[jitsi-users] Help me set up Jitsi with Kamailio/Asterisk


#1

Hi Everyone,I have been lurking around on the jitsi website and trying to convince the people at my small company that Jitsi would be the ideal solution for our secure videoconferencing/desktop sharing requirements. I have been given a go ahead to try setting up a SIP server of my own and I am very excited even though I am not much familiar with system administration. Please be patient with me.I found a tutorial at http://kb.asipto.com/kamailio:skypelikeserviceinlessthanonehour and I was convinced that my task was easy. Alas, if only it was that simple! I got hold of a Ubuntu 12.04 machine with a live IP and followed the steps through. The installation gave no errors but when I tried to start the Kamailio server, it said that it could not find the modules (all of them). Someone would have pointed it out on the above page, but comments are not enabled on the page. Ultimately I figured out that it must have something to do with the config file and I solved that problem by putti
ng the correct address to the modules. Except one. corex.so was missing. I tried to locate it. Downloaded it from other source and put it in the folder but it did not help. Ultimately I commented out the lines that needed in the hope that it might be something that I might not need for the current purpose. Nope. I was not able to login from Jitsi using the credentials I created on Kamailio.Then I headed for Asterisk. Installed it through Synaptic on my Ubuntu and it automatically installed the dependencies. I made the following addition to sip.conf because I found it in a sample:[general]context=defaultallowguest=no[user1]type=friendhost=dynamicsecret=user1234context=demo[user2]type=friendhost=dynamicsecret=user2345context=demoI was happy to note that for the first time I was able to log in to Jitsi using two independent machines that are under a firewall. But that was about it. I added the other user into the accounts but they do not appear to be online even though asterisk
  x "sip show peers" tells me that both users have logged in. I am not able to communicate with them in any way.Please help me out on this one. My basic requirement is:* SIP server on a Live IP machine (Without firewall)* Jitsi on multiple machines under firewall.* Maximum number of concurrent user: 5 (so I don't need MySQL, I guess)* Secure and Peer to Peer communication (video calls as well as videobridge conference)I can post my config files etc. if required. Please point me to some working samples.Thank you all in advance.Cheers!GauravDear users! Get Yourself a cool, short @in.com Email ID now!


#2

Hello,

Hi Everyone,
I have been lurking around on the jitsi website and trying to convince the people at my small company that Jitsi would be the ideal solution for our secure video-conferencing/desktop sharing requirements. I have been given a go ahead to try setting up a SIP server of my own and I am very excited even though I am not much familiar with system administration. Please be patient with me.

I found a tutorial at http://kb.asipto.com/kamailio:skype-like-service-in-less-than-one-hour and I was convinced that my task was easy. Alas, if only it was that simple! I got hold of a Ubuntu 12.04 machine with a live IP and followed the steps through. The installation gave no errors but when I tried to start the Kamailio server, it said that it could not find the modules (all of them). Someone would have pointed it out on the above page, but comments are not enabled on the page. Ultimately I figured out that it must have something to do with the config file and I solved t hat problem by putting the correct address to the modules. Except one. corex.so was missing. I tried to locate it. Downloaded it from other source and put it in the folder but it did not help. Ultimately I commented out the lines that needed in the hope that it might be something that I might not need for the current purpose. Nope. I was not able to login from Jitsi using the credentials I created on Kamailio.

I expect that the path of modules was with lib64 if you installed on a 64bit or other folder on the file system.

You should send an email to kamailio mailing list with the error messages from the syslog and I am sure you will get help there. Corex is a module installed by default, so either you have a version that doesn't include it or there is some other error done during installation. So, add your version string of kamailio when sending the email.

Cheers,
Daniel

···

On 25/09/14 15:03, Gaurav Kumar wrote:

--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda


#3

I'm not familiar with kamailio so can't help you there. For the most part, I use freePBX as a front-end GUI to asterisk, as it makes life a lot easier, especially for a new user. There is an asterisk mailing list, as well as an IRC channel on freenode.

It sounds like you may have a NAT issue. In your below example, you'll likely want to set NAT=yes in extensions.conf - and possibly in sip.conf as well. I'm not an expert on it by any means so if you don't feel like experimenting, hit up the asterisk mailing list and/or IRC channel for a better answer. You'll likely be asked to "show the call", so be prepared to show your asterisk logfile trimmed to the specific call with "sip set debug on" and "core set verbose 10". There's actually a variety of issues you could be having in regard to your network setup in relation to asterisk and jitsi, NAT is just the most obvious.

···

On 09/25/2014 06:03 AM, Gaurav Kumar wrote:

Hi Everyone,
I have been lurking around on the jitsi website and trying to convince the people at my small company that Jitsi would be the ideal solution for our secure video-conferencing/desktop sharing requirements. I have been given a go ahead to try setting up a SIP server of my own and I am very excited even though I am not much familiar with system administration. Please be patient with me.

I found a tutorial at http://kb.asipto.com/kamailio:skype-like-service-in-less-than-one-hour and I was convinced that my task was easy. Alas, if only it was that simple! I got hold of a Ubuntu 12.04 machine with a live IP and followed the steps through. The installation gave no errors but when I tried to start the Kamailio server, it said that it could not find the modules (all of them). Someone would have pointed it out on the above page, but comments are not enabled on the page. Ultimately I figured out that it must have something to do with the config file and I solved t hat problem by putting the correct address to the modules. Except one. corex.so was missing. I tried to locate it. Downloaded it from other source and put it in the folder but it did not help. Ultimately I commented out the lines that needed in the hope that it might be something that I might not need for the current purpose. Nope. I was not able to login from Jitsi using the credentials I created on Kamailio.

Then I headed for Asterisk. Installed it through Synaptic on my Ubuntu and it automatically installed the dependencies. I made the following addition to sip.conf because I found it in a sample:

[general]
context=default
allowguest=no

[user1]
type=friend
host=dynamic
secret=user1234
context=demo

[user2]
type=friend
host=dynamic
secret=user2345
context=demo

I was happy to note that for the first time I was able to log in to Jitsi using two independent machines that are under a firewall. But that was about i t. I added the other user into the accounts but they do not appear to be online even though asterisk -x "sip show peers" tells me that both users have logged in. I am not able to communicate with them in any way.

Please help me out on this one. My basic requirement is:

* SIP server on a Live IP machine (Without firewall)
* Jitsi on multiple machines under firewall.
* Maximum number of concurrent user: 5 (so I don't need MySQL, I guess)
* Secure and Peer to Peer communication (video calls as well as videobridge conference)

I can post my config files etc. if required. Please point me to some working samples.

Thank you all in advance.

Cheers!
Gaurav

Dear *users!* Get Y ourself a cool, short *@in.com* Email ID now! <http://www3.in.com/sso/commonregister.php?ref=IN&utm_source=invite&utm_medium=outgoing>

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