Hello,

it seems that the method or module is not available (as indicated also from the 
error message).

Check in your Kamailio.cfg if you load the permissions module.

Cheers,

Henning

--
Henning Westerholt – https://skalatan.de/blog/
Kamailio services – https://gilawa.com<https://gilawa.com/>
Kamailio Merchandising – https://skalatan.de/merchandising

From: sr-users <[email protected]> On Behalf Of Sujit Roy
Sent: Thursday, November 21, 2019 11:14 AM
To: Kamailio (SER) - Users Mailing List <[email protected]>
Subject: Re: [SR-Users] Adding SIP Gateway Endpoint in Kamailio

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 
<[email protected]<mailto:[email protected]>> 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
email: [email protected]<mailto:[email protected]>
phone: +34669448337


On Fri, Nov 15, 2019 at 2:31 PM Sujit Roy 
<[email protected]<mailto:[email protected]>> 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 
<[email protected]<mailto:[email protected]>> 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.

https://dopensource.com/2017/05/31/installing-configuring-rtpproxy/

Should help you getting started.

Regards,

David Villasmil
email: [email protected]<mailto:[email protected]>
phone: +34669448337


On Fri, Nov 15, 2019 at 1:21 PM Sujit Roy 
<[email protected]<mailto:[email protected]>> 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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_______________________________________________
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--
Regards
===================
Sujit Roy

_______________________________________________
Kamailio (SER) - Users Mailing List
[email protected]<mailto:[email protected]>
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
Kamailio (SER) - Users Mailing List
[email protected]<mailto:[email protected]>
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users


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

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