PRI is an abbreviation for Primary Rate Interface and is a telecommunication standard for carrying multiple data or voice DS0 transmissions between 2 physical locations.
All data and voice channels are (ISDN) and operate at 64 kbit/s.
North America and Japan use a T1 of 23 B channels and one D channel. Europe, Australia and most of the rest of the world use a slightly higher capacity E1 of 30 B channels and one D channel.
A B-channel is used to transmit the voice or data and is sometimes also called a user channel, a D channel is used for control messages and signalling.
In some T1 configurations, no D-channels might be used, instead the signalling will be sent inband over the B channels, this is called in-band signalling or bit robbing, resulting in lower transmission rates than the E-carrier system. This resulted in many US ISDN installations only having an effective data rate of 56 kbit/s over a nominal 64 kbit/s channel. See also A&B. This depends on the framing format used.
In the states its common to order a fractional T1, with less than 23 B channels.
1.3. signalling
On both E1 and T1, one timeslot is usually reserved for a D-channel for call setup and call teardown. (called signalling).
- CAS: Channel associated Signalling: with this kind of signalling, a set of bits is used to replicate opening and closing the circuit (as if picking up the telephone receiver and pulsing digits on a rotary phone), or using tone signalling which is passed through on the voice circuits themselves.
- CCS: Common Channel signalling: A more recent kind of signalling, (ISDN signalling and ss7 signalling are a subgroup of CCS signalling.) In this kind of signalling, short messages are sent over the signalling channel, with more information about the call, including caller ID, type of transmission required, etc. etc.
1.4. Framing
- HDB3
- AMI: Aternate Mark Inversion
- B8ZS:
1.5. Timing or clock sources
A PRI connection needs a timing device on one of both ends.
A PRI line can be clocked internally or can be clocked by the telco.
Sunday, December 11, 2011
Sunday, December 4, 2011
USB 2.0 port support heavy telephony traffic
The USB 2.0 interface provides a theoretical speed of 480,000,000 bits per second. A typical uncompressed phone conversation uses about 64,000 bits per second per direction, plus some overhead; in total less than 200,000 bits per second per phone call is used. Thus, the theoretical concurrent number of conversations that the USB 2.0 interface can handle is 480,000,000 divided by 200,000: roughly 2,400 calls for a single USB 2.0 port.
This is a theoretical number, of course, and the typical host processor will not be able to handle this amount of concurrent calls, but it gives a clear answer to the question: "Can the USB port support heavy telephony traffic with multiple Astribanks?"
Connecting multiple Astribank units to a single USB 2.0 port is easy: simply use a USB 2.0 Hub to connect many Astribank units.
This is a theoretical number, of course, and the typical host processor will not be able to handle this amount of concurrent calls, but it gives a clear answer to the question: "Can the USB port support heavy telephony traffic with multiple Astribanks?"
Connecting multiple Astribank units to a single USB 2.0 port is easy: simply use a USB 2.0 Hub to connect many Astribank units.
Xorcom Astribank XPP Technology
Xorcom has developed a revolutionary concept in telephony interfaces for Asterisk systems which we call Astribank XPP Technology. Using USB 2.0 ports to connect to any Asterisk server, Astribank eliminates the requirement for a PCI (E1/T1) card, and even for PCI slots.
The USB 2.0 interface provides a theoretical speed of 480 Mbits per second, which for typical uncompressed phone conversations translates to roughly 2,400 calls per port. The USB architecture employed by Xorcom products affords many advantages, including the ability to build large systems that support hundreds of analog extensions, rapid set-up and easy maintenance.
The USB 2.0 interface provides a theoretical speed of 480 Mbits per second, which for typical uncompressed phone conversations translates to roughly 2,400 calls per port. The USB architecture employed by Xorcom products affords many advantages, including the ability to build large systems that support hundreds of analog extensions, rapid set-up and easy maintenance.
Monday, November 28, 2011
Asterisk ACDs Call Centers
Automatic Call Distributors (ACDs) allow call centers to handle thousands of simultaneous calls, routing them to agents based on caller input, dialed number, load and other factors. ACD systems typically cost tens if not hundreds of thousands of dollars and require specialized training to install and operate. With Asterisk you can build a powerful ACD for the cost of the server hardware and phones.
Asterisk Supported Protocols
Asterisk® supports a wide range of protocols for the handling and transmission of voice over traditional telephony interfaces including H.323, Session Initiation Protocol (SIP), Media Gateway Control Protocol (MGCP), and Skinny Client Control Protocol (SCCP).
Using the Inter-Asterisk eXchange (IAX™) Voice over IP protocol Asterisk® merges voice and data traffic seamlessly across disparate networks. The use of Packet Voice allows Asterisk® to send data such as URL information and images in-line with voice traffic, allowing advanced integration of information.
Asterisk® provides a central switching core, with four APIs for modular loading of telephony applications, hardware interfaces, file format handling, and codecs. It allows for transparent switching between all supported interfaces, allowing it to tie together a diverse mixture of telephony systems into a single switching network.
Using the Inter-Asterisk eXchange (IAX™) Voice over IP protocol Asterisk® merges voice and data traffic seamlessly across disparate networks. The use of Packet Voice allows Asterisk® to send data such as URL information and images in-line with voice traffic, allowing advanced integration of information.
Asterisk® provides a central switching core, with four APIs for modular loading of telephony applications, hardware interfaces, file format handling, and codecs. It allows for transparent switching between all supported interfaces, allowing it to tie together a diverse mixture of telephony systems into a single switching network.
Monday, October 31, 2011
Systm Episode 5 - Asterisk
What is Asterisk? Asterisk is an insanely powerful, yet easy to use, open source voip telephony toolkit (server software) that runs on virtually any platform. It allows anyone to manage VoIP services in their own home in more powerful ways than the most expensive PBX (public branch exchange). For example, it can serve as a gateway from a VoIP service to your copper phone lines, forward calls to your mobile device from wherever you are, allow you to create VoIP connections to anyone on the Internet, manage voicemail, prompts ... a virtually limitless capability depending on how far you want to take it.
Sunday, October 30, 2011
Asterisk 10 Released
Asterisk 10, also known as Asterisk version 1.10, is a substantial update focused on the platform's media engine, according to Steve Sokol, Asterisk marketing director at Digium. Developers added a number of new codecs to the platform, including Skype's SILK codec, 32kHz Speex support and pass-through support for CELT, and it can support all types of audio. According to Digium, Asterisk previously operated on 8kHz and 16kHz sampled audio, but can now support 8, 12, 16, 24, 32, 44.1, 48, 96 and 192 kHz rates for audio.
"'The new version is actually capable of negotiating the specific details of media,' Sokol explained. 'That's the biggest thing: a rip-and-replace of the media engine so that it's now capable of doing any kind of media or any video at almost any quality rate.
"'The new version is actually capable of negotiating the specific details of media,' Sokol explained. 'That's the biggest thing: a rip-and-replace of the media engine so that it's now capable of doing any kind of media or any video at almost any quality rate.
Thursday, October 27, 2011
Asterisk 1.8.8.0-rc2 Now Available
The Asterisk Development Team has announced the second release candidate of
Asterisk 1.8.8.0. This release candidate is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
Asterisk 1.8.8.0. This release candidate is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
Saturday, October 22, 2011
Asterisk with FXO/FXS interface cards
FXO/FXS interface cards
The Wildcard X100P is an FXO device which will use fxs signalling with Asterisk.
The Wildcard S100U is an FXS device which will use fxo signalling with Asterisk.
The Wildcard TDM400P may have a combination of FXO and FXS modules installed in its four slots. Note that the Wildcard TDM400P may be referred to by different names according to how many modules and what kind of modules are installed on it. If it has one FXS module on it then it is called TDM10B. If it has two FXS modules then it is TDM20B and so on. TDM400P is sometimes used to refer to just the board with no modules on it.
Configuration for the Wildcard X100P:
fxsks=1: if you have one Wildcard X100P installed
fxsks=1-2: if you have two Wildcard X100P cards installed
fxsks=1-3: if you have three Wildcard X100P cards installed, and so on.
Configuration for the Wildcard S100U:
fxoks=1: for one card
fxoks=1-2: for two cards
fxoks=1-3: for three cards, and so on.
Configuration for the Wildcard TDM400P:
If your card is a TDM10B (one FXS module), put: fxoks=1
If your card is a TDM20B (two FXS modules), put: fxoks=1-2
If your card is a TDM30B (three FXS modules), put: fxoks=1-3
If your card is a TDM04B (four FXO modules), put: fxsks=1-4
If your card has modules FXO, FXO, FXS, FXS (in that order), put:
fxsks=1-2
fxoks=3-4
If your card has modules FXS, FXO, FXO, FXS (in that order), put:
fxoks=1,4
fxsks=2-3
If your card has modules FXS, (empty), FXO, FXS (in that order), put:
fxoks=1,4
fxsks=3
If you have one Wildcard TDM40B and one Wildcard TDM22B installed, put:
fxoks=1-4 (for the TDM40B)
fxsks=5,8 (for the two FXO modules on your TDM22B)
fxoks=6,7 (for the two FXS modules on your TDM22B)
NOTE: the names are reversed. For FX0 modules you use fxsks= and for FXS modules you use fxoks=
The Wildcard X100P is an FXO device which will use fxs signalling with Asterisk.
The Wildcard S100U is an FXS device which will use fxo signalling with Asterisk.
The Wildcard TDM400P may have a combination of FXO and FXS modules installed in its four slots. Note that the Wildcard TDM400P may be referred to by different names according to how many modules and what kind of modules are installed on it. If it has one FXS module on it then it is called TDM10B. If it has two FXS modules then it is TDM20B and so on. TDM400P is sometimes used to refer to just the board with no modules on it.
Configuration for the Wildcard X100P:
fxsks=1: if you have one Wildcard X100P installed
fxsks=1-2: if you have two Wildcard X100P cards installed
fxsks=1-3: if you have three Wildcard X100P cards installed, and so on.
Configuration for the Wildcard S100U:
fxoks=1: for one card
fxoks=1-2: for two cards
fxoks=1-3: for three cards, and so on.
Configuration for the Wildcard TDM400P:
If your card is a TDM10B (one FXS module), put: fxoks=1
If your card is a TDM20B (two FXS modules), put: fxoks=1-2
If your card is a TDM30B (three FXS modules), put: fxoks=1-3
If your card is a TDM04B (four FXO modules), put: fxsks=1-4
If your card has modules FXO, FXO, FXS, FXS (in that order), put:
fxsks=1-2
fxoks=3-4
If your card has modules FXS, FXO, FXO, FXS (in that order), put:
fxoks=1,4
fxsks=2-3
If your card has modules FXS, (empty), FXO, FXS (in that order), put:
fxoks=1,4
fxsks=3
If you have one Wildcard TDM40B and one Wildcard TDM22B installed, put:
fxoks=1-4 (for the TDM40B)
fxsks=5,8 (for the two FXO modules on your TDM22B)
fxoks=6,7 (for the two FXS modules on your TDM22B)
NOTE: the names are reversed. For FX0 modules you use fxsks= and for FXS modules you use fxoks=
Asterisk Sip Nat
If a peer is configured with nat=yes, it causes Asterisk to ignore the address information in the SIP and SDP headers from this peer, and reply to the sender's IP address and port. nat=yes enables a form of Symmetric RTP and SIP Comedia mode in Asterisk.
