Thursday, April 28, 2011
Asterisk VS sipXecs PBX
sipXecs is an Enterprise SIP PBX that comes complete with voice mail and auto-attendant. It can also be used as a high performance enterprise toll-bypass SIP router. sipXecs combines all common calling features, XML-based SIP call routing, Web-based configuration, and integrated management and configuration of the PBX and attached phones and gateways. It is a modular server-based solution that does not require any additional hardware, as it interoperates with any SIP compliant gateway, phone, or application.
Wednesday, April 27, 2011
Asterisk Queue
Asterisk queues are wealth! You can do anything with the incoming calls using the queues. Have you ever wanted to route incoming calls to another person, or to a []voicemail, without the knowledge of the customer? With the Asterisk queues you can balance incoming call workload among your employees. Have you ever wanted one person to have more priority in the calls, or a whole group of callers?
The main Asterisk configuration files are located in /etc/asterisk/. In this directory is located queues.conf - /etc/asterisk/queues.conf.
There are two possible ways to configure the settings in this file.
The first one, is the so called "static" way. For this configuration you do not need the agents.conf file.
You will say directly, which users to answer the incoming call. In our example, we have created a context with the name [simple-queue]. In this context we have written the following: member => IAX2/idefisk and member => IAX2/idefisk-rulez. This means that these agents will be responsible for the answering of the incoming calls in the queue simple-queue. In the same way you can add even more than two users, which you would like to answer in this queue. However in our case we will have only two users. The advantage of this method, is that you do not have to use the AgentLogin or the AgentCallbackLogin applications in order to log the users in the queue. It will be logged in automatically, when it turns on its phone. The disadvantage is that these users can use, only the phones, on which they are registered. That is why the method is called "static". You can see a screenshot below.
The other way is the so called "dynamic" way. For this configuration you will need the agents.conf file.
Unlike the "static" method, here we have to write not the name of the user, but the number of the agent, as written in the agents.conf file. In our example we have written the following: member => Agent/8888. This means that we want the agent with number 8888, to answer the incoming calls in the queue test. If you decide to use this method you have to add in your dialplan either the AgentLogin or the AgentCallbackLogin application. This is necessary, because unlike the "static" method, here is not enough just to turn on your phone. The user has to log in itself in the queue, manually. The advantage of this method is that the user could log in from any place and phone, which is connected with the Asterisk PBX.You can see a screenshot below.
Here you can read detaily about agents.conf.
All available settings in /etc/asterisk/queues.conf can be found here.
By default Asterisk queues are using FIFO (First In, First Out) or called in the asterisk world – first-come, first-served. The behavior is: what comes in the queue first it is handled first, what comes in the queue next waits until the first is finished, etc…
There are also two other queues by theory:
1) LIFO (Last In First Out) - what comes in first is handled first, what comes in next waits until the first is finished, etc.
2) Generalized Processor Sharing - customers are served equally. Network capacity is shared between customers and they all effectively experience the same delay.
In /etc/asterisk/extensions.conf you should use the Queue() application, by which you can put calls in the queue.
/etc/asterisk/extensions.conf:
Simple:
[idefisk]
exten => 78,1,Answer
exten => 78,n,Queue(simple-queue)
exten => 78,n,Hangup
/etc/asterisk/queues.conf:
[simple-queue]
context = idefisk
member => IAX2/idefisk
member => IAX2/idefisk2
If any in the system dial 78, the call will be put to the queue named simple-queue.
The main Asterisk configuration files are located in /etc/asterisk/. In this directory is located queues.conf - /etc/asterisk/queues.conf.
There are two possible ways to configure the settings in this file.
The first one, is the so called "static" way. For this configuration you do not need the agents.conf file.
You will say directly, which users to answer the incoming call. In our example, we have created a context with the name [simple-queue]. In this context we have written the following: member => IAX2/idefisk and member => IAX2/idefisk-rulez. This means that these agents will be responsible for the answering of the incoming calls in the queue simple-queue. In the same way you can add even more than two users, which you would like to answer in this queue. However in our case we will have only two users. The advantage of this method, is that you do not have to use the AgentLogin or the AgentCallbackLogin applications in order to log the users in the queue. It will be logged in automatically, when it turns on its phone. The disadvantage is that these users can use, only the phones, on which they are registered. That is why the method is called "static". You can see a screenshot below.
The other way is the so called "dynamic" way. For this configuration you will need the agents.conf file.
Unlike the "static" method, here we have to write not the name of the user, but the number of the agent, as written in the agents.conf file. In our example we have written the following: member => Agent/8888. This means that we want the agent with number 8888, to answer the incoming calls in the queue test. If you decide to use this method you have to add in your dialplan either the AgentLogin or the AgentCallbackLogin application. This is necessary, because unlike the "static" method, here is not enough just to turn on your phone. The user has to log in itself in the queue, manually. The advantage of this method is that the user could log in from any place and phone, which is connected with the Asterisk PBX.You can see a screenshot below.
