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sipproxy

This is project for proxy (sip and rtp) calls.

Instructions for https://hub.docker.com/repository/docker/dalsearan/proxy_for_sip/general

for expirienced users to dig in config - branch master

Minimum server requirements:

  • Debian 10/11
  • At least 10 GB of free disk space
  • 1 GB of RAM
  • Public static ip-address
  • Installed git

Installation instructions:

  1. Clone this repo to a local VDS/VPS:

    # git clone https://github.com/Dalsearan/sipproxy.git

  2. Go to the sipproxy directory:

    # cd sipproxy

  3. Run command to make executable:

    # chmod +x setup-sipproxy.x

  4. Then run as a root on Debian system:

    # ./setup-sipproxy.x

  5. Follow the installation instructions.

Or you can use a one-line command:

# apt update && apt -y install git && git clone https://github.com/Dalsearan/sipproxy.git && cd sipproxy && chmod +x setup-sipproxy.x && ./setup-sipproxy.x

Using of sipproxy:

SIP phone:

Set the ip-address and port 25060 of sipproxy as an outbound proxy in your SIP phone settings.

Asterisk:

SIP:

Set the parameter "outboundproxy=ip-address of sipproxy:25060" for the SIP-trunk in sip.conf, for example:

outboundproxy=192.168.0.1:25060

PJSIP:

https://wiki.asterisk.org/wiki/display/AST/PJSIP+with+Proxies

FreeSwitch:

In a SIP trunk parameter file such as "sip-trunk.xml", add the following parameter , for example:

<include>
  <!--gateway name required, you will use this in the outbound dialplan -->
    <gateway name="sip-trunk">
      ...
      <param name="outbound-proxy" value="192.168.0.1:25060"/>
      ...
    </gateway>
</include>