Hello David

I have successfully installed kamailio with rtp proxy.

I also have added incoming sip endpoint as below

kamctl address add 1 XX.YY.ZZ.177  0

but when i am doing reload i m getting below output.

[root@unassigned ~]# kamctl address reload
which: no gdb in (/usr/local/sbin:/usr/local/bin:/usr/sbin:/usr/bin:/usr/local/bin/rtpproxy:/root/bin:/)
{
  "jsonrpc":  "2.0",
  "error":  {
    "code": 500,
    "message":  "Method Not Found"
  },
  "id": 10086
}

Can u please let me know what should I do now ?

Thanks


On Fri, Nov 15, 2019 at 10:52 PM David Villasmil <david.villasmil.work@gmail.com> wrote:
Hello,

I already shared a pretty well explained tutorial.

It's pretty straightforward starting from the default kamailio configuration https://github.com/kamailio/kamailio/blob/5.2/etc/kamailio.cfg (that's for 5.2, use whichever version you have)
You need to install rtpproxy locally: https://blog.voipxswitch.com/2015/06/18/rtpproxy-compiling-installing-on-debian-8/

Then you enable:

#!define WITH_MYSQL
#!define WITH_AUTH
#!define WITH_NAT
#!define WITH_PSTN

Then you need to set the B server IP and port

pstn.gw_ip = "B-SERVER-IP" desc "PSTN GW Address"
pstn.gw_port = "B-SERVER-PORT" desc "PSTN GW Port"

and add your users to the db via kamctl utility (you need to configure /etc/kamailio/kamctlrc)

Hope that helps.

Regards,

David Villasmil
phone: +34669448337


On Fri, Nov 15, 2019 at 2:31 PM Sujit Roy <sujitroydhk@gmail.com> wrote:
Thanks for the good suggestion.

Can you kindly suggest the configuration in kamailio for both A & B SIP Gateway endpoint ? 

Thanks

On Fri, Nov 15, 2019 at 7:31 PM David Villasmil <david.villasmil.work@gmail.com> wrote:
If you're only using Asterisk as a media server, why use it? Why not just use rtpproxy or mediaproxy? it'd be much simpler and you'd achieve the same thing.
Unless you need to do something specific in Asterisk, there's really no need.


Should help you getting started.

Regards,

David Villasmil
phone: +34669448337


On Fri, Nov 15, 2019 at 1:21 PM Sujit Roy <sujitroydhk@gmail.com> wrote:
Hello

Here is my call flow scenario.

SIP Gateway (A)
Kamailio (K)
Asterisk (AST)
SIP Gateway (B)

Now i want to send calls from A -> B by using Asterisk as media server.
Kamailio shall be used to authenticate A and allow A to send calls to B.

What are the configurations i need to make in Kamailio and Asterisk ? 

Thanks in advance.

--
Regards
===================
Sujit Roy

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--
Regards
===================
Sujit Roy

_______________________________________________
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sr-users@lists.kamailio.org
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_______________________________________________
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--
Regards
===================
Sujit Roy