Voice Over Internet VoIP

Verizon announced at Spring 2006 VON Wednesday that it is providing service-level agreements on the quality of its VoIP services.

SipReality Limited, a developer and distributor of Voice Over Internet Protocol (VoIP) softswitch and end user device provisioning systems is pleased to announce the Commercial Release of it's TotallySip Softswitch platform.

TotallySip has been developed as a carrier grade softswitch solution to service both the emerging VoIP Internet Service Providers (VISP) and much larger and geographically dispersed Competitive Local Exchange Carriers (CLEC) as well the Incumbent Carriers (ILEC). The system is fully scalable from a single node operating with end user provisioning to Class IV and Class V switch features. TotallySip integrates easily with many vendor gateway solutions allowing for maximum flexibility in using existing infrastructure to augment services into the VoIP arena.

Low cost of entry combined with flexible integration and of course extensive features makes TotallySip stand out as one of the most economically viable Softswitches on the market today. Clustering between nodes in geographically disperse or centralized locations is simple and efficient while still allowing for single point management of the entire system. Least Cost Routing (LCR) is done via NPA/NXX with OCN based support coming soon. (LERG subscription required for OCN based routing).

Paul Falcon, President of SipReality stated, "We are very happy to reach our goal of delivering a high availability yet reasonably priced softswitch solution to the market. Today we hit a milestone. Expect to see more modules and add-ons for TotallySip to be available in the coming months extending the ease of use and functionality far beyond any competitive product."

Click Here for the Full Release

Digium Inc., the creator of Asterisk(TM), and pioneer of open source telephony, today announced the availability of new hardware solutions to enhance Asterisk transcoding and echo cancellation performance for VoIP and PSTN gateways. These new products include the TC400P VoIP transcoding card and the TE420P and TE415P four-port T1/E1/J1/PRI cards with onboard hardware echo cancellation.

"Our product team is always working to develop solutions like these that ultimately further the open source movement in VoIP," said Mark Spencer, president of Digium. "Not only are we constantly striving to improve Asterisk's performance, but we also want to contribute to the overall VoIP experience, while keeping costs low."

Click Here for the Full Release

I am currently developing a asterisk ncurses interface using the manager API. The project is currently awaiting sourceforge's approval but I have a beta online at: http://sig.lange.googlepages.com/assman

The projects real home will be assman.sf.net. This project really consists of two parts, libassman is a C manager API and assman is the ncurses portion. It's still beta but I have been running it for quite some time on a production server w/o any major glitches. Soon as the sf.net approves the project I will have SVN and the latest versions online.

Feedback is welcome as well as requested features.

Thanks.

--
Sig Lange
http://www.signuts.net/

Great day for the callingcard-fan ! Just a little mail to let you know that a new version of A2Billing 1.1 (Asterisk2Billing) is available! Many features have been added, lot of bugs solved and hundreds of good improvement made, so there we go -> http://www.asterisk2billing.org

The key newest features :
* Ecommerce product with API addons - Integration with OsCommerce
* Speeddial-support for UIs (Customer & Admin)
* Add DB backup/restore tool
* Currencies support management - yahoo financial (cront for auto update) Add new model for update currencies from Yahoo , now currencies are in Database in cc_currencies table. Remove rates.inc and any information about.
* Signup autocreates SIP/IAX
* New features for PEAK & OFF-PEAK
Add new model for ratecard , removing week day and adding starttime and endtime instead.
* Add Voip Provider
* Add the RATECARD SIMULATOR
* Add support for Jiax web phone
* notenoughcredit_assign_newcardnumber_cid
IF the CARD doesn't have enough credit, request to enter a new cardnumber.
* Assign the CallerID to the new cardnumber
* Predictive Dialer Features
* Manage Campaign, Phonelist, Import Phonelist.
* Customer Interface (Agent) have the ability to call a predefined amount of Phone numbers.
* Support call at Zero-Cost & Negative cost (plus param = maxtime_tocall_negatif_free_route)
* CallerID authentication improvement
- (new param : notenoughcredit_cardnumber ;
cid_auto_assign_card_to_cid ; cid_auto_create_card ;
cid_auto_assign_card_to_cid)
* Popup Select Card - avoids long load (issue for user that have create lot of cards)
* PAYPAL SUPPORT - IPN - Customer can buy credit through paypal
* DID SELLING SUPPORT + DID monthly billing - features to sell to your customer preconfigured DID.
Customer would have the opportunity to redirect those to his phonenumber and even deploy a Follow-Me
* and lot of bug fixed and much more fancy stuff...


