Hello
I have followed an installation video for installing OpenSIPS on Ubuntu.
When I run: sudo opensips
I get "Listening on udp: 127.0.0.1:5060
What is my next step?
My first goal is to develop a simple Android app that connects to the OpenSIPS
server.
I guess I need to somehow
We have installed OpenSIPs - we have two PBX's setup - they communicate to one
another perfectly... now, we are ready to point OpenSIPs to our up-stream
SIPTrunk... could someone explain how to do this in just a couple steps?
Everything we are reading and trying, seem to be overly complex.
Than
Hi,
Unless you are on the same host trying to connect with clients, else
you should have an IP accessible to your clients.
For a start you can use opensipsctl script to create users with default
opensips script, and try to connect your clients.
Use xlog in opensips.cfg for each statement to u
-boun...@lists.opensips.org] On Behalf Of Salman Zafar
Sent: 2. april 2014 14:46
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Newbie question
Hi,
Unless you are on the same host trying to connect with clients, else you
should have an IP accessible to your clients.
For a
gt; users-boun...@lists.opensips.org] *On Behalf Of *Salman Zafar
> *Sent:* 2. april 2014 14:46
> *To:* OpenSIPS users mailling list
> *Subject:* Re: [OpenSIPS-Users] Newbie question
>
>
>
> Hi,
>
> Unless you are on the same host trying to connect with clients, else
> y
There's no standard way to route via a SIP trunk. In the end, you need to
rewrite the RURI of the incoming INVITE and there are several way to do
that. If you have the two PBXs connected via OpenSIPS, then you already
know that.
If you want to use OpenSIPS, you need to understand SIP and how it wor
Hi All,
Just installed opensips yesterday, using the default config file i was
able to do some user to user calling, and then adding some URI checking
and rewritehost i was able to send calls to my gateway. now i would like
to test the call control for prepaid, my question is simply do i need
On Tuesday 07 April 2009, Nhadie wrote:
> Hi All,
>
> Just installed opensips yesterday, using the default config file i was
> able to do some user to user calling, and then adding some URI
> checking and rewritehost i was able to send calls to my gateway. now i
> would like to test the call contr
Greetings,
I have a quick newbie question, how can I view the raw SIP Mesages
coming in/out of opensips?
I have set the debug logging to 9, and I'm seeing lots of debug
messages (I've redirected it to /var/log/opensips). But even in the
debug log I don't see the raw SIP messages. Do I n
Do a quick google search for
Opensips + ngrep
Ngrep is all u need to run on the opensips box
Sent via BlackBerry by AT&T
-Original Message-
From: Brendan Sterne
Date: Tue, 13 Apr 2010 10:54:24
To:
Subject: [OpenSIPS-Users] Newbie Question: View Raw SIP Mess
try a tool like tshark (part of ethereal/wireshark)
On Tue, Apr 13, 2010 at 10:54 AM, Brendan Sterne wrote:
> Greetings,
>
> I have a quick newbie question, how can I view the raw SIP Mesages coming
> in/out of opensips?
>
> I have set the debug logging to 9, and I'm seeing lots of debug message
Apr 2010 10:54:24
> To:
> Subject: [OpenSIPS-Users] Newbie Question: View Raw SIP Messages?
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
&g
At this point, I can make basic SIP calls (i.e. xlite to xlite) =)
No experience with dialplan at this point, so I'm curious to know if anyone has
OpenSIPS working with Avaya CM?
If so, any generic config/ tips that can be shared?
Thx
--
2010/3/19 Christian Vo :
> No experience with dialplan at this point, so I’m curious to know if anyone
> has OpenSIPS working with Avaya CM?
>
> If so, any generic config/ tips that can be shared?
OpenSIPS is a SIP proxy/registrar (along with other features). No
special config is required in orde
sl_send_reply("403", "No permissions for internat. calls");
> exit;
> };
>};
>
>
384a435,439
>route[4] {
># PSTN route #
> rewritehostport("172.16.217.242");
> route(1
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