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.


Core Features

USB channel bank for AsteriskSupports FXS, FXO, BRI, PRI and R2Native Asterisk – no extra software or hardware requiredInput/Output ports for connecting peripheral devicesFrom eight ports to hundreds of portsTelecom connection optionPower drive for ISDN phones

Wednesday, December 22, 2010

Christian ‘’Food for the Hungry’’ selects 8x8 hosted VoIP services

Business VoIP communications solutions provider Company 8x8 Inc. has announced that one of Non Governmental Organization (NGO) Christian relief and development organization Food for the Hungry has deployed its Virtual Office VoIP phone service to ensure communications among all local, regional and international detachments.

As per Company’s officials, to maintain domestic and overseas staff and volunteers daily connectivity, the charity organization chose the 8x8 VoIP Virtual Office phone application, thus providing all necessary global coverage, versatility and saved the organization capital and ongoing operational expense. Nowadays charity organization subscribes more than 150 Virtual Office Unlimited extensions along with 8x8 virtual and toll free numbers and fax extensions. Field offices in the UK, Guatemala, Bolivia and Haiti, as well as individual volunteers operating from distant locations, are connected over 8x8's VoIP provided network, making them to call one another by dialing an extension, in this way preventing any international fees imposed. Non profit ‘’Food for the Hungry’’ is vigorously helping disaster victims Worldwide and currently training volunteers to fight the cholera epidemic that rampaging in Haiti. With all this relief work, they need some reliable communications applications, so they've employed 8x8's hosted VoIP solution to execute the duties. The 8x8 Virtual Office phone service fits the bill for the international non-profit as it needs to connect its workers both domestically and all over the world. With a low entrance fee due to nature of hosted VoIP, the price is right for a relief organization that would rather spend their dollars of helping people in need.

New mobile video calling technologies from Fring

Company fring announced the release of network-optimized dynamic video quality (DVQ) mobile video calling technology. As per company’s officials, this technology adjusts video bit rate and frame rate as being required by the specific device to match particular network bandwidth while a call is in progress. It’s targeted to provide the best possible video picture quality for currently available bandwidth between call ends.

As it is reported by ‘’fring’’, once the network capacity is becoming constrained, its DVQ technology dynamically adjust the video picture quality, instead of losing the video call, without any distortions or compromising of VoIP based audio quality. Moreover, by means of the integrated DVQ indicator, users are informed about changes in their network strength in real time. As the fring’s strategy describes, mobile ecosystem resources are the pivot of company’s product design and implementation: DVQ technology adapts to match bandwidth at every single moment of every video call. This versatile nature enables the changing of call parameters and network availability which results into more connected calls and fewer dropped calls. It is of huge importance, when users enjoy impeccable video calling according to network congestion and in line with carriers’ bandwidth availability – at any traffic hours. DVQ technology is compatible with all mobile data bearers: 3G, 4G, WiMAX and Wi-Fi. It’s available from the iPhone App store and the Android market. “In the year since pioneering mobile video calls, we’ve seen that users network conditions change dramatically during and between video calls. That’s the nature of mobile experiences in heterogeneous networks," said Alex Nerst, co-founder and CTO of fring. "DVQ lets users make the best use of the peer-to-peer network capacity available at any moment during a video call, regardless of if they're stepping into an elevator, commuting on a train or simply walking away from their Wi-Fi hub."

Promotion of VoIP Business Telephone Solutions by Toshiba America Information Systems

As it is described in the last press release, Toshiba America Information Systems Inc., Telecommunication System Division, and LEAF Financial Corporation (LEAF), the commercial finance subsidiary of Resource America Inc. have initiated a vendor finance program agreement to foster the customer financing and leasing requirements of the Authorized Toshiba Dealer Network.

The Program will be named as Toshiba Telecom Finance, in order to be the tool for authorized Toshiba Dealers to provide comprehensive lease finance offering, targeted to make Toshiba VoIP business telephone solutions more available to their small to medium business customer pool. "We are very excited to work with LEAF Financial to provide our customers with flexible financing solutions from an industry leader in the small and middle-market telephony leasing space,” said Larry Meyer, vice president of Sales and Marketing of Toshiba America Information Systems for the Telecommunication Systems Division. “We are committed to providing the Authorized Toshiba Dealer Network with innovative products and programs that will assist them in maximizing their telecom sales efforts." "We are pleased that Toshiba selected LEAF to provide financial services to Authorized Toshiba Dealers and their customers and to support Toshiba Telecom Finance," said Crit DeMent, CEO of LEAF, in a press release. "As companies invest in leading-edge communication solutions to stay competitive, Toshiba Telecom Finance will help them achieve those goals with affordable financing solutions. LEAF's highly responsive service and support capabilities will make it extremely convenient for the Authorized Toshiba Dealer Network to utilize lease financing as a powerful closing tool." LEAF Financial Corporation is headquartered in Philadelphia, Penn., being the commercial finance and asset management company. Its business model is designed to emit small to middle sized equipment leases and loans in a variety of asset ranging through specialized strategic business units. After origination of the leases, LEAF operates these leases for its own account, institutions, and individual investors by means of investment partnerships and other different forms of investment.

Hosted VoIP services provider Junction Networks acquires the 10,000th Account

Company Junction Networks announced that enterprise MBLM NYC has become the company's 10,000th customer of its hosted VoIP services. As it is declared, the New York-based branding enterprise selects to deploy OnSIP Hosted PBX service to quickly enable a complete communications solution in Unified Communications concept and save on upfront investment.

VoIP solution ‘’OnSIP’’ is available at an industry low average cost per user monthly of under $20 and offers a complete UC set: hosted HD calling and conferencing, voicemail-to-email, instant messaging, telepresence, and many others. "When we merged with a larger company and needed to quickly deploy an additional 16 workstations, we decided to move from a PSTN Gateway service with Junction Networks to OnSIP, in order to save on upfront cost and get up and running quickly," said Olaf Kreitz, Partner, MBLM NYC. "We love that the service was easy to set up and also offers such a robust feature set. We did a lot of shopping around and found OnSIP to be the best value. The My.OnSIP feature has also become a fast favorite for us, as it allows each individual employee to manage their extension in so many ways." "The addition of our 10,000th customer is a tremendous milestone for OnSIP and Junction Networks. OnSIP is quickly becoming the go-to hosted VoIP service for small businesses because it's simple to use, a cinch to set up, and extremely cost effective," said Michael Oeth, CEO of Junction Networks. "We will continue to increase our service offerings and keep costs down to provide small businesses the benefits of an enterprise phone system for a fraction of the cost."

AMI Partners releases updated research on VoIP market for small business

Research Company AMI-Partners has released a report predicting growth in the VoIP market among small and medium businesses (SMBs). The research says that more than 30 percent of small businesses (1 – 99 employees) and 50 percent of medium businesses (100-999 employees) determines that VoIP will become basic form of voice communications in their business operations.

