstLinux Customized version of Linux designed to run from compact flash or small Hard drive. Magiclink Asterisk Appliance: Blackfin embedded Asterisk hardware with 8 ports FXS/FXO released. VoIPBox: For WRAP, Live CD, PXE Image, with web GUI in German and English. Planned: USB stick, Via EPIA Asterisk hardware Soekris running Asterisk on Soekris hardware. Asterisk hardware Gumstix Running Asterisk on Gumstix (World's Smallest VoIP PBX) MetrixCommunication:Company that offers a complete soekris/asterisk kit, and offers customer support for configuration as well Asterisk fli4l: OPT package for the router-on-a-diskette fli4 Limey Linux: A custom CF Asterisk/Linux Distribution. Fits on a 128MB or 512MB compact flash card. Optimized for VIA Mini-ITX boards with 512MB of DDR RAM. Recently upgraded to use the 2.6 kernel and Asterisk 1.4. Allows Asterisk, Libpri, and Zaptel to be compiled on the target system. Pebble Linux: CF Linux distribribution - no longer supported. It has been superseded by Pyramid Linux. Phone systems - A comprehensive guide to phone systems Pyramid Linux: Linux distribution focused on embedded wireless networking applications Embedded Asterisk: Step by step guide for installing an embedded * system using Debian, CF, i586 Build your own Asterisk hardware based on the Free Telephony Project (Aug 2006) IP04 Open Hardware IP-PBX: An open (free as in speech) four port IP-PBX design that can be built for around $100 and will retail for around $400. Asterisk Management GUI for OpenWrt, Unslung and DD-WRT firmwares AskoziaPBX, ~10MB image, * 1.4, FreeBSD 2.6, derived from monowall, images for soekris, wrap, standard pc, vmware HP Thin Client Run Asterisk and/or OpenSER on a solid state HP Thin Client using a standard Debian install.
Go back to Asterisk

Asterisk Voicemail

Password protected Separate away and unavailable greetings Default or custom greetings Multiple mail folders Web interface for checking of voicemail E-mail notification of voicemail with audio file attachment Voicemail forwarding Visual message waiting indicator (MWI) Message waiting stutter dialtone
The number of messages for each voicemailbox is limited to 99 messages. Starting with Asterisk 0.7.1 the system now plays a prompt warning the user that the mailbox is full and cannot record any more messages. New features (as of April 28, 2004)
Option to have the CID of the caller heard before the voicemail Option to allow an operator to be reached after leaving a voicemail Option to review, rerecord, or save voicemails after leaving them Option to review, rerecord, or save busy, unavailable, and name prompts. Option to allow dialing out from within voicemail Option to allow calling back of the person who left voicemail (with the correctly formatted contexts in extensions.conf) Group voicemail (voicemail broadcast)
The context defined in the voicemail.conf corresponds to entries in the channel configuration file, i.e. zapata.conf and sip.conf which require the context to be appended to the mailbox when using VoiceMail2.

Configuration lines for a voicemailbox in voicemail.conf:
[YourVoicemailContext]
210 => 5555,John Smith,jsmith@yourdomain.com

Line in zapata.conf or sip.conf:
mailbox=210@YourVoicemailContext

The location of saved messages also changes with the context:
/var/spool/asterisk/voicemail/YourVoicemailContext/210/INBOX

When listening to a message you'll hear when it was recorded. Voicemails are saved on the server in all formats specified in voicemail.conf, but only the first format specified is sent as an attachment to the email. The /contrib/scripts directory of /usr/src/asterisk contains a Perl script that can be applied to regularly expire too old voicemail messages
You can follow this article Asterisk - Setup voicemail to send email with mp3 attachment to get a detailed explaination how to setup Asterisk to send voicemail by email with mp3 audio attachment. This method uses only bash script and usual linux tools.

note: my is /var/lib, some peoples will be /usr/share.
1. ensure "attachfmt=wav" and "format=wav|gsm"
2. Put the below script in: //asterisk/agi-bin/mp3vm.pl
3. chmod a+rx mp3vm.pl
4. modify mailcmd=perl //asterisk/agi-bin/mp3vm.pl
5. install lame if you don't already have it
6. check the path to perl, base64, dos2unix and lame and modify the script and mailcmd as needed. (If you are getting 0kb files, your path to lame is likely wrong.)

