OpenSIPS's Blog

Repair Mysql Database

2011-08-02T08:19:00.000-07:00

How to repair MYSQL database?

mysqlcheck -u root -p --auto-repair --check --optimize --all-databases

Compilation of Download Links

2011-02-17T00:11:00.001-08:00

http://www.adobe.com/support/downloads/product.jsp?product=1&platform=Windows

Successfully Configured Asterisk

2010-01-27T16:50:00.000-08:00

Adobe PhotoShop Tutorial!

Asterisk configuration was successful, I am using Asterisk Version 1.4.28. There was some confusion on which driver to use for my X101P card. The version I am using right now deprecated the zap commands on Asterisk CLI, although I am still using the zaptel driver.

I was able to configure the SIP channel successfully, but calls using SIP channel has a lot of noise. A simple dialplan has been created. Still looking on ways how to minimize noise during calls.

New Project Details

2010-01-21T18:39:00.000-08:00

Our group was able to successfully configure OpenSIPS as a Proxy Server (stateful) and Registrar. Test calls were clear and nat traversal was working. Additional planned capabilities of OpenSIPS such as Presence support using OpenXcap has been cancelled due to lack to time. There were a lot of things to do, but we don't have time.

I have decided to continue our project to learn more about OpenSIPS and its capabilities. What I am trying to now is to configure OpenSIPS as a proxy, load balancer and registrar. I am also planning to integrate Asterisk as a Media Server and PSTN Gateway. Currently I am studying Asterisk and was able to purchase Digium Wildcard X101P.

Blog Changes

2010-01-21T18:30:00.000-08:00

After several months of being inactive due to a lot of pressure in school and graduation preparation I got finally sometime to write again. This time I would be working alone on this project and expect some changes. Anybody who is interested can post their questions and comments here.

Using SIPp to Test OpenSIPS

2009-10-08T19:18:00.000-07:00

This a good reference for testing your OpenSIPS

OpenSIPS VS Asterisk as registrar (SIPp test)

Read this article on the OpenSIPS mailing list. The script for creating the CSV file for this testing scenario is available for down load at the above link.

Load Balancing for OpenSIPS

2009-09-24T19:02:00.000-07:00

The group started integrating the load balancing module in the opensips.cfg file. Having problems in setting the resources rules in the load_balancer table in the opensips database.

The group believes that this module is very important in handling load balancing using multiple servers.

Any information and/or references would be appreciated.

OpenSIPS webinar3 video

2009-08-27T17:23:00.000-07:00

Guys the video on OpenSIPS webinar 3: Types of route is now available at this link http://www.opensips.org/html/docs/video/webinar003/

Important Linux Command for Managing OpenSIPS and Other application

2009-08-10T17:00:00.000-07:00

The following are useful commandline for managing opensips and other related applications.

python -c "import sys, gnutls; print gnutls.__version__; print.modules['gnutls']"
command to display the gnutls version

ngrep -p -q -W byline port 5060 > filename.txt
command to capture sip datapackets and store it into a text file

find . -exec grep -H "keyword" {} \; -print
command to locate a file containing the keyword on the entire computer

python -c \
'import platform; print platform.python_version()'
display python version
RTPProxy file = /etc/sysconfig/rtpproxy
OPTIONS="-u rtpproxy -l 172.16.100.135 -s udp:172.16.100.135:12345"

OpenSIPS Boot Camp in South East Asia!

2009-08-07T17:42:00.000-07:00

The group believes that there are people interested in attending OpenSIPS Boot Camp in South East Asia (e.g. Singapore, Hong Kong, Thailand, Philippines, Malaysia, Indonesia). OpenSIPS Boot Camps that were held in Brazil and California and the upcoming boot camp in Germany this coming September are very inconvenient for interested people in this region. First and foremost the location is inconvenient for those who are working, second financial constraints and Visa requirement.
The group has informed people on the OpenSIPS mailing list and they have showed interest making this event possible.

The following is an excerpt from an email sent by Bogdan

Hi,

We would like to give it a try, but we have no contacts to help with the logistics in the area....Where are you located and will you be willing to support in organizing such an edition ?

Thanks and regards,
Bogdan

bay2x1 wrote:

Is it possible that an OpenSIPS boot Camp be held in South East Asia,
(Singapore, Hong Kong, Philippines, Malaysia etc)? I know that there are a lot
of people who are interested in OpenSIPS specially in this region, and if
you are interested please let the organizers know so that they will be
encourage to start planning for the Boot Camp in South East Asia...