Here you can read detaily about agents.conf.
All available settings in /etc/asterisk/queues.conf can be found here.
By default Asterisk queues are using FIFO (First In, First Out) or called in the asterisk world – first-come, first-served. The behavior is: what comes in the queue first it is handled first, what comes in the queue next waits until the first is finished, etc…
There are also two other queues by theory:
1) LIFO (Last In First Out) - what comes in first is handled first, what comes in next waits until the first is finished, etc.
2) Generalized Processor Sharing - customers are served equally. Network capacity is shared between customers and they all effectively experience the same delay.
In /etc/asterisk/extensions.conf you should use the Queue() application, by which you can put calls in the queue.
/etc/asterisk/extensions.conf:
Simple:
[idefisk]
exten => 78,1,Answer
exten => 78,n,Queue(simple-queue)
exten => 78,n,Hangup
/etc/asterisk/queues.conf:
[simple-queue]
context = idefisk
member => IAX2/idefisk
member => IAX2/idefisk2
If any in the system dial 78, the call will be put to the queue named simple-queue.
Asterisk Desktop Assistant
Asterisk Desktop Assistant is a desktop call management application for Windows PCs. Asterisk Desktop Assistant (ADA) is Digium?s first step towards offering a comprehensive Computer Telephony Integration (CTI) suite. It makes dialing and handling phone calls simpler and faster by adding click-to-call functionality into popular desktop applications. It also adds call notifications directly to the Windows desktop.
Asterisk AppKonference
AppKonference is a high-performance Asterisk voice/video conferencing module. It's basically a drop in replacement for meetme - although does things a little differently and doesn't require a timing source.
Asterisk VMukti
VMukti is leading Asterisk/ Yate enabled p2p Video IP Communications Suite for Web / PSTN. These serverless broadband ready platform enable OS community to save 90% on capital & operating costs over proprietary software for conferencing & Call Center.
Asterisk with X-Lite Softphone
CounterPath's X-Lite is the market's leading free SIP based softphone available for download. Combining voice and video calls, Instant Messaging and presence management in an intuitive and familiar interface, X-Lite helps you seamlessly transition from a traditional hard phone environment into the world of Voice over IP.
If you are already a VoIP or Broadband service subscriber (i.e. you receive VoIP service from your local Service Provider or local IP-PBX), X-Lite 3.0 lets you make audio calls to any softphone, mobile or landline number, and make video calls and send IMs to any softphone which supports these functions.
Whether you are are having a simple voice conversation, enjoying a video conference call or engaging with friends over IM, X-Lite leads you down the new path of personalized communications.
If you are already a VoIP or Broadband service subscriber (i.e. you receive VoIP service from your local Service Provider or local IP-PBX), X-Lite 3.0 lets you make audio calls to any softphone, mobile or landline number, and make video calls and send IMs to any softphone which supports these functions.
Whether you are are having a simple voice conversation, enjoying a video conference call or engaging with friends over IM, X-Lite leads you down the new path of personalized communications.
Monday, April 25, 2011
Amanda *Star PBX
Amanda *Star PBX - is The Amanda Company's version of Asterisk, the open source PBX. Asterisk is the world's leading open source telephony engine software, and arguably, the world's leading telephone system. Open source software is free to download, use, and to customize. Hence there are now literally thousands of programmers continuously implementing new features, extending capabilities, and innovating the system, bringing ever evolving benefits to the market for anyone to use. Google it and see for yourself!
Imagine installing the last telephone system you will ever need.
Virtually any communication capability you have experienced is likely to have already been implemented by someone, somewhere with Asterisk. If you have imagined something new, or especially creative for your situation, Asterisk's extensibility and open source approach allows you to develop it or have it developed as you would want it. Try asking one of those Telecom Giants to do that for you!
Amanda *Star PBX combines the Asterisk system with an Amanda designed, browser-based GUI. There are also many extensions that make installing, administrating, and customizing your Amanda *Star PBX easier, more sensible, and most importantly, as you might expect with a telephone system rather than a computer program.
Extensions such as:
• Shared Line Appearances on supported SIP phones for easier hold - announce - call pick-up
• Plug-and-Play Auto-Provisioning of supported SIP phones, including Cisco, Polycom, Mitel, Snom and Aastra phones
• Full Extension Mobility allows you to on-the-fly map an extension to any phone so now your calls are not chained to your desk phone
• Time-based Routing gives you automatic control of your usual routines for handling incoming calls not only for your business but also to individual extensions
• Web-based User Managed Access for customized, automatic, or, on-the-fly changes for call routing, voicemail access and extension mobility at the individual level
And with Amanda *Star PBX you are not constrained by specific hardware, phones, or even service providers. You can use any of the hundreds of already available SIP phones on the market, any PC platform you want (although we think ours is pretty good!), and even any telephone lines you want from POTS, ISDN to SIP trunks.
Break Free from traditional, restrictive, proprietary telephone systems that imprison you with their software, their hardware, their support, and their problems. Take control of what you want and how you want to communicate with Amanda *Star PBX.