Comedia mode means that Asterisk will ignore the IP and port in the received SDP from the peer and will wait for incoming RTP. This RTP should arrive to the port that Asterisk replied in the "200 OK" SDP. After that, Asterisk already knows where to send its RTP.
This make communication possible with UA's behind NAT which don't solve NAT problem in client side (STUN, ICE, ALG enabled router, etc). This options works properly in conjuntion with qualify=yes option in order to keep open the connection from Asterisk to the peer behind NAT.
Asterisk 1.8:
The 'nat' option has now been been changed to have yes, no, force_rport, and comedia as valid values. Setting it to yes forces RFC 3581 behavior and enables symmetric RTP support. Setting it to no only enables RFC 3581 behavior if the remote side requests it and disables symmetric RTP support. Setting it to force_rport forces RFC 3581 behavior and disables symmetric RTP support. Setting it to comedia enables RFC 3581 behavior if the remote side requests it and enables symmetric RTP support.
Comedia mode means that Asterisk will ignore the IP and port in the received SDP from the peer and will wait for incoming RTP. This RTP should arrive to the port that Asterisk replied in the "200 OK" SDP. After that, Asterisk already knows where to send its RTP.
This make communication possible with UA's behind NAT which don't solve NAT problem in client side (STUN, ICE, ALG enabled router, etc). This options works properly in conjuntion with qualify=yes option in order to keep open the connection from Asterisk to the peer behind NAT.
Asterisk 1.8:
The 'nat' option has now been been changed to have yes, no, force_rport, and comedia as valid values. Setting it to yes forces RFC 3581 behavior and enables symmetric RTP support. Setting it to no only enables RFC 3581 behavior if the remote side requests it and disables symmetric RTP support. Setting it to force_rport forces RFC 3581 behavior and disables symmetric RTP support. Setting it to comedia enables RFC 3581 behavior if the remote side requests it and enables symmetric RTP support.
Wednesday, October 19, 2011
Essentra BAX
Service providers are looking for ways to leverage the burgeoning broadband access market in order to introduce new revenue-generating IP communications services. Enabling the delivery of residential and hosted enterprise VoIP services over any broadband infrastructure, VocalTec’s Essentra BAX Broadband Access Server helps service providers take advantage of evolving IP opportunities.
Based on over a decade of leadership in VoIP solutions, Essentra BAX, VoIP Application Server, offers service providers a cost-effective entry into broadband VoIP services, with the capability of scaling up to millions of subscribers over time. The carrier-grade turnkey solution supports both traditional subscriber calling features, including call waiting and call forward, and a range of cutting-edge features such as call screening and click-to-dial.
For enterprise customers, Essentra BAX supports VoIP virtual private networks (VPNs) and an array of IP-Centrex features. Equipped with a Web-based subscriber self-provisioning interface, the access server enables subscribers to control their own services, thereby reducing the operating expenses (OPEX) of service providers.
Based on over a decade of leadership in VoIP solutions, Essentra BAX, VoIP Application Server, offers service providers a cost-effective entry into broadband VoIP services, with the capability of scaling up to millions of subscribers over time. The carrier-grade turnkey solution supports both traditional subscriber calling features, including call waiting and call forward, and a range of cutting-edge features such as call screening and click-to-dial.
For enterprise customers, Essentra BAX supports VoIP virtual private networks (VPNs) and an array of IP-Centrex features. Equipped with a Web-based subscriber self-provisioning interface, the access server enables subscribers to control their own services, thereby reducing the operating expenses (OPEX) of service providers.
Wednesday, October 12, 2011
OpenVBX
The open-source nature of OpenVBX can be a little confusing. Like WordPress, OpenVBX is a downloadable software package that you upload to your own server and are free to tweak as much as you’d like. However, unlike WordPress, which offers a hosted solution at WordPress.com, Twilio isn’t offering a hosted service yet (though it’s quite obvious that they’re working on launching one in the future). For the time being you’ll have to host it yourself, though Lawson notes that Dreamhost currently has a 1-click install setup, which should make things pretty simple.
Of course, OpenVBX relies heavily on Twilio’s service, tapping into Twilio SMS and telephony APIs. When you use OpenVBX you’re charged Twilio’s normal rates, which run $1 per phone line per month ($2 for a toll-free number) with usage charges of 3 cents per minute (or SMS). Lawson says that it is technically feasible for someone to use OpenVBX with a backend other than Twilio, but that it would take quite a bit of configuration.
Twilio has always been focused on catering to developers, and OpenVBX isn’t any different. While the product will obviously appeal to businesses, Twilio is encouraging developers to take the technology and make it their own by integrating tweaks and plugins. For example, an enterprising developer could tweak OpenVBX so that it’s perfect for restaurants, and then resell it as their own service. Twilio still gets paid through their per-minute and phone line charges, and the developer can charge a hefty premium on top of it.
Of course, OpenVBX relies heavily on Twilio’s service, tapping into Twilio SMS and telephony APIs. When you use OpenVBX you’re charged Twilio’s normal rates, which run $1 per phone line per month ($2 for a toll-free number) with usage charges of 3 cents per minute (or SMS). Lawson says that it is technically feasible for someone to use OpenVBX with a backend other than Twilio, but that it would take quite a bit of configuration.
Twilio has always been focused on catering to developers, and OpenVBX isn’t any different. While the product will obviously appeal to businesses, Twilio is encouraging developers to take the technology and make it their own by integrating tweaks and plugins. For example, an enterprising developer could tweak OpenVBX so that it’s perfect for restaurants, and then resell it as their own service. Twilio still gets paid through their per-minute and phone line charges, and the developer can charge a hefty premium on top of it.
Tuesday, October 11, 2011
Benefits of Voice and Unified Communications
Voice and Unified Communications combine communications into a single solution to help your employees, customers, suppliers, and partners communicate quickly and easily.
Unified Communications benefits people by connecting teams and information, and helping to enable comprehensive and effective collaborative experiences. This helps your company:
- Connect co-workers, partners, vendors, and customers with the information and expertise they need
- Access and share video on the desktop, on the road, and on-demand, as easily as making a phone call
- Facilitate better team interactions, dynamically bringing together individuals, virtual workgroups, and teams
- Make mobile devices extensions of the corporate network so mobile workers can be productive anywhere
- Innovate across the value chain by integrating collaboration and communications into applications and business processes
Unified Communications benefits people by connecting teams and information, and helping to enable comprehensive and effective collaborative experiences. This helps your company:
- Connect co-workers, partners, vendors, and customers with the information and expertise they need
- Access and share video on the desktop, on the road, and on-demand, as easily as making a phone call
- Facilitate better team interactions, dynamically bringing together individuals, virtual workgroups, and teams
- Make mobile devices extensions of the corporate network so mobile workers can be productive anywhere
- Innovate across the value chain by integrating collaboration and communications into applications and business processes
What is Unify Communication
According to the International Engineering Consortium, unified communications is an industry term used to describe all forms of call and multimedia/cross-media message-management functions controlled by an individual user for both business and social purposes. This includes any enterprise informational or transactional application process that emulates a human user and uses a single, content-independent personal messaging channel (mailbox) for contact access.
The essence of communication is breaking down barriers. In its simplest form, the telephone breaks distance and time barriers so that people can communicate in real time or near real time when they are not together. There are now many other barriers to be overcome. People can use many different devices to communicate (wireless phones, personal digital assistants [PDA], personal computers [PC], thin clients, etc.), and there are now new forms of communication as well, such as instant messaging. The goal of unified communications involves breaking down these barriers so that people using different modes of communication, different media, and different devices can still communicate to anyone, anywhere, at any time.
The essence of communication is breaking down barriers. In its simplest form, the telephone breaks distance and time barriers so that people can communicate in real time or near real time when they are not together. There are now many other barriers to be overcome. People can use many different devices to communicate (wireless phones, personal digital assistants [PDA], personal computers [PC], thin clients, etc.), and there are now new forms of communication as well, such as instant messaging. The goal of unified communications involves breaking down these barriers so that people using different modes of communication, different media, and different devices can still communicate to anyone, anywhere, at any time.
Thursday, October 6, 2011
Asterisk 10.0.0-beta2 Now Available
Submitted by asteriskteam on Tue, 09/27/2011 - 14:41
The Asterisk Development Team is pleased to announce the second beta release of
Asterisk 10.0.0. This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
With the release of the Asterisk 10 branch, the preceding '1.' has been removed
from the version number per the blog post available at
http://blogs.digium.com/2011/07/21/the-evolution-of-asterisk-or-how-we-a...
All interested users of Asterisk are encouraged to participate in the
Asterisk 10 testing process. Please report any issues found to the issue
tracker, https://issues.asterisk.org/jira. It is also very useful to see
successful test reports. Please post those to the asterisk-dev mailing list.
All Asterisk users are invited to participate in the #asterisk-testing
channel on IRC to work together in testing the many parts of Asterisk.
Asterisk 10 is the next major release series of Asterisk. It will be a
Standard support release, similar to Asterisk 1.6.2. For more
information about support time lines for Asterisk releases, see the Asterisk
versions page: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions
A short list of features includes:
T.38 gateway functionality has been added to res_fax.
Protocol independent out-of-call messaging support. Text messages not
associated with an active call can now be routed through the Asterisk
dialplan. SIP and XMPP are supported so far.
New highly optimized and customizable ConfBridge application capable of mixing
audio at sample rates ranging from 8kHz-192kHz
Addition of video_mode option in confbridge.conf to provide basic video
conferencing in the ConfBridge() dialplan application.
Support for defining hints has been added to pbx_lua.
Replacement of Berkeley DB with SQLite for the Asterisk Database (AstDB).
Much, much more!
A full list of new features can be found in the CHANGES file.
http://svnview.digium.com/svn/asterisk/branches/10/CHANGES
For a full list of changes in the current release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-...
Thank you for your continued support of Asterisk!
The Asterisk Development Team is pleased to announce the second beta release of
Asterisk 10.0.0. This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
With the release of the Asterisk 10 branch, the preceding '1.' has been removed
from the version number per the blog post available at
http://blogs.digium.com/2011/07/21/the-evolution-of-asterisk-or-how-we-a...
All interested users of Asterisk are encouraged to participate in the
Asterisk 10 testing process. Please report any issues found to the issue
tracker, https://issues.asterisk.org/jira. It is also very useful to see
successful test reports. Please post those to the asterisk-dev mailing list.
All Asterisk users are invited to participate in the #asterisk-testing
channel on IRC to work together in testing the many parts of Asterisk.
Asterisk 10 is the next major release series of Asterisk. It will be a
Standard support release, similar to Asterisk 1.6.2. For more
information about support time lines for Asterisk releases, see the Asterisk
versions page: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions
A short list of features includes:
T.38 gateway functionality has been added to res_fax.
Protocol independent out-of-call messaging support. Text messages not
associated with an active call can now be routed through the Asterisk
dialplan. SIP and XMPP are supported so far.
New highly optimized and customizable ConfBridge application capable of mixing
audio at sample rates ranging from 8kHz-192kHz
Addition of video_mode option in confbridge.conf to provide basic video
conferencing in the ConfBridge() dialplan application.
