[OpenSIPS-Users] OpenSIPs & CP - Documentation (Dynamic Routing, DialPlans, CP)

Rick McGill - ₪ rick at netrovoip.com
Fri Mar 12 06:53:01 EST 2021

Dear OpenSIPs Community,

I'm very new to OpenSIPS and have a new fresh install and have started to
customize it.
So if my question seems so very basic this is why.

I have the following:
Debian 1.7 (Buster)
OpenSIPs 3.1
OpenSIPs Control Panel

I have a couple of uses setup and can call with softphone from User to user
on OpenSIPS.
I have an FreePBX/Asterisk SIP Trunk setup to OpenSIPS.   With outbound
routes and all configured and working properly.  
I also have on OpenSIPS a Route/Gateway or Trunk from OpenSIPs to CommPeak
For Outbound calls.

The issue I'm having is learning or finding documentation on how exactly to
setup Dialplan, Dynamic Routes, Permissions and such so that I can get a
call from Ext on Asterisk routed to OpenSIPS then out to CommPeak for an
outbound International call.

Currently when I try to make this type of call OpenSIPS says for example
6628888888 at sip.netrovoip.com    userloc does not exists and hangs up.
Sip.netrovoip.com is my OpenSIPs server
So apparently with all my efforts my Dialplan, Permissions, and Dynamic
Routes is not working no matter what changes I keep trying in them.

FYI I have not yet edited or customized the file

I cannot find out what code and customization I should do for attrib and
what I can use standard in OpenSIPS-CP

<------>// name , description
<------>"a" => "Descr a",
<------>"b" => "Descr b",
<------>"c" => "Descr c",

So My question is:
Where can I find some comprehensive detailed documentation (if it exists) so
I can learn more about the following?:
Dynamic Routes
And whatever else I need to learn to get this kind of basic Dynamic Routing
I have gone thru the OpenSIPS-CP documentation and the OpenSIPs.org too but
there it seems a bit basic or outdated.

Also I do have my Regular Expressions Quick Reference v1.2 but no matter
what I program in CP it does not seem to work.
I think I'm missing some very basic but major bit of information on how to
get this simple Asterisk -> OpenSIPS -> CommPeak outbound call basic setup
configured / working.

Again sorry if all this sound like such a basic knowledge set of questions.

Rick McGill – CEO
Rick at NetroVOIP.com     |     Rick at NetropolitanWorks.com 
Thailand: +66-2105-4262  x1001  |   USA: +1-737-237-2030   |    Mobile:
Support:: +66-97047-2000  |  SKYPE & LINE ID:  NetroVOIP  |
Support at NetroVOIP.com
  ₪  www.NetroVOIP.com  Telecommunications / Video Consulting & Solutions

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