19 Oct
2017
19 Oct
'17
2:37 p.m.
On Thu, Oct 19, 2017 at 12:13:47PM +0100, Timothy oladapo olawuyi wrote:
Kamailio will act as our SIP control while Freeswitch
will act media server
for incoming calls.
Freeswitch will send all outgoing calls to Kamailio for onward transfer to
our SIP provider network.
No registration, presence, location Accounting, Authentication etc. are
required.
I have gone through the book SIP ROUTING WITH KAMAILIO and FreeSwitch and
Kamailio integration sample config available at
http://kb.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms am still
unable to figure out how to go with the configuration.
That sample does all the things you don't want to implement.
I will appreciate any one who can provide guideline
for achieving the above
scenario.
Your required config is a simple proxy.
https://kamailio.org/docs/modules/stable/modules/dispatcher.html#dispatcher…
contains a config example, which sends all traffic to 1 dispatcher. You
need to modify this a little. Add a second dispatcher so you have 1 for
your upstream and 1 for your freeswitch. If a message arrives from 1
dispatcher send it to the other. For example something like:
route[DISPATCH]
{
if(ds_is_from_list("1"))
{
ds_select_dst("2", "4");
}
else if(ds_is_from_list("2"))
{
ds_select_dst("1", "4");
}
else
{
sl_send_reply("403","Forbidden");
exit;
}
...