Support for defining hints has been added to pbx_lua.
Replacement of Berkeley DB with SQLite for the Asterisk Database (AstDB).
Much, much more!
A full list of new features can be found in the CHANGES file.
http://svnview.digium.com/svn/asterisk/branches/10/CHANGES
For a full list of changes in the current release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-...
Thank you for your continued support of Asterisk!
Asterisk 1.8.7.0 Now Available
Submitted by asteriskteam on Fri, 09/23/2011 - 13:12
The Asterisk Development Team announces the release of Asterisk 1.8.7.0. This
release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.8.7.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
Please note that a significant numbers of changes and fixes have gone into
features.c in this release (call parking, built-in transfers, call pickup,
etc.).
NOTE:
Recently, we were notified that the mechanism included in our Asterisk source
code releases to download and build support for the iLBC codec had stopped
working correctly; a little investigation revealed that this occurred because of
some changes on the ilbcfreeware.org website. These changes occurred as a result
of Google's acquisition of GIPS, who produced (and provided licenses for) the
iLBC codec.
If you are a user of Asterisk and iLBC together, and you've already executed a
license agreement with GIPS, we believe you can continue using iLBC with
Asterisk. If you are a user of Asterisk and iLBC together, but you had not
executed a license agreement with GIPS, we encourage you to research the
situation and consult with your own legal representatives to determine what
actions you may want to take (or avoid taking).
More information is available on the Asterisk blog:
http://blogs.asterisk.org/2011/09/19/ilbc-support-in-asterisk-after-goog...
The following is a sample of the issues resolved in this release:
Added the 'storesipcause' option to sip.conf to allow the user to disable the
setting of HASH(SIP_CAUSE,) on the channel. Having chan_sip set
HASH(SIP_CAUSE,) on the channel carries a significant performance
penalty because of the usage of the MASTER_CHANNEL() dialplan function.
We've decided to disable this feature by default in future 1.8 versions. This
would be an unexpected behavior change for anyone depending on that SIP_CAUSE
update in their dialplan. Please refer to the asterisk-dev mailing list more
information:
http://lists.digium.com/pipermail/asterisk-dev/2011-August/050626.html
Significant fixes and improvements to parking lots.
(Closes issues ASTERISK-17183, ASTERISK-17870, ASTERISK-17430, ASTERISK-17452,
ASTERISK-17452, ASTERISK-15792. Reported by: David Cabrejos, Remi Quezada,
Philippe Lindheimer, David Woolley, Mat Murdock. Patched by: rmudgett)
Numerous issues have been reported for deadlocks that are caused by a blocking
read in res_timing_timerfd on a file descriptor that will never be written to.
A change to Asterisk adds some checks to make sure that the timerfd is both
valid and armed before calling read(). Should fix: ASTERISK-18142,
ASTERISK-18197, ASTERISK-18166 and possibly others.
(In essence, this change should make res_timing_timerfd usable.)
Resolve segfault when publishing device states via XMPP and not connected.
(Closes issue ASTERISK-18078. Reported, patched by: Michael L. Young. Tested
by Jonathan Rose)
Refresh peer address if DNS unavailable at peer creation.
(Closes issue ASTERISK-18000)
Fix the missing DAHDI channels when using the newer chan_dahdi.conf sections
for channel configuration.
(Closes issue ASTERISK-18496. Reported by Sean Darcy. Patched by Richard
Mudgett)
Remove unnecessary libpri dependency checks in the configure script.
(Closes issue ASTERISK-18535. Reported by Michael Keuter. Patched by Richard
Mudgett)
Update get_ilbc_source.sh script to work again.
(Closes issue ASTERISK-18412)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.7.0
Thank you for your continued support of Asterisk!
The Asterisk Development Team announces the release of Asterisk 1.8.7.0. This
release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.8.7.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
Please note that a significant numbers of changes and fixes have gone into
features.c in this release (call parking, built-in transfers, call pickup,
etc.).
NOTE:
Recently, we were notified that the mechanism included in our Asterisk source
code releases to download and build support for the iLBC codec had stopped
working correctly; a little investigation revealed that this occurred because of
some changes on the ilbcfreeware.org website. These changes occurred as a result
of Google's acquisition of GIPS, who produced (and provided licenses for) the
iLBC codec.
If you are a user of Asterisk and iLBC together, and you've already executed a
license agreement with GIPS, we believe you can continue using iLBC with
Asterisk. If you are a user of Asterisk and iLBC together, but you had not
executed a license agreement with GIPS, we encourage you to research the
situation and consult with your own legal representatives to determine what
actions you may want to take (or avoid taking).
More information is available on the Asterisk blog:
http://blogs.asterisk.org/2011/09/19/ilbc-support-in-asterisk-after-goog...
The following is a sample of the issues resolved in this release:
Added the 'storesipcause' option to sip.conf to allow the user to disable the
setting of HASH(SIP_CAUSE,) on the channel. Having chan_sip set
HASH(SIP_CAUSE,) on the channel carries a significant performance
penalty because of the usage of the MASTER_CHANNEL() dialplan function.
We've decided to disable this feature by default in future 1.8 versions. This
would be an unexpected behavior change for anyone depending on that SIP_CAUSE
update in their dialplan. Please refer to the asterisk-dev mailing list more
information:
http://lists.digium.com/pipermail/asterisk-dev/2011-August/050626.html
Significant fixes and improvements to parking lots.
(Closes issues ASTERISK-17183, ASTERISK-17870, ASTERISK-17430, ASTERISK-17452,
ASTERISK-17452, ASTERISK-15792. Reported by: David Cabrejos, Remi Quezada,
Philippe Lindheimer, David Woolley, Mat Murdock. Patched by: rmudgett)
Numerous issues have been reported for deadlocks that are caused by a blocking
read in res_timing_timerfd on a file descriptor that will never be written to.
A change to Asterisk adds some checks to make sure that the timerfd is both
valid and armed before calling read(). Should fix: ASTERISK-18142,
ASTERISK-18197, ASTERISK-18166 and possibly others.
(In essence, this change should make res_timing_timerfd usable.)
Resolve segfault when publishing device states via XMPP and not connected.
(Closes issue ASTERISK-18078. Reported, patched by: Michael L. Young. Tested
by Jonathan Rose)
Refresh peer address if DNS unavailable at peer creation.
(Closes issue ASTERISK-18000)
Fix the missing DAHDI channels when using the newer chan_dahdi.conf sections
for channel configuration.
(Closes issue ASTERISK-18496. Reported by Sean Darcy. Patched by Richard
Mudgett)
Remove unnecessary libpri dependency checks in the configure script.
(Closes issue ASTERISK-18535. Reported by Michael Keuter. Patched by Richard
Mudgett)
Update get_ilbc_source.sh script to work again.
(Closes issue ASTERISK-18412)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.7.0
Thank you for your continued support of Asterisk!
Wednesday, September 14, 2011
Asterisk Hosted PBX
Hosted Asterisk or Hosted PBX solutions allow customers access to feature rich VoIP PBX solutions without the costly investment of an on-site PBX. With the growth and availability of high speed broadband access, customers can easily access advanced Unified Communications systems that were previously only available to large enterprises.
Hosted PBX phone systems are easy to operate and install, as well as being affordable for businesses of all sizes. From the one person home business that wants to appear to be large and prosperous to the corporation that wishes to make the most of company resources, hosted PBX phone systems are a great option.
Hosted PBX phone systems are easy to operate and install, as well as being affordable for businesses of all sizes. From the one person home business that wants to appear to be large and prosperous to the corporation that wishes to make the most of company resources, hosted PBX phone systems are a great option.
Asterisk VQM solution
We are happy to announce update to Asterisk VQM solution! This update includes new features and more parameters for call quality tracking. Asterisk VQM is a customer proved solution for voice quality monitoring in VoIP, PSTN, Mobile and converged networks, load testing, conference bridge testing, termination testing. Please consider some screen shots of the new release test results representation.
Overview
* Asterisk-powered dialer software
* Web Interface
* UNIX/Linux Cron-based Schedule Logic
* Open-Source Code
* Graphing Monitoring Stats
* MySQL Database for Call Records
Current Features
Asterisk_Dialer_2 * Dial by SIP or PSTN - Asterisk base capable of dialing via any medium
* Blast-Dialing - send multiple calls to 1 trunk for specified duration
- No QoS/MOS scoring performed, designed for load testing
* QoS/MOS Test-Dialing
- Ad-Hoc - perform on-demand test dials
- Scheduled - unlimited schedules - hardware/software limited
- Multiple "Test Applications" Possible: Current system designed to access conference bridge
* Simple Customized sequencing could allow testing through IVR menus, or other applications as required.
- Dial MIRROR/Echo: Each dialer configured to respond to calls from another dialer by CallerID
* Pre-Deployment Option: Perform simple inter-dialer testing
* Reporting
- Reports performance of scheduled tests over time
- Detailed graphs over time
* MOS, PESQ, R-Value, Volume/Amplitude Difference
- Sortable tables of results for all calls
- Detailed View of each test call
* Embedded web player; compare source, reference WAVs
* Initiate Dials via Web Service
- All calls are initiated by HTTP POST (even internally)
Upcoming Features
* Roll-up Reporting Dashboard
* Scheduled email reports
* Email notifications
- Threshold definitions per schedule, email notifications
Overview
* Asterisk-powered dialer software
* Web Interface
* UNIX/Linux Cron-based Schedule Logic
* Open-Source Code
* Graphing Monitoring Stats
* MySQL Database for Call Records
Current Features
Asterisk_Dialer_2 * Dial by SIP or PSTN - Asterisk base capable of dialing via any medium
* Blast-Dialing - send multiple calls to 1 trunk for specified duration
- No QoS/MOS scoring performed, designed for load testing
* QoS/MOS Test-Dialing
- Ad-Hoc - perform on-demand test dials
- Scheduled - unlimited schedules - hardware/software limited
- Multiple "Test Applications" Possible: Current system designed to access conference bridge
* Simple Customized sequencing could allow testing through IVR menus, or other applications as required.
- Dial MIRROR/Echo: Each dialer configured to respond to calls from another dialer by CallerID
* Pre-Deployment Option: Perform simple inter-dialer testing
* Reporting
- Reports performance of scheduled tests over time
- Detailed graphs over time
* MOS, PESQ, R-Value, Volume/Amplitude Difference
- Sortable tables of results for all calls
- Detailed View of each test call
* Embedded web player; compare source, reference WAVs
* Initiate Dials via Web Service
- All calls are initiated by HTTP POST (even internally)
Upcoming Features
* Roll-up Reporting Dashboard
* Scheduled email reports
* Email notifications
- Threshold definitions per schedule, email notifications
Monday, August 29, 2011
ARRIS Telephony
To succeed in the residential communications services market, cable operators need to be able to supply and support reliable voice and data services for their subscribers. Critical to delivering these services is the quality and reliability of the endpoint device, the E-MTA (Embedded Multimedia Terminal Adaptor). ARRIS understands the importance of this unit in the successful deployment of combined voice and data services, and has worked countless hours with MSO’s worldwide to ensure compatibility and quality of service from their devices. Each E-MTA offers carrier grade voice service quality, to meet and exceed subscribers expectations; battery back-up, for telephony service in the case of a power outage; multiple diagnostic tools to measure and monitor any issues with the unit’s battery, the home’s inside wiring, the home’s telephony equipment, or other network problems. These key features keep the cable operator informed and alerted to any instances of issues with the subscriber’s service related to the product.
