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.

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/

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=

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.

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.

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.

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

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.

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!

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!