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Tuesday, February 28, 2012

Kamailio 2011 Awards

End of winter, therefore it's now time for the 5th edition of Kamailio Project Awards, granted for the activity during 2011, like in the past, each category has two winners.



First, I want to thank to everyone contributing to and using Kamailio/SER during 2011, their effort made possible to release a new major version with tons of new features - see release notes for v3.2.0.

The unconventional winners of 2011 are:
  • the project itself - it succeeded to survive over 10 years since it started. There were many critical moments in the past (forks, renames and such), but it went through, becoming even stronger now, with an exceptional support from an amazing community.
  • the development team - it succeeded to get on board lot of new contributors. This made it very hard to select the winners for Developer Remarks awards. According to Ohloh statistics, there were over 30 registered developers contributing code during the past 12 months.


The new category this year is Friends of Kamailio, for persons that helped for many years in various occasions, in background or foreground, and were always promoting and saying good stories about the project.

Next are presented the categories and the winners.

 Blogging:
  • CSDN - the "Chinese Software Developer Network" - for spreading out news and tutorials about Kamailio, our Asian user base grew substantially lately.
  • Jason Penton and Richard Good (the two being colleagues, were tied together here) - for many useful posts of how to deal with Kamailio on Solaris. The two are contributing code to the IMS extensions in our project.
Related Projects:
  • Gemeinschaft - a project that aims to provide a secure, scalable and easy to use open source PBX platform built with Kamailio and FreeSWITCH, developed by Amooma and sponsored by the German Federal Office for Information Security (BSI).
  • Homer - SIP Capturing Server started by Alexandr Dubovikov, developed first time for Kamailio SIP Server project and adopted by other VoIP applications. In one side, Kamailio can be configured to mirror the SIP traffic, on the other side Kamailio can receive the mirrored traffic and save it to database from where it can be searched for various keywords via a web interface. Then, it can build diagrams from results or the results can exported in pcap format.
Technical Support:
  • Jon Bonilla - for setting up the Debian/Ubuntu APT repositories, updating the .deb specs and maintaining them. It made the spreading through packages and installation lot more easier for these distributions.
  • Laura Testi - for substantial feedback, testing and many patches on our SIP SIMPLE Presence extensions. I I would award development topics, SIMPLE Presence extensions would have been among the top ones in 2011 (user presence, dialog states (aka blinking lamps), resource list services, embedded XCAP server, SIP-XMPP presence gateway). Laura helped a lot to improve and develop further these components.
 New Contributions:
  • dmq - Distributed Message Queue on top of SIP - the module presents huge potential and building a native distributed platform with Kamailio, personally I expect several modules to use it soon as underneath layer for data exchange between SIP server nodes. The developer is Marius-Ovidiu Bucur.
  • json - JSON Parser and JSONRPC Client - json format has become very popular, being able to interact directly from the config file, opens the way to integrate easier with backend structures used for Web 2.0 services. As a plus, JSONRPC client uses the asynchronous processing framework added starting with v3.0.0. The developer is Matthew Williams.
Developer Remarks:
  • Peter Dunkley - he and Crocodile RCS team contributed major enhancements to SIP SIMPLE Presence extensions towards RCS/RCSe (e.g., user presence services, resource lists services, embedded xcap server), as well as scalability. For a fair acknowledgement of contributed work, I am mentioning also his colleagues Paul Pankhurst, Hugh Waite and Andrew Miler.
  • Timo Terras - by contributing SQLite database connector, running Kamailio on embedded systems entered in a new era. The primary choice in the past as DB backed for embedded devices was dbtext module, but it has limitations dealing with large records. With SQLite, any module can be fully functional on very small devices, from user authentication and authorization, sqlops, to location and presence services.
Advocating:
  • Carsten Bock - for participating at, organizing and sponsoring several events related to Kamailio SIP server as well as publishing blogs and news about the project.
  • Henning Westerholt - another year with Henning on traveling to most of our public events, speaking about the over 3 000 000 VoIP users platform he is involved in, with special credits for taking care of our LinuxTag 2011 presence, booth and slides.
VoIP Services:
  • Portugal Academic Network - for running a large grid of Kamailio and Asterisk servers (about 500) to provide communication services. Credits to Ruben Sousa (at Astricon 2010) and Olle E. Johansson (at 10 Years SER event - the slides from previous link) for sharing their experiences in deploying this VoIP network with us.
  • Telio - one of the oldest companies with development contributions to SER/Kamailio project, Telio continued to have in 2011 good revenue growth. I am adding also a special remark for its movement towards mobile networks, with launch of Goji application.
Business Initiatives:
  • Crocodile RCS - one of the few players in the vendors market with a clear target for Rich Communication Services, their team was very active in the development process of Kamailio during last year. Mobile operators are in a rush to roll out value added services, especially in the social networking area - real time instant messaging and presence are going to play a crucial role for operators in their battle with the big Internet companies.
  • Frafos - with a business oriented mainly to SIP Express Media Server (SEMS), the project started in the same place as SIP Express Route (SER), having many common developers, the company can help deploying addition components in Kamailio-based VoIP platform, to achieve functionalities such as media server (voice mail, conferencing, IVR), back to back user agents or session border controllers.
Events:
  • 10 Years SER - the project crossed first time the age of 2 digits, a good opportunity to meet and look at the past, present and future. About 50 people spent a great day at FhG Fokus in Berlin, from first developers to our latest users. There were 15 presentations during the day and a relaxing grill party in the evening.
  • LinuxTag - we were several time in a raw at LinuxTag with a booth for the project and presentations about it, but the last year we had the opportunity to meet in the same place 5 out of 11 members of the management team, pretty rare situation as we are distributed over many countries.
Academic Environment:
  • Columbia University - for using our SIP server to conduct various research projects since 2002, results from some of them being useful and relevant for development and scalability of SER/Kamailio platforms. The Green VoIP paper showing interest results, like ability to cope with about 43 000 active TLS connections on a server with 2GB memory allocated for the SIP server. Worth to mention that Jan Janak, one of the major contributors to the SIP server, has been involved in some of the research projects.
  • FhG Fokus Research Institute - for being a host of 10 Years SER event, but overall for starting in 2001 and funding the development of our SIP server project for many years. Also, during the recent past years, Fokus hosted several developer meetings. Another relevant contribution to the VoIP environment coming from Fokus is the OpenIMSCore project, developed on top of SER/Kamailio, several components from it being integrated in our project.
Friends of Kamailio:
  • Olivier Taylor - for helping to organizes our meetings in Brussels, at Fosdem events. By now, for the last several years in a raw, Olivier did his best to select nice places to meet and enjoy great time with folks around Kamailio project and VoIP arena.
  • Suzanne Bowen - an exceptional friend of open source telephony applications, Suzanne was ready to help always with written interviews and podcasts, free invitations to conferences and exhibitions, setting up connections, recommendations and discussions to various business entities, to reveal in the best way the benefits and strong points of Kamailio.
As of Personal Facts related to the project, I continued to release complete tutorials of using Kamailio, very useful and actual would be Bridging IPv4-IPv6 VoIP Networks, Secure Communications with Kamailio (building own Skype service alternative) - see all of them at:
This is it for 2011. If you want to check the previous turn of awards, visit:

Monday, February 27, 2012

Social Networking Event, London, March 6, 2012

Several folks involved in Kamailio SIP Server and VoIP in general (including me, Daniel-Constantin Mierla, co-founder Kamailio) are organizing a social networking event in London, during the evening of 6th of March.

If you want to participate, send us a short note via registration form:
Be sure your email is valid in order to send you the details of the location.

The event will be dinner/pub-drinks style, with the goal of discussing the latest developments of Kamailio project as well as what is new in VoIP and Unified Communications world.

Wednesday, February 15, 2012

Unified Communications Expo 2012

UC Expo 2012 takes place in London, UK, between March 06-07, 2012. Asipto representatives will be present this year as well at the event, meeting many of our UK customers base that will exhibit at the show.

UC Expo describes itself as the show mirroring the diversity of Unified Communications by bringing together all the key technologies and key people of this rapidly evolving world.

