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CIITIX-VoIP 1.0 How-To - Page 2


User Configuration

1. On the command prompt as a super user, create users as follows:

[email protected]:~# kamctl add omar 123

User "omar" with password "123" should be created. Likewise create other users.

2. Users could be removed as follows:

[email protected]:~# kamctl rm <user name>

3. Passwords can be changed as follows:

[email protected]:~# kamctl password <user name> <newpa*sword>

Client Configuration


You can try any SIP client (Soft-phone/Hard-phone) with xcap protocol support.

This demo uses sip-communicator from http://www.sip-communicator.org/index.php/Main/Download (which is a free client but still in alpha stage). Download the one for your architecture & Operating System.

You can also try Bria or eyebeam soft-phones from http://www.counterpath.com/


Run "sip-communicator", and choose "Add New Account" from the "File" menu.

From the "Network" comb box choose “SIP”. Add the user credentials. "omar"/"123" in our case. You can use the IP address of the server or the FQDN if you have a DNS setup. Click the "Advanced" button afterwards. Click the "Presence" Tab & set the entries accordingly. The suffix from "5060" onwards should remain the same. (Default port of SIP is 5060 if you have changed it use that one). This step is imperative if you wish to maintain a permanent list of your buddies (sort of like roaming profile thing). Click the "Next" button afterwards. Click "Sign in" if all the information is right. You should be logged in after 1-2 seconds. Adding contacts/buddies is also pretty straight forward. Just choose the "Add Contact" from the "File" menu. Add your contact.
Note: You cannot search for users. You just have to know about your contact's username beforehand. In this example a buddy named "osman" is being added. And yes he also has to be registered on the same server. The status of your contacts will be displayed according to their active status. Congratulations! You have set up CIITIX-VOIP successfully. 




By: dk

Awan, it's nice to see a non-digium FOSS for VoIP receiving some attention but one of the principle aspects of configuring a PBX for VoIP is setting up external calls via SIP trunking or other means so that calls can be placed to and from the system. If 2 internal users can call each other that may be fine as a demo but I can't say that I believe the CIITIX system is "configured" to any degree of completion if it isn't able to place external calls or receive calls from an outside line. Just my two cents and no disrespect intended - I'd just like to see this how-to completed.


Not an unfair question.

CIITIX-VoIP can definitely place & receive calls from an outside line, provided it has been coupled with PSTN gateway or you have signed up with DiD thing or some other voice service.

You can configure/place/re-route your calls to your PSTN gateway which could be hardware based or software based (call weaver, asterisk, freeswitch etc..). But that would be your configuration. no ?

Like wise you can configure/place/re-route your calls to your purchased/subscribed DiD/voice serivce  & it will be your specific configuration.

I hope to you get it. I mean yeah I could as an example demonstrate that with maybe asterisk or some DiD  & I didn't. We are busy in coupling GUI in the 1.1 release & are HR deficient. Maybe you could chip-in. (seriously :) )


By: swmds

does anyone have a copy of ciitix-voip laying around that i can get. the servers for download are down so it seems.

Thanks Bunches