Imagine installing the last telephone system you will ever need.
Virtually any communication capability you have experienced is likely to have already been implemented by someone, somewhere with Asterisk. If you have imagined something new, or especially creative for your situation, Asterisk's extensibility and open source approach allows you to develop it or have it developed as you would want it. Try asking one of those Telecom Giants to do that for you!
Amanda *Star PBX combines the Asterisk system with an Amanda designed, browser-based GUI. There are also many extensions that make installing, administrating, and customizing your Amanda *Star PBX easier, more sensible, and most importantly, as you might expect with a telephone system rather than a computer program.
Extensions such as:
• Shared Line Appearances on supported SIP phones for easier hold - announce - call pick-up
• Plug-and-Play Auto-Provisioning of supported SIP phones, including Cisco, Polycom, Mitel, Snom and Aastra phones
• Full Extension Mobility allows you to on-the-fly map an extension to any phone so now your calls are not chained to your desk phone
• Time-based Routing gives you automatic control of your usual routines for handling incoming calls not only for your business but also to individual extensions
• Web-based User Managed Access for customized, automatic, or, on-the-fly changes for call routing, voicemail access and extension mobility at the individual level
And with Amanda *Star PBX you are not constrained by specific hardware, phones, or even service providers. You can use any of the hundreds of already available SIP phones on the market, any PC platform you want (although we think ours is pretty good!), and even any telephone lines you want from POTS, ISDN to SIP trunks.
Break Free from traditional, restrictive, proprietary telephone systems that imprison you with their software, their hardware, their support, and their problems. Take control of what you want and how you want to communicate with Amanda *Star PBX.
SoftSyl Introduced VoiceXML Platform for Asterisk
SoftSyl, a company providing Interactive Voice Response (IVR) solutions for small and medium-size call centers and businesses, released AsterUVX, a premier VoiceXML platform specifically designed to integrate with Asterisk and other open-source telephony platforms.
The newly launched AsterUVX enables open-source platforms to work as full call center suites that are capable of handling millions of simultaneous calls. Through this launch, SoftSyl expects to provide cost-effective, enterprise-grade IVR solutions at a fraction of the cost of other providers.
AsterUVX is worked out to be used as a cloud-based IVR solution but can also be licensed for traditional on-premises deployments. Whether customers choose a hosted or on-site IVR system, SoftSyl stated that it customizes each turn-key solution package by including any necessary professional engineering services.
When combined with SoftSyl’s Unified Voice Exchange (UVX) platform and any number of open-source telephony platforms, clients can dramatically reduce the total cost of ownership for an IVR system without sacrificing industry compliance and portability. This, according to the company, ensures greater operational efficiency and business profitability.
SoftSyl has been delivering customized IVR solutions with built-in reliability, scalability and application multi-tenancy. SoftSyl’s products, including AsterUVX, are accessible to both the end-user and call center vendors.
The newly launched AsterUVX enables open-source platforms to work as full call center suites that are capable of handling millions of simultaneous calls. Through this launch, SoftSyl expects to provide cost-effective, enterprise-grade IVR solutions at a fraction of the cost of other providers.
AsterUVX is worked out to be used as a cloud-based IVR solution but can also be licensed for traditional on-premises deployments. Whether customers choose a hosted or on-site IVR system, SoftSyl stated that it customizes each turn-key solution package by including any necessary professional engineering services.
When combined with SoftSyl’s Unified Voice Exchange (UVX) platform and any number of open-source telephony platforms, clients can dramatically reduce the total cost of ownership for an IVR system without sacrificing industry compliance and portability. This, according to the company, ensures greater operational efficiency and business profitability.
SoftSyl has been delivering customized IVR solutions with built-in reliability, scalability and application multi-tenancy. SoftSyl’s products, including AsterUVX, are accessible to both the end-user and call center vendors.
Friday, April 22, 2011
The Zaptel Drivers
When running an Asterisk PBX under FreeBSD, certain telephony features such as conferencing (aka the MeetMe feature) require an external timer in order to function properly. This timer is often provided by the Zaptel hardware drivers. However, if you are using a non-Zaptel configuration, such as on a completely SIP-based telephony setup, you probably won't have the drivers for Zaptel hardware installed and configured. To make matters worse, the Zaptel drivers have only recently been ported to FreeBSD in a stable manner, so they seem to be a recent development on the platform.
A few steps are required to get the drivers running. Below are my experiences adding the Zaptel drivers to an existing FreeBSD-based Asterisk PBX in early July 2006. You should also be aware that some reports indicate the Zaptel drivers for FreeBSD only work correctly under FreeBSD v5.x or v6.x, so you may need to upgrade your base installation.
Step 1 : Install the Zaptel Drivers
Ensuring your ports collection is up to date, you can install the Zaptel drivers from /usr/ports/misc/zaptel.
