Last stable series is 3.0.x (Jan 11, 2010, see release notes), the first based on SIP-Router.org project. Among brand new features, it lets you run mixed Kamailio (OpenSER) and SIP Express Router (SER) modules in the same SIP server instance, giving you the most powerful tools to build stable, very performant and features rich VoIP and Unified Communication platforms.

Next class Kamailio Advanced for Carriers, organized by Asipto in collaboration with IPCom Network and NAP of the Americas, focuses on the features that allow you to build a carrier SIP infrastructure:

• least cost routing
• load balancing
• traffic dispatching
• DID routing
• prefix based routing
• SIP trunks and peering
• traffic shaping
• topology hiding
• flood detection
• enum
• authentication and authorization
• accounting
• redundancy
• high availability and failover

Location: Miami, USA
Date: June 21-23, 2010
Trainer: Daniel-Constantin Mierla
Registration price starting at: 1795 USD

More details and registration at:

http://www.asipto.com/index.php/kamailio-advanced-training/

Teachers:

Daniel-Constantin Mierla – co-founder of Kamailio (former OpenSER) project in 2005, currently core-developer and member of project’s management board

Olle Johansson – Asterisk developer and member of the Digium Asterisk Advisory Board.

End of 2008, Kamailio (OpenSER) and SIP Express Router (SER) started a joint collaboration under http://sip-router.org project, bringing together valuable developers and architects of SIP servers. Kamailio (OpenSER) is now at version 3.0.0 (released on January 11, 2010), being based on SIP-Router.org project.

Learning to configure the SIP server is not easy, but is the key for a successful and secure VoIP business. The flexibility of SIP routing engine allows you to implement in no time innovative services, IP telephony, Instant Messaging, Presence and beyond. Asterisk comes to complete with rich media services and applications. Doing everything designed right and scalable saves time and money.

We create the opportunity for you, guided by experienced instructors, to learn how to build an Unified Communication platform from scratch using the SIP server engine and Asterisk.

Click here for course details and registration.

Siremis enables straightforward management of subscriber profiles, least cost routing and load balancing rules, communication at runtime with the SIP server, displays monitoring charts. You find detailed list of feature at:

http://siremis.asipto.com

v1.0.0 is a major release introducing many new features:

• presence service management – watchers, active watchers and presentity
• sip trace view – access sip_trace module records, highlight headers and group by call-id
• user location statistics charts – online user agents, supported SIP methods, contacts and nat related charts
• update of the core components to latest stable versions
• update to work with latest Kamailio (OpenSER) v3.0.0
• patch to work with older Kamailio (OpenSER) v1.5.x
• time-based X-axis for charts print labels as HH:MM
• charts types supported: line, line dot and area

Download and installation steps:

http://siremis.asipto.com/install/

Some screenshots specific for this version:
http://www.asipto.com/gallery/v/siremis/siremis_22.png.html
http://www.asipto.com/gallery/v/siremis/siremis_23.png.html
http://www.asipto.com/gallery/v/siremis/siremis_24.png.html

More screenshots:

http://www.asipto.com/gallery/v/siremis/

Demo site (it works on a database with random data, username: admin, password: admin):

http://siremis.asipto.com/demo/

The class duration is six months and gives the opportunity to learn how to write proper configuration files for SIP Router and Kamailio (OpenSER) SIP servers, how to take care of VoIP security and scale the service.

Target attendees:

• VoIP administrators willing to learn and have a support channel for SIP Router configuration file

• System and network administrators willing to enhance their portfolio with VoIP knowledge

Please use the contact form for registration details.

Daniel-Constantin Mierla, co-founder Kamailio (OpenSER), announced availability of version 3.0.0 of the open source SIP server.

It is a major release of Kamailio (OpenSER), following ten months of development and heavy testing. Represents a special release, being based on SIP Router project, therefore enables admins to blend Kamailio (OpenSER) and SIP Express Router (SER) core features and modules in same configuration file.

Given the above, this version brings out an impressive number of new features and improvements, read the release notes at:

http://www.kamailio.org/mos/view/Kamailio-OpenSER-v3.0.0-Release-Notes

Asipto will release soon a new version of SIREMIS, the web administration interface for Kamailio, to work out of the box with v3.0.0. ]]> http://siremis.asipto.com/2010/01/11/kamailio-openser-3-0-0-released/feed/ 0 http://siremis.asipto.com/2009/10/05/siremis-v0-9-4-released/ http://siremis.asipto.com/2009/10/05/siremis-v0-9-4-released/#comments Mon, 05 Oct 2009 10:32:07 +0000 admin http://siremis.asipto.com/?p=181 A new version of SIREMIS Web Management Interface for Kamailio (former OpenSER) is available as v0.9.4.

Siremis enables straightforward management of subscriber profiles, least cost routing and load balancing rules, communication at runtime with Kamailio, displays monitoring charts. You find detailed list of feature at:

http://siremis.asipto.com

v0.9.4 is a patch release fixing bugs reported since v0.9.3.