#!/usr/bin/perlopen(VOICEMAIL,"|/usr/sbin/sendmail -t");open(LAMEDEC,"|/usr/bin/dos2unix|/usr/bin/base64 -di|/usr/local/bin/lame --quiet --preset voice - /var/spool/asterisk/tmp/vmout.$$.mp3");open(VM,">/var/spool/asterisk/tmp/vmout.debug.txt"); my $inaudio = 0;loop: while(<>){ if(/^\.$/){ last loop; } if(/^Content-Type: audio\/x-wav/i){ $inaudio = 1; } if($inaudio){ while(s/^(Content-.*)wav(.*)$/$1mp3$2/gi){} if(/^\n$/){ iloop: while(<>){ print LAMEDEC $_; if(/^\n$/){ last iloop; } } close(LAMEDEC); print VOICEMAIL "\n"; print VM "\n"; open(B64,"/usr/bin/base64 /var/spool/asterisk/tmp/vmout.$$.mp3|"); while(){ print VOICEMAIL $_; print VM $_; } close(B64); print VOICEMAIL "\n"; print VM "\n"; $inaudio = 0; } } print VOICEMAIL $_; print VM $_;}print VOICEMAIL "\.";print VM "\.";close(VOICEMAIL);close(VM);#CLEAN UP THE TEMP FILES CREATED#This has to be done in a separate cron type job#because unlinking at the end of this script is too fast,#the message has not even gotten piped to send mail yet

Yeastar IP PBX

Yeastar specializes in the developing and manufacturing IP-PBX products and is committed to the distribution of new generation technology products in the field of enterprises' communications. With professional and high quality team, Yeastar designs products to worldwide applications and maintains the long-term stability of products to greatly benefit users. Yeastar welcomes the cooperation from various kinds of companies and will sincerely treat them to create the multi-win situation together.

Our Products:

MyPBX is a standalone embedded hybrid PBX for small businesses and remote branch offices of larger organizations (2-100 users per site). MyPBX also offers a hybrid solution (a combination of VoIP applications using legacy telecom equipment) alternative for enterprises who are not yet ready to migrate to a complete VoIP solution. Users: 100
Concurrent Calls: 22
Voicemail: 4000min

Interface:
   Up to 16 Analog Ports (FXO/FXS)
   Up to 4 GSM Ports(Quad-Band GSM/GPRS850/900/1800/1900MHz)
   Up to 8 BRI Ports
Flash: 512 MB Onboard Flash
RAM: 128 MB Onboard RAM
LAN: 1 (10/100MBps)
WAN: 1 (10/100MBps)

Size: 280x175x33 mm
Weight: 700g
Power Supply: AC 100~240V/50~60Hz(DC 12V, 5A)

Protocol: SIP(RFC3261), IAX2
Codec: G.711 A/u-law, G.726, G.729A, GSM, Speex, H261, H263,H263p, H264 ,MPEG4.
LED: Red for FXO/GSM, Orange for BRI, Green for FXS
Network: PPPoE, DHCP Client, Static IP DHCP Server, VLAN
Multiple Languages Support: Chinese, English, Hebrew, Portuguese, Russian,Spanish

