Hello Daniel,
I will try your recommendation and revert.
Regards
Olawuyi Timothy Oladapo | IT/VAS Operation | Information System Dept |Pethahiah Rehoboth
INTL. Limited | 2nd Floor Rubby Block All seasons Plaza , Lateef Jakande Rd, Agidingbi,
Ikeja, Lagos. | Mobile: +2348052612001,+2348098797928 |Skype: daptims |
htttp://www.pethahiah.com
-----Original Message-----
From: sr-users [mailto:sr-users-bounces@lists.kamailio.org] On Behalf Of Daniel Tryba
Sent: Thursday, October 19, 2017 12:37 PM
To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org>
Subject: Re: [SR-Users] Kamailio and FreeSwitch integration
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;
}
...
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