[SR-Users] Adding SIP Gateway Endpoint in Kamailio

[Sujit Roy]

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 at 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 at 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
> email: david.villasmil.work at gmail.com
> phone: +34669448337
>
>
> On Fri, Nov 15, 2019 at 2:31 PM Sujit Roy <sujitroydhk at 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 at 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.
>>>
>>> https://dopensource.com/2017/05/31/installing-configuring-rtpproxy/
>>>
>>> Should help you getting started.
>>>
>>> Regards,
>>>
>>> David Villasmil
>>> email: david.villasmil.work at gmail.com
>>> phone: +34669448337
>>>
>>>
>>> On Fri, Nov 15, 2019 at 1:21 PM Sujit Roy <sujitroydhk at 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
>>>>
>>>> _______________________________________________
>>>> Kamailio (SER) - Users Mailing List
>>>> sr-users at lists.kamailio.org
>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>>>
>>> _______________________________________________
>>> Kamailio (SER) - Users Mailing List
>>> sr-users at lists.kamailio.org
>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>>
>>
>>
>> --
>> Regards
>> ===================
>> Sujit Roy
>>
>> _______________________________________________
>> Kamailio (SER) - Users Mailing List
>> sr-users at lists.kamailio.org
>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>
> _______________________________________________
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>


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