OpenSIPS 1.6 eBootcamp, Get Ready For A Torrent Of SIP Technology

OpenSIPS 1.6 eBootcamp http://snapvoip.blogspot.com/

OpenSIPS is holding a eBootcamp on OpenSIPS 1.6. This bootcamp is unique in itself because it is held over the Internet for over 7 weeks. Participants will learn from downloading installing and configuring and on to administration of OpenSIPS. It will start on September 19th 2011. The ebootcamp will accept late registrations up until 12th September 2011. So you better hurry. Contact bootcamp@opensips.org for registration and more information

The users will learn how to authenticate users, install a GUI to help with daily administration, forward calls to the PSTN through Dialplan, integrate Asterisk and Voice Mail, Presence agent, Load Balancing, Traverse Nat for SIP and generate CDR records to a Radius Server. At the end, you will learn how to use troubleshooting tools to solve end user problems.

This is basically the same OpenSIPS Bootcamp but help over a broadband connection over a longer period, seven weeks to be exact.

"The live classes will be taken online by web-conference every Tuesday and Thursday 03:00PM GMT, 11:00AM EDT 08:00AM, PDT. To attend this training you will need to have broadband Internet access. You are going to receive a DVD with a virtual machine to run the labs. The virtual machine will be available in the VMWARE format. You can download the free VMWARE player to run the VM. We suggest that you have one separate desktop or server for your VM and at least one IP Phone/ATA in your private labs to complete the training. A LMS (Learning management system) will be available with forums, quizzes and support materials."

The course of learning will cover key objectives;

Install OpenSIPS on a Linux Machine

Routing basics and the default configuration

OpenSIPS authentication using MySQL and Memcache

Install OpenSIPS control Panel.

Connect to the PSTN using Dialplan and Dynamic Routing

Voicemail integration using Call Forward and AVPs

Implement a presence agent

Understand important aspects of load balancing and high availability

Implement SIP NAT traversal using RTPProxy

Account Calls to MySQL

How to use test and monitoring tools to check your configuration

OpenSIPS

Blink Cocoa and Blink QT Cool SIP Clients From AG Projects.

#Blink Cocoa and Blink QT Cool #SIP Clients http://snapvoip.blogspot.com/
AG Projects have been mentioned many a time on this blog for their Free SIP Accounts, OpenSIPS and the SIP Simple Testbed.
Blink is a very good SIP client and works on MAC, Windows and Linux platforms. Blink Cocoa runs on MacOSX 10.5 Leopard and MacOSX 10.6 Snow Leopard, wwhile Blink QT runs on Windows and Linux platforms.
Some of the features of Blink includes;

• Support for multiple SIP accounts
• Detection of IP address changes
• Encryption for signaling and media
• Self-update capability
• One-click SIP account sign-up at http://sip2sip.info (listed Free SIP Accounts, Revisited.)
• Integration with AG Projects Multimedia Service Platform
• Integration possible with third-party SIP service providers
• Multi-party conferencing with unlimited number of participants
• Drag and Drop contacts to conference
• SIP, DNS and MSRP protocol trace to file or window
• Wideband Audio (G722 & speex)
• Multiple parallel calls
• Answering machine with real-time takeover capability
• Play hold tone and disconnect tone
• In-band DTMF support for legacy devices 

You can download from  icanblink

OpenSIPS will Be At SIPit 26, To Test Interoperability.

SIPit 26 http://snapvoip.blogspot.com/
The SIPit, or Session Initiation Protocol Interoperability Test, is a week long event held to assist people and companies involved in SIP implementations to ensure they work well with each other.
OpenSIPS project will be there on May 17-21 to perform tests on IPv6, presence integration and presence aggregation and B2BUA functionality of OpenSIPS.
SIPit 26 will be held in Kista - the telecom valley outside Stockholm, Sweden on May 17-21, and will be hosted by Edvina and TANDBERG and sponsored by Ingate, Intertex and .se.
I will not be attending SIPit this year.

Register at SIPit at EDVINA

OpenSIPS Certified Professional (OCP) Program Launched.

OpenSIPS (OpenSER) http://snapvoip.blogspot.com/
If You are serving your clients solutions based on OpenSIPS (formarly OpenSER), now you can put them at ease with the OpenSIPS Certified Professional (OCP). The OCP credentials will show that you can design, implement and administer a SIP solution based on the OpenSIPS SIP Server.