Other good stuff as well :
- WIKI -> http://wiki.asterisk2billing.org/
I hope it will help to build quickly a serious user manual, I know
that it's pain in $%& to understand the soft.
- FORUM -> http://forum.asterisk2billing.org/
Damn !!! Que demande le peuple !!!
- DEMO -> http://demo.asterisk2billing.org/
- UNLIMITED FREE CALL ON PSTN -> ... forgot the link!


Seriously bit of helps (documenting, dev...) would be greatly appreciated so if someone is willing to help/contribute, please contact me directly! Enough talk it's time to enjoy this new version, have fun and don't forget to send me your comments :P

Cheers,
/Areski

Break out your SSH terminals ladies. If you don't have a base setup don't read further and go back to look at my other help articles. When you are done with this article you should be able to set up an IAX2 soft client and configure Asterisk to route this call to our sip phones a the office. We will use GSM as the single codec to keep it simple. Our options in the iax.conf file will be tailored to low bandwidth connections.


1) Open a terminal window. If you need to access the server remotely Download an SSH (Secure Shell) client to access the Asterisk server. You can use Secure Shell from a vareity of Microsoft Windows clients freely available on the world wide web. If you have Linux or Mac OS X just read the man files from a terminal. From the shell change directory to the Asterisk config folder.

Example:

[matt@localhost ~]$ cd /etc/asterisk/



2) Change to root user and open up the iax.conf file with your favorite text editor. When you are done the iax.conf should look like below. What we are doing here is defining the base options for our remote soft phone to be able to register. Once we finish this we will move to the extensions.conf to add in an IAX context.

Example:

[matt@localhost asterisk]$ su
Password:
[root@localhost asterisk]#nano iax.conf

[general]
port=4569
bandwidth=low
disallow=all
allow=gsm
jitterbuffer=yes
tos=lowdelay

[9252]
type=friend
context=from-iax
secret=2005
host=dynamic
allow=gsm
auth=plaintext,md5,rsa
callerid=9252
mailbox=9252

3) Next we open up the extensions.conf, and add a route to the IAX user we just defined. Your extensions.conf will look like below. These are the same extensions as used in previous help articles.

Example:

[root@localhost asterisk]# nano extensions.conf
[general]

static=yes ; These two lines prevent the command-line interface

writeprotect=yes ; from overwriting the config file. Leave them here.

[bogon-calls]

exten => _.,1,Congestion

[from-sip]

exten => 9250,1,Dial(SIP/9250,20)
exten => 9250,2,Voicemail(u9250)
exten => 9250,102,Voicemail(b9250)

exten => 9250,103,Hangup

exten => 9251,1,Dial(SIP/9251,20)
exten => 9251,2,Voicemail(u9251)
exten => 9251,102,Voicemail(b9251)
exten => 9251,103,Hangup

exten => 2999,1,Voicemailmain()
exten => 2999,2,Hangup

include => from-iax

[from-iax]

exten => 9252,1,Dial(IAX2/9252,20)
exten => 9252,2,Voicemail(u9252)
exten => 9252,102,Voicemail(b9252)
exten => 9252,103,Hangup

include => from-sip

4) Finally, we need to add in our user to voicemail.conf.