-----
http://n2.nabble.com/OpenSIPS-Boot-Camp-in-South-East-Asia!-td3360752.html#a3360752

Understading Presence Support for OpenSIPS using OpenXcap

2009-08-06T16:54:00.000-07:00

The group is using a GNU License softphone client and has added some of its feature to support Presence using OpenXcap. The group had a difficulty in implementing presence support for the software and has to consult different RFC Documentation. After some painstaking work currently the group was able to send the PUBLISH and SUBSCRIBE request to the Proxy and the OpenXcap server.

Webinar on Different Routing in OpenSIPS

2009-08-05T16:18:00.000-07:00

Here is the registration link for the OpenSIPS webinar on different routing scheduled on August 12, 2009 https://www2.gotomeeting.com/register/403938514
Routing is one of the most essential part in configuring OpenSIPS, although the group has already configured OpenSIPS, they still consider attending this webinar important.

Publish Request for OpenSIPS

2009-07-24T00:14:00.000-07:00

After several days of trying the group was able to process the PUBLISH request for OpenSIPS using a prototype code for SIPS client. The group's aim was to develop a C++ base code for a sip client to send a PUBLISH request to the OpenSIPS proxy server. The group consulted several RFC's (Request for Comments) related to presence support in SIP.

The following is the partial list of the RFC documentations:
1. RFC 3856: A Presence Event Package for the Session Initiation Protocol (SIP)
2. RFC 3857: A Watcher Information Event Template-Package for
the Session Initiation Protocol (SIP)
3. RFC 3858: An Extensible Markup Language (XML) Based Format for Watcher
Information
4. RFC 3863: Presence Information Data Format (PIDF)

RFC 3863 was very helpful in creating the standard xml format for sending presence related information.

The next task is to be able to process the subscription request and understand it process.

For any questions you can contact the group at opensips101@gmail.com

OpenSIPS: OpenXcap Presence Support

2009-07-19T21:17:00.000-07:00

The group is currently reviewing the presence support using OpenXcap. Initial testing using the test.py python script, the OpenXcap configuration passed the 47 tests. Basic presence support has been enabled in the opensips.cfg configuration. Both ekiga and xlite softphones where able to detect presence.

OpenSIPS Webinar: Video Post

2009-07-17T17:53:00.000-07:00

The video of the Routing Webinar has been posted on the net. http://www.opensips.org/html/docs/video/webinar002/. The Webinar was conducted last July 8, 2009.

Call Control: Accounting Internally Generated BYE Request

2009-07-15T18:48:00.000-07:00

This issue has been documented on the previous Post for Call Control.

When a caller prepaid account reached zero balance during a call, call control immediately send BYE request to both ends to terminate the call. Since this request is internally generated, accounting should take place on the local_route.

The group used the code below to account the BYE

local_route {

if (is_method("BYE") ) {
acc_db_request("Internally generated BYE", "acc");
acc_rad_request("Internally generated BYE"); #// This is sending an Bye to the radius server to create the stop time.
}
}

The code was placed before the main routing block.

CDRTool Call Price Computation Problem

2009-07-11T16:49:00.000-07:00

After installing CDRTool 6.8.0 and configuring the rating engine properly the group was expecting that call price will be computed for each call. After several testing it was discovered that all the calls made were considered as a free calls. The global.inc configuration was reviewed several times and modified but still the issue was not resolved.

Further reading regarding the problem revealed that the issue was on the E164 class, which by default uses the European class. By just dialing the subscriber username e.g. 1000, the rating engine fails to determine the destination id for the call. Using the testing procedure at the ag-projects wiki the group discovered that if 0031(default country code for Netherlands) was appended to the username, price for the call were computed. The rating engine considers all the calls as local calls thus it considers all calls as free call. This was encountered because the group failed to integrate the dial plan and determining LCR (Least Cost Route). Without appending the destination id on the canonical uri ($avp(s:can_uri) = $ru;) the rating engine fails to compute for the price.

Command Syntax:

ShowPrice From=sip:123@example.com To=sip:0031650222333@example.com Gateway=10.0.0.1 Duration=59

Result

0.2173
Duration: 59 s
App: audio
Destination: 31650
Customer: domain=example.com
Connect: 0.0000
StartTime: 2009-06-26 04:47:41
–
Span: 1
Duration: 59 s
ProfileId: grn_premium / weekday
RateId: grn_premium / 0-24h
Rate: 0.2210 / 60 s
Price: 0.2173

The command above was done by telneting to the rating engine to port 9024. By typing help all rating engine commands will be displayed with sample syntax. It can be observed that the syntax of the command above is exactly the same with the given example for the ShowPrice command. If your rating engine failed to produce the same result above the problem might be your global.inc configuration or you have not properly importated the sample data for the CDRTool Rating engine. (importRatingTables.php)

Call Control Problem

2009-07-10T03:29:00.000-07:00

Integrating the call control module on the opensips.cfg file makes opensips crash. The group is currently assessing the opensips configuration on what might be the problem. Previous configuration is being reviewed.