Step 2 : Copy over the Zaptel startup script
Upon installation, the zaptel drivers install a startup script in /usr/local/etc/rc.d. This will likely be alongside your Asterisk startup script, and any others you may already have installed there. The zaptel installation process asks you to rename zaptel.sh.sample to zaptel.sh in order to have the script start properly now that the drivers have been installed.
Step 3 : Edit the Zaptel startup script
For some reason, the zaptel.sh startup script provided doesn't include the necessary lines in order to load and unload the ztdummy.ko kernel module. For our purposes, this is important, since that ztdummy module is the entire reason for installing these drivers. Supposedly ztdummy provides a suitable dummy device which can be used by Asterisk for the objectives we are seeking here. So, we need to add a few lines to this startup script to ensure that ztdummy kernel module gets loaded and unloaded properly.
A few steps are required to get the drivers running. Below are my experiences adding the Zaptel drivers to an existing FreeBSD-based Asterisk PBX in early July 2006. You should also be aware that some reports indicate the Zaptel drivers for FreeBSD only work correctly under FreeBSD v5.x or v6.x, so you may need to upgrade your base installation.
Step 1 : Install the Zaptel Drivers
Ensuring your ports collection is up to date, you can install the Zaptel drivers from /usr/ports/misc/zaptel.
Step 2 : Copy over the Zaptel startup script
Upon installation, the zaptel drivers install a startup script in /usr/local/etc/rc.d. This will likely be alongside your Asterisk startup script, and any others you may already have installed there. The zaptel installation process asks you to rename zaptel.sh.sample to zaptel.sh in order to have the script start properly now that the drivers have been installed.
Step 3 : Edit the Zaptel startup script
For some reason, the zaptel.sh startup script provided doesn't include the necessary lines in order to load and unload the ztdummy.ko kernel module. For our purposes, this is important, since that ztdummy module is the entire reason for installing these drivers. Supposedly ztdummy provides a suitable dummy device which can be used by Asterisk for the objectives we are seeking here. So, we need to add a few lines to this startup script to ensure that ztdummy kernel module gets loaded and unloaded properly.
Asterisk OpenBTS
OpenBTS or Asterisk. Using OpenBTS linked to an open-source voice server running software such as Asterisk, a distributed network now has the ability to make voice calls without going back to the centralized core network of a wireless or wireline carrier. If one hooks a server running OpenBTS to an Asterisk server on a Commotion network, then voice calls via VoIP are now available via the existing GSM radios on the phones, even without using something like Serval.
Thursday, April 21, 2011
Asterisk Pandaboard Open Source Platform
The OMAP 4 platform-based PandaBoard includes an OMAP4430 processor with two ARM® Cortex™-A9 processors (running at 1 GHz each) delivering symmetric multiprocessing (SMP) performance, as well as rich multimedia and 3D graphics support. The board also includes WLAN and Bluetooth® connectivity technologies. PandaBoard offers the ideal platform to support projects leveraging all its capabilities within the tight mobile power budget. More “wild” out-of-the-box technical features of PandaBoard include:
Feature Spec
OMAP4430
Dual-core ARM Cortex-A9 MPCore (cores run at 1 GHz each)
Ample MIPS for rich applications
Multimedia
Full 1080p30 multi-standard HD record and playback
OpenGL® ES v2.0 graphics engine
Memory
1GB LPDDR2 (DRAM)
Wireless connectivity
WiLink™ 6.0 (802.11b/g/n, Bluetooth® v2.1 +EDR)
Display
HDMI v1.3 connector for HD display
DVI-D connector for simultaneous dual display
Expansion
USB host
USB OTG host
Support
Open-source hardware design
Open-source community engagement
Feature Spec
OMAP4430
Dual-core ARM Cortex-A9 MPCore (cores run at 1 GHz each)
Ample MIPS for rich applications
Multimedia
Full 1080p30 multi-standard HD record and playback
OpenGL® ES v2.0 graphics engine
Memory
1GB LPDDR2 (DRAM)
Wireless connectivity
WiLink™ 6.0 (802.11b/g/n, Bluetooth® v2.1 +EDR)
Display
HDMI v1.3 connector for HD display
DVI-D connector for simultaneous dual display
Expansion
USB host
USB OTG host
Support
Open-source hardware design
Open-source community engagement
Wednesday, April 20, 2011
Asterisk with JumpBox
Asterisk is a toolkit for building telephony-based applications. It contains all the necessary components to build complex phone interfaces with features like IVR trees, speech recognition and synthesis, call recording and routing. With over two million users it's the world's most popular open source telephony project.
JumpBox provides very simple to use virtual appliances based around Open Source software. Any JumpBox will run on VMWare as well as Parallels Desktop and Server, Microsoft Virtual PC, Microsoft Virtual Server, Hyper-V, VirtualBox and Xen Open Source. With a Pro subscription you can also run on Amazon EC2 and with the features of the JumpBox platform can easily move your installation from a local install to an install in the cloud and back at any time.