AMI’s 2010 report delivers an in-depth analysis of VoIP implementing by U.S. small and medium businesses. AMI considers that the market for IP-based voice communications, i.e. VoIP, IP PBX, will merge finally. However, last several years have been under the tensions of a tough recession and many small and medium sized companies put serious technology decisions on stand by mode. Nevertheless, the migration of voice and data will continue, the assets to leverage data networks with voice communications will get the boost, and the boarders for greater market penetration of IP solutions among the small business community remains wide open. Karen Nielsen, senior consultant with AMI, foresee that the facts and findings will be good news for the industry. “Most SMBs have limited IT resources; they will rely heavily on channel partners for VoIP installation and turnover, and to help with the crossover from analog to digital," Nielsen said. "The moment of conversion from analog to digital voice is the single biggest pain point for SMBs. Suppliers and channel partners should be prepared to make this switchover seamless. Partners should also understand and be able to prove that economic benefits stem not only from lower ongoing costs but also from a lower TCO." “From a supplier perspective, it is still early," AMI said in a prepared statement. The players are not fully known. The architectures are not completely defined. The markets are not structured. But AMI believes that down the road, the same things that are important to analog/TDM users will be important to digital/IP users: reliability, security, and quality. The winners will be those providers who can provide not only a reliable service, but an alternate back up, as well as the channel expertise to serve SMBs from soup to nuts if needed."

Tuesday, December 21, 2010

How is the leader in call center World?

The Philippines are expected to overtake India as the call center capital of the world, a recent study by Everest Research Institute indicated.

The Philippines' call center common revenues are going to reach $5.7 billion this year or $200 million more compared to India's $5.5 billion. Business Process Outsourcing revenues are predicted to outreach $9.5 billion, thus catching-up to India's $12.4 billion. The Business Processing Association of the Philippines (BPAP) noted that surpassing India in the call center business was not "far-fetched" as more companies based there are moving their operations in the Philippines. The migrating to the Philippines is associated with basic factors: the country's existing and potential workforce has better English-speaking performance, the country’s telecommunications networks are of considerably better performance than of India, and Filipinos' affinity with the American mentality. Of course, in terms of sizes, India far outweighs the Philippines. India has started to work in the BPO industry 10 years ahead of the Philippines. The Philippines’ workforce is no match to India’s, given their total population of 92 million and 1 billion, respectively, a BPAP executive has said in a recent interview. Among the list of services currently offered by outsourcing companies, such as those stationed in the Philippines, the voice-based or the traditional call center sector has one of the lowest profit margins. Amongst several transfigurations of the BPO business model is to have the low-end call center services in the Philippines, which has cultural affinity with the mostly American clients, while the up-market technical services are located in India. A multi-country model also fits well for outsourcing companies concerned about managing their business' local country risks. The Indian industry group National Association of Software and Service Companies, or NASSCom, and the BPAP keep ties for partnerships and business exchanges. Most of the BPAP member companies still offer call center services but treat this as just their entry point to get a biggest portion of the clients' more profitable non-voice services. Call centers may be low-end and low-margin, but for the Philippines it meant an employment boon. So, BPAP officials noted that the growing local BPO industry should not only translate to higher economic growth, but also more jobs. In the call center industry lone, some 60,000 jobs will be up for filling in 2011. The BPO industry has just released its latest roadmap, wherein its revenues and workforce are targeted to reach $25 billion and 1.3 million, respectively, from $9 billion and 500,000 today.

British Telecom launches new cloud computing services in Ireland

British telecom has announced the launch of two new products in Ireland, aimed for companies and organizations that desire to get the advantages of cloud computing when they expand their next-generation ICT services.

One of the products from line is Virtual Data Centre (VDC): it is an open hierarchy virtual data centre that ensures huge savings over traditional hosted infrastructure. Second product is called ‘’Next Generation Contact Centre (NGCC)’’, being on- demand solution that provides multi-channel contact centre capabilities without the capital investment. Clients are able to leverage British Telecom new products innovation to access technical innovations while cutting relayed upgrading and maintenance costs. They can become more versatile with cloud services that drastically simplify massive infrastructure while delivering greater flexibility to meet growing and ever changing business requirements. "BT’s cloud services offer significant total cost of ownership savings over traditional approaches for example; savings of up to 40pc can be recognized with VDC, while mitigating the risks associated with investing in new technologies," explained Peter Evans, strategy director, BT Ireland. Virtual Data Center ensures really vivid and virtualized infrastructure basis that enables consumers to implement their IT and networking infrastructure as a particular service and ensures foundation for future cloud services. It consists of virtual components on a pre-configured infrastructure; it sort out in a efficient way all virtual servers, storage, security and networking capabilities. It is being hosted in a British Telecom data centre, can be reached via an online portal, and supported by a single service desk, the infrastructure can be changed in real-time, allowing customers to scale their computing, network and storage requirements according to the business need. Company AMEY was the customer to test the product and successfully utilized it to implement a disaster recovery (DR) solution. In case of emergency it can restore and run all its applications within hours. "There is no operational overhead and we only pay for full DR if it is has to be used," said Delia Campbell, AMEY.The flexibility and total cost of ownership of VDC is very compelling." NGCC is an "out-of-the-box" product that conjoins multi-channel contact centre capabilities in a pre-configured product, hosted by British Telecom. It can be flawlessly integrated into any existing call centre application and is very versatile in order to grow with the business. Being embedded in basic global network, enterprises with regional locations elsewhere in the world can use it to deliver contact centre services to any place, with seamless performance and fast roll out to agents. Having an open hierarchy, it can be stepped up and down, with new agents added or removed depending on seasonal requirements. The contact centre platform has enhanced geographic performance and provides 24/7 availability. Supervisors and administrators can manage operations centrally from any internet connection. Manufacturer offers a flexible pricing policy on NGCC to suite every type of contact centre operation. Customers only pay for what they need on a per minute basis. "Businesses today face the dual challenge of ensuring excellent customer service while keeping capital expenditure costs under control," Evans explained. "This is precisely what NGCC was designed to address."

FeatureTel for business VoIP communications of pharmaceutical professionals

Company Stategic Pharma Solutions, a communications company managing pharmaceutical industry, has selected VoIP business phone services provider FeatureTel in order to provide communications services for Triangle-based business.

"When we opened our Raleigh office suite last year, we were starting from square one in selecting a phone service provider," says Theresa Moore, operations manager for Strategic Pharma Solutions. "We received referrals from an office furniture vendor we were working with, and they recommended FeatureTel for our telecom."  Moore notes that although they were scrutinizing huge amount of other providers, FeatureTel jumped up to the top of the list in a rapid way: "They were extremely responsive and went above and beyond to deliver what we needed in a timely manner."  She described all set of functionalities, such as phone calls that find and follow individuals from desk to mobile phones, were crucial for their office, where some staff are part-time and others travel quite a bit. Moore also said that the ease of expanding services was important, since they plan to add new staff in 2011, as was the built in disaster recovery and business continuity services.  FeatureTel was born in 2003 in Apex, N.C., and company offers hosted VoIP business telephone service over private networks for companies of all sizes across the Carolinas, providing a cost-effective, feature-rich alternative to traditional phone service. Removing the need for investment in a telephone system, FeatureTel customizes voice solutions delivered to a wide range of clients. FeatureTel was the first Hosted VoIP carrier in North Carolina to receive the Competitive Local Exchange Certification (CLEC) by the state's Utilities Commission and was named to the 2010 ranking of the USA fastest-growing private companies.

Partnership between VoIP company Shoretel and Americom Inc.