Once certified, the credential will cover the abilities of the professional;

knowledge of Installation, SIP Routing, OpenSIPS Scripting Skills, Authentication, OpenSIPS Control Panel, Connection to the PSTN, Call Forwarding and Media Server Integration, NAT Traversal, SIP Server Load Balancing, Accounting and Monitoring Tools. 

To earn the OCP on OpenSIPS Server, you must pass a single exam:

• OpenSIPS Certified Professional Exam

Exam Details:

The OpenSIPS exam is a 60 minute test with 70 questions.

Register for an exam via exams@opensips.org .

Further information is available at OpenSIPS certification information page.

Take control of your OpenSIPS (Open SIP Server) With OpenSIPS Control Panel 4.0 Released

OpenSIPS-CP 4.0 snapvoip.blogspot.com
OpenSIPS org has released the the latest control panel, OpenSIPS CP 4.0 for the Open SIP Server. As the OpenSIPS team mentions, it is not just a system provisioning tool, but a complete OpenSIPS provisioning interface.
OpenSIPS-CP 4.0 brings users and admin management tools allowing you to add, edit, remove OpenSIPS users (see online/offline users) as well as add administrators (with permissions) for the OpenSIPS CP.
Also OpenSIPS CP 4.0 brings you more system tools like "dialog" tool (for monitoring and terminating ongoing dialogs) and "permissions tool (for IP based authentication for OpenSIPS).

Download is available under http://opensips-cp.sourceforge.net/
Take control of your OpenSIPS!  Also do not forget to check out the OpenSIPS webinars to learn more about OpenSIPS.

Free SIP based VoIP Service From OpenSIPS

Free SIP Accounts 
OpenSIPS will be present at Astricon, and all the links at the end of the post snapvoip.blogspot.com
Now I can add another Free SIP Account provider to the list as OpenSIPS is providing a free VoIP Service with support from Voice Systems.
The service is a full blown residentioal like service and provides an opensips.org SIP address and enables to place VoIP calls to any SIP network.
current features include;

Calling

• aliases
• speeddial
• DND service
• anonymous and anonymous blocking
• click to call from web interface

Call forwarding (on server)

• permanent forward
• forward on not registered
• forward on timeout (not available)
• forward on busy

Call filtering

• inbound and outbound call filtering
• white/black lists approach
• reg exp based filtering

IM

• end-2-end messaging
• message storing on server if off-line
• read/delete off-line messages from web.

Presence

• presence server
• XCAP support (optional)
• view presence status via web interface

Phonebook

• click to call from web interface
• IM from web interface
• search users in white pages

CDRs

• incoming call records
• outgoing call records
• missed call records 

and conferencing and voicemaill will be following up soon.
I have registered myself and it was painless two minute affair.
To register for a new account follow this link
Information on Astricon Presence

Integrating Asterisk And OpenSIPS In Realtime : Tutorial

Integrating Asterisk And OpenSIPS

We have been working with Asterisk and OpenSIPS for a long time (Even when OpenSIPS was OpenSER!). But we had to hammer out solutions as they fit and there were no methodology to it. Various engineers had their notes and were collected at one place only to be pulled out when the need arises to bring these two Open Source Telephony servers together.
SIP server like OpenSIPS usually provides registration and call handling, either by routing, forwarding or direct connections. The Asterisk on the other hand is a full blown telephony server, providing IVR, Voicemail, conferencing and announcements in addition to call handling, So by bringing these two servers (services) together, one will be able to provide full blown telephony solution integrating SIP, and Asterisk services.
So without saying that our method(s) are messy, I am going to point you to a very nicely written (Also very nicely thought out) article on how to integrated Asterisk and OpenSIPS.
Bogdan Andrei Iancu, a core OpenSIPS developer has written the tutorial. Go read it, Print it and save it! It is very good!
PS The tutorial currently addresses Asterisk 1.4 or 1.6 and OpenSIPS 1.5x. Once the OpenSIPS 1.6 is released the documentation will be updated appropriately.
Realtime OpenSIPS - Asterisk Integration

OpenSIPS 1.5.3 Ready For Download

OpenSIPS 1.5.3 is released
OpenSIPS (Formaerly OperSER) has released OpenSIPS 1.5.3, a minor release from the OpenSIPS 1.5 branch addressing bug fixes. There are no changes to no functionality, scripting or interfacing made.

The bug fixes are related to transaction, dialog and presence support. The team advise us all to upgrade, as this OpenSIPS 1.5.3 provides stability improvements

OpenSIPS 1.5.3 could be downloaded from OpenSIPS web site and SourceForge. Visit the Downloads page.