Download and installation steps:

http://siremis.asipto.com/install/

View screenshots:

http://www.asipto.com/gallery/v/siremis/

Demo site (it works on a database with random data, username: admin, password: admin):

http://siremis.asipto.com/demo/

From InfoWorld site:

“Award winners in network and network management are old favorites Cacti and Nagios, the IPCop firewall, Kamailio SIP proxy server, KeePass password manager, Openfiler SAN/NAS appliance, OpenNMS enterprise monitoring system, PacketFence network access control solution, Puppet configuration management framework, and Untangle network security gateway.”

“Kamailio is the open source SIP proxy server formerly known as OpenSER. Used with an Asterisk IP PBX server for phone features, plus a hardware gateway for connection to the outside world, Kamailio brings important call handling and scalability benefits to Asterisk, while also removing the Asterisk server as a single point of failure. Larger organizations get the phone features they need, as well as the added safety of VoIP calls surviving an Asterisk server outage.”

On the other hand, Kamailio project is approaching the time for a new major release — a matter of days to enter the testing phase — this one will be versioned Kamailio 3.0 to reflect the major enhancements, among them: the http://sip-router.org core, asynchronous TCP, asyncronous SIP message processing API, memcached backend, native topology hiding, nat traversal using kernel space for media relaying … see more at:

http://sip-router.org/benefits/

http://sip-router.org/wiki/features/new-in-devel

Teachers:

Daniel-Constantin Mierla – co-founder of Kamailio (former OpenSER) project in 2005, currently core-developer and member of project’s management board

Olle Johansson – Asterisk developer and member of the Digium Asterisk Advisory Board.

By end of 2008, Kamailio (OpenSER) and SIP Express Router (SER) started a joint collaboration under http://sip-router.org project, bringing together valuable developers and architects of SIP servers. Kamailio 3.0 and SER 3.0 (to be released soon) become compatible in terms of configuration file and extensions.

Learning to configure the SIP server is not easy, but is the key for a successful and secure VoIP business. The flexibility of SIP routing engine allows you to implement in no time innovative services, IP telephony, Instant Messaging, Presence and beyond. Asterisk comes to complete with rich media services and applications. Doing everything designed right and scalable saves time and money.

We create the opportunity for you, guided by experienced instructors, to learn how to build an Unified Communication platform from scratch using the SIP server engine and Asterisk.

Click here for course details and registration.

• communication with SIP-Router.org and SIP Express Router (SER) via  XMLRPC (RPC command interface)
  • pear package XML_RPC must be installed
• communication with FreeSwitch via event socket
  • no extra library required, implemented as library using TCP sockets
• support to print rich html formatting of MI, XMLRPC and FreeSwitch command response (config option)
  • makes use of rich text editor features, output can be formated using the HTML languages (previous version displayed the output as text plain in a textarea)
• stand-alone PHP classes for MI, XMLRPC and FS Event Socket, making further development easier
  • the code for communication with SIP applications is now separate of the view php code
  • php classes can be easy pulled and used in other PHP applications
• output of commands can be filtered for nicer display (e.g, see FS  help, xmlrpc MI commands)
  • current architecture makes easier to highlight parts of the command output, now just couple of commands benefit of pretty formating, the others print raw format (e.g., xmlrpc response)
• MI, XMLRPC and FS Event Socket groupped under Cmds Panel tab
  • dedicated tab for external commands
• updated phpopenbiz core
• fix for re groups db table
• link referencing on username between group view, db aliases view and subscriber view
  • click on an username value jumps to subscriber view matching that user
• more options to sort the views using column names
  • more column names become clickable to sort the records
• auto-suggestion for group name when adding a new record
  • auto-completion as you type the group name

Download and installation steps:

http://siremis.asipto.com/install/

View screenshots:

http://www.asipto.com/gallery/v/siremis/

Demo site (it works on a database with random data, username: admin, password: admin):

http://siremis.asipto.com/demo/

Teachers:

Daniel-Constantin Mierla – co-founder of OpenSER/Kamailio project in 2005, currently core-developer and member of project’s management board

Olle Johansson – Asterisk developer and member of the Digium Asterisk Advisory Board.

By end of 2008, Kamailio (OpenSER) and SIP Express Router (SER) started a joint collaboration under http://sip-router.org project, bringing together valuable developers and architects of SIP servers. Kamailio 3.0 and SER 3.0 (to be released soon) become compatible in terms of configuration file and extensions.

Learning to configure the SIP server is not easy, but is the key for a successful and secure VoIP business. The flexibility of SIP routing engine allows you to implement in no time innovative services, IP telephony, Instant Messaging, Presence and beyond. Asterisk comes to complete with rich media services and applications. Doing everything designed right and scalable saves time and money.

We create the opportunity for you, guided by experienced instructors, to learn how to build an Unified Communication platform from scratch using the SIP server engine and Asterisk.

Click here for course details and registration.