Partnership between VoIP company Shoretel and Americom Inc.Manufacturer of enterprise level VoIP phone systems ShoreTel has recognized repeatedly AmeriCom, Inc. for reaching unsurpassed level of professionalism and customer care in all aspects of network management, system design, project management, as well as post sale technical support. This recognition is based on specified ShoreTel customer satisfaction surveys, completed by independent and fully objective third party in 2010.

“Our 95.5% rating is a 4.5% increase over last year,” states AmeriCom’s Vice President Craig Yancich. “This increase is a direct result of AmeriCom’s effort of attaining 100% customer satisfaction. When a company provides a great product and excellent customer service, at a fair price, the result is happy customers.”  During the last decade VOIP business phone systems and solutions have performed extremely high level of competitiveness in market; ShoreTel VOIP phone systems ride the crest of the wave. AmeriCom performs brilliantly at both the technical and customer oriented fields of fitting the right ShoreTel VOIP business phone system to an organization in need of communications masterpiece. AmeriCom’s aims and targets on excellence are reflected in their high customer satisfaction results. AmeriCom has been offering business VoIP communication solutions to New England, Boston area for 25 years and has been a ShoreTel Certified Partner since 2005. AmeriCom’s ShoreTel VOIP customers have phones deployed from the east to west coast in the United States as well as in Australia, the UK, Russia and India.

Innovative VoIP gadget from Inspan

Indian company Inspan Infotech, describing itself as being listed in the Indian top 20 IT distribution enterprises and specializing on motherboards and hardware components launched VoIP mouse ‘’Genius Navigator 380’’, which is designed to merge functionalities of Web-mouse and VoIP phone. So, gadget can be used at public places for Internet telephony implementing VoIP technologies and having, as per company’s officials, quality sound and confidentiality of the conversation.

“Genius Navigator 380 is a right product to fulfill this need. It will provide the required convenience to the high flying executive” stated Sudhir S, MD, Inspan Infotech. “Partners would benefit from this kind of hybrid products, because there is always a section of customers who would want to buy such products,” adds Sudhir. The manufacturer described the innovative hardware as the mouse with a high resolution 1200 dpi optical engine; the LED indicator notifies when there is an inbound call. This VoIP mouse supports most prominent IMs such as Skype, Google Talk, MSN, Yahoo, and others, which can be merged in to one window.  The machine is available in a light green color; the mouse is equipped with seven utility buttons for different functionalities. Used as a mouse, like other pointing devices from Genius, this mouse stands as the best alternative for touch pads of the notebook, stated the company. The VoIP mouse is priced at 675 Indian rupees, is having a warranty of three years and is available immediately.

The First Open Source Telephony Conference in 2011

The real beautiful thing about open source telephony is when everybody talks about it and benefits from it. I am happy to know that the VoIP88.com has announced the 1st open source telephony conference in Shenzhen China in the coming year. And I do think it’s also the 1st one in the worldwide community since it will be launched on 10th Jan, 2011.

voip88_logo

VoIP88.com is now the largest open source telephony technical forum in China and is authorized to release the Elastix Chinese Version to the community. They are now providing free technical support in its forum and Elastix solution nationwide in local language.

What to Expect in the Conference?
The first Open Source Telephony Conference will be focus on promoting the most popular open source telephony software in China including Asterisk, AsterCC, Elastix, freeSWITCH, FreePBX and so on. Meanwhile, you will also expect to meet the new solutions and products provided by the hardware manufacturers such as Digium, OpenVox, Sangoma.

Who to Meet in the Conference?
Well, you will meet VoIP providers, PBX developers, Call Centre Developers, System Integrators of Open Source Technology, Asterisk users, hardware providers.

Conference Speakers:
Doug Vilim from Sangoma Technologies Corp
Solo from AsterCC
Tony Zhang from VoIP88.com

Click HERE for the original conference news in Chinese.

AsterCC · asterisk · Asterisk China · China Asterisk Conference · digium · elastix · freepbx · freeswitch · open source · OpenVox · sangoma · telephony · VoIP · voip88

OpenVox Announces Free Support for Asterisk Configuration >>

Monday, December 20, 2010

Asterisk cdr odbc

There is a module called cdr_odbc that supports the unixODBC library. ODBC is a standard interface between an application and a database. The unixODBC library supports many databases, from MySQL and MS SQL Server to text files. The cdr_odbc also adds functionality for a more stable connection to the database: If the database connection is lost, cdr_odbc automatically reconnects.

Note: There is a backport for Asterisk 1.4 of the Adaptive CDR ODBC that was introduced with Asterisk 1.6. This feature permits to write to arbitrary column names to the CDR table, i.e. you can invent your own additional fields as you like.


First you need unixODBC installed. In the past, lots of people have had problems with use ODBC from RPMs, so when possible, please use the ODBC source package. You can install it with the --disable-gui if you don't have/want X.

On Debian or Ubuntu systems, the standard packages should work fine ("sudo apt-get install unixodbc").


unixODBC install instructions:


 cd /usr/src
 tar zxf unixODBC-2.2.7.tar.gz
 cd unixODBC-2.2.7
 ./configure --disable-gui
 make
 make install
 

Now you need to configure your datasource. If you choose to use MySQL you need to install MyODBC (Debian: "apt-get install libmyodbc"). http://dev.mysql.com/doc/connector/odbc/en/faq_2.html http://dev.mysql.com/downloads/connector/odbc/3.51.html http://www.iodbc.org/index.php?page=downloads/index
Basic odbcinst.ini:

This is where you configure your various odbc drivers. On Debian systems, the drivers' entries will be created automatically when you install the distribution-supplied odbc driver.


 [MySQL]
 Description     = MySQL ODBC MyODBC Driver
 Driver          = /usr/lib/libmyodbc3.so
 FileUsage       = 1


 [Text]
 Description     = ODBC for Text Files
 Driver          = /usr/lib/libodbctxt.so
 Setup           = /usr/lib/libodbctxtS.so
 FileUsage       = 1
 CPTimeout       =
 CPReuse         =


 [PostgreSQL]
 Description     = PostgreSQL driver for Linux & Win32
 Driver          = /usr/lib/libodbcpsql.so
 Setup           = /usr/lib/libodbcpsqlS.so
 FileUsage       = 1


 [DB2]
 Description     = DB2 Driver
 Driver          = /opt/IBM/db2/V8.1/lib64/libdb2.so
 FileUsage       = 1
 DontDLClose     = 1
 DMEnvAttr       = SQL_ATTR_UNIXODBC_ENVATTR={DB2INSTANCE=db2inst1}


Note: you may have to change the path to the .so files depending on where they are installed.
Default location for Suse 10: /usr/lib/unixODBC


Basic odbc.ini:


This is where you configure your datasources.