A JumpBox requires one quick step of configuration via a web browser with no need to touch a command line. Each JumpBox also includes a simple web based management interface that includes the JumpBox Backup System. The JumpBox Backup System allows you to export the state to NFS, Windows File Sharing or Amazon S3 (This provides an easy means offsite backup and entry-level disaster recovery).
JumpBox appliances are updated on a regular basis with all the latest components of the software stack along with application fixes.
JumpBox provides very simple to use virtual appliances based around Open Source software. Any JumpBox will run on VMWare as well as Parallels Desktop and Server, Microsoft Virtual PC, Microsoft Virtual Server, Hyper-V, VirtualBox and Xen Open Source. With a Pro subscription you can also run on Amazon EC2 and with the features of the JumpBox platform can easily move your installation from a local install to an install in the cloud and back at any time.
A JumpBox requires one quick step of configuration via a web browser with no need to touch a command line. Each JumpBox also includes a simple web based management interface that includes the JumpBox Backup System. The JumpBox Backup System allows you to export the state to NFS, Windows File Sharing or Amazon S3 (This provides an easy means offsite backup and entry-level disaster recovery).
JumpBox appliances are updated on a regular basis with all the latest components of the software stack along with application fixes.
Tuesday, April 19, 2011
Asterisk asterCC
asterCC is a software package, in which we provide two asterisk solutions for now:
asterCRM, an open source contact center solution for asterisk
asterBilling, a realtime billing solution for asterisk, could be used for hosted callshop, asterisk pbx billing
all above in the package will use a linux daemon script named astercc, which could grab realtime CDR from asterisk, and it’s welcomed that if you want to develop your own application based astercc daemon.
The astercc daemon connect to asterisk via AMI(Asterisk Management Interface), so it could work with all kinds of asterisk solutions, and would not affect your original asterisk.
Here’re the benefits of solutions using astercc daemon
brilliant performance
we have tested that astercc could support more than 240 calls in asterisk
good compatibility
astercc could work with all asterisk based solutions, and it support both asterisk 1.2.X and 1.4.X
distributed solution
no need install astercc daemon on your asterisk server, even an embedded asterisk product could use astercc for expand
asterCRM, an open source contact center solution for asterisk
asterBilling, a realtime billing solution for asterisk, could be used for hosted callshop, asterisk pbx billing
all above in the package will use a linux daemon script named astercc, which could grab realtime CDR from asterisk, and it’s welcomed that if you want to develop your own application based astercc daemon.
The astercc daemon connect to asterisk via AMI(Asterisk Management Interface), so it could work with all kinds of asterisk solutions, and would not affect your original asterisk.
Here’re the benefits of solutions using astercc daemon
brilliant performance
we have tested that astercc could support more than 240 calls in asterisk
good compatibility
astercc could work with all asterisk based solutions, and it support both asterisk 1.2.X and 1.4.X
distributed solution
no need install astercc daemon on your asterisk server, even an embedded asterisk product could use astercc for expand
Asterisk OpenERP
This module has 2 main features. The first one : it adds a dial button in the partner address view so that users can directly dial a phone number through Asterisk. This feature is usually known as click2dial. Here is how it works :
In OpenERP, the user clicks on the dial button next to a phone number field in the Partner address view.
Asterisk makes the user's phone ring.
The user answers his own phone (if he doesn't, the process stops here).
Asterisk dials the phone number found in OpenERP in place of the user.
If the remote party answers, the user can talk to his correspondent.
The second main feature is : it adds the ability to show the name of the calling party on incoming phone calls if the presented phone number is present in the Partner addresses of OpenERP. Here is how it works :
On incoming phone calls, the Asterisk dialplan executes an AGI get_cid_name_timeout.sh.
The get_cid_name_timeout.sh script calls the get_cid_name.py script with a short timeout.
The get_cid_name.py script will make an XML-RPC request on the OpenERP server to try to find the name of the person corresponding to the phone number presented by the calling party.
If it finds the name, it is add as CallerID name to the call, so as to be presented on the IP phone of the user.
In OpenERP, the user clicks on the dial button next to a phone number field in the Partner address view.
Asterisk makes the user's phone ring.
The user answers his own phone (if he doesn't, the process stops here).
Asterisk dials the phone number found in OpenERP in place of the user.
If the remote party answers, the user can talk to his correspondent.
The second main feature is : it adds the ability to show the name of the calling party on incoming phone calls if the presented phone number is present in the Partner addresses of OpenERP. Here is how it works :
On incoming phone calls, the Asterisk dialplan executes an AGI get_cid_name_timeout.sh.
The get_cid_name_timeout.sh script calls the get_cid_name.py script with a short timeout.
The get_cid_name.py script will make an XML-RPC request on the OpenERP server to try to find the name of the person corresponding to the phone number presented by the calling party.
If it finds the name, it is add as CallerID name to the call, so as to be presented on the IP phone of the user.
Call Tracking Software
SourceTrak Basic
Place a unique local or toll-free phone number anywhere (e.g. yellow pages, infomercial, direct mail) and view real-time reports online about the number of calls from each ad.