Auto-provision
Blind Transfer
BLF Support
Blacklist
CDR(Call Detail Records)
Call Forward
Call Parking
Call Recording
Call Pickup
Call Routing
Call Transfer
Call Waiting
Caller ID
Conference
Define Office Time
DISA(Direct Inward System Access)
DIDs
Distinctive Ringtone
DND(Do Not Disturb)
Firewall
Follow me
IVR (Interactive Voice Response)
Intercom/Zone Intercom
Music On Hold
Music On Transfer
Paging/Zone Paging
PPPoE
Queue
Ring Group
Route by Caller ID
Skype Integration (Skype Connect)
SRTP
Three-way Calling
TR069
FAX (T.38)
Voicemail
Voicemail to email
Voicemail forwarding
Web based Control Panel
Yeastar - BizPBX is Windows-based software PBX solution, which is the world first communication system that realizes the free connection among three different networks Skype, PSTN and SIP. Yeastar - BizPBX business class system not only includes complete and rich functions of PBX, but also contains Instant Messaging and E-Mail system for company internal communication. The simplicity and easy-to-use of this Yeastar - BizPBX solution makes it especially suitable for small and medium sized company.
1. Support three kinds of trunks,PSTN,SIP and Skype.
2. Support three kinds of extensions,Analog Phone,SIP Phone, Softphone.
3. Yeastar - BizPBX Client software can act as Softphone.
4. Yeastar - BizPBX Client software accessible for every extension.
5. Contacts Listing.Internal Members Listing and displays status of every member.
6. Contacts Listing is sharable among members.
7. Internal E-Mail & Internal Instant Messenger.
8. Voice Mail:Let caller leaves voice message when you are busy or away.
9. Remote Office: Work Yeastar - BizPBX Client software as an extension when working in remote office.
10.Caller ID Profile:Show caller's information in real-time.
11.Click to Call:Click on 'name' to call and save precious time on dialing.
12.Conference Call: Both sides are able to invite others into conference.
13.Free Interoffice Trunking: Make free calls among multiple branches through the interoffice trunking.
14.Web Call: Receive concurrent calls from website with One Skype ID.
15.Virtual Office: Apply local numbers in different countries to reach Yeastar - BizPBX. Authentication
Automated Attendant
Blind Transfer
Call Booking
Call Detail Records
Call Forward
Call Monitoring
Call Parking
Call Queuing
Call Recording
Call Restriction
Call Retrieval
Call Routing
Call Transfer
Call Waiting
Call Remind
CallerID
Database Integration
Dial by Name
Direct Inward System Access(DISA)
Direct Line
Do Not Disturb
Extension Group
Interactive Voice Response(IVR)
Local and Remote Call Agents
Music On Hold
Protocol Conversion
Remote Call Pickup
Remote Office Support
Roaming Extensions
Share Contacts BizFAX allows consumers to save greatly on resources such as paper, electricity, time and money.
1. Cost reductions
BizFAX delivers a significant reduction in fax and document delivery costs by eliminating the need for paper-based delivery, stand-alone fax machines and associated maintenance, supply, equipment and labor costs.
2. Time savings
BizFAX significantly reduces the time it takes your employees to exchange documents with customers or suppliersâ??resulting in faster sales cycles, quicker customer service and improved efficiency.
3. Document security
BizFAX offers electronic document delivery solution that helps you comply with regulations and safeguard information security. Faxes are securely and immediately delivered to your client. Privacy is ensured.
4. Reliability & Mobility
BizFAX runs on embedded OS and is a trusted and reliable fax server. No more paper jams, lost pages or toner mishaps on office environment. And itâ??s also a mobile fax machine, send and receive faxes even outside your office.
5. Full series
BizFAX family consist of 6 models that are E100, E200, S200, S400, M800 and M3000(E1), in which the smallest one supports 1 line only and the biggest one supports 30 lines to receive and send faxes simultaneously. BizFAX can easily meet the unique needs of different sizes of companies. 1. Install BizFAX Client on a computer, and then you own a Virtual Fax Machine.
2. The FXS port is to connect a physical fax machine for scan, print or fax directly.
3. Compatible with PSTN (FXO) .
4. Auto convert documents as doc, xls and pdf to fax format tiff by BizFAX Printer
5. Auto create a single fax document with multiple pages from multiple documents
6. Send a fax to multiple numbers at the same time
7. Customizable fax page header
8. User is able to make annotation, signature and stamp through BizFAX Editor
9. If the other party manually receives a fax, user can make a call to talk with the other party at first by virtual fax extension and then send fax after the call.
10. FAX to Email.
11. Can save up quantity of fax documents. E serial memory is about 500M ( about 14000 pages) and expandable by Net Disk; S serial and M serial takes with infinite memory.
12. Virtual fax extensions can make calls; send emails and instant messages to each other.

Digium TDM410P Analog Interface Card Overview

The Digium® TDM410 is a half-length PCI 2.2-compliant modular gateway card for connecting analog telephone stations and analog POTS lines through a PC. It supports a combination of up to four station or trunk modules for a total of 4 lines.

Using Digium’s Asterisk® software and standard PC hardware, one can create a telephony environment that includes all of the sophisticated features of a high-end business telephone system.

Using an industry-standard bursting, bus-mastering interface chip that is found within millions of PC systems worldwide, and Digium VoiceBus™ technology, the TDM410 eliminates the requirement for external gateways, with industry-leading performance and price. The trunk and station modules are interchangeable, allowing the creation of any combination of interfaces. The optional hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior voice quality on both trunk and station interfaces. Scaling of this solution is accomplished by adding additional TDM410 or other Digium analog interface cards.