 [MySQL-asterisk]
 Description     = MySQL ODBC Driver Testing
 Driver          = MySQL
 Socket          = /var/run/mysqld/mysqld.sock
 Server          = localhost
 User            = username
 Password        = password
 Database        = database
 Option          = 3
 #Port           =


 [Text-asterisk]
 Description     = ODBC for Text Files test
 Driver          = Text
 Database        = /root/flatfile
 Trace           = Yes
 Tracefile       = /root/trace.log


 [PostgreSQL-asterisk]
 Description         = Test to Postgres
 Driver              = PostgreSQL
 Trace               = Yes
 TraceFile           = sql.log
 Database            = asterisk
 Servername          = localhost
 UserName            = username
 Password            = password
 Port                = 5432
 Protocol            = 7.4
 ReadOnly            = No
 RowVersioning       = No
 ShowSystemTables    = No
 ShowOidColumn       = No
 FakeOidIndex        = No
 ConnSettings        =


[DB2-asterisk]
 Database = astdb
 Driver = DB2


cdr_odbc.conf:

 [global]
 dsn=MySQL-asterisk
 username=username
 password=password
 loguniqueid=yes ;; Required to use the userfield
 dispositionstring=yes ;; Required to use disposition like ANSWERED and FAILED


Note: If you don't use dispositionstring=yes, see the relationship above:
ANSWERED=8, NO ANSWER=4, BUSY=2, FAILED=1 and 0 for none disposition (maybe when the caller hangup the call before somebody pickup the call).



IMPORTANT adaptive_odbc information: The tables below will cause you problems with the calldate field. In cdr_adaptive_odbc, call setup, answer, and teardown are stored in each of three different fields; start, answer, and end, and the calldate field is not used. The "default now()" will of course auto-populate the calldate and unless you are using usegmtime, you won't notice it until something really breaks. Additionally, "end" is a keyword to some databases (namely PostgreSQL) and the adaptive code does not quote it, causing the query to die and tearing down the entire odbc connection. The work-around for this is to alias end to calldate, which you will want to do anyway if you want to maintain the functionality of the old cdr_odbc.

Table Setup for Specific DB's:


Postgres:


CREATE TABLE cdr (
  calldate timestamp with time zone NOT NULL default now(),
  clid varchar(80) NOT NULL default '', 
  src varchar(80) NOT NULL default '', 
  dst varchar(80) NOT NULL default '', 
  dcontext varchar(80) NOT NULL default '', 
  channel varchar(80) NOT NULL default '', 
  dstchannel varchar(80) NOT NULL default '', 
  lastapp varchar(80) NOT NULL default '', 
  lastdata varchar(80) NOT NULL default '', 
  duration bigint NOT NULL default '0', 
  billsec bigint NOT NULL default '0', 
  disposition varchar(45) NOT NULL default '', 
  amaflags bigint NOT NULL default '0', 
  accountcode varchar(20) NOT NULL default '', 
  uniqueid varchar(32) NOT NULL default '', 
  userfield varchar(255) NOT NULL default '' 
);


Mysql:


CREATE TABLE cdr (
 calldate datetime NOT NULL default '0000-00-00 00:00:00',
 clid varchar(80) NOT NULL default '',  
 src varchar(80) NOT NULL default '',  
 dst varchar(80) NOT NULL default '',  
 dcontext varchar(80) NOT NULL default '',  
 channel varchar(80) NOT NULL default '',  
 dstchannel varchar(80) NOT NULL default '',  
 lastapp varchar(80) NOT NULL default '',  
 lastdata varchar(80) NOT NULL default '',  
 duration int(11) NOT NULL default '0',  
 billsec int(11) NOT NULL default '0',  
 disposition varchar(45) NOT NULL default '',  
 amaflags int(11) NOT NULL default '0',  
 accountcode varchar(20) NOT NULL default '',  
 uniqueid varchar(32) NOT NULL default '',  
 userfield varchar(255) NOT NULL default ''  
);


Microsoft SQL:


CREATE TABLE cdr (
calldate      datetime              NOT NULL ,
clid          varchar (80)          NOT NULL ,
src           varchar (80)          NOT NULL ,
dst           varchar (80)          NOT NULL ,
dcontext      varchar (80)          NOT NULL ,
channel       varchar (80)          NOT NULL ,
dstchannel    varchar (80)          NOT NULL ,
lastapp       varchar (80)          NOT NULL ,
lastdata      varchar (80)          NOT NULL ,
duration      int                   NOT NULL ,
billsec       int                   NOT NULL ,
disposition   varchar (45)          NOT NULL ,
amaflags      int                   NOT NULL ,
accountcode   varchar (20)          NOT NULL ,
uniqueid      varchar (32)          NOT NULL ,
userfield     varchar (255)         NOT NULL
);


DB2:


CREATE TABLE cdr (
 calldate timestamp NOT NULL default current timestamp,
 clid varchar(80) NOT NULL default '',
 src varchar(80) NOT NULL default '',
 dst varchar(80) NOT NULL default '',
 dcontext varchar(80) NOT NULL default '',
 channel varchar(80) NOT NULL default '',
 dstchannel varchar(80) NOT NULL default '',
 lastapp varchar(80) NOT NULL default '',
 lastdata varchar(80) NOT NULL default '',
 duration bigint NOT NULL default 0,
 billsec bigint NOT NULL default 0,
 disposition varchar(45) NOT NULL default '',
 amaflags bigint NOT NULL default 0,
 accountcode varchar(20) NOT NULL default '',
 uniqueid varchar(32) NOT NULL default '',
 userfield varchar(255) NOT NULL default ''
);

Sunday, December 19, 2010

Cheap international phone calls using Asterisk/FreePBX and local numbers

One of the easiest ways to make cheap international phone calls is to set up a local access number that automatically forwards to an international number using VOIP.

This is a great way to keep in touch with friends and family in other countries (or to let them keep in touch with you!)

It’s very simple to do in FreePBX in two easy steps.

Create a Misc Destination for the number you want to call -

then create an Inbound Route with your local number as the DID Number -

and set the destination to your Misc Destination -

You can then call the local number and be connected to the international number, only paying inexpensive VOIP rates. The local number can be dialled in the normal way, including from a mobile phone.

Give it a go by ringing this London number – 020 3455 4080 – which will connect you to the Naval speaking clock in Washington DC.

If you’re looking for free UK local numbers then try here – http://www.ukddi.com/

Related posts: Cheap international phone calls with AsteriskGetting started with FreePBX – Part 6 Cheap phone calls using DISA and Callbackvoip.ms with Asterisk

What is IPv6 ?

Over the past few years there has been increasing concern about the shortage of IPv4 addresses. It’s now starting to grab the attention of the media, and for good reason.


At one point it must have seemed like the 4 billion IPv4 addresses would be more than enough but there are currently less than 10% left, and they’re going fast.


Exactly when they will run out is difficult to predict. The widespread usage of NAT has helped to prolong the availability of public addresses, but an interesting page here predicts only 86 days left at the time of writing. This will probably be extended as organisations hand back unused blocks, but they won’t last long.


What’s the answer? IPv6!


IPv6 provides 340 billion billion billion billion IP addresses so we really shouldn’t run out of those. However, IPv4 and IPv6 are incompatible. They do not work together, so soon we are going to have 2 separate Internets. There is going to be a huge demand over the transition period (which will take years) for people that understand IPv4 and IPv6 and how to make them work together.


So if you’re looking for something new to learn – start playing with IPv6! Your skills will definitely be in demand.

FreePBX vs A2Billing

When you start looking at control panels for Asterisk it can be difficult to decide what you should be using – FreePBX or A2Billing.

While they are both web GUIs for setting up Asterisk, they are used for different things and which one to choose depends on your needs.