SourceTrak Dynamic
Assign unique phone numbers to specific online traffic sources (e.g. PPC, SEO, keywords, domains, ads). Based on the referral source of a visitor to your website, Ifbyphone dynamically displays the phone number associated with that traffic source. View reports online to analyze conversion rates for different keywords, search engines, or online ads.
SourceTrak Dynamic Groups
For larger online marketing campaigns, assign a group of phone numbers to a group of traffic sources (e.g. keywords within a PPC Adgroup). Based on the referral source of a visitor to your website, Ifbyphone dynamically displays a phone number associated with that group of traffic sources.
SourceTrak Session
Use Session based Call Tracking to track an unlimited number of traffic sources to the keyword-level without preassigning phone numbers to specific sources. This solution provides the easiest configuration but limits your control over custom greetings and call routing.
Place a unique local or toll-free phone number anywhere (e.g. yellow pages, infomercial, direct mail) and view real-time reports online about the number of calls from each ad.
SourceTrak Dynamic
Assign unique phone numbers to specific online traffic sources (e.g. PPC, SEO, keywords, domains, ads). Based on the referral source of a visitor to your website, Ifbyphone dynamically displays the phone number associated with that traffic source. View reports online to analyze conversion rates for different keywords, search engines, or online ads.
SourceTrak Dynamic Groups
For larger online marketing campaigns, assign a group of phone numbers to a group of traffic sources (e.g. keywords within a PPC Adgroup). Based on the referral source of a visitor to your website, Ifbyphone dynamically displays a phone number associated with that group of traffic sources.
SourceTrak Session
Use Session based Call Tracking to track an unlimited number of traffic sources to the keyword-level without preassigning phone numbers to specific sources. This solution provides the easiest configuration but limits your control over custom greetings and call routing.
Secure Remote Access
Secure application publishing and secure socket layer (SSL) virtual private network (VPN) enable organizations to make their key end-user focused IT infrastructure components—such as computers running Microsoft Exchange Server, Microsoft SharePoint Portal Server, file shares, and other resource servers—accessible to remote users over a fully protected application-layer connection. By pre-authenticating users before they gain access to any published servers, inspecting even encrypted traffic at the application layer in a stateful manner, and providing automated publishing tools, Forefront edge security and access products make it easier to provide security for corporate applications accessed over the Internet.
HOB PPP Tunnel
The HOB PPP Tunnel is part of the comprehensive security solution HOB RD VPN.
With the PPP Tunnel the user gets complete network access to all resources in the central corporate network.
This access is also bi-directional. i.e., from the central network, one can access all resources on the client as well.
Compression makes this access highly performant. Strong authentication and SSL encryption ensure that the access is secure.
Using SSL as the transmission protocol has the advantage that it is supported by all network devices; this is not the case with comparable IPsec solutions.
Momentary interruptions in the connection to the client trigger an automatic reconnect.
Currently, the HOB PPP Tunnel supports Windows Vista as the operating system on the client; support for Apple MAC, Linux, FreeBSD or Solaris will be available soon.
The PPP Tunnel in HOB RD VPN also has the abbreviation HOB-PPP-T1.
With the PPP Tunnel the user gets complete network access to all resources in the central corporate network.
This access is also bi-directional. i.e., from the central network, one can access all resources on the client as well.
Compression makes this access highly performant. Strong authentication and SSL encryption ensure that the access is secure.
Using SSL as the transmission protocol has the advantage that it is supported by all network devices; this is not the case with comparable IPsec solutions.
Momentary interruptions in the connection to the client trigger an automatic reconnect.
Currently, the HOB PPP Tunnel supports Windows Vista as the operating system on the client; support for Apple MAC, Linux, FreeBSD or Solaris will be available soon.
The PPP Tunnel in HOB RD VPN also has the abbreviation HOB-PPP-T1.
Oslec Echo Canceller
Oslec is an open source high performance line echo canceller. When used with Asterisk it works well on lines where the built-in Zaptel echo canceller fails. No tweaks like rxgain/txgain or fxotrain are required. Oslec is supplied as GPL licensed C source code and is free as in speech.
Oslec partially complies with the G168 standard and runs in real time on x86 and Blackfin platforms. Patches exist to allow any Zaptel or DAHDI compatible hardware to use Oslec. It has been successfully tested on hardware ranging from single port X100P FXO cards to dual E1 systems.
Oslec partially complies with the G168 standard and runs in real time on x86 and Blackfin platforms. Patches exist to allow any Zaptel or DAHDI compatible hardware to use Oslec. It has been successfully tested on hardware ranging from single port X100P FXO cards to dual E1 systems.
Asterisk PBX Integration Zimlet
Asterisk PBX Integration Zimlet is an Extension for Zimbra Collaboration Suite. The Zimlet does Interface with the Asterisk Manager Interface to integrate with Asterisk PBX. The main focus is dial-on-click for Phone numbers inside Contacts and Emails.