Digium Analog cards, with there interchangeable single and quad FXS and FXO modules can eliminate the requirement for seperate channel banks or access gateways. Digiums commercial, toll-free quality High Performance Echo Cancellation (HPEC) software is available to our analog customers at no additional cost. The optional VPMADT032 hardware echo cancellation module provides the same toll-quality HPEC but without the perfoamce inpact of a software based solution. Scaling of an analog card solution is accomplished by adding additional cards and FXO / FXS modules.

Core Features

4 Ports for connecting analog telephones or POTS linesHalf-length Analog CardUp to 4 FXS or FXO ModulesHigh Performance Echo Cancellation (HPEC) Software (Optional)TDM410 for use with a PCI 2.2 compliant slot

Valcom PagePro VIP204 Paging Gateway Overview

The PagePro IP Model VIP-204 is a SIP based paging server designed for use with virtually all IP based telephone systems. The unit provides 8 zones, groups thereof and all call paging. Up to 4 simultaneous non conflicting pages. This PagePro IP unit can drive up to 65,000 IP speakers or 65,000 analog speakers or provide the audio source for legacy paging systems. It can simultaneously drive any combination of these paging applications. The VIP-204 can form the foundation of most paging projects. The 8 zones can be accessed individually or in flexible groups. Multiple VIP-204 devices can be grouped together for expandability. Provides feedback elimination.

Valcom PagePro IP VIP-204 allows paging from any SIP compliant telephone to 8 SIP page destinations. The SIP page destinations can be any of the VIP-204’s four analog outputs, any combination of those outputs, or any
combination of page groups from other Valcom VIP units. The four analog outputs on the VIP-204 are line level –10dBm audio outputs. These outputs are designed to directly drive up to 150 Valcom self-amplified speakers.
The outputs are also suitable for connection to an auxiliary input on a utility amplifier of a legacy paging system.

Group Paging Features
The PagePro IP Model VIP-204 allows SIP based phone systems to page any number of Valcom IP speakers, Valcom Amplified Speakers and Valcom VIP series analog paging gateways. Valcom multicast group paging features robust grouping and priority controls. These controls make it easy to distribute low priority announcements yet allow high priority paging to override ongoing announcements. The PagePro IP Model VIP-204 provides 8 IP zones and 1 analog output capable of driving a mix of Valcom self amplified overhead paging speakers and horns. You can assign one SIP access phone number for the analog port. The VIP-204 provides 8 fully customizable SIP identities that can access any combination of IP zones and the analog port. Analog paging functions can be seamlessly combined IP paging functions allowing powerful combinations of analog and IP paging.Group Paging Priority Feature
Any page group or port can be assigned a priority. These priorities allow a logical scaled approach to group paging. An absolute priority can be assigned to allow for emergency override in any circumstance. Group paging priorities work locally or across ganged units in the same fashion allowing facility wide emergency paging. Red Phone functionality is easily achieved with either IP phone or analog handsets via a Valcom gateway.Feedback Elimination Feature
The PagePro IP Model VIP-204 provides programmable store and forward capability for all paging accesses. This features records up to 30 seconds of audio, and then replays the audio when the page call is terminated. The number of times the recorded audio is played back can be set from 1 to 5. Some paging applications require live announcements. For these applications the store and forward capability can be turned on/off per page zone.Wiring Supervision Features
The PagePro IP Model VIP-204 provides for supervision of field wiring. If a fault is detected in the wiring connecting the analog output to either speakers or amplifiers it can be reported to an external device or syslog facility.Audio Supervision Features
The PagePro IP Model VIP-204 provides internal monitoring of the audio path from the network connection to the analog output connector and can report failures to an external device or syslog facility.Legacy Applications
The PagePro IP Model VIP-204 provides 1 analog output that can easily be conditioned to drive any existing overhead paging system. Legacy paging features can be seamlessly combined allowing powerful combinations of legacy analog and IP paging.Easy To InstallLive or Recorded Messages65,000 Speakers, Expandable To 256 IP Paging ZonesEasily Connect Loudspeaker System To Existing Computer NetworksCompatible With Cisco, Nortel, Avaya, Iwatsu And Most IP Telephone SystemsSupports IP And Analog Amplified SpeakersSupports IP Talkback SpeakersSupports Legacy Paging SystemsSynchronized Clock CompatibilityPoE (802.3af) Or Local PowerSIP And MulticastNetwork SupervisionFeedback EliminatorNight RingDHCP

Vendor of Asterisk VoIP application Digium prepares for ITEXPO

Company’s Digium CEO Danny Windham has been added to the list and schedule of Keynote Presenters at ITEXPO East, February 2-4, 2011 at the Miami Beach Convention Center, Miami, Florida.