To ensure top level support to operators, ARRIS prepares all their own software and firmware for product operation, and thus can diagnose and rectify product problems locally as opposed to those with products designed and developed in other parts of the world with no local technical support. With 10’s of millions of lines being operated today on ARRIS telephony modems by cable operators worldwide, ARRIS takes pride is supporting each and every customer deploying an ARRIS branded device.
To ensure top level support to operators, ARRIS prepares all their own software and firmware for product operation, and thus can diagnose and rectify product problems locally as opposed to those with products designed and developed in other parts of the world with no local technical support. With 10’s of millions of lines being operated today on ARRIS telephony modems by cable operators worldwide, ARRIS takes pride is supporting each and every customer deploying an ARRIS branded device.
Firefly wireless phone
The new Firefly wireless phone is designed for kids. It is easy to use. Just press a couple of buttons to place a call. There is no need to remember numbers or dial. The special phone book makes this possible. It can hold up to 22 pre-programmed numbers. Use parental control options to limit incoming and outgoing call to the numbers in the phone book.
With just 5 (front) keys, it brings mobiles back to what they were originally designed to do - keep you connected. Firefly phones also keep it fun with lights, sounds, colours and animations.
Most importantly, it has PIN-protected features so you can control who the user receives calls from and, who they speak to.
Firefly Fireworks™ display flashes when making and receiving calls, during standby, and when charging.
With just 5 (front) keys, it brings mobiles back to what they were originally designed to do - keep you connected. Firefly phones also keep it fun with lights, sounds, colours and animations.
Most importantly, it has PIN-protected features so you can control who the user receives calls from and, who they speak to.
Firefly Fireworks™ display flashes when making and receiving calls, during standby, and when charging.
Tuesday, August 23, 2011
LumenVox Speech Engine
LumenVox Speech Engine is an API that performs recognition on audio data from any audio source, and allows for dynamic language, grammar, audio format, and logging capabilities.
LumenVox Speech Engine can be used with both Asterisk Business Edition and open source Asterisk.
The LumenVox Speech Engine is speaker and hardware independent, supports industry standards such as SRGS and Semantic Interpretation, and includes highly efficient barge-in and noise reduction technology for better performance in challenging environments.
The LumenVox Speech Engine is the core technology that handles the "recognition" of words and phrases. It is speaker-independent and does not require any special training. The engine relies on Grammars, which are a list of expected words or phrases that a caller might speak, to successfully recognize words. For example, a Call Router/Auto Attendant at a company will have all the names of the employees and perhaps "Sales" and "Support" loaded into the list of grammars.
LumenVox Speech Engine is directly and seamlessly integrated with the Asterisk platform and Dial Plan through the unique Asterisk Connector Bridge. Now application developers can easily build speech-enabled IVR's by using the familiar Dial Plan scripting language or the C-API.
LumenVox Speech Engine can be used with both Asterisk Business Edition and open source Asterisk.
The LumenVox Speech Engine is speaker and hardware independent, supports industry standards such as SRGS and Semantic Interpretation, and includes highly efficient barge-in and noise reduction technology for better performance in challenging environments.
The LumenVox Speech Engine is the core technology that handles the "recognition" of words and phrases. It is speaker-independent and does not require any special training. The engine relies on Grammars, which are a list of expected words or phrases that a caller might speak, to successfully recognize words. For example, a Call Router/Auto Attendant at a company will have all the names of the employees and perhaps "Sales" and "Support" loaded into the list of grammars.
LumenVox Speech Engine is directly and seamlessly integrated with the Asterisk platform and Dial Plan through the unique Asterisk Connector Bridge. Now application developers can easily build speech-enabled IVR's by using the familiar Dial Plan scripting language or the C-API.
AsteriskNOW
Install Asterisk and start building custom telephony applications with AsteriskNOW. AsteriskNOW installs a complete Linux distribution, Asterisk, the DAHDi driver framework, and your choice of the FreePBX administrative GUI, the AsteriskGUI or no GUI at all.
AsteriskNOW makes it easy to create custom telephony solutions by automatically installing the "plumbing". Much of the complexity of Asterisk and Linux is handled by the installer and the administrative GUI. Application developers and intergrators can concentrate on building their solution.
Among the many applications you can create with AsteriskNOW are:
VoIP Gateway IVR Server
Skype Gateway Voicemail System
IP PBX Call Recorder
Call Center ACD Fax Server
Conference Bridge Speech Server
AsteriskNOW users have two options for support. There is an active community of AsteriskNOW users, integrators and developers who provide community support on the AsteriskNOW forums and mailing list.
Digium offers commercial support subscriptions for AsteriskNOW. If your application requires direct manufacturer support, check out the subscriptions page.
Features And Functions
* Quick and easy installation. Typically installs in 15 minutes or less.
* Requires no in-depth knowledge of Linux. Safe for Mac and Windows users.
* Web-based configuration interface makes most management tasks point-and-click.
* Build data-driven applications with integrated support for ODBC and HTTPS
* Wizard-driven configuration of VoIP connections makes it simple to connect.
* Install pre-built, packaged applications using the Application Manager.
* Automatic detection and configuration of Digium analog and digitial telephony hardware.
* Sound Manager makes it easy to create, install and manage system prompts and recordings
* Dialplan and AEL script editor with syntax highlighting and validation simplifies development.
* Log and Call Detail Record (CDR) viewer gives instant access to system activity.
* Integrated help text for applications, functions, CLI, AGI and AMI commands.
* Realtime monitoring and debugging console helps simplify the development process.
AsteriskNOW makes it easy to create custom telephony solutions by automatically installing the "plumbing". Much of the complexity of Asterisk and Linux is handled by the installer and the administrative GUI. Application developers and intergrators can concentrate on building their solution.
Among the many applications you can create with AsteriskNOW are:
VoIP Gateway IVR Server
Skype Gateway Voicemail System
IP PBX Call Recorder
Call Center ACD Fax Server
Conference Bridge Speech Server
AsteriskNOW users have two options for support. There is an active community of AsteriskNOW users, integrators and developers who provide community support on the AsteriskNOW forums and mailing list.
Digium offers commercial support subscriptions for AsteriskNOW. If your application requires direct manufacturer support, check out the subscriptions page.
Features And Functions
* Quick and easy installation. Typically installs in 15 minutes or less.
* Requires no in-depth knowledge of Linux. Safe for Mac and Windows users.
* Web-based configuration interface makes most management tasks point-and-click.
* Build data-driven applications with integrated support for ODBC and HTTPS
* Wizard-driven configuration of VoIP connections makes it simple to connect.
* Install pre-built, packaged applications using the Application Manager.
* Automatic detection and configuration of Digium analog and digitial telephony hardware.
* Sound Manager makes it easy to create, install and manage system prompts and recordings
* Dialplan and AEL script editor with syntax highlighting and validation simplifies development.
* Log and Call Detail Record (CDR) viewer gives instant access to system activity.
* Integrated help text for applications, functions, CLI, AGI and AMI commands.
* Realtime monitoring and debugging console helps simplify the development process.
Sunday, August 14, 2011
Asterisk and Cell Phone
If you travel frequently, you’ll often find yourself in an airport, hotel, or eatery that offers wi-fi access. Combined a few other technologies, you’ll be able to use that wi-fi to make and receive phone calls using your company’s phone system rather than your cell. You’ll save cell minutes and your battery, plus have access to the power and flexibility of the company PBX. Here’s what you’ll need:
Asterisk-based VoIP business phone system (PBX)
softphone, such as CounterPath’s eyeBeam installed on your notebook.
Bluetooth headset (the same type used with most cell phones)
and optionally, VPN software, such as the open-source OpenVPN.
Asterisk
If you’re reading this post, you’re probably already familiar with Asterisk. But if not: Asterisk is an extremely popular, open source PBX application that runs atop of a Linux-based server. It can be configured to use your business’s existing phone lines and phone numbers, and replaces the phone system that is likely hanging on the wall in your telcom closet. RightBrain Networks is a Michigan-based an Asterisk integrator and can assist you any additional questions you have about about the system.
Softphone
The softphone is a computer application that simply mimics a physical office phone. CounterPath’s basic eyeBeam product is less than $34 per copy and is well worth the money. However, if you’d like to test this setup or would just like to save a few bucks, you can also use their free softphone, X-Lite. Once you’ve downloaded and installed the softphone, it will have to be configured to talk to your Asterisk server. You will need to know your SIP username and password, as well as the hostname of your Asterisk server. The person or organization that maintains your Asterisk server will be able to assist you with these steps.
Asterisk-based VoIP business phone system (PBX)
softphone, such as CounterPath’s eyeBeam installed on your notebook.
Bluetooth headset (the same type used with most cell phones)
and optionally, VPN software, such as the open-source OpenVPN.
Asterisk
If you’re reading this post, you’re probably already familiar with Asterisk. But if not: Asterisk is an extremely popular, open source PBX application that runs atop of a Linux-based server. It can be configured to use your business’s existing phone lines and phone numbers, and replaces the phone system that is likely hanging on the wall in your telcom closet. RightBrain Networks is a Michigan-based an Asterisk integrator and can assist you any additional questions you have about about the system.
Softphone
The softphone is a computer application that simply mimics a physical office phone. CounterPath’s basic eyeBeam product is less than $34 per copy and is well worth the money. However, if you’d like to test this setup or would just like to save a few bucks, you can also use their free softphone, X-Lite. Once you’ve downloaded and installed the softphone, it will have to be configured to talk to your Asterisk server. You will need to know your SIP username and password, as well as the hostname of your Asterisk server. The person or organization that maintains your Asterisk server will be able to assist you with these steps.
New Fedora Linux Project Leader Comes From Asterisk Roots
The Red Hat sponsored Fedora Linux community is an open source development effort that includes a diverse set of participants. At the top of the organizational chart for Fedora is the position of Fedora Project Leader, the person tasked with overseeing the general direction and operations of the Fedora project.
In July, Jared Smith took up the position of Fedora Project Leader, replacing the outgoing Paul Frields. Among Smith's first jobs is to guide the development and release of the upcoming Fedora 14 Linux distribution, set for general availability in November. Building the Fedora Linux distribution is one of Smith's key responsibilities as Fedora Project Leader, but it involves more than just pure code.
"A lot of the time we think of Fedora as just the bits and the bytes that we burn on a CD every six months and ship out, but Fedora is more than that, it has to be a community," Smith said. "As such we have to concentrate on building that community and taking care of the community as much as we take care of the bits and bytes."
Smith's vision for Fedora is about ensuring that the Fedora community is an inclusive place where multiple views and contributions are welcome. Smith doesn't necessarily have any new or unique tools for building community, but he does bring a different background to the position than past Fedora Project Leaders.
In July, Jared Smith took up the position of Fedora Project Leader, replacing the outgoing Paul Frields. Among Smith's first jobs is to guide the development and release of the upcoming Fedora 14 Linux distribution, set for general availability in November. Building the Fedora Linux distribution is one of Smith's key responsibilities as Fedora Project Leader, but it involves more than just pure code.