Here is a brief description of both to help you decide -

FreePBX

Used for setting up extensions and trunks for inbound and outbound callsIncludes lots of features of a traditional PBX – voicemail, IVRs, ring groups, queues etc.Includes Call Detail Records (CDR) that logs all calls, their destination and duration

A2Billing

Used for billing for callsCan be used to charge for calling card, sip user or regular outbound callsThe heart of A2Billing are the rate cards that include the per minute cost for all destinations allowed to be calledLeast cost routing with multiple rate cards with the cheapest route being chosenAdmin and customer interfaces

So FreePBX is used to setup Asterisk with the features of a ‘traditional’ PBX and A2Billing focuses on billing for different types of calls.

It is also possible to combine the two and use A2Billing to account for outbound calls for extensions setup within FreePBX.

A2Billing is more complicated to setup than FreePBX. While it’s possible to setup an extension and trunk in FreePBX and start making calls very quickly there is quite a learning curve with A2Billing. Managing rate cards which hold all destinations and their cost can be quite complex.

All VPS templates offered by SysAdminMan include FreePBX – SysAdminMan, Trixbox, Elastix and PBX-in-a-flash

The following two templates include A2Billing – SysAdminMan and Elastix

Related posts: Least Cost Routing (LCR) with Asterisk and A2BillingA2Billing 1.4 development UK VPSUsing A2Billing to account for extension calls in Elastix

Nortel 1535 IP phone with Asterisk and O2 broadband

I recently ordered a Nortel 1535 SIP phone as recommended by Ward Mundy here – http://nerdvittles.com/?p=703


At £35 (price seems to vary up and down) from e-bay they are definitely a bargin. There’s a good thread on the PBX-in-a-Flash forum about setting it up here – http://pbxinaflash.com/forum/showthread.php?t=8273



They come with the default language set to Turkish so that needs changing to English.


I also really struggled with a couple of things setting it up with my O2 router. By default the WiFi would not work and also it would not register with Asterisk.


Wifi not working


I have a O2 Wireless Box II and by default the Wifi security “WPA-PSK Version:” was set to “WPA+WPA2?. The phone would not connect to the WiFi until I set this to just “WPA2? (and entered the passphrase). Even then the WiFi setup on the phone is a bit flakey. Manually assigning the ESSID seems to help too.



Phone would not register with Asterisk


Even more tricky to track down was the fact that the O2 Wireless Box II tries to do something clever with SIP connections to port 5060. I had to telnet to the router (default address – 192.168.1.254) and remove the SIP settings.


To do this -


telnet 192.168.1.254
User = SuperUser password = O2Br0ad64nd
:connection unbind application=SIP port=5060
:saveall


and then reboot the router. This information was taken from here – http://forum.o2.co.uk/viewtopic.php?t=1313


After that the phone would register following the setup directions in the PBX-in-a-Flash forum thread mentioned above.

voip.ms with Asterisk

I’ve been testing voip.ms, an ITSP based in Canada, for the past few weeks.


They have quite a few interesting features including a server located in London (which has been producing good results for me),  multiple ways to download rate information (CSV, Excel, XML API …) and the ability to set up reseller accounts.


It works well with Asterisk and below are an example of settings that can be used for a FreePBX trunk -


Trunk Name:


Dial Rules alter the way that the dialled number is passed to the call provider. This can be used to convert a number dialled in the local country format to an ‘international’ format. For instance, in the UK, you would want to remove the leading zero and replace it with the country code, 44.


Dial Rules:

44+0|[1-3].
44+0|7[56789].
44+0|8.
011+00|.

Trunk Name:


PEER Details:

username=USERNAME
type=friend
trustrpid=yes
sendrpid=yes
secret=PASSWORD
qualify=yes
insecure=port,invite
host=london.voip.ms
fromuser=USERNAME
disallow=all
context=from-trunk
canreinvite=nonat
allow=ulaw

Register String:

USERNAME:PASSWORD@london.voip.ms:5060

Obviously you need to replace the USERNAME and PASSWORD with your own voip.ms account details.

Friday, December 17, 2010

3CX Versus Asterisk

We all know the capabilities and endless feature set of Asterisk. It’s a powerful, software based PBX, that’s possibilities are endless. Asterisk is distributed in a number of ways, the first being the open sourced distribution that is command line driven. This distribution, 1.6.2.11 being the latest stable version can be downloaded from here for free, then compiled with a version of Linux, such as UBUNTU or Centos, and then administered from the command Line CLI interface.


Then we have your open sourced asterisk-based distributions freely available to the public for download. These versions would include trixbox CE, Elastix, FreePBX, PBXinaFlash, and Asterisk Now, which just names a few. The separator between these software applications as compared to the command line asterisk distribution is that while asterisk source code is running in the background, the majority of configuration changes can be made via a Graphical Web GUI interface from both a user and administrators perspective. These software applications also allow for asterisk cmd line access or CLI access in order to make changes if required. Companies such as these have taken the asterisk source code and “rounded” it to their own unique application.


And last but not least, we have our commercial asterisk based PBX’s. Most notability would be Switchvox SMB or SOHO software which is asterisk based, but no access is granted to the asterisk CLI. Rather all configuration changes are done via a web GUI interface. Software from Switchvox is considered a “paid” commercial product, which distinguishes itself from any of the open source based distributions mentioned above.


With all flavors of asterisk, each meets its own unique need and fulfillment in individual VoIP deployments. When it comes to purchasing an IP PBX, you have your choice of asterisk software and also hardware to go along with it. In all regards, whether you choose the free open source cmd line version of asterisk or pay for Switchvox’s award winning SMB software, all solutions present you with a powerhouse feature set and customizability to meet your unique business needs. A short list if asterisk’s feature sets can be found here . The list is too long to fit in this blog.


Until most recently, the majority of IP PBXs being deployed were of an asterisk flavor based on a Linux operating system. While companies such as trixbox, Elastix, Switchvox, and many others have made great efforts to take the Linux programming piece out of the picture, the solutions are still Linux based meaning when it came time for your choice of hardware or server to run your asterisk solution, you needed to make sure the motherboard and peripheral components were compatible. See my previous blog posts relating to Open Source Hardware Requirements for more information. If you were not designing your own server platform, of course you could always purchase a Switchvox appliance for your SOHO or SMB solution, or a Rhino or Phoneboch’s appliance preloaded with your desired version of trixbox CE, Elastix, or other open sourced based software. This however adds to the cost of your solution.


For the following comparison, I would like to focus on Switchvox SMB software and its comparisons against a new windows-based PBX, 3CX. The reason for this is that both solutions are “paid” commercial software based applications. Pitting these 2 up against one another would make the most sense from a cost, feature set, and positioning standpoint, so onward to the comparisons.


3CX is a Windows based IP PBX platform that is becoming very popular in the VoIP world. A few benefits of 3CX include, “its windows based”, most of us are used to Windows based operating systems and applications; it is completely software based and allows you to use existing Windows servers in your existing environment if available. For a complete list of compatible windows based operating systems, see here. Below are a few more benefits, 3CX has to offer:


-Windows Application Integration support- applications such as Microsoft Exchange, Outlook, Unified Messaging , Microsoft CRM , and SalesForce CRM are all fully supported on 3CX version 9.