Update: July 13, 2009

After replacing the opensips.cfg file with an older configuration and re-intergrating the call control script of the opensips.cfg file call control is now working. The group is still comparing the two configuration files to determine why the newer configuration was not working. The group is checking on the possibility on the sequence of loading the modules (still in the process on determining what module).

Initial testing on the call control reveals that if the balance reaches below 4.00, the module will consider it low account balance hence, calls can not be made by the user. The group still exploring on how to set this into a much lower value.

Update: July 14, 2009 - 7:30 a.m.
The issue on why call control is not allowing calls if it reach certain amount is not related to any minimum balance which was considered by the group as a possibility. Further readings on the documentation regarding call control module call control has no minimum balance requirement. The group is still trying to determine on this is happening.

Update: July 14, 2009 - 9:30 a.m.
After reviewing the opensips.cfg configuration, the group has mistaken inserted the call control script before the session of the call has been fully established. After editing the opensips.cfg file call control was able to calculate the maximum time allowed for the call. Testing shows that call control was able to terminate the call when the is no longer enough for the call.

Another issue has been encountered however, when the call control module sends the BYE to both User Agent, the transaction is not properly accounted. Although the session is ended, radius is unable to update the accounting record for the call.

OpenSIPS Webinar on Routing

2009-07-08T21:48:00.000-07:00

Last night July 9, 2009 at 1:00 a.m. Philippine time (July 8, 2009 8:00-9:00 p.m. EEST), a webinar on routing was conducted by Bogdan Iancu and Flavio Goncalves. The webinar was very informative specially for those who are starting to study OpenSIPS. The discussion was very brief and concise and the participants where given the chance to ask questions after the discussion. The speaker discussed how to properly route the different requests by using record routing and sample scripts.

Considering the time (one hour), it was not enough to tackle all the subjects on routing specially on NAT Traversal. They are planning to conduct more webinars regarding OpenSIPS. If the speakers and sponsors organized OpenSIPS webinars more frequently, it will encourage a lot of people to adopt OpenSIPS on their telephony systems.

OpenSIPS: FreeRadius Installation

2009-07-08T07:27:00.000-07:00

FreeRadius is used for accounting in conjunction with the CDRTool. FreeRadius version 1.1.7 was installed using installation packages for Fedora Core 10.

Syntax:

yum install freeradius, freeradius-mysql

After the installation make sure radiusclient-ng support is available on your system.

The following configuration files need to be modified.

On /etc/raddb folder
clients.conf
radiusd.conf
dictionary
sql.conf

On /etc/radiusclient-ng folder
servers
radiusclient.conf

---- to be continued --- unfinished

Opensips Migration: Updating the OpenSIPS Database:

2009-07-08T04:01:00.000-07:00

Database migration is advised if you have upgraded from OpenSIPS 1.4 to 1.5. There have been changes on some tables on the opensips databases.

How to Migrate:

Command line syntax:
opensipsdbctl migrate old_database_name new_database_name

Example:

opensipsdbctl migrate opensips opensips2

After the migration be sure to modify your opensips.cfg and other related configuration files to point to the new database. Please note that the saved data on the old opensips database will be copied into the new database.

July Status

2009-07-07T04:52:00.000-07:00

Finished installing OpenSIPS 1.5 with tls support. The opensips.cfg configuration file has been configured to use mysql database and authentication.

CDRTool 6.8 is also installed on the same machine which uses FreeRadius version 1.1.7 for accounting and mediaproxy as media relay. Currently, accounting and billing of OpenSIPS calls are working properly. Call Control has also been installed for prepaid support but is still in configuration stage.

OpenXcap has also been installed for presence support. Initial testing using the testing procedure found at the ag-projects wiki, showed that the installation passed all the 47 tests.

MSRP Relay will be used for Chat Support.

OpenSIPS: A Case Study

2009-07-07T04:08:00.000-07:00

Case Study Overview

As a part of a course requirement in Electronic Communication Engineering this study is undertaken to explore and document the capabilities of OpenSIPS. The group is also aiming to gather relevant information in the implementation of OpenSIPS using different topologies. This blog has been created to document their progress and find other interested people who are willing to share their technical abilities in solving documented problems of the group.

The Group's Technical Skills and Background
The group is composed of five fourth (4th) year students with computer programming skills on C++, C and php. Basic knowledge on Linux Operating System and Administration.

OpenSIP Set-up Information

The OpenSIPS case study is using three (3) computers in a local area network. Two computers serve as the User Agents and the other computer serves as the proxy and registrar. All three computers have the same hardware specifications, Intel Core 2 Duo with 1 Gig Memory.