If you want to meet with me (Daniel-Constantin Mierla of Asipto, co-founder and core developer of Kamailio SIP Server project), feel free to contact us:

Tuesday, February 14, 2012

Call Center World 2012

I will be present at Call Center World congress in Berlin, February 27 – March 01, 2012. If happens for you to be around and want to meet, feel free to contact us.

The event gathers over 250 exhibitors from around the globe, mainly focusing on help desk and support solutions, integrating SIP/IP and TMD for call centers. Along with the exhibition, there are workshops and the conference.

Asipto’s offerings include reliable solutions to scale and enhance security as well as add new features to call center oriented systems. For example, the load balancing solutions can be used to scale call center capacity in a transparent and flexible manner.

Thursday, February 9, 2012

CeBIT 2012

CeBIT 2012, the biggest digital show, takes place in Hanover, Germany, March 06 – 10, 2012. Daniel-Constantin Mierla (me in other words) of Asipto, co-founder and core developer of Kamailio SIP Server project, is visiting the event.

The exhibition has dedicated pavilions for Telecommunication Industry, from equipment providers to software integrators. Asipto is glad to see several customers exhibiting there and we will be delighted to meet and discuss with the other participants at the event, as well.

If you want to schedule a meeting during the CeBIT 2012, don’t hesitate to contact us:

Wednesday, February 8, 2012

Presentation at Fosdem 2012

Fosdem 2012 included a DevRoom for Open Source Telephony, Daniel-Constantin Mierla of Asipto participated and presented “Secure SIP Communication with Kamailio”.

First part focused on an overview of the project, history and latest new features, then continued to present the tools offered by Kamailio to achieve strong security in your VoIP deployments.

You can find the slides (pdf) at:
If you have questions related about SIP security or look for consultancy in this area, our experienced team can help you, feel free to contact us.

Kamailio at VoIPUsersConference, Feb 17, 2012

Daniel-Constantin Mierla (co-founder) and Alex Balashov (member of management board) will coordinate a new session of VoIP Users Conference (VUC) about Kamailio SIP Server, on Friday, the 17th of February, 2012, starting at 17:00GMT. There will be other developers and users around, ready to answer whatever questions you may have.

VUC is the well know weekly online conference that allow everyone to connect via SIP (voice) and/or IRC (text) to interact with the guests. The host and moderator will be Randy Resnick. You can read more details about this weekly event at:
To participate, choose your preferred way to connect:
  • SIP: 200901@login.zipdx.com
  • Skype: vuc.me or ld.vuc.me
  • IRC: #vuc channel on irc.freenode.net
  • PSTN: +15672522286
The session targets first to give an update on latest developments: asynchronous TLS layer, asynchronous SIP processing, SIMPLE presence extensions, embedded HTTP-XCAP server and MSRP relay, no-SQL storage systems (Redis, Cassandra), embedded interpreters (Lua, Python, C#), a.s.o.

Besides hearing what is new, it is a good chance for everyone to ask and learn about how Kamailio can be used by service providers to meet today’s communication demands: integrated voice, video, instant messaging and presence services, load balancing and least cost routing, security and confidentiality, scalability and redundancy, SIP in IPv6, interaction with web 2.0 and social networking services, …

The previous VUC session about Kamailio and SIP Express Router was done almost two years ago, you can read more and listen the recorded podcast at:

Tuesday, January 31, 2012

Kamailio v3.2.2 Released

Kamailio SIP Server v3.2.2 stable is out – a minor release including fixes in code and documentation since v3.2.2 – configuration file and database compatibility is preserved.
Kamailio (former OpenSER) 3.2.2 is based on the latest version of GIT branch 3.2, therefore those running previous 3.2.x versions are advised to upgrade. There is no change that has to be done to configuration file or database structure comparing with older v3.2.x.
Resources for Kamailio version 3.2.2
Source tarballs are available at:
Detailed changelog:
Download via GIT:
# git clone –depth 1 git://git.sip-router.org/sip-router kamailio
 # cd kamailio
 # git checkout -b 3.2 origin/3.2
 # make FLAVOUR=kamailio cfg
Binaries and packages will be uploaded at:
Modules’ documentation:
What is new in 3.2.x release series is summarized in the announcement of v3.2.0:

Sunday, January 29, 2012

Presentations at Fosdem 2012

Fosdem 2012 includes again a dev room for Open Source Telephony, Kamailio SIP Server Project having a dedicated presentation “Secure SIP Communication with Kamailio”, by me (Daniel-Constantin Mierla).