-Feature rich- functions and feature sets of 3CX version 9 are very comparable to an Asterisk distribution. See here for a complete set of features. Please look at the “paid commercial” versions.


-Free and Paid commercial versions- are available to experiment with and test. Free versions of 3CX software are easily upgradable to any commercial paid version located here, through a unique activation license key provided to you via email after your purchase. You can then apply it directly from the 3CX admin GUI configuration. Updates are instantaneous after you apply them. They are based upon the maximum number of concurrent calls you will have on your solution, and can be easily upgraded to higher amount of simultaneous calls if needed. Once again, changes are done via the web GUI and are instantaneous.


-User Dashboards– the 3CX assistant provides each 3CX user a PC based presence application where they can easily set their calls rules, answer calls; listen to voicemails, and chat. More information on 3CX Assistant can be found here.


-Hotel Billing and PMS integration- On top of their paid software platforms, administrators can add the 3CX Hotel Billing, or the full-blown Hotel PMS integration module to easily integrate Hotel PMS platforms with 3CX. This is a huge hit in the hospitality industry. More information on this support can be found here. For pricing on the software add-ons, see the billing and PMS products on VoIPSupply.com.


-Non proprietary, SIP Standards based- Since the 3CX solution is based upon SIP protocol, administrators have the ability to use any SIP based endpoints such as IP Phones, Gateways, ATA’s ,and DECT Phones . Furthermore 3CX has fully tested a large number of these endpoints and offers full support including auto-provisioning. See here. If you are looking to add some PCI cards to the solution, you will want to go with a Sangoma PCI FXO or T1 Netborder Express card with echo cancellation.


-Smartphone Apps- 3CX offers and fully supports a soft-phone application specifically designed for Android’s, and is beta testing a new app for Apple products such as the iphone, ipod, and ipad running IOS 4 or higher.


Now onto the Switchvox SMB platform.


-Third Party Application Integration support– Switchvox SMB integrates very easily with Sales Force CRM, Sugar CRM, Fire Dialer for Firefox users, Notifier Suite to integrate Switchvox to outlook and other Windows applications, and finally an Outlook plug-in to allow click to call from your outlook contacts. More information can be found here. If that’s not enough, Switchvox offers a developer API


-Feature Support- I always like to refer the SMB platform as Switchvox’s “Big Daddy” PBX, the reasonin behind it is simple; it is jammed packed with features for both users and administrators. A full list of SMB feature set can be found here. Make sure your looking in the SMB column.


-Multiple software versions available- If SMB is too much for you, you can opt for Switchvox SOHO or HOME versions which are more limited in functionality and cheaper in costs as compared to the SMB solution. Also, Digium offers a FREE 60 Day Trial of the full-blown SMB software, so you can “test drive” the actual SMB solution before making your purchase. If you are interested in test driving the SMB solution, contact our dedicated Sales Representative Arthur Miller at 716-250-3871 to discuss the details.


-User Dashboards- Personally, The Switchvox Switchboard, ONLY available with SMB is one of the most intuitive presence applications on the market, and it all comes bundled with your SMB solution. From the Switchboard, users can drag and drop calls to other users, see other users real-time call state, access VM messages, customize to see Google Maps, integration with CRM accounts, Queue status, CDR, Chat, and the list goes on. If you would like to see the switchboard in action, just watch it here.


-Non proprietary SIP based endpoints- While SMB has built-in auto provisioning support via is Provisioning Token system for all Polycom and Snom IP Phones, it is SIP based, so any SIP endpoint including IP Phone, gateway, ATA, etc will function. If you are looking to add some PCI cards to the solution, you will want to go with any Digium PCI card for your FXO/FXS, or T1 connectivity. Echo cancellation models should also be considered.


-Smartphone Apps- Digium Switchvox offers a Smartphone application for both apple based products and Blackberry’s. For you Droid users, sit tight, you will get yours soon. The application allows users to create and modify their call rules, access VM, and call into the Switchvox PBX. More information can be found here.

Buying the right Polycom phone

If you are ready to buy a VoIP phone and have done the research and decided on Polycom, congratulations!  Now you just need to narrow down your selection from the 9 Polycom phone models offered and the 2 variations of each model, so here we go!

Knowing what kind of call volume your business has or will have is imperative to deciding which Polycom phone to install.  Sure, any Polycom phone will perform exceptionally well even if the user is on the phone 24/7, but it won’t necessarily help the user be more effective or satisfy your customers who are calling in.

The IP321, 331, and 335 have 2 SIP lines, 3 soft keys, and a 2.5mm headset port (IP335 has an RJ9 headset port).  If your user will rarely have more than 1 call coming in at a time and will have little need to transfer, this phone will work just fine.  Yes the IP321/331 and IP335 are capable of call transfer, but there is only one hard-key for busy lamp field (BLF) so multiple “favorite extensions” is not possible.  Also with only 3 soft-keys and the basic hold, headset, speaker, menu, and dial hard key, the use of special features like conference, transfer, and redial are a bit more cumbersome.  So the Polycom SoundPoint IP321, IP331, and IP335 are recommended for a desktop where there is very low call volume, or common areas like break rooms, kitchens, copy rooms, and lounges.

Polycom SoundPoint IP335 Polycom SoundPoint IP335

For most deskbound positions with medium to high traffic volume (think 12+ calls a day) the IP450, IP550, or IP560 is a sound, budget conscious choice.  The Polycom SoundPoint IP450 and IP550/IP560 have 3 and 4 SIP lines, respectively, offer HD Voice wideband sound on both the speakerphone and handset, which is awesome, have RJ9 headset ports, and have a large backlit graphical LCD display.  Four context sensitive soft-keys give the user more options at their fingertips.  For those requiring more complex phone functionality like transfer, conference, company directory, and Do Not Disturb, the IP550 or IP560 is recommended over the IP450. The IP550/IP560 has 4 context sensitive soft-keys and a Mute, message, redial, speaker, handset, and hold, like the IP450, but it also has dedicated hard-keys for Directories, Applications, Conference, Transfer, and Do Not Disturb.  I have a Polycom IP550 at my desk and I love it lots. It offers 4 SIP lines, 2 of which I have assigned to my extension and the other two I use as a BLF for my 2 most dialed extensions.

Phone users who would be considered “super users” would most definitely need an IP650 or IP670. These super users, such as receptionists, operators, and call center managers need more than a few Busy Lamp Fields for transferring to multiple extensions and monitoring others’ phone status. The IP650 and IP670 offer 6 SIP lines that can be used as BLF extensions right out of the box, however, you can add up to 3 sidecar modules to these bad boys to give you an additional 12 BLF extensions each.  The Polycom sidecars are nicer than most because instead of using paper inserts, the labels are an LCD screen (monochrome for the IP650 and color for the IP670).  This offers flexibility for the administrator to change extensions quickly and for the user to see the changes in real time without having to print out and insert a new paper label.

Consider carefully where you will be placing your Polycom SoundPoint IP phones before you decide on a model.  First question you should ask yourself is will these phones be attached to a PoE (Power over Ethernet) switch?  If so, all Polycom SoundPoint IP Phones are PoE, so you are cool there, but don’t spend more than you need to.  Most Polycom SoundPoint models come in two variations, with (part number ends in -001) and without an AC Power adapter (ends in -025). If your network is already PoE, make sure you buy a Polycom model that ends in -025. Why pay for a power adapter you don’t need?