"A lot of the time we think of Fedora as just the bits and the bytes that we burn on a CD every six months and ship out, but Fedora is more than that, it has to be a community," Smith said. "As such we have to concentrate on building that community and taking care of the community as much as we take care of the bits and bytes."
Smith's vision for Fedora is about ensuring that the Fedora community is an inclusive place where multiple views and contributions are welcome. Smith doesn't necessarily have any new or unique tools for building community, but he does bring a different background to the position than past Fedora Project Leaders.
Saturday, August 13, 2011
AskoziaPBX Asterisk PBX System
AskoziaPBX is a complete PBX (Private Branch Exchange) on a small live CD that can be installed to your hard drive. It makes Asterisk, the number one open source PBX, install in minutes. Coupled with a little bit of telephony hardware or a SIP phone and a VOIP gateway account, you can have a complete phone system running in less than half an hour using AskoziaPBX.
Thursday, August 11, 2011
Asterisk CTI Integration Tool
TTTelecom has developed a CTI-integration tool for the Open Source Asterisk PBX. ADAT ("Asterisk Dial & Announce Tool) enables users to monitor and control their extension. CRM-integration is also part of the software.
Features:
Call notification
Incoming and outgoing calls are announced using a tray icon balloon.
On incoming calls the calling party name and number is announced (as log as Asterisk provides this information)
CRM-integration
CRM Customer details can be shown when a call is announced by clicking on the balloon
The CRM-integration url's can be defined seperately for inbound and outbound calls
Manual PC Dialer
Using the ADAT dialer form, any number can be manually entered (or pasted) and dialed.
ADAT will setup the calll for you by dialing your extension first. When you answer the outbound call will be setup.
Copy & Paste Dialer
Faster dialing with a simple copy & paste allows you to use any source of information (Windows program that allows select and copy)
Just paste the number into ADAT and dial.
Click-To-Dial using callto: links
It is possible to let ADAT handle CallTo URL's, resulting in fast click-to-dial functionality.
ADAT will automatically setup a call between your extension and the CallTo URL number you click on.
The use if CallTo URL's on the public web is growing, but can easily be implemented in your own personal Intranet and/or CRM enviroment
Internet Explorer and Firefox context-menu dialer
ADAT includes an Internet Explorer plugin that can be used for context-menu dialing on any web page.
Just select the number, right-click and choose "Dial using ADAT" to initiate a dial out.
The Firefox plugin is available as a separate .XPI download.
Call history
Using the call history you can lookup placed and received calls.
You can also easily (re)dial any number from your history.
Auto-answer
ADAT can be configured (available through .ini setting at the moment) to send a special SIP header which instructs SIP phones to auto-answer the call request.
Your phone answers the ADAT call-request immediately (putting it on speakerphone) and then dials the requested desitination without touching the phone!
The auto-answer feature has been tested with Grandstream phones. Other SIP phones may work also.
Features:
Call notification
Incoming and outgoing calls are announced using a tray icon balloon.
On incoming calls the calling party name and number is announced (as log as Asterisk provides this information)
CRM-integration
CRM Customer details can be shown when a call is announced by clicking on the balloon
The CRM-integration url's can be defined seperately for inbound and outbound calls
Manual PC Dialer
Using the ADAT dialer form, any number can be manually entered (or pasted) and dialed.
ADAT will setup the calll for you by dialing your extension first. When you answer the outbound call will be setup.
Copy & Paste Dialer
Faster dialing with a simple copy & paste allows you to use any source of information (Windows program that allows select and copy)
Just paste the number into ADAT and dial.
Click-To-Dial using callto: links
It is possible to let ADAT handle CallTo URL's, resulting in fast click-to-dial functionality.
ADAT will automatically setup a call between your extension and the CallTo URL number you click on.
The use if CallTo URL's on the public web is growing, but can easily be implemented in your own personal Intranet and/or CRM enviroment
Internet Explorer and Firefox context-menu dialer
ADAT includes an Internet Explorer plugin that can be used for context-menu dialing on any web page.
Just select the number, right-click and choose "Dial using ADAT" to initiate a dial out.
The Firefox plugin is available as a separate .XPI download.
Call history
Using the call history you can lookup placed and received calls.
You can also easily (re)dial any number from your history.
Auto-answer
ADAT can be configured (available through .ini setting at the moment) to send a special SIP header which instructs SIP phones to auto-answer the call request.
Your phone answers the ADAT call-request immediately (putting it on speakerphone) and then dials the requested desitination without touching the phone!
The auto-answer feature has been tested with Grandstream phones. Other SIP phones may work also.
Friday, August 5, 2011
Video Conference Solution
The video-conferencing industry has seen great traction in the enterprise community as organizations are realizing the value it delivers. Synergy Research Group forecasts the industry to near $2.5 billion in 2011, up from $1.1 billion in 2010. But the real question remains how large a role SMBs will play in this market adoption? For the SMB market, the two core challenges of incorporating a telepresence offering into daily communications remain cost and quality. While high quality usually means high costs, there are some solutions on the market that have reconciled this issue to make telepresence an affordable option.
So how does the channel take advantage of telepresence solutions to remain competitive and grow businesses more effectively? How should your organization present and offer these solutions? Consider the following:
Self-engage with the system. Possession is 9/10 of ownership. Being an early adopter is an essential step in the selling process. Fully understanding the benefits and potential challenges your customers could encounter will help support the sale and continued use of the product. More importantly, it positions you as a credible authority and deepens your connection with your clients, which can support future sales efforts.
Seeing is believing. Increase your customer's comfort level by communicating using the videoconferencing system. An IDC study found that companies incorporating video collaboration into their business processes enjoyed the following benefits:
So how does the channel take advantage of telepresence solutions to remain competitive and grow businesses more effectively? How should your organization present and offer these solutions? Consider the following:
Self-engage with the system. Possession is 9/10 of ownership. Being an early adopter is an essential step in the selling process. Fully understanding the benefits and potential challenges your customers could encounter will help support the sale and continued use of the product. More importantly, it positions you as a credible authority and deepens your connection with your clients, which can support future sales efforts.
Seeing is believing. Increase your customer's comfort level by communicating using the videoconferencing system. An IDC study found that companies incorporating video collaboration into their business processes enjoyed the following benefits:
Sunday, July 17, 2011
Redfone foneBridge 2
Redfone makes hardware and software products for open-source VoIP implementations, including bridges and gateways. The foneBridge 2 itself is a T1/E1 PRI-to-Ethernet bridge enabled for Asterisk, trixbox, FreeSwitch, Elastix and other platforms, providing Ethernet communication using Asterisk TDMoE (Time-division multiplexing over Ethernet) on Asterisk servers, instead of T1/E1 connections. The appeal of TDMoE for customers is that it offers traditional TDM features over Ethernet instead of pricey TDM hardware.
Freeside Billing
Freeside's open-source billing, CRM, trouble-ticketing and automation software copped a Best of Show award at this month's Digium Asterisk World / IT Expo in Miami -- the second year in a row it's collected the honor. It's as described: open-source software for ISPs, VoIP, hosted solutions, service providers and other customers, including Web-based businesses, that don't want to pay through the nose for comparable proprietary systems. Freeside itself began as an internal application developed by a regional ISP in the mid-90s, from which its founders released the first open-source version of its code and turned the application into a consultancy and provider of pre-configured hardware.
OrecX Oreka TR
OrecX makes voice recording, screen capture and quality monitoring products to be used with call centers. Its basic Oreka GPL recorder integrates with any phone system and computer and is free. The total recorder (TR) is available for $199 a year and is OrecX's value-add version, offering the ability to record VoIP SIP sessions through network packets, record multiple channels from standard sound devices like microphones, and segment and split audio into different, retrievable files. According to OrecX's, call replay is accessed through the Web and available on all standard browsers without a software upgrade.
Friday, July 15, 2011
Fonality trixbox Pro
Fonality's trixbox Pro, the business version of its IP-PBX software trixbox, comes in four editions: Pro Standard Edition (SE), Enteprise Edition (EE), Call Center Edition (CCE) and Unified Agent Edition (UAE). Fonality uses what it calls a hybrid-hosted architecture: a split, hosted-plus-premise model that includes free VoIP calling and monitoring like most hosted solutions provide, but also PSTN connectivity and call privacy like you'd find in a more traditional PSTN set-up. Started by enterpreneur Chris Lyman about five years ago, Fonality first came to many solution providers' notice with its Asterisk-based PBXtra.
Asterisk Dialogic DiaStar
Dialogic's DiaStar Server is a platform for app dev -- Asterisk-based or otherwise -- around functions like call progress analysis, fax, fax over IP, SS7 and Sigtran. The goal, according to Dialogic, is to get developers as much flexibility as possible, whether they want to leverage PSTN protocols like ISDN, SS7 or CAS or provide rich applications around voice and video. For LAN environments, Dialogic uses Woomera, the open-source IP-based signaling protocol.
Wednesday, July 13, 2011
Asterisk Synway
Asterisk based IP-PBXs are becoming more available and popular on the PBX business market, threatening the traditional proprietary PBX manufacturers position continuously. IP-PBX has proved itself to be efficient because it can reside on the current network using existing LAN. Besides, it can greatly lower operational costs especially for systems with a higher number of users.
Synway specializes in providing superior Media Processing & Signaling Technologies, Telephony Hardware and Integrated Multimedia Switch in use for convergence (voice/data/video) communications. Since 1995, over 1000 software developers and system integrators have integrated Synway's offerings to deliver a broad range of TDM and VoIP-based applications, including unified communications, call center, mobile VAS, media gateway, fax, conferencing, call recording, Asterisk-based open source applications for operators and enterprises worldwide.
Synway specializes in providing superior Media Processing & Signaling Technologies, Telephony Hardware and Integrated Multimedia Switch in use for convergence (voice/data/video) communications. Since 1995, over 1000 software developers and system integrators have integrated Synway's offerings to deliver a broad range of TDM and VoIP-based applications, including unified communications, call center, mobile VAS, media gateway, fax, conferencing, call recording, Asterisk-based open source applications for operators and enterprises worldwide.
Asterisk Skype
Skype for Asterisk is an add-on channel driver for Asterisk-based PBX systems. Adding Skype for Asterisk to any Asterisk server enables complete access to the Skype community, including low cost PSTN access and calling to more than 500 million registered Skype users. Use of Skype for Asterisk requires a Skype Manager account.
Key Features
• Make Skype to Skype calls
• Calls to landlines and mobile phones
• Receive calls with SkypeIn
• Make world-wide PSTN calls with SkypeOut
• Make and receive multiple concurrent Skype calls from the same Skype account
• DTMF support for incoming and outgoing calls
• Read Skype profile fields from incoming calls
• Set and retrieve online status
• Set privacy settings
• Handle incoming Skype calls using all Asterisk applications (voicemail, ACD, MeetMe conferencing, etc.)
• Simultaneous access from both Asterisk and the Skype desktop client
• Use Asterisk phone for voice and Skype desktop client for IM, video
• Supports G.711 and G.729 (included) codecs
• Send and receive text chat messages on the Skype network
Key Features
• Make Skype to Skype calls
• Calls to landlines and mobile phones
• Receive calls with SkypeIn
• Make world-wide PSTN calls with SkypeOut
• Make and receive multiple concurrent Skype calls from the same Skype account
• DTMF support for incoming and outgoing calls
• Read Skype profile fields from incoming calls
• Set and retrieve online status
• Set privacy settings
• Handle incoming Skype calls using all Asterisk applications (voicemail, ACD, MeetMe conferencing, etc.)