Example:

[general]

format=wav
attach=no
[default]
9250=1000,matt, matt@somecompany.com
9251=1000,joel,joel@somecompany.com
9252=1000,gerald,gerald@somecompany.com

5) We're almost done we need to download a soft client. Get your speakers and microphone ready. Go to
http://www.laser.com/dante/diax/diax.html. Download the latest version of Diax. After it is installed, right click on one of the small boxes that goes from right to left in numerical order. Each one of these boxes contains the possible servers the client can register too. In our case we have one server so we'll click on the first box and enter our login information. Remember that we orginally set up GSM as the codec. So select GSM and unselect the other codecs. Leave the context blank. Oh by the way, the name and number fields on the bottom are for Caller ID information.





Thats it folks. Fire up Asterisk and enjoy.


Note: My name is Matt Birkland, I work as a VoIP Engineer for VoiceIP Solutions an Asterisk Provider in Washington State. Every week I will be submitting a one page Asterisk/VoIP tip of the week on the blog.

Digium Inc., the original creator of Asterisk and pioneer of open source telephony, and Zimbra, a leader in open source next-generation collaboration and messaging, today announced the integration of Voice-over-IP calling capabilities into the Zimbra Collaboration Suite (ZCS), by leveraging Asterisk. With this partnership, Digium and Zimbra are leading the way to the first open source Unified Messaging platform.

"We're really pumped that Zimbra and Digium are able to provide Unified Messaging this quickly by leveraging open platforms. Now, I can initiate a conference call with my engineering team, quickly access my voicemail or call home from the road - all through my e-mail," said Satish Dharmaraj, Zimbra co-founder and CEO. "What's beautiful about this is that it's all based on open standards. We've implemented a SIP integration with ZCS that has been tested and proven on Digium's open source Asterisk VoIP system."

Click Here for the Full Release

Esnatech, a provider of enterprise real-time communications solutions, today announced it has joined the Digium(TM) Partner Program to deliver enterprise based Unified Communications solutions integrated with Asterisk(TM) IP telephony platforms.

Esnatech's Telephony Office-LinX is a next-generation, real-time communications platform integrated within an organization's telephony network. The release of IP integration with Asterisk will provide small business organizations a truly Unified Communications platform incorporating IP telephony with integrated IP applications targeted specifically for the enterprise business marketplace.

Digium is the creator and primary developer of Asterisk, the industry's first Open Source PBX. Used in combination with Digium's PCI telephony interface cards, Asterisk offers a strategic, highly cost-effective approach to voice and data transport over TDM, switched and Ethernet architectures. The Digium Partner program is designed to promote Digium and Asterisk related products. The goal is to form a closer relationship between Digium and companies who have incorporated Digium and Asterisk technology into their products.

Esnatech's Asterisk integration provides pure SIP-based Unified Communications solution. It provides secure server-based wireless messaging with access to messages conveniently from virtually any communication device, including office telephones, cell phones, PDAs, PCs or any Web browser. It bundles a suite of communication solutions including location-based routing, fax, online and offline access to presence management and text messaging, speech-enabled routing and corporate dialing, desktop call control, IVR and CRM integration, all bundled into one integrated SIP based platform.

Click Here for the Full Release

Excerpt: "Asterisk@Home 2.7 has hit the street with some new goodies and a new release of Asterisk. So here we go again. With this update, you get version 1.2.5 of Asterisk as well as the latest and greatest version of:

-Linux
-CentOS 4.2, the latest Festival Speech Engine (1.96)
-Latest version of the Asterisk Management Portal (1.10.010)
-Flash Operator Panel (version 0.24)
-Open A2Billing
-Digium card auto-configuration
-Loads of AGI scripts including weather forecasts and wakeup calls
-xPL Support
-Latest SugarCRM Contact Management System with the Cisco XML Services interface
-Click-to-Dial support
-Microsoft File Sharing and Networking support through Samba (3.0.10)
-Plus dozens and dozens of free utility software applications for Asterisk compliments of Nerd Vittles.

And, yes, Asterisk@Home 2.7 still fits on a single CD! By popular request, we've also added something new in this tutorial: some tips and tricks to assist those upgrading from a previous version of Asterisk@Home."

Click Here for the Full Nerd