If you need more information, a complete Changelog is available here.

Tags: OpenSIPS, OperSER, OpenSIPS 1.5.3

AstLinux 0.6.7 Ready For Download

AstLinux Distribution
Kristian Kielhofner has released a newer version of AstLunux distribution, AstLinux 0.6.7. He has made a quite a few changes and those are available from his site.
AstLinux is a custom Linux distribution built around Asterisk, the Open Source IP PBX. But you will see more than Asterisk with AstLinux. Many VoIP applications, many of them, my favorites, like OpenSIPS and FreeSWITCH are also made available from the project.
AstLinux is an ideal choice for embedded and / or commercial Asterisk based solutions. I have been running an AstLinux server for a while now.
If you already have AstLinux, you can upgrade via upgrade function in the web interface. You can also do the same from the command line:
"upgrade-run-image upgrade http://mirror.astlinux.org" sans quotes. If you are new, download the complete ISO or the img.
More information available from AstLinux site.

Tags: ASTLinux, OpenSIPS, FreeSWITCH, Asterisk

SIP Routing, A Webinar By OpensSIPS

Routing in SIP
We here are big fans of OpenSIPS (Formally OpenSER). In fact it is our favorite SIP engine. We have been involved with this SIP Server ever since it forked from SER.
The OpenSIPS is very capable SIP engine and is capable of handling large volumes of users and traffic. But we tend to make mistakes due to lack of understanding and the OpenSIPS have steped in to help us all. The second in series of webinars will be presented this month.
(If you need further information on OpenSIPS, you can visit OpenSIPS website and There is also a Recorded webinar, Introduction to OpenSIPS. Links all at the end).

Next Webinar;

Routing In SIP

Date: Monday, July 8, 2009
Time: 8:00 PM - 9:00 PM EEST (5:00 PM - 06:00 PM GMT)

This webinar will help you to understand SIP routing concepts. SIP routing is very often a misunderstood concept. It is essential for anyone trying to script OpenSIPS to fully understand these concepts.

Registration form : https://www2.gotomeeting.com/register/429289130

Registratio is free but the seating is limited. So register early.

Former webinar held on June 8th June, Introduction to OpenSIPS, and a video recording of this webinar will help you to get in tune..

Tags: OpenSER, OpenSIPS, SIP Routing, SIP Education, SER, Routing In SIP, Introduction to OpenSIPS

OpenSIPS CP 2.0 Is Out

OpenSIPS Control Panel 2.0 is released

OpenSIPS-CP version 2.0 is a major release from the OpenSIPS Control Panel project. OpenSIPS-CP is a Web Control Panel Application for OpenSIPS server and is currently compatible with OpenSIPS 1.5.x.

In addition to the fixes for several bugs, the new release enhances the existing tools and have streamlined DB scheme (definitions and interwork with OpenSIPS DB). Another improvement is the extension of the drouting tool by adding list-based route handling, giving the freedom of choosing between creating your own list of gateways or using a predefined list.

What you find new in this version of control panel is;

• dispatcher - provisioning of the dispatcher lists via DB, cache reload via MI
• nathelper - provisioning of the RTPproxy sets via DB, cache inspection and reload (MI), RTPproxy enable/disable (via MI), ping enable/disable (MI).
• permissions - provisioning via DB of the trusted set of addresses.

OpenSIPS-CP 2.0 is now available for download on SourceForge.

Tags: OpenSIPS-CP 2.0, OpenSIPS, drouting, RTPproxy sets via DB, dispatcher lists via DB, OpenSIPS Control Panel

OpenSIPS 1.5.0

OpenSIPS 1.5.0 Released
This slipped under our notice but the OpenSIPS team released the latest OpenSIPS server version 1.5.0 on March 23rd, about six month since the last major version release.
While there are are major improvements in Database, Management interface and dialog support, the new additions to the SIP server stands out. These include cache support, Load Balancing, PrePaid support, SIP Identity , Dynamic Routing, IP geo location, to name a few but you can find all the new additions on what is new on 1.5.0.
The team has compiled a manual to help version 1.4 users to migrate to the new OpenSIPS 1.5.0 and is very helpful as there are some major changes need to be addressed such as the new database structure. (but it is much easier than it sounds!)
A set of links, release notes, OpenSIPS Download and other helpfule links are available from the release news page.