• Simultaneous access from both Asterisk and the Skype desktop client
• Use Asterisk phone for voice and Skype desktop client for IM, video
• Supports G.711 and G.729 (included) codecs
• Send and receive text chat messages on the Skype network
Asterisk PRI
PRI is the standard for connections to offices. It's based on a T1 line in the US,
and E1 line in Europe. The T1 PRI consists of 24 channels, the E1 PRI of 32.
PRI has varying number of channels depending on the country.
In North American and Japan it's 23xB + 1xD (23 64Kbps digital channels + 1 64Kbps signalling/control channel) on a T1 1.544 Mbps
In Europe and Australia it's 30xB + 2xD on an E1 2.048 Mbps
(One timeslot on the E1 is not available for the user since it is used for internal synchronization purposes.)
PRI is also available in partial variants - ie 10xB + D.
Question: What about outside US and Europe - what's being used?
Answer: Most of the rest of the world use E1. Japan uses a variant of T1.
NFAS (Non Facility Associated Signalling) allows sharing one D channel accross multiple PRI.
As well as 'standard' signalling there are vendor specific extentions to PRI. The relevant standards include:
National ISDN v1 aka. NI-1
National ISDN v2 aka. NI-2 (see SR-4994)
Nortel DMS100, DMS250 (see NIS-A211-1)
Lucent 4ESS, 5ESS (see TR 41459)
some recent versions of the 5ESS support protocol extensions that Lucent refers to as Natianal ISDN v3 aka. NI-3
and E1 line in Europe. The T1 PRI consists of 24 channels, the E1 PRI of 32.
PRI has varying number of channels depending on the country.
In North American and Japan it's 23xB + 1xD (23 64Kbps digital channels + 1 64Kbps signalling/control channel) on a T1 1.544 Mbps
In Europe and Australia it's 30xB + 2xD on an E1 2.048 Mbps
(One timeslot on the E1 is not available for the user since it is used for internal synchronization purposes.)
PRI is also available in partial variants - ie 10xB + D.
Question: What about outside US and Europe - what's being used?
Answer: Most of the rest of the world use E1. Japan uses a variant of T1.
NFAS (Non Facility Associated Signalling) allows sharing one D channel accross multiple PRI.
As well as 'standard' signalling there are vendor specific extentions to PRI. The relevant standards include:
National ISDN v1 aka. NI-1
National ISDN v2 aka. NI-2 (see SR-4994)
Nortel DMS100, DMS250 (see NIS-A211-1)
Lucent 4ESS, 5ESS (see TR 41459)
some recent versions of the 5ESS support protocol extensions that Lucent refers to as Natianal ISDN v3 aka. NI-3
Tuesday, July 12, 2011
Telemarketing Techniques
A ringing phone is the result of your marketing efforts, which you pay good money to develop and implement and the fact that your phone rings indicates that you’re marketing is working. Don’t squander your hard-earned marketing dollars (and reputation) by underestimating the value of each phone call.
The value in any business is its customer and prospect list and the fastest way to build that list is to ask for contact information.
Have you heard - - cold-calling doesn’t work - - but “warm-calling” does. Warm-calling happens when you call someone who has already had some type of experience with you. Whenever you call a prospect, have a valid reason to call.
People call you because they want something. Let me say that again, people call you because they want something. They might want a price, an explanation, to report a problem, to speak with a manager etc. Whatever the case may be, they want something.
The key to selling is to give people what they want. What better time to listen to what somebody wants than when they call you up? If you listen well enough, your prospects will tell you just what you need to know to sell them your product or service.
One of the most common challenges to telephone marketing is how to handle price checkers, people who call you up just to get a price. This is especially dangerous if they have not yet received enough information to determine value.
It leaves you to compete price alone. This leaves you at a disadvantage unless you sell a bottom-end, cheap-as-dirt product. If you do, hopefully you have a cost advantage over your competitors.
Perhaps the most detrimental result of giving out your price over the phone is that you have just taken away the single most powerful motivation for people to come into your business and speak with you personally. You’ve just set yourself up to be price shopped.
A phone call without an appointment is like building a website and not getting visitors. Why work to make your phone ring if you’re not going to try and get an appointment? You should ask for an appointment on most every call.
There’s no better time to market to somebody then when they are waiting to speak to you. Instead of telling the person waiting on the phone about how great you are, help them to solve their problems. Here is a typical message that I hear often: “Thanks for calling ABC Pools, our store hours are from 9:00 a.m. to 6:00 p.m. Please call back during our normal business hours. Thank you.” Click.
Your telephone can be a powerful marketing tool if used correctly. With some know-how training you’ll be able to see results immediately.
The value in any business is its customer and prospect list and the fastest way to build that list is to ask for contact information.
Have you heard - - cold-calling doesn’t work - - but “warm-calling” does. Warm-calling happens when you call someone who has already had some type of experience with you. Whenever you call a prospect, have a valid reason to call.
People call you because they want something. Let me say that again, people call you because they want something. They might want a price, an explanation, to report a problem, to speak with a manager etc. Whatever the case may be, they want something.
The key to selling is to give people what they want. What better time to listen to what somebody wants than when they call you up? If you listen well enough, your prospects will tell you just what you need to know to sell them your product or service.
One of the most common challenges to telephone marketing is how to handle price checkers, people who call you up just to get a price. This is especially dangerous if they have not yet received enough information to determine value.
It leaves you to compete price alone. This leaves you at a disadvantage unless you sell a bottom-end, cheap-as-dirt product. If you do, hopefully you have a cost advantage over your competitors.
Perhaps the most detrimental result of giving out your price over the phone is that you have just taken away the single most powerful motivation for people to come into your business and speak with you personally. You’ve just set yourself up to be price shopped.
A phone call without an appointment is like building a website and not getting visitors. Why work to make your phone ring if you’re not going to try and get an appointment? You should ask for an appointment on most every call.
There’s no better time to market to somebody then when they are waiting to speak to you. Instead of telling the person waiting on the phone about how great you are, help them to solve their problems. Here is a typical message that I hear often: “Thanks for calling ABC Pools, our store hours are from 9:00 a.m. to 6:00 p.m. Please call back during our normal business hours. Thank you.” Click.
Your telephone can be a powerful marketing tool if used correctly. With some know-how training you’ll be able to see results immediately.
CoreDial's VoiceAxis
CoreDial's VoiceAxis is a turnkey software package targeted to enterprise-level businesses and service providers. The suite is designed for the management, billing and provisioning of hosted PBX, VoIP, SIP trunking and other related products -- essentially a one-stop management tool that Asterisk lovers have embraced for its ease-of-use with Asterisk PBXes. Version 3.0 of VoiceAxis arrived in October 2009.
Aheeva's Asterisk-based Contact Center Suite (CCS)
Aheeva's Asterisk-based Contact Center Suite (CCS) 2.2 is designed as an all-in-one contact center offering that ties together call routing, contact management functions and support and research functions -- all of the traditional call center features -- and latches on to an existing IP infrastructure to provide those things. According to Aheeva, it's fully scaleable and requires only a Web browser for access, which ramps up its appeal as a services play for remote monitoring, management and network analytics. Aheeva's message is that it's bringing functionality of a traditional telephony switch-based call center at a fraction of what proprietary systems cost.
Aksys' KONNECT Office Phone System
Aksys' KONNECT Office Phone System copped a best of show award at last year's IT Expo, the umbrella conference that includes Digium Asterisk World. It's presently available in two models: the KONNECT 600PL VoIP and landline phone and the 600P IP phone, which includes the 600PL's features minus a landline (FXO connector), and includes three-way conference calling, public address, paging to multiple phones, 3-digit dialing between sites and the ability to support as many as 14 concurrent calls per phone. Aksys markets KONNECT as an "office phone system inside a phone" and has gained traction with small businesses looking to integrate VoIP and landline functions and home and remote offices.
Tuesday, June 21, 2011
VoiceAxis Asterisk Implementation
VoiceAxis is multi-tenant PBX and VoIP management software designed to enable Service Providers and Enterprises to install and manage a large scale Asterisk implementation. The software controls a unique VoiceAxis Asterisk Cluster environment which utilizes Asterisk as the feature server or “SoftSwitch”. VoiceAxis works with both Asterisk Business Edition and the Asterisk Community Edition (asterisk.org), and leverages several other open source and commercially supported technologies.
Saturday, June 18, 2011
ElastixWorld 2011
For the second consecutive year ElastixWorld will take place, this time the event that will bring together the whole Elastix community will take place in Mexico City on the 3rd and 4th of November.
ElastixWorld 2011 will consist of two days of high-level conferences; new advances made on our distro and the launch of new products will be presented as well as awards be given. This is the opportunity to share experiences between customers, manufacturers, resellers and developers worldwide.
Stay tuned, you will be able to sign in soon through ElastixWorld's official website.
"After testing Elastix with OpenVox hardware, we certainly recommend this good product for PBX projects"
Belief Mo
OpenVox Communications Co.
Hong Kong
Friday, June 17, 2011
Elastix inside an Alcatel Enterprise Network
C.A.G is a refrigeration company. This company provides some refrigeration elements all over Europe. They have several employees on site for commercial work, an others outside. It is a middle company with about 150 employees.
Elastix inside an Alcatel OmniPCX Enterprise (OXE) Network with a Codian Bridge.
C.A.G
France
Author: Franck Danard - ECE
This company wants perform video conferences with their employees outside the complex easily. Due to the requirement to be connected at a video conference, every employee had an ISDN access and a video device (Video Phones, Video bridge system).
This company already has implementations with a Codian Bridge and an IPPBX network (Alcatel OXE), all over France (Some nodes in France, and some remotes A.C.T.). They dispose equally a Codian Bridge (ISDN & SIP) and a LAN network with a WAN access. The client has a SIP phone GrandStream GXV 3000 to make some test with the video conference and all the X-Lite extensions.
The client required to be able to make a video conference to the Codian Bridge from anywhere in France and Europe. He also required to be able to make an outgoing call to the Alcatel IPPBX OXE, and to be able to receive incoming calls from the Alcatel IPBX (from internal and an ISDN trunk).
The best system available to complete the requirement is a system based on Asterisk. We choose Elastix as the complementary solution.
The actual amount of concurrent calls between the Elastix, the Codian Bridge and the Alcatel OXE (across the WAN access) is 10. Maybe in the future, more SIP extensions would be added.
The technology used by every employee is an X-Lite – CounterPath softphone. This Softphone is free and provides lots of advantages including a complete list of audio and video codecs, video capability, easy installation process, Multi-OS (Windows, Linux). It’s also easy to use and has a good audio quality.
We used different technologies such as SIP and UA protocol (Alcatel standard).
The SIP technology is employed for the VoIP / ToIP, based on Ethernet OS. The UA technology is from Alcatel and is proprietary, based without IP protocol. These devices are connected on the UAi16 which will be able to receive 16 phones.
In addition, an ISDN trunk is included to make outbound calls to the operator and a SIP Trunk to make any call between the Alcatel OXE and the Elastix server.
