Posted in Tutorials by admin |
Obviously, this is really an old procedure, actually this is my documentation way back early 2006 when it’s my first time to work with asterisk. This procedure was implemented before in about 30-50 call center agents using softphones and it quite worked well.
So many things have changed, as we all know, from asterisk 1.2 to asterisk 1.6 , and Asterisk Management Portal is popularly known now as FreePBX. Other packages were also updated now, many changes but still asterisk is standing there as the best open source telephony system. We had many community and commercial PBX softwares today that are asterisk-based, that’s how big asterisk right now comparing back 2006 
This is a preview of
Installing Asterisk 1.2 and Asterisk Management Portal
.
Read the full post (903 words, estimated 3:37 mins reading time)
Posted in Quick Tips & Tricks by tux |
Want to test your Asterisk PBX system if it can sustain load and large traffic? Then you can use this tool.
Sipp is a performance testing tool for the SIP protocol. Its main features are basic SIPStone scenarios, TCP/UDP transport, customizable (xml based) scenarios, dynamic adjustement of call-rate and a comprehensive set of real-time statistics.
Sipp can be used to test real SIP equipments and very useful to emulate thousands of user agents calling your SIP system.
Installation:
1. Download the stable version of Sipp ( sipp-xxx.tar.gz)
2. Uncompress the tarball file
#tar zxvf sipp-xxx.tar.gz
#cd sipp
#make