Andreas Granig, also a developer of the project, will present another talk about SIP:Provider solution, which has Kamailio as the core components for SIP routing.

Related to our eco-system, Stefan Sayer will talk about SIP Express Media Server and its SBC functionality.

The schedule for telephony dev room is available at:

Friday, January 27, 2012

Kamailio & friends dinner at Fosdem 2012

Fosdem 2012 is approaching and we are going to have our traditional dinner at the event on the evening of Saturday, February 4.

At this moment should be over 15 participants, many Kamailio developers and community members, among them:
  • Henning Westerholt
  • Andreas Granig
  • Daniel-Constantin Mierla
  • Stefan Sayer
  • Olivier Taylor
  • Peter Dunkley
  • Raphael Coeffic
If you want to join us, send an email to registration@lists.sip-router.org. Register as early as possible, since we need to make a reservation, also the place cannot accommodate too many people. All friends of Kamailio, SER, SEMS, Asterisk, FreeSWITCH and SIP/VoIP are welcome!

Tuesday, January 17, 2012

New Module – Cassandra DB Connector

Courtesy of Anca Vamanu, of 1 & 1 Internet AG, Germany, a new module is available in the development version of Kamailio SIP Server (to become the 3.3.0 release). Here is an adapted excerpt done for this web news from the announcement sent to mailing list.

“The module is named db_cassandra and offers a DB interface that can be used by other modules to perform DB operations instead of other DB modules (like db_mysql for example).
Because Cassandra is a NoSQL storage system, it is not possible to run all kind of SQL-like queries on it and this is the reason why the module has some limitations.

It is especially suited for applications that store large data or that require data distribution, redundancy or replication. One usage example is a distributed location system in a platform that has a cluster of SIP servers, with more proxies and registration servers accessing the same location database.

This was actually the main usage we had in mind when implementing the module. It has been tested with usrloc and auth_db modules, but it can also be used with other modules that have similar queries.
You can read more about this module in the README file:
Inside the module directory you can find an example that modifies the default configuration file to enable a location service with a Cassandra backend. If you have a Cassandra installation, it should be very easy to test it.

We hope you find this module useful and are glad to receive any feedback, comments about it.”

Monday, January 16, 2012

New module – embedded MSRP relay

A new module in development branch of Kamailio SIP Server, named msrp, provides a MSRP routing engine, a.k.a. MSRP relay. The core specification of MSRP (Message Session Relay Protocol) is defined by RFC4975, the extensions for a MSRP Relay being covered in RFC4976. One of typical use case for MSRP is to do Instant Messaging sessions negotiated with SIP via INVITE-200OK-ACK.

The msrp is controlled from configuration file via actions in event_route[msrp:frame-in]. The module is a full, embedded MSRP relay, it does not require any external application nor library. It uses the core transport layer components, thus it benefits of the scalable and asynchronous TCP/TLS support implementation already existing in the project for many years now.

Kamailio, with msrp module loaded, can handle SIP and MSRP traffic received on the same port. But you can configure Kamailio as a stand alone instance to deal only with MSRP traffic, leaving the SIP traffic to another Kamailio instance. Also, another option is to configure Kamailio to listen on different TCP/TLS sockets (e.g., different ports or IP/network interfaces) and direct SIP and MSRP to different ports — then in the config file you can take care of filtering (dropping) inappropriate content on specific ports. With all this flexibility, you can choose a configuration that will not affect at all the routing of SIP messages with Kamailio.