Now, we are ready to add our Elastix server into the LAN network. This server is a simple personal computer with 40 GB of Hard-Disk, 1 GB of RAM, an Ethernet device 100 BT, and a CPU of 1.5 GHz.
The client gave us a free IP address for this server. We need a SIP trunk for OXE connection, and a WAN access for any remote user extensions (managed by the client). About OXE, we need to make a T2 trunk with a Qsig GF protocol, all on a SIP gateway. The IPBX Alcatel OXE needs some licenses keys to enable the SIP protocol, the SIP gateway, and the SIP user (Not free).
Apart from this there is nothing in particular about this implementation. The bandwidth is enough qualified for every calls with each codecs (Audio and video) until 10 concurrent's calls. More, if the users not use the video!
The selected bandwidth is enough to make 10 concurrent video calls. The bandwidth will be greater if the call does not use video.
Alcatel OXE can use 3 codecs: G711a, (G711µ for export series), G729, and G273. For this configuration, it was selected a G711a codec, (the best MOS quality). The OXE accept the ‘canreinvite’ (RTP direct) on the SIP trunk, very useful.
About SIP trunk part Elastix, it's very simple; the configuration is like this in (example #1). For test, no user and password is required. In the future, it will select a user with a secret code, and why not, a md5 authentication.
If you look at the example #1, you could see that the context is 'from-internal'. The trunk needs to use it to select this context to be recognized as an internal call. If the trunk uses another context like ‘from-pstn’ or ‘from-trunk’, every call then will be recognized like an external incoming call. This is very important.
About the dialplan with the Alcatel OXE, it must be common at least with the DID.
The Codian Brige is included like a simple SIP extension with the H263 and H263+ video codec and G711a (alaw) audio codec. Every X-Lite extension should have the same configuration, and same thing about the GrandStream GXV 3000 video phone.
Every SIP remote extensions should have the 'nat=yes' parameter to be able to make incoming calls with each internal phone to the OXE and be able to connect to the Codian Brige. It is important to remember that the router should forward every SIP port and RTP port to the Elastix server. This part is managed by the client.
The investments are not expensive at the Elastix side, but it could be increased by the license keys required by the Alcatel OXE. Every X-lite soft-phone is free. The server hardware is not expensive compared with the capabilities.
Service improvement
Every employee can participate at any video conference. Before the implementation it was difficult, and even impossible with the video and the client should have another video conference system or an ISDN video-phone. Today, with this solution and a WAN access, it's easier.
Every employee can make calls. (There is no need to pay any call to another operator. The outbound call is directly send from the siege).
Every employee can call any UA internal phones. (There is no need to use an operator).
With this configuration, the client:
Can receive incoming calls directly from ISDN with a DID number.
Can connect to the video conference from anywhere.
The implementation it’s easy to use, it can be improved on an already existing IPPBX, and be useful for every employees of the company.
Elastix Marketplace beta now available
Elastix Marketplace beta now available 
Now all the users can count with a space for adding new solutions fully compatible with Elastix. Elastix Marketplace will allow users to integrate third-party software to extend functionalities in the distro.
This feature can be found today in the addons module available in the 2.0.4 beta version. The main idea of this release is that with a simple process the user has access to new, easy to install applications, for an Elastix customization.
From now and on numerous software developers could offer solutions, to the installed Elastix server base, which will hopefully enrich the user’s experience with Elastix. We have developed this solution in the most intuitive way for the user providing easy installations and removals.
The only requirement prior to downloading a commercial addon is to register your Elastix. This registration will generate an installation id necessary to make transactions in the Marketplace.
Our main goal is that Elastix Marketplace could be a meeting place between developers and users, having Elastix as a platform.
We would like to thank the first certified addons:
- FOP2
- Visual Dialplan
- QueueMetrics
We are on the process of including more solutions and will continue to work on new versions to include a greater number of available solutions, your feedback is welcome as always.
Elastix MarketPlace purchase procedure
"After testing Elastix with OpenVox hardware, we certainly recommend this good product for PBX projects"
Belief Mo
OpenVox Communications Co.
Hong Kong
Elastix Beta with Asterisk 1.8 Available
Elastix Beta with Asterisk 1.8 Available As we announced at the end of April, we are launching today an update for Asterisk 1.8 in Elastix.
This update comes installed by default on 2.0.4-B4 our newest beta version of Elastix. There is an update available also for 2.0.3 version in the Elastix-Beta repositories, that can be can be activated at the Elastix Web Interface (System->Updates->Repositories). This update is available for all users of that version who want to test this new launch.
Elastix 2.0.4-B4 also includes Dahdi 2.4.1.2
Among the features you will find on Elastix are the following:
- Support for Secure RTP
- Support for IPv6
- Support for Google Talk and Google Voice
- Calendar Integration with support for CalDav, iCalendar, and others
Elastix 2.0.4-B4 - http://elx.ec/204beta4
We wait for your comments and feedback on this trial phase.
"After testing Elastix with OpenVox hardware, we certainly recommend this good product for PBX projects"
Belief Mo
OpenVox Communications Co.
Hong Kong
CATEG implements a Call Center with Elastix
CATEG implements a Call Center with Elastix CATEG, Guayaquil Electricity Company, needed bill processing and debt consultations from their clients through an inbound call center. They also needed to unify the dialing plan between headquarters and all of its branches with a High Availability solution.
CATEG
Ecuador
PaloSanto Solutions
Author: Alfredo Zambrano - ECE
The Solution was implemented in three CATEG agencies, two of them with an ELX-025 Elastix Appliance, and one agency with a cluster solution consisting of one ELX-3000 and one ELX-025. This solution works with a 2E1 Redfone, for the integration of 2 E1 PRI from the telephony provider.
The North agency initially had an ELX-025 with 25 concurrent calls capacity, but they saw a substantial increase in users and therefore integrate the ELX-3000 to expand the solution.
The objective of the three implementations was to allow the client to consult billing information at a new CATEG call center. To achieve this objective an IVR was configured to allow the subscriber to enter the invoice number and get the amount of the debt. When the customer enters the requested information, a connection is made with an Oracle database to get information and results. This database is managed by the system department of every agency.
According to statistics obtained by the reporting module, the demand for this application is around 100 to 150 visits per day.
"Elastix allows us to gain a stable platform that is fully compatible out-of-the-box with modern hardware such as PCI express and Serial ATA and allows proper use of interrupt management features like APCI. All of this without losing the PBX features you have come to love. In addition with Elastix you gain new features such as integrated HylaFax server, fully integrated communications and more flexibility through the Web UI"
Rhino Equipment Corp.
USA
Elastix 2.0 has been released!
It has been over two years since we released Elastix 1.0. Almost from that same moment we began to plan and develop what would become Elastix 2.0. It was a long period of hard work, but the new distro is finally ready to see the light. Thanks to the whole community for the incredible support. Special thanks go to the beta testers mailing list, whose members tested the product for several months before its release.
Elastix 2.0 includes many suggestions received from our users as well as many new ideas around Unified Communications. The ability to extend new functionalities through add-ons will allow Elastix to adapt to the requirements of broad scenarios, while at the same time enabling the integration with third party applications.
There are lots of new functionality included in this release. We will highlight the most relevant in this article.
Renovated Dashboard
The new Elastix dashboard allows the user to be aware of the system state, its performance, missed calls, emails, news from Elastix and much more. The user can select what applets to show and how to organize them. The applets can be organized by drag and drop in the browser window itself without the need of a plug-in.
Add-ons
The new Elastix add-ons module enables the installation of Elastix-integrated applications in a straightforward and intuitive manner. This new functionality will also allow third party application writers to integrate their apps with Elastix very easily.
In addition, the user will be kept informed of any new updates as well as newly available add-ons.
Elastix Operator Panel
Elastix Operator Panel is an application embedded within Elastix which allows for the management of calls in real time as well as providing a snapshot of the current activity. The Elastix Operator Panel does not require a flash plug-in to be installed in the browser. Everything is drag and drop!
Elastix Conference Room
The Elastix Conference Room is ideal for virtual meetings. Currently, one can set up a voice conference, share presentations in several formats, establish a file repository and chat with meeting attendees at the same time.
Mail Module improvements
It is now easier to integrate Elastix with an external mail server. Just fill a form and the integration would be ready without the need to edit a single text file. Another improvement is the integration with the Mailman mailing-list engine so one can create a mailing list straight from the Elastix interface.
Web-based Faxing
Starting in Elastix 2.0 users can send faxes directly from the web interface. Users can write the fax content directly in the web interface or upload files in PDF, TIFF or plain text format.
DHCP Module improvements
The DHCP server now allows the user to see the list of assigned IPs and their MAC addresses. Also, Elastix now has the capability of assigning static IPs based on the MAC address of the IP phone.
Automatic Backups
Backups can now be scheduled in advanced so they can occur daily, weekly or monthly. In addition, the interface now allows for a file to be uploaded or downloaded with all data and configurations to/from an external FTP server with drag and drops.
Agenda Module improvements
Agenda events can now be organized and downloaded in iCal format. The user can also move the events or modify their duration by drag and drop.
The complete detail of the changes in Elastix 2.0 can be viewed in the changelog. Elastix 1.6 will be mantained for bug correction and security issues only. However, there will not be any new functionality added to the 1.6 versions in the future.
Download:
Thursday, June 16, 2011
Rhino Ceros line moves to Elastix
Rhino Ceros line moves to Elastix
Last Friday, the Rhino team published the following news:
“Starting in January 2011, all Rhino Ceros systems will ship pre-loaded with the Elastix Unified Communications Server. Based on customer demand, we have decided to install Elastix by default. Elastix boasts over 1 million downloads and is in production environments around the world. With dedicated developers working to provide a stable, professional integrated communications environment, you can rest assured that the system that is stable today will be stable tomorrow and for years to come. Elastix also has a large community of active members providing support, tips, and advice to ensure users get the most from their Elastix systems.
With the native branding features of the Ceros line and the professional unified look of Elastix you can take your sales to the next level. […]”
See More…
Elastix has been collaborating closely with Rhino in the past, and we are including this partner in the Hardware Certification Program we have been performing this year.
We would like to welcome this initiative and continue working closely with Rhino.
"After testing Elastix with OpenVox hardware, we certainly recommend this good product for PBX projects"
Belief Mo
OpenVox Communications Co.
Hong Kong
IP Telephony Implementation at Villa Maria National University
Villa Maria National University is an education organization established at Villa Maria city in Cordoba State, Argentina. Two years ago directives of the university order the transfer of the existing telephony network to an IP platform, with a preliminary evaluation of technical and economical feasibility.
Universidad Nacional de Villa María
Argentina
Mariano Martín
Villa Maria National University is an education organization established at Villa Maria city in Cordoba State, Argentina. Two years ago directives of the university order the transfer of the existing telephony network to an IP platform, with a preliminary evaluation of technical and economical feasibility.
Different versions of Asterisk were analized in several Linux distributions at virtual environments. At the end the solution chosen was an integral/complete distribution.
The initial tests were performed on Elastix v1.2 32 bits in a real server using Digium clone cards (modems with Motorola/Intel chipset) with FXO ports connected to the PSTN. At this stage Softphones and IP phones were use for users.