Secondly, will the desktop or location you are installing the phone have 1 or 2 RJ45 jacks?  If there is only 1 and there will be a PC there as well, you need to choose a Polycom phone that has a dual RJ45 switch integrated into it. This will allow you to plug the phone into the single wall jack, and then the PC into the phone, passing connectivity through to both appliances without having to install expensive additional wiring.  Polycom IP Phones that offer the dual RJ45 jacks are the IP331, IP335, IP450, IP550, IP560, IP650 and IP670.  So if you only have one network jack and need a phone and PC there, stay away from the IP321.

Another consideration here is speed. If you are using your Polycom IP Phone to pass data through to your PC (dual RJ45 jacks) you should consider the speed of your server switch.   If you have a 10/100/1000 network switch and want to offer gigabit speed to your PC, you need a Polycom phone not only with dual RJ45 ports, but with dual 10/100/1000 RJ45 ports.  That narrows down your selection quite a bit more.  If you are looking for a gigabit Polycom phone, your choices are the IP560 or IP670.

The Polycom IP650 and IP670 are virtually the same phone with 2 major differences. First, the IP670 has gigabit support (boring).  But even cooler is that the IP670 and its sidecar support a full color LCD, so you can choose color schemes or set pictures as your background. In addition, the IP670 has a USB port for local call recording. That’s pretty cool, and necessary if you are in the legal or financial industry where you may need record of phone calls on the fly.

Polycom IP670 Polycom IP670

You should also consider Polycom’s HD Voice.  Polycom HD Voice equipped phones not only support the G.722 wideband codec, but they also build up their hardware in the handset and speakerphone to enhance the HD voice sound.  It is really better than actually being there.  Today, HD Voice is really a cool factor, the sound of it will just amaze you, but you only really experience the rich sounds when both you and the other party are on HD Voice phones, the call is completely IP, and all PBX’s involved support HD Voice.  For today that may only involve internal calls and outbound calls with a few clients. However buying a Polycom phone that is equipped with HD Voice will be future proofing your infrastructure for the next 5 years. More and more phones are coming standard with HD voice support, and as the world continues to migrate to IP PBX’s, most already support HD voice and if they don’t, it is in their upcoming upgrades.

Because you asked, and even if you didn’t I am going to tell you anyway, I recommend the Polycom IP450 for most office applications.  I am a moderate phone user and I have the IP550. The extra hard keys for Directory, Applications, Conference, Transfer, and redial are nice, but I don’t really use them.  My IT guy has provisioned some of these options into the context sensitive soft-keys company-wide for some of the users who don’t have the IP550 and I use those for transfer and redial.  Yeah, the IP550 is pretty cool, but if you are on a limited budget, the IP450 should still do you well.  Of course, for high volume users like secretaries and receptionists, you would need the IP650 and the IP650 sidecar.  IP670 is a must in legal or financial scenarios where local call recording is required.

Thursday, December 16, 2010

Yealink VP-2009P Overview

yealink vp-2009pAs the popularity of video telecommunications continues to grow, so do the makes and models of IP video phones.  To go along with the more well known offerings from Polycom and Grandstream, Yealink has jumped into the market with their own entry – the VP-2009P.  The Yealink VP-2009P has the sleek look and style of the other Yealink desk phones and is reasonably priced, with a price point sitting between similar models from Grandstream and Polycom.  With a large 7” touchscreen LCD display, support for both h.263 and h.264 video codecs, as well as support for the wideband G.722 audio codec, the Yealink VP-200P is looking to position itself as one of the leaders in the desktop video phone market.

While this innovative IP multimedia phone offers the most advanced video and audio compression technology for superb voice and picture quality, it also supports IPTV, VoD, media player, web browser, message, digital photo frame and IM as well.  The VP-2009P not only enhances face-to-face communications with a plethora of other features, it does so while only requiring minimal bandwidth.  Grandstream and Polycom better be on their game, because Yealink wants them to know they are coming and they are coming hard!

VIDEO

Touch screen7” digital TFT-LCD with 800×480 pixels resolutionRotatable CMOS sensor camera with 300K pixelsVideo codec:  H.263 and H.264 baseline profilePicture codec:  JPEG, GIF, PNG, BMPFrame rate up to 30fps with resolution of D1 (720×480) or CIF (352×288)Bandwidth:  128kbps – 1MbpsAdaptive bandwidth adjustment

VOICE

Voice codec:  G.711, G.723.1, G.729AB, G.722 (wideband)VAD, CNG, AEC, PLC, AJB, AGCFull-duplex hands free speakerphone with AEC

VIDEO APPLICATION

Full screenPicture-in-pictureLocal video controlColor ring/imageTV output when video callOnline/offline advertisementWork with IP CameraWork with doorbell

PHONE FEATURES

Video and voice call selectionCall forward, Call waiting, Call transferHold, DND, Mute, Auto answerDial plan, Dial-now, Speed dialCaller ID display, Call historyPhonebook, Black list, Group manageVolume control, Ring tone selectionWall paper, Screen saver, SleepDate & Time manually or automaticallyIcon-driven menuMulti-language27 keys including 4 soft keys1xUSB 2.0 port1xSD card slot2.5mm headset portPoE Support (IEEE 802.3af)A/V out:  RCA style stereo audio and composite video output

As you can see, the Yealink VP-2009P is feature-rich and has most everything you could ask for from an IP multimedia phone.  For more information on this or any Yealink product, or video telecommunications in general, please contact me at 716-250-3873 or email at jshanahan@voipsupply.com.

Grandstream GXV3140 Now Skype Certified

As Skype continues to evolve as a service provider it’s inevitable that they will continue to find ways to move away from the desktop and onto the desktop.

Over the summer, Grandstream released a new firmware version that allow it’s GXV3140 to natively support Skype and Skype video calling. Today, Grandstream has taken this one step further by announcing that they Grandstream GXV3140 is now Skype certified.

What’s this mean?

Well, unlike this summers announcement, it means that Skype has officially given the Grandstream GXV3140 its blessing, making it the second videophone on the market for use with Skype. And according to Grandstream,

“Skype certification on the GXV3140 IP multimedia phone creates a seamless video-enabled communications experience from PC to desk phone, giving Skype users the utmost flexibility and convenience for video calling. Grandstream’s GXV3140 bridges the gap between convenient-to-use dedicated voice/video end-points and the increasingly popular world of PC-based Internet multimedia communication offerings like Skype.”

The idea of seamless video communications is a major hurdle to widespread video calling adoption. For the most part there is little interoperability between video calling services and hardware, leaving most companies interested in video calling hesitant to pull the trigger for fear of living in a walled garden.

While this announcement certainly won’t change this, it is a step in the right direction and a sign that more seamless video communications solutions are coming.

Switchvox Mobile on your iPhone or Blackberry

Are you mobile?  Do your clients have to juggle 2 or more phone numbers to reach you because you are sometimes at your desk but most of the time all over the place?  Don’t you wish your business card were a little less cluttered with just one contact phone number that you can be reached at all the time?  You can have these things now, and you probably won’t even have to buy anything!