Tags: OpenSIPS 1.5.0, SIP Server

Load Balance SIP Servers With OpenSIPS

http://snapvoip.blogspot.com/
OpenSIPS could be used to load balance media servers with the "Load Balance" module. The module provides capability to rout traffic based on the load.
The LB module defines the offered resources, and for each resource, it defines the capacity / maximum load as number of concurrent calls the destination can handle for that resource.
For an example derived form OpenSIPS docs;The destinations are assigned with their resources ready to share and as mentioned the capacities. OpenSIPS Load Balancer Module documentation on OpenSIPS Load Balancing - how it works at OpenSIPS provides detailed directions to setup load balancing a set of Yate / Asterisk servers.
Tags:OpenSER,OpenSIPS, YATE Project,

OpenSIPS At Amoocon (Former AsteriskTag)

http://snapvoip.blogspot.com/
OpenSIPS project will be presenting papers on May 4th - 5th, 2009 at AMOOCON (Formerly AsteriskTag)in Rostock (Germany). The bilingual conference on Open-Source VoIP and Open-Source communication. AMOOCON was formally Asterisk-Tag and thay changed the name to fit the changing field of Open Source VoIP. AMOOCON now includes Yate and FreeSwitch in addition to Asterisk.

Among many hot topics related to Open Source VoIP will be presented by various organizations and OpenSIPS will focus its efforts on very interesting subjects.

The papers already submitted include subjects covering,

Load-balancing (using the new load-balancer module) for media servers, Workshop

Deploying and delivering Presence based Services with OpenSIPS, Workshop

SIP clustering (OpenSIPS + PBX integration), Workshop

All are invited.
For more information visit OpenSIPS and AMOOCON

Tags: OpenSIPS, Open Source VoIP, AMOOCON, AsteriskTag, Asterisk-Tag, Yate, FreeSwitch, SIP clustering, Presence based Services with OpenSIPS, Load-balancing Media Servers

OpenSIPS 1.4.4 And A OpenSIPS Bootcamp, Awaits you.

http://snapvoip.blogspot.com/
In the footsteps od OpenSIPS 1.4.3 (production) release, OpenSIPS is offering us a new minor release, OpenSIPS 1.4.4. This release fixes a set of major and minor fixes on the OpenSIPS 1.4 branch.
The OpenSIPS team recommends to upgrade to this release, as it brings stability improvements to the already a fantastic system.
For those who are interested, complete change log could be found here.
OpenSIPS 1.4.4 is now available for download at OpenSIPS and Sourceforge.

For those interested in training on the OpenSIPS system, there is a OpenSIPS Bootcamp in sunny Brazil, 16-20 February, 2009, Florianopolis.

Tags: OpenSIPS Bootcamp, OpenSIPS 1.4.3, OpenSIPS 1.4.4, Sourceforge, Open Source VoIP, OpenSIPS, SIP

OpenSER 1.3 and MediaProxy 1.0 Has Reached End Of Life

http://snapvoip.blogspot.com/

Multimedia Service Platform based on OpenSER 1.3 and MediaProxy 1 has reached its end of life. The new platform is based on OpenSIPS 1.4 and MediaProxy 2.0. Customers are required to schedule a platform upgrade to migrate to the latest software versions. The migration requires changes in the SOAP/XML provisioning client. Customers can update their SOAP/XML clients by using AG Projects testing facility at:

https://secure.dns-hosting.info/ngnpro/wsdl

The changelog for the latest software versions is found at:

https://secure.dns-hosting.info/debian/changelogs.txt

Tags: OpenSER, OpenSIPS, MediaProxy, OpenSER 1.3, MediaProxy 1.0, OpenSIPS 1.4, MediaProxy 2.0

Call Control (Prepaid Support) In OpenSIPS

http://snapvoip.blogspot.com/

Call Control is a new application for OpenSIPS (formerly OpenSER) and used together with call_control module and CDRTool rating engine to manage the time duration of SIP sessions based on a prepaid amount. One is also able to control the time duration of SIP sessions without having to deal with prepaid accounting.

Both the above are handled by Call Controll by using a timer to track maximum time allocated and to sending a BYE signel to both the ends of the SIP session, when the limit is reached. Call control could also be used to forcibly close a session by a third party.

Following features are listed by the release notes and you can download Call Control from here.

• Parallel sessions using one balance per subscriber
• Support for sessions that have timeout without BYE
• Support for sessions that have timeout for media (using MediaProxy)
• Manual session stop from server terminal
• Overview of ongoing sessions in the web page or server terminal
• Graceful restart without loosing track of sessions
• Detailed logging of all performed actions using syslog
• Web page provisioning for prepaid accounts (using CDRTool)

Also a very well written installation guide, TXT file, PDF File.