The embedded MSRP relay, built on top of the SIP server, offers many benefits such as:
  • reuse mature code tested over the past 10 years, msrp module itself being really small piece of code in regards to MSRP protocol
  • MSRP is done over TCP/TLS, thus implicitly the forwarding is done asynchronously, offering great performances
  • IPv4 and IPv6 support
  • MSRP is for transmission of a SIP session content, going to be used by the SIP users in your UC platform — there is no need to manage a different user profile
  • the configuration and MSRP routing is done via the same flexible language and format as for SIP traffic, you being in control of what is passing through your server
  • access to all existing extensions that are related to SIP request routing, for example: IP address checking, flood detection, many database connectors, accounting, a.s.o.
You can read more about the msrp module in the documentation file:
At this moment, Kamailio offers a set of extensions that allows building a complete Unified Communication platform, within a single SIP server instance for small deployments as well as a grid of servers, each one doing particular functions:
  • voice, video, screen sharing, etc. sessions with content communication via RTP
  • end to end presence – this is purely SIP routing
  • SIMPLE-based presence (aka, presence server or presence agent model) via presence* and pua* modules — user presence, dialog states notification (aka, blinking lamps), resource lists service (including OMA/RCS extensions), user location states notification and replication, audio/video conference mixer notifications, a.s.o.
  • embedded XCAP server – management of user contact lists, presence policies, user agent configuration files, a.s.o. There is also an XCAP client extension
  • embedded HTTP server – for admin and user interaction with the service via pure HTTP or XMLRPC requests
  • embedded MSRP relay – for relaying and fine controlling of the message-based content of SIP sessions
  • IRC-style instant messaging conference via imc module
  • storage of instant messages for offline users and relay to them when they become again online via msilo module
All above components are built on the same solid foundation, practically is Kamailio core plus a selected set of modules, no extra dependencies, just configuration options.

Saturday, January 14, 2012

Per socket number of worker processes

The development branch of Kamailio SIP Server has a new feature that allow setting number of worker processes to handle received traffic per listen socket.

So far there were global parameters that were applied to all sockets (e.g., ‘children’ value set the number of workers for all udp sockets). So far each UDP and SCTP socket had its own pool of workers (e.g., children=4 and 2 udp sockets resulted in 8 processes), while for TCP and TLS was a single pool of workers (e.g., having children (or tcp_children) set 8, resulted in 8 processes no matter how many TCP/TLS sockets).

The new features is based on using a new config parameter, named “socket_workers“, before a “listen” parameter. The value of “socket_workers” will overwrite the value of appropriate “*children” parameter. For UDP and SCTP will result in creating a number of “socket_workers” processes. For TCP and TLS will add an extra set of “socket_workers” processes, that will handle traffic only on those specific sockets.

The value of “socket_workers” is reset with the next listen socket added. If “socket_workers” is not set, the value of “*children” parameter is used in backward compatible fashion.
Some typical scenarios where this feature may become handy:
  • set a lower number for loopback or internal/replication sockets, as the traffic there is low (e.g, maybe for keepalive monitoring on loopback, or it is only REGISTER requests replication done over the replication sockets)
  • set a dedicated group of tcp/tls workers for handling HTTP/XMLRPC/XCAP traffic – handling such traffic may be time consuming, in this way you avoid delays on routing SIP over TCP/TLS
  • fine tune the number of over all forked processes by a SIP server instance, thus controlling better the resources used from the physical server (e.g., overall private memory used by sip server is a matter of how many forked processes are there)
You can see details about the new parameter and examples on the wiki page:

Wednesday, January 11, 2012

Fancy time recurrence matching in config

A new module for Kamailio SIP Server named for now tmrec allow matching of time recurrences based on definitions specified by Internet Calendaring and Scheduling Core Object Specification (Calendar COS – RFC 2445).

It becomes trivial to match current time against rules such as working hours, weekend, up to complex conditions such as the interval from 18:00 to 20:00 of the 98th day of every other year if it is a Thursday.

Here is an example of how to match the working hours 8:30am to 6:30pm on business days:

 if(tmrec_match("20120101T083000|10H|weekly|||MO,TU,WE,TH,FR")
  xdbg("it is within working hours\n");

The rule can be specified via a config variable (e.g., load from user profile stored in database via sqlops). A typical use case is time based routing policies.

