Hi, If everything is setup as you expected it then don't read this > http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb, else follow the article from start till end and implement it. Don't worry about the name asterisk, just replace that all with FreeSWICTH. It will work like that too. Once that is implemented then apply dispatcher module on the route which says [REGFWD] or [TOASTERISK]. Also explain your topology abit more and features required as well that may help in telling which extra module you will require in order to make things work. In your current implementation are you sure SIP phones are registering on FS and not on Kamailio?and that both end points making call to each other are on same FS? -- Regards, Sammy 2011/11/14 Henrik Aagaard Sørensen <henrikaagaardsorensen@gmail.com>
Hi everyone. In regards to my earlier question regarding with FreeSwitch behind Kamailio Dispatcher, I've attached a call from extension 1001 to 1002, which fails. It just hangs for some time and then says that 1002 cannot be found, and then the voicemail for it comes up.
2011/11/13 Henrik Aagaard Sørensen <henrikaagaardsorensen@gmail.com>
I'm trying to get the setup Kamailio Dispatcher -> FreeSwitch to work.
I've setup Kamailio via this: http://www.kamailio.org/docs/modules/stable/modules_k/dispatcher.html
I've installed FreeSwitch from scratch on Ubuntu via: http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start
Now, when registering extensions via Kamailio Dispatcher I'm able to call to FreeSwitch and listen to hold music. But that's it. I'm not able to call between extensions etc.
Can anyone help me setting up FreeSwitch to accept registration, calls etc. from Kamailio and everything else that is needed to use FreeSwitch behind a load balancer?
I'm very new to FreeSwitch, but I'm trying to use the terminal (without any GUI etc.) as I want the installation to be as clean as possible. So I would prefer very precise help, as I'm still getting hold of FreeSwitch.
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