Tags: Call Control, Prepaid Support, OpenSIPS, CDRTool, MediaProxy

Media Relay MediaProxy 2.1.0 Released!

http://snapvoip.blogspot.com/
The development team responsible for developing and managing to MediaProxy project has release MedaProxy 2.1.0. MediaProxy is a media relay for RTP/RTCP and UDP streams that works in tandem with OpenSIPS to provide NAT traversal capability for media streams from SIP user agents located behind NAT.

When using MediaProxy, NAT traversal for RTP media will work without any settings in the SIP User Agents or the NAT router.


Main Features
--------

- Scalability of thousands of calls per server limited only by the Linux kernel networking layer and network interface bandwidth
- Supports multiple chained relays as long as each has a public IP
- TLS encryption between the relays and dispatcher
- T.38 fax support
- Graceful shutdown capability
- Modular design for custom selection algorithms for the relay location
- Automatic load balancing and redundancy among all media relays
- Real-time sessions statistics
- Configurable IP and UDP port range
- Support for any combination of audio and video streams
- Ability to use OpenSIPS' MI interface to close a call that did timeout
- Radius accounting of IP network traffic
- Database accounting of complete media information including all streams, their type, codecs and duration.

Tags: mediaproxy, OpenSIPS, NAT traversal, SIP user agents, T.38 fax

OpenSIPS 1.5.0 Gets A New Routing Module.

http://snapvoip.blogspot.com/
OpenSIPS announced that OpenSIPS 1.5.0 (development branch) has a new module for routing. Even though it is similar to LCR module in basic functionality like routing based on rules, the major difference seems to be in ability to scale and high performance.

The OpenSIPS site noted that the new module is following capabilities and enhancements;

Internal data organized as a search tree which is very fast
Unlimited number of rules and gateways (Tested on a production setup with 200K rules and with no search penalties)
A rule may point to set of GWs to be used for;
(1) sequential selection, (2) random selection , (3) weight selection
Inbound processing - prefix and strip for calls originated by the GWs

The module already sports an intuitive web interface for provisioning which will released in the near future, along with an OpenSIPs Control Panel Web interface.

A comprehensive documentation and description available from the OpenSIPS development site.

Tags: OpenSIPS, OpenSER

OpenSIPS At SIPIT and My Loyalty!

Solarion seems to be head over heals with OpenSIPS but after digging deep into the story behind OpenSIPS, I tend to siding with Solarion. Like Solarion, I also got my SIP feet wet with SER. Even though stiff and don't feel like at home was all around IPTel, We hung around because there were not many other resources providing SIP down to nuts and bolts SER did.
So When OpenSER came out with new development and management philisophy, I was over the wall! I started converting all my SER servers (quite a few) in to OpenSER servers. Even while doing so I could not belive the version numbers climbing up. I was so used to SER being stuck on version 0.7 I think, I had a party when OpenSER reached version 1.0. Because I knew then SER was history stuck is version 0.9.
Now the tables have turned and for what ever the politics or other reasons, OpenSER has been forked and two new entities, OpenSIPS and Kamailio (I feel good now as I thought I was bad at picking names!).
If you have noticed, I still have a link to SER, on the links on the right. I guess it will be there for giving me first real taste of SIP. But otherwise, I have moved on. And like the Solarion, I am leaning towards OpenSIPS and I will be playing with OpenSIPS 1.4 this weekend. May be Kamailio afterwards.

But I know where I will see OpenSIPS. as they have promised to be at SIPIT, so will I. I will see my Mom and then drive to Lannion as I have not done this great past time I learned in the USA, Driving long distances! Yes I know it is expensive and environmentally incorrect, but like Mr. Gore, I pay for my Carbon foot print else where in the world, I have done so even befor being green became the buzz word of the century.

So I picked up the following information from OpenSIPS site;

OpenSIPS at SIPIT

OpenSIPS project is registered to participate the 23th edition of SIPIT - SIP Interoperability Test event in Lannion, France, October 13-17, 2008.
Compliance and interoperability are essential parts of project testing and participating to such large testing events is a natural step in the QA process.

If there are people interesting in being part of the OpenSIPS test team, please let us know (via the users mailing list).

Tags: OpenSER, OpenSIPS, Kamailio, SIPIT, Lannion, Going Green, SIP Interoperability Test