Businesses running Switchvox 4.6 or 4.5 already have the ability to stream the Switchvox desktop phone functionality to iPhones and Blackberries.  If you are using Switchvox for your desktop phone system and you have an iPhone or a Blackberry, you are less than 10 minutes from enjoying all the features your desk phone has to offer on your SmartPhone!

Ready? Let’s get mobile!

Once you enter the App store, search for Switchvox Mobile.  This App is a FREE DOWNLOAD!!!! (Don’t you just love FREE)

Switchvox Mobile for Blackberry Switchvox Mobile for Blackberry

-          For Blackberry go to www.BlackBerry.com/AppWorld 

-          For iPhone go to www.apple.com/iphone

(This is neat…. If you don’t have Switchvox yet, but want to see how Switchvox mobile works, you can download the App and use it in DEMO mode!!!! )

-iPhone Users – click here for easy to follow instructions on how to set up your iPhone for Switchvox Mobile (with Pictures!)

-BlackBerry Users – click here for easy to follow instructions on how to set up your Blackberry for Switchvox Mobile

See details, listen to your voicemails, and return calls at the touch of a button.Access your Switchvox Company directory without logging into the internet.Remotely set your greetings without a chain of touch-tone commands.Make calls out either from your SmartPhone number or your Switchvox extension.Set up call rules using an easy GUI while away from the office.See call history from your SmartPhone while nowhere near your desk. Even see call history on your SmartPhone for calls you made from your desk!

SOHO DECT Systems

DECT Solutions for your SOHO Environment

So after reading my first two posts on DECT technology and how to design a DECT system, you are now ready to move forward with a solution that fully meets your needs from a hardware perspective. There are a few solutions that are viable to meet your needs with more on the brink. Since DECT phones are becoming more popular by the minute in the VoIP industry, we are seeing manufacturers entering this market and delivering robust cordless solutions that can meet the needs of any SOHO application. Below, we will detail a few of these solutions.

panasonic_voip

The first solution is actually a new solution and actually a new “player” in the SIP DECT world. Panasonic who is well known for their analog cordless handsets which are very popular in your large retail chain stores such as Best Buy, Target, or Wal Mart have introduced 2 cordless SIP DECT Solutions to the market. The first solution is the KX-TGP550 which is a corded desk IP phone that ships with 1 additional DECT handset which pairs to the base phone. If you are familiar with the Aastra CT solutions, this is very similar EXCEPT, the KX-TGP550 can supply up to 6 additional KX-TPA50 DECT handsets and EACH handset can be allocated as its own individual unique SIP extension unlike the Astra solution that only allows 4 handsets and each handset inherits Line1 configuration of the base phone. Here is a good look at how the KX-TGP550 and KX-TPA50 handsets function.

panasonic_dect_system

The solution allows for 8 individual SIP extensions but SIP extensions may also be shared between each handset and/or base unit. A good look at how this is configured via the web GUI interface is below:
panasonic_dect_screenshot

The KX-TGP550 base phone can also act as BLF/Speed dial application as it contains 6 keys which are each dedicated to a unique handset paired to it. Receptionists can use these buttons to see if the person is on a call or available to take a call and even press that users handset button to speed dial them. They can also press a “ALL” button which essentially calls every handset on the DECT network.  A good look at how this is setup and displayed is below:

panasonic_dect_system_closeup

The next solution offered by Panasonic in the SIP DECT world is the KX-TPG500 which offers a DECT base station that physically connects to your LAN network via cat5 and transmits DECT signal as well handles the SIP registration between the paired DECT KX-TPA50 handsets and the SIP Call server or provider. The KX-TPG500 ships with a single base station and handset. It can support a total of 6 DECT KX-TPA50 handsets and 8 SIP registrations. Here is a good look at how this solution is setup:
panasonic_dect_key_system
Both solutions are viable for SOHO applications and can completely transform your home or small office into an entire cordless mobile phone system. The only downside to these solutions is that there are currently no DECT repeaters available to extend DECT signaling to greater ranges.

snom_logo

The next DECT SOHO solution is manufactured by Snom and has been around for quite some time. The Snom DECT solution is called the Snom M3 . The M3 ships with a DECT base station that transmits DECT signal and also handles SIP registration between the 8 additional M3 handsets that can be paired to it. It ships with 1 handset and each handset can be programmed with its own unique SIP extension. The base can handle 3 concurrent calls at any time and unlike most DECT solutions, the M3 DECT solutions offers DECT repeater options that can be spread out within the main base’s DECT frequency to offer a greater range of DECT signal coverage. The base can support up to 6 additional DECT repeaters which can be purchased separately. Below is a good example of the Snom M3 solution and how the DECT repeaters offer greater signal coverage.

snom_m3_dect

Each Snom handset supports a speakerphone option, 2.5mm headset jack, customizable ring/vibrate options, color LCD, out of range notification, low battery notification, and ships with its own desktop charger.
aastra_logo

Aastra also offers a SIP DECT Solution called the MBU-400. The MBU-400 ships with a single MBU-400 DECT base station that broadcasts DECT frequency and handles 8 unique SIP registrations between each paired Aastra DECT 420d handset. The MBU-400 ships with 1 420d handset and can handle up to 8 total 420d handsets which can be purchased separately. The base is unique to other DECT solutions because it can handle 4 concurrent calls, 3 VoIP over the LAN connection, and 1 PSTN via its FXO port. This is also useful in cases where your network may go down. The FXO port is also utilized as a lifeline and calls are automatically routed over to the FXO port if the base loses its SIP registration or network connectivity.

aastra_sip_dect

Each 420d handset offers a speakerphone, 2.5mm headset jack, ring and vibrate options, out of range indications, low battery indications, ships with its own desktop charger, color LCD, distinctive ringing, customizable phonebook, and over the air updates. The downside to the Aastra MBU-400 is it doesn’t offer any DECT repeater option.

siemens_gigaset_logo

And last but not least, we have Siemens Gigaset which offers 2 SOHO SIP DECT solutions, the A580IP and S675IP. Pictured below the A580 IP ships with (1) A58H DECT handset and 1 DECT base station that interfaces with your LAN network. The base can support up to 6 total Siemens Gigaset DECT handsets and Siemens provides a few different models linked and pictured below. All handsets support Wideband voice quality through HDSP technology which is a feature that sets this solution apart from any of the DECT solutions talked about previously.
gigaset_a580

Additional Handsets that are supported on the A580 IP solution and can be mixed and matched include the following:

siemens_gigaset_dect_handsets

The second SOHO DECT solution offered by Siemens Gigaset is the S675 IP. The S675 IP solution ships with 1 S67H DECT handset and like the previous solution can support up to 6 total DECT handsets paired to the S675 IP base unit. The base unit contains a single LAN port to connect to your network as well as 1 FXO port to connect to your local land line. It supports 3 concurrent calls, 2 VOIP and 1 through the FXO port via PSTN. This is also used for failover. Below is a picture of the S675 IP and a listing of its supported HDSP DECT handsets.
siemens_gigaset_s675_ip
Additional Handsets that are supported on the A580 IP solution and can be mixed and matched include the following:

siemens_gigaset_s675_handsets

And that about does it for SOHO IP DECT solutions. Stay tuned for my nest post which will describe a few enterprise level DECT solutions.