Windham's presentation is planned to be provided on February 2, 2011 at 3:30 PM, addressing how Asterisk continues to grow up as a telephony platform, performing extensive expansion in business, government, and network carrier markets. The session will be open to all ITEXPO attendees, appearing also to be a core element of so called ‘’Digium Asterisk World ‘’ exhibition in frames of ITEXPO. "I'm excited to return to Miami for ITEXPO and Digium|Asterisk World, as these three days will provide an unprecedented training opportunity for anyone interested in learning more about open source telephony," said Danny Windham. "Asterisk is the world's most widely used open source telephony software, and I'm looking forward to spending time with our growing community of developers, resellers, and enterprise executives who are basing their solutions on Asterisk." Digium is the producer and primary operational sponsor of Asterisk. The product lines of Digium include a broad spectrum of firmware and software that provides capabilities for resellers and entrepreneurs to design their requirements tailored VoIP systems. The main responsibility of Danny Windham is setting the Digium’s corporate strategy, and ensuring efficient daily business operations. Before joining the Digium, Windham worked as president and chief operating officer of company ADTRAN, a worldwide provider of networking and telecommunications equipment. He started his ADTRAN career in 1989, when ADTRAN successfully absorbed the Processing Telecom Technologies, a company Windham co-founded in 1986. Prior to becoming Digium’s CEO in 2005, Windham served as the senior vice-president and general manager of the Enterprise Networks Division. Windham possesses the Bachelor of Science degree in Electrical Engineering from Mississippi State University, where he was awarded as Distinguished Engineering Fellow in 2001. He also holds an MBA from the Florida Institute of Technology. ITEXPO has initiated the registration process; its collocations will include: 4GWE Conference; Smart Grid Summit; M2M Evolution Conference; Cloud Communications Summit; Social CRM Expo; SIP Tutorial; VIPeering; Business Video Expo; Digium|Asterisk World; ChannelVision Expo (CVx); StartupCamp; and Super WiFi Summit.

Xorcom Astribank Interface Overview

Astribank is a versatile and powerful channel bank that was specifically designed for the Asterisk IP-PBX. Astribank supports all the common telephony lines and trunks: FXS, FXO, BRI, PRI and R2. The Astribank driver is a part of the standard Asterisk distribution. With Xorcom Rapid Live CD™, a fully operational Asterisk server with preconfigured FXS, FXO, BRI, PRI and R2 ports can be up and working in less than 3 minutes! The USB 2.0 interface supplied with the Asteribank server guarantees a reliable, extremely high bandwidth, hotpluggable solution that is not dependent on a PRI PCI card. The USB 2.0 plug-and-play connection can support, theoretically, over 3,000 concurrent calls on a single USB 2.0 port. The reliable, industry-standard USB connection eliminates the costly, cumbersome PRI cards that are needed for standard channel bank support, and adds unique features such as reliable fax support, transparent synchronization between Astribank units, an auxiliary system interface, and more.

Astribank Chassis
Each 19” 1U rack-mountable Astribank chassis accommodates up to four modules. Multiple Astribank units may be connected to a single Asterisk IP-PBX using different USB 2.0 ports, a USB 2.0 hub or a USB 2.0 PCI card.

Telecom Connector Option (TCO)
This option is available in units with 8 or more FXS ports. To streamline installation of multi-port systems, the TCO unites all FXS ports on the Astribank unit into a single telecom connector on the real panel. All front panel indicators and RJ11 connectors remain active and enable easy monitoring and maintenance. A cable for connecting the telecom connector to the patch-panel is included.

Input/Output (I/O) Ports
Analog units with FXS modules support I/O ports. The two output ports enable activation/deactivation of peripheral devices by dialing an extension number. A popular example of output port usage is to open a door by dialing an extension. The four input ports create an off-hook event in Asterisk. The dial plan can be configured to use this function to dial an extension, play a message, send an email, activate another device using the output ports, etc. Examples of possible input port usage include forwarding a call to specific numbers, and playing a pre-recorded message if the fire/burglar alarm goes off.