Configuring SIP client(s)(soft phone) to connect to the local freePBX server(SIP server)

Tried out softphones  - linphone, microsip, ekiga

There are various open source soft phones available which can be used to register with Asterisk PBX(freePBX).
In the current scenario the already existing and unused extension(106) was used for SIP client.
images of the freepbx extention

In the case of ekiga, remove the default account created by ekiga like echo test etc

Initially
used 106 for sip client <sip:106@192.168.100.28:5060 > where 192.168.100.28 is the pc IP addr and the 5060 is the port to communicate
but wasn't able to establish the connection<investigate - wireshark>
Eth was given static IP
which gave Ekiga loop detected ERROR also the microsip was giving timeout msg
so changed the eth IP to dynamic and the error was gone
Make sure that only one SIP client is active at a time ****

Ekiga makes use of SIP or H.323 compatible hardware and software. It supports many audio and video codecs and all modern voip features for both sip and H 323. Ekiga was formerly known as gnomemeeting.

Error:- Ekiga loop detected
http://blog.sviluppoweb.eu/2013/07/19/ekiga-registration-error-loop-detected/
for this the static ip of the machine was changed to the dynamic ip
screenshots of the ekiga and settings

screenshots of linphone

Screenshots of MicroSIP