OpenBTS Project
The OpenBTS Project is an effort to construct an open-source GSM basestation using the USRP and the Asterisk VoIP PBX. Our goal is to enable a new type of hybrid GSM/VoIP cellular network for greenfield deployments in the developing world.
Druid open source unified communications platform
Druid is an open source unified communications platform, built around technology such as Asterisk, IMAP, XMPP. Druid gives your organization access to the best available IP communications platform that bringing together voicemail, VOIP, mobile phone, faxes and instant messaging.
Vicidial inbound/outbound call center
VICIDIAL is a set of programs that are designed to interact with the Asterisk Open-Source PBX Phone system to act as a complete inbound/outbound call center suite.
The agent interface is an interactive set of web pages that work through a web browser to give real-time information and functionality with nothing more than an internet browser on the client computer.
The agent interface is an interactive set of web pages that work through a web browser to give real-time information and functionality with nothing more than an internet browser on the client computer.
Friday, April 15, 2011
Asterisk Queue/ CDR Log Analyzer
The Asterisk Queue (and CDR) Log Analyzer is a set of PHP scripts which allow selecting, listing and graphing of records from the Asterisk Queue and CDR logs via a WEB interface.
For easier access to select specific log records, the Queue and CDR logs need to be in a MySQL database. Asterisk itself records (specified in a conf file) CDR data into a MySQL database table. A Python utility program called loadq.py is provided with this package which can be used to load queue log records (as they are created) into a MySQL database table.
For easier access to select specific log records, the Queue and CDR logs need to be in a MySQL database. Asterisk itself records (specified in a conf file) CDR data into a MySQL database table. A Python utility program called loadq.py is provided with this package which can be used to load queue log records (as they are created) into a MySQL database table.
Asterisk XiVO PBX
XiVO is a full PBX solution based on Asterisk with a user-friendly web interface, provisioning tools for many types of phones, CTI daemon and CTI client for Windows, Linux and MacOS. All solutions are released under GPLv3. Currently only released in French, but easy to translate in other languages, XiVO provides administrators with a simple to configure phone system.
You can find a demo on https://demo.xivo.fr with login root and password proformatique so you can try it out for yourself.
On the first page you have a dashboard for monitoring all processes and you can choose PBX on the menu to administer your PBX. It's not based on Freepbx or other existing software, it's a full development from scratch.
You can find a demo on https://demo.xivo.fr with login root and password proformatique so you can try it out for yourself.
On the first page you have a dashboard for monitoring all processes and you can choose PBX on the menu to administer your PBX. It's not based on Freepbx or other existing software, it's a full development from scratch.
Asterisk Web/PHP Event Monitor
The Asterisk Event Monitor WEB/PHP Interface was created to view the current state of Asterisk and all Asterisk Events via a WEB interface. It does not poll Asterisk for these events, instead it collects them in a MySql database via an Asterisk Manager API python script. AJAX (Javascript) is used to display the events from the database almost as they occur. All code is released under the GNU GPL license.
Asterisk With BroadVoice
BroadVoice™ Internet phone service allows residential and business customers to use their cable modem, DSL modem, or other BroadBand Internet connection to make and receive Voice over IP (VoIP) phone calls using an ordinary touchtone telephone. Bring Your Own Device™ (BYOD™) plans allow customers to connect their own SIP devices, including IP phones, softphones, and Asterisk PBXs. BroadVoice utilizes our SecureSIP™ technology to ensure accurate connectivity throughout the user experience. SmartSIP™ technology by BroadVoice is used to optimize the routing of network voice traffic, provides the best possible quality voice transmission for each customer's phone device, and automatically configures BroadVoice Authorized BYOD™ devices
Thursday, April 14, 2011
Auto-Dialer Application for Switchvox or Asterisk
StarCONNECT, Starnet’s Managed Autodialer offering, serves to address the needs of smaller organizations who want to improve customer retention, satisfaction and loyalty but who are not in a position to make a large upfront capital expenditure to get started.
StarCONNECT provides customers the software development, implementation and ongoing management of the outbound dialing operation as a ‘Technology As A Service’ offering.
With StarCONNECT organizations can have information from their customer database be used to generate the appropriate outbound reach, relaying information to customers that the customers need to interact more effectively with the organization.
As a Managed Service, the organization's initial capital outlay is minimized and all ongoing support is covered by the competitive monthly fee.
For those organizations that want to manage their own dialing environment, StarCONNECT is available for purchase under a perpetual use license as well.
StarCONNECT provides customers the software development, implementation and ongoing management of the outbound dialing operation as a ‘Technology As A Service’ offering.
With StarCONNECT organizations can have information from their customer database be used to generate the appropriate outbound reach, relaying information to customers that the customers need to interact more effectively with the organization.
As a Managed Service, the organization's initial capital outlay is minimized and all ongoing support is covered by the competitive monthly fee.
For those organizations that want to manage their own dialing environment, StarCONNECT is available for purchase under a perpetual use license as well.