You can read more about the new extension in the documentation:

Tuesday, January 3, 2012

New Year, New Extension – Embedded execution of managed code (C#)

The first committed module for Kamailio SIP Server in 2012 is app_mono, which offers embedded execution of manage code (e.g., C#/.NET) via Mono project (http://www.mono-project.com/).
The readme of the new module is available at:
Current API which exported by SIP server for usage in C# application is documented at:
Besides C#, app_mono should be able to run managed code compiled from applications written in other languages such as VisualBasic.NET, Java, Java Script, Python, … more are listed at:
A primary use is integration with several widely used Microsoft technologies and APIs, for example C# having up-to-date libraries to connect to Active Directory LDAP service or MS SQL Server.

Sunday, January 1, 2012

Happy 2012!

Same type of activity for me in the past 10 years and still enjoying it! Looking forward to the 11th!

Watch Kamailio SIP Server project news, lot of cools stuff is coming out!

Have a great 2012 friends!

Wednesday, December 14, 2011

Siremis v3.2.0 Released

Siremis v3.2.0 is out – the web management interface for Kamailio SIP Server (former Openser) and SIP Express Router (SER). This is a major release, compatible with Kamailio v3.2.x.
This release brings a large set of new features. Among them:
  • SQL-based CDR rating engine for billing purposes
    • stored procedure to compute the costs of calls
  • Management of billing rates
    • longest prefix rate selections
    • rating rules can be grouped to allow many sets of values
    • time unit is configurable per rating rule
  • Management of remote registration records (uacreg table)
  • Managment of mtree module (mtree and mtrees tables)
  • Management of dialog variables table
  • Update of LCR and SIP Trace views for compatibility with Kamailio 3.2.x
  • Tools to generate new database table views in a wizard fashion
    • create new views to database table with a command line tool in 5 steps
  • Charts drawing statistics of accounting records
    • graphics to show the evolution of accounting records during the past hours
    • graphics to show the types of INVITEs (call setup) during the past hours
  • Tables presenting summary of accounting records
    • count the number of INVITEs and BYEs in the past hours
    • present the top activity of accounting records – e.g., top 5 caller and callee
    • more can be added from configuration file
  • More SIP server activity charts (e.g., SIP requests traffic load)
    • e.g., default chart presents how many requests are received in intervals of 10 minutes
  • Buttons to switch to command pannel to reload Dispatcher or PDT records in SIP server cache
    • once new records are added, in two clicks they get in the cache of SIP server
  • Views for managing global black lists table
  • Many improvements to user interface
    • selection of local domain is done via select box or picker form (e.g., in aliases, user preferences, pdt, …)
    • selection of local username is done via picker form (e.g., user black lists, user preferences, aliases, …)
    • group names can be set in config file and selected from a list box
    • many static values are given as option to select from a list box (e.g., dispatcher flags, lcr options)
  • More targets in Makefile to make administration easier
Step by step installation tutorial, screenshots and demo are available on the web at:
Siremis is used during Kamailio Advanced Training classes for management of SIP server, a good oportunity to learn about Siremis itself, check for next locations at:

Friday, December 2, 2011

Migrating project from BerliOS to GitHub

BerliOS, the open source software forge, announced ending of its life by Dec 31, 2011. Although some time later, there was another announcement that the service will continue, to be operated by a non-profit foundation, I thought anyhow of copying two projects I had there to GitHub.

The two small projects I wrote long time ago, while being a researcher at Franhofer Fokus, were not really maintained, but anyhow it would be a pity to lose the code, parts of it may be useful in the future. If anyone wanders, here is short the description of these projects:
  • pocketsipmsg - this project was used a lot durin 2002-2005 to make demos of SIP instant messaging using iPaq running Windows CE. It is basically a SIP user agent capable of sending and receiving text messages, developed in Visual C for Windows CE
  • tmrec - this project offers a C library and command line tool for matching time recurrences defined by iCal RFC2445. The code is actually used, being embedded in cpl-c module of Kamailio SIP Server
Initially they were stored in CVS repository of BerliOS, but I switched them to SVN some time ago. Therefore, I was looking how to migrate SVN repository from BerliOS to a GIT repository on GitHub.