Once the tests concluded, the authorities of the University took the decision of change the platform in their policies.
A Network adjustment was made in order to work with VoIP, also a there was a replacement of some switches by others with Power over Ethernet.
The infrastructure of the main Communication server was installed in an Intel Xeon/ HD 500 GB server with Elastix v1.3 32 bits.
At first we install 10 extensions using Grandstream IP Phones model GXP2000. We also used a Sangoma A200 card with 4 FXO ports to access the PSTN. This implementation was executed at one of the extensions of the University Campus.
In the next stage we include a Sangoma A200 telephony card with echo cancelation to expand the telephony integration to a total of 24 FXO ports. With this expansion we create more users to the system that include 100 extensions at the Dean Headquarter using different IP phone models (GXP2000, GXP1200, GXP280, and BT200). We also added console operators for more than 100 BLF indicators.
In addition to the 100 users, 10 Grandstream GXV3000 IP VIDEO phone where included. For this purpose, we enabled video support and all necessary codecs.
In the following stages the University incorporated a new Elastix v1.3 de 32bits server with the same hardware characteristics. for the Central Campus The telephone integration was made with a Sangoma A200 card with 24 FXO ports. Both servers were interconnected with IAX2 Trunks and a dedicated wireless link with 22Mbps and 100 extensions where added.
The main goal of having two servers with their own access to the PSTN is to avoid the lack of communication between the extensions if the link failed at some point.
For this functionality we create 15 users and used telephone adapter switches (ATA) for Fax machines. After that we created 20 virtual FAX (Hylafax), all of them redirected to several emails. There is also an access to FAX from the PC with Winprint Hylafax.
Some of the following features where enabled:
A freepbx CUSTOM-CONTEXT module was installed to manage different access to the PSTN for internal groups.
Conference Call, Call transfer, Call Pick-up, Call forwarding, Ringing groups, IVR, etc…
A call billing feature was enable, this was possible by including a destiny code to the bill. A2Billing was configured in order to work with different users with a billing process and prepaid call system.
Currently there is a development in AGI-PHP to use a text to speech application to allow teachers and students to make consultations of a database. This prototype is already working.
This implementation is part of a project that includes a complete implementation of IP telephony and Unified communications at the Villa Maria University which include a lot of work and further conclusions.
Some load data where obtained by creating modifications in the PHP code of the following module to reports and graphics in Elastix:
/admin/common/graph_hourdetail.php
This made possible to obtain the MAX LOAD value of the server in concurrent calls. We modified the code to obtain that value in a regular day and then for a complete month. We got the following results:
Dean Headquarter IP-PBX Server
Here you can appreciate that January is the month with the lowest activity due to the summer vacations and the higher activity was detected on December 2009 with 21 concurrent calls registered in the Principal server and 14 concurrent calls registered in the Campus server.
Elastix allowed us to reduce the implementation times. It is also useful to the system administration allowing the addition of more users with an easy training thanks to the information available.
At the moment of the publication of this case we have 4 servers interconnected with IAX2 and SIP.
The total number of extensions is 250 and we are working on a main project that will include VoIP platforms of other Argentinean public Universities to our system through a Proxy SIP Server.
The coordination of the telephony network that includes 18 Nacional Universities is performed at The Villa Maria National University and San Luis.
PaloSanto Solutions silver sponsor at Asterisk World
PaloSanto Solutions silver sponsor at Asterisk World Elastix and PaloSanto Solutions will be present at the next ITExpo East on the 2nd of February, 2011.
ITExpo is part of a dual event that TMC develops in Miami and Los Angeles, at the beginning and midyear, respectively. ITExpo gathers the most representative exponents of the telecommunication industry with keynote conferences and product exhibition.
Elastix will be present and you will have the opportunity to visit our stand and talk with important members of the Project and Gold Resellers of the region. See exhibitor list.
As part of our participation we will make the launch of some products, enlist the certified hardware available at the moment and will announce the 2010 community awards winners.
Elastix is also Silver Sponsor for Asterisk World, a collocated event inside ITExpo and dedicated to this important telephony distribution. We will present a keynote there the 3rd of February at 12h30 Miami.
http://www.tmcnet.com/voip/conference/digium-asterisk-world/east-11/
Join us at this event, we will be at Booth #1009. For a free pass to visit our booth you can register here
We hope to see you all
Sincerely
"Elastix is a stable, reliable and flexible system, built to serve both: integrators and end users. For these reasons we chose it as the default operating system in our IP-PBX solutions, and market feedback confirms this was a good decision."
Eran Gal - CEO
Xorcom Ltd
Thursday, May 26, 2011
Zentyal As A Gateway: The Perfect Setup
Zentyal is the Linux Small Business Server, it lets you manage all your network services through one single platform. It's a Network Gateway, as well as an Infrastructure, UTM (Unified Threat Manager), Office and Communications Server. All these features are fully integrated and easy to configure, it truly helps to save system administrators time.
In this tutorial you will see how to set up a Zentyal Server to act as a gateway in a very common scenario. Zentyal will provide basic network infrastructure, load balancing between two Internet providers, firewall and HTTP proxy caching and content filtering. All these steps are well explained in the Zentyal Documentation, which is a really recommended reading. The following example network layout is used:
Zentyal runs on top of Ubuntu Server so it will work on the same hardware. You can take a look at the Ubuntu-certified hardware page for more information. There are two ways to install Zentyal:
Using Zentyal installer that you can download from the project website. This is the recommended choice, it includes all package dependencies for offline install and also makes some custom configuration.Install on top of a working Ubuntu Server, you can find detailed info and URL for the repository in the Zentyal Installation Guide.
If you install Zentyal using the installer you will see this screen when booting from CD-ROM and a couple of wizards will guide you through the process. You can choose default settings in all of them.
Zentyal provides a web administration interface, after the installation a Firefox browser will show up giving you access to it (you can also access Zentyal from any client browser typing: https://zentyal_server_ip). User and password are the same you entered during installation.
Now you can select the desired packages to install, for this tutorial you should install the Gateway package. Later DHCP and DNS modules will also be installed by using the Software Management module.
After this step all the necessary packages are installed, now setup will guide you through configuration wizards for installed modules, in this case Network and Users. We can skip network configuration for now, so if you start this tutorial from an already installed Zentyal you can still follow it.
Zentyal Server is now installed. By following the next steps you will configure each module.
As shown in the scenario, you have to configure three network interfaces, two external routers and one for the internal network. Zentyal will balance traffic between the two Internet connections.
Go to Network -> Interfaces and configure each interface by introducing its IP and netmask. Don't forget to mark external interfaces because Zentyal uses this info in firewall rules. In the next image you can see configuration for one of the external interfaces and the internal one.
Now you have to set up both gateways in the gateways table (Network -> Gateways):
Go to Network -> Balance Traffic to enable load balancing between the gateways.
Zentyal As A Gateway: The Perfect Setup - Page 2Importing iptables Configurations Into Firewall Builder
Author: Mike Horn
http://www.fwbuilder.org
Firewall Builder is a firewall configuration and management GUI that supports configuring a wide range of firewalls from a single application. Supported firewalls include Linux iptables, BSD pf, Cisco ASA/PIX, Cisco router access lists and many more. The complete list of supported platforms along with downloadable binary packages and soure code can be found at http://www.fwbuilder.org.
Import of existing iptables configurations was greatly improved in the recently released Firewall Builder V4.2. Features like object de-duplication and expanded rules recognition make it even easier to get started using Firewall Builder to manage your iptables configurations.
For this tutorial we are going to import a very basic iptables configuration from a firewall that matches the diagram shown below.
Firewall Builder imports iptables configs in the format of iptables-save. Script iptables-save is part of the standard iptables install and should be present on all Linux distribution. Usually this script is installed in /sbin/.
When you run this script, it dumps the current iptables configuration to stdout. It reads iptables rules directly form the kernel rather than from some file, so what it dumps is what is really working right now. To import this into Firewall Builder, run the script to save the configuration to a file:
iptables-save > linux-1.confAs you can see in the output below, the example linux-1.conf iptables configuration is very simple with only a few filter rules and one nat rule.
# Completed on Mon Apr 11 21:23:33 2011# Generated by iptables-save v1.4.4 on Mon Apr 11 21:23:33 2011
*filter
:INPUT DROP [145:17050]
:FORWARD DROP [0:0]
:OUTPUT DROP [1724:72408]
:LOGDROP - [0:0]
-A INPUT -m state --state RELATED,ESTABLISHED -j ACCEPT
-A INPUT -i eth1 -s 10.10.10.0/24 -d 10.10.10.1/32 -p tcp -m tcp --dport 22 -m state --state NEW -j ACCEPT
-A FORWARD -m state --state RELATED,ESTABLISHED -j ACCEPT
-A FORWARD -o eth0 -s 10.10.10.0/24 -p tcp -m tcp --dport 80 -m state --state NEW -j ACCEPT
-A FORWARD -o eth0 -s 10.10.10.0/24 -p tcp -m tcp --dport 443 -m state --state NEW -j ACCEPT
-A FORWARD -j LOGDROP
-A LOGDROP -j LOG
-A LOGDROP -j DROP
COMMIT
# Completed on Mon Apr 11 21:23:33 2011
# Generated by iptables-save v1.4.4 on Mon Apr 11 21:23:33 2011
*nat
:PREROUTING ACCEPT [165114:22904965]
:OUTPUT ACCEPT [20:1160]
:POSTROUTING ACCEPT [20:1160]
-A POSTROUTING -s 10.10.10.0/24 -o eth0 -j MASQUERADE
COMMIT
# Completed on Mon Apr 11 21:23:33 2011
If you are running Firewall Builder on a different system than the one that is running iptables copy the file linux-1.conf from the firewall to the system where Firewall Builder is running.
Launch the Import wizard by selecting the File -> Import Firewall menu item.
Click Browse to find the file named linux-1.conf.
Click the Continue button to move to the next step of the import process.
The next window shows a preview of the configuration file that will be imported and the type of firewall that Firewall Builder has detected it to be.
Next you need to enter a name for the firewall. This is the name that will be used in Firewall Builder to refer to the firewall after it is imported. When you click the Commit button the configuration data will be read.
By default, Firewall Builder attempts to detect if there are items, like IP addresses, used in the rules that match existing items in the object tree. If there is a match the existing item is used, if there is no match a new object is created. This feature can be disabled by unchecking the box next to "Find and use existing objects" which will result in objects being created for evry item used in the imported rules regardless of whether it already exists in the object tree or not.
After the import is complete, Firewall Builder displays a log showing all the actions that were taken during the import. Warning messages are displayed in blue font and error messages are displayed in red.
The program tries to interpret the configuration file rule by rule and recreates the equivalent rule in Firewall Builder. Note that rules imported into Firewall Builder may not always be optimized since features like defining multiple source and/or destinations are supported by Firewall Builder, but not by iptables.
The progress window displays warning and error messages, if any, as well as some diagnostics that shows network and service objects created in the process.
As you can see from the import process log, Firewall Builder detected that there are rules in the iptables configuration that allow RELATED and ESTABLISHED traffic through the firewall. This behavior can be controlled by a setting in Firewall Builder, so a warning message is shown.
Click the Done button to complete the firewall import. Next we will go through some common post-import actions.
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