Wednesday, April 13, 2011
Microbase CTI Pro
Microbase CTI Pro is a software system that provides Asterisk PBX with advanced features like dynamic prioritized call queues for call distribution, modular IVR system, detailed statistics, power (progressive) dialer, campaign management and sophisticated monitoring and administration capabilities.
It is a communications software which enables automated handling of inbound and outbound traffic, allows the integration of these functions into 3rd party applications, keeps detailed statistics for all associated activity and provides interfaces for real-time monitoring of all the aspects of its operation.
The operation of Microbase CTI is divided into 4 categories: agent management, the IVR system, inbound call handling and outbound call handling. There are also two monitoring facilities, the logging and statistics database schema for offline monitoring and the real-time operation monitoring system (operating through performance counters).
Microbase CTI’s basic advantage is that it has been designed from scratch to be simple, easy to use, problem free and covering all basic CTI needs of any organization, however complicated this may be.
In a few words, Microbase CTI is the software needed by an Asterisk PBX to be able to offer all the advanced features which are required by a constantly increasing number of customers.
The feature of Predictive Dialing is also available with the Microbase CTI Enterprise suite.
It is a communications software which enables automated handling of inbound and outbound traffic, allows the integration of these functions into 3rd party applications, keeps detailed statistics for all associated activity and provides interfaces for real-time monitoring of all the aspects of its operation.
The operation of Microbase CTI is divided into 4 categories: agent management, the IVR system, inbound call handling and outbound call handling. There are also two monitoring facilities, the logging and statistics database schema for offline monitoring and the real-time operation monitoring system (operating through performance counters).
Microbase CTI’s basic advantage is that it has been designed from scratch to be simple, easy to use, problem free and covering all basic CTI needs of any organization, however complicated this may be.
In a few words, Microbase CTI is the software needed by an Asterisk PBX to be able to offer all the advanced features which are required by a constantly increasing number of customers.
The feature of Predictive Dialing is also available with the Microbase CTI Enterprise suite.
AsterUVX - VoiceXML IVR Platform for Asterisk
AsterUVX is SoftSyl’s IVR platform that is powered by VoiceXML to natively integrate with the Asterisk open source telephony system. AsterUVX transforms Asterisk into a robust enterprise-grade IVR solution. This combination is the most cost efficient way to create and run any type of call center while maintaining the strict industry standards required by Fortune 500 companies. This platform allows IVR applications that leverage use of open standards to go live, bringing call center technology to real-world companies. Call volumes in excess of several millions to billions per year can easily be handled by this product, allowing for enterprise grade functionality at an affordable price.
SoftSyl's solutions are powered by a combination of VoiceXML, Voice Over IP (VoIP), telephony and web technologies all working together to create a final powerhouse to handle and route any imaginable call volume.
SoftSyl's solutions are powered by a combination of VoiceXML, Voice Over IP (VoIP), telephony and web technologies all working together to create a final powerhouse to handle and route any imaginable call volume.
Monday, April 11, 2011
Ingate System - Ingate Siparator
Ingate SIParator® is a device that connects to an existing network firewall to seamlessly enable SIP Communications (Session Initiation Protocol). While traditional firewalls block SIP traffic – including mission-critical applications like Voice over IP – the SIParator® resolves this problem, working in tandem with your current security solutions. Ingate SIParators® are available in a range of models to meet the needs of the entire enterprise market.
Ingate Product - Internet Telephony Service provider
With an Ingate product at the customer network edge,
Internet Telephony Service Provider (ITSP) can interface with any SIP-enabled IP-PBX,
eliminating the need to achieve interoperability with every IP-PBX vendor and address the
widest number of customers. Ingate Firewalls and SIParators serve as a “normalization
engine” between the IP-PBX and ITSP, paving the way for a successful SIP trunk roll-out.
Ingate products have proven seamless interoperability with the leading IP-PBX
vendors, ensuring that SIP trunks are deployed quickly and with a minimum of effort.
Ingate’s SIP proxy integrated with its NAT and fi rewall provides advanced routing
capabilities for deploying SIP trunks. Combined with the optional Back-to-Back User Agent
functionality, all service provider requirements for authentication and signaling from any
IP-PBX can be supported.
Internet Telephony Service Provider (ITSP) can interface with any SIP-enabled IP-PBX,
eliminating the need to achieve interoperability with every IP-PBX vendor and address the
widest number of customers. Ingate Firewalls and SIParators serve as a “normalization
engine” between the IP-PBX and ITSP, paving the way for a successful SIP trunk roll-out.
Ingate products have proven seamless interoperability with the leading IP-PBX
vendors, ensuring that SIP trunks are deployed quickly and with a minimum of effort.
Ingate’s SIP proxy integrated with its NAT and fi rewall provides advanced routing
capabilities for deploying SIP trunks. Combined with the optional Back-to-Back User Agent
functionality, all service provider requirements for authentication and signaling from any
IP-PBX can be supported.
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