Googling gave lot of useful tutorials about migrating SVN to GIT, I am writing here just to show the specific case of migration from BerliOS to GitHub - it could save time for some people interested in same kind of operation.

Personally I used a Mac OS X, so first I installed git-svn from ports:

# port -v -d install git-core +svn

On a Debian/Ubuntu, the command should be:

# apt-get install git-svn

You have to create a file to map BerliOS username (used for SVN commits) to name and email address (to be used by Git). I named it authors.txt, the content:

dcm = Daniel-Constantin Mierla <me@xyz.com>
(no author) = Daniel-Constantin Mierla <me@xyz.com>
root = Daniel-Constantin Mierla <me@xyz.com>

For some reasons, which I didn't want to spend time to investigate why, there were two other authors appearing to have committed in SVN: root and (no author) -- so simply I mapped them to myself as well.

Next step I used git svn to clone the berlios project - here is the command used for pocketsipmsg project:

# git svn clone http://svn.berlios.de/svnroot/repos/pocketsipmsg --no-metadata -A authors.txt -t tags -b branches -T trunk pocketsipmsg

You will have to replace pocketsipmsg with your project ID on BerliOS.

I wanted to get the tags and branches from SVN. In the cloned directory, pocketsipmsg, you can list the branches with:

# git branch -r

You will notice that the SVN tags are now branches, to get them back to tags, you have to execute for each tag (named next as $tagname):

# git tag $tagname tags/$tagname
# git branch -r -d tags/$tagname

On GitHub you have to create the repository for storing the project - see http://help.github.com/create-a-repo/ to learn how to do it, if you haven't done it yet. In my case, I named it also pocketsipmsg.

Then add GitHub as remote repository and push to it:

# git remote add origin git@github.com:miconda/pocketsipmsg.git
# git push origin master --tags

That's all, your project is now stored on GitHub!

In summary, if git-svn is installed, your project does not have tags, you created authors file and added the repository $PROJECTNAME on GitHub under user $USERID, the commands you have to run for migration of $PROJECTNAME from BerliOS to GitHub are:

# git svn clone http://svn.berlios.de/svnroot/repos/$PROJECTNAME --no-metadata -A authors.txt -t tags -b branches -T trunk $PROJECTNAME
# cd $PROJECTNAME
# git remote add origin git@github.com:$USERID/$PROJECTNAME.git
# git push origin master

Enjoy!

Thursday, December 1, 2011

Kamailio v3.2.1 Released

Kamailio SIP Server v3.2.1 stable is out – a minor release including fixes in code and documentation since v3.2.0 – configuration file and database compatibility is preserved.

Kamailio (former OpenSER) 3.2.1 is based on the latest version of GIT branch 3.2, therefore those running previous 3.2.0 versions are advised to upgrade. There is no change that has to be done to configuration file or database structure comparing with v3.2.0.

Resources for Kamailio version 3.2.1

Source tarballs are available at:

Detailed changelog:

Download via GIT:

 # git clone –depth 1 git://git.sip-router.org/sip-router kamailio  # cd kamailio  # git checkout -b 3.2 origin/3.2  # make FLAVOUR=kamailio cfg

Binaries and packages will be uploaded at:

Modules’ documentation:

What is new in 3.2.x release series is summarized in the announcement of v3.2.0:

Tuesday, November 29, 2011

Kamailio v3.2.0 Developer Guide

Development guide for Kamailio SIP Server has been updated for v3.2.0 – it goes through internal components, presenting the APIs for pkg/shm memory, locking/synchronization, config file interpreter, database connectors, a.s.o., as well as guiding how to write a new module.

There is a section trying to collect hints about upgrading a module developed for old versions 1.x to newer architecture and APIs in versions 3.x.

The tutorial is available online at:

A mirror is hosted at:

Looking forward to your contributions to Kamailio SIP Server!