[Freeswitch-users] Where to go from here?

[Bote Man]

I assume that this is a reply to my previous message, but since you created
a new message it broke the thread.

There is no boiler plate guide to receiving calls from a VoIP provider, each
one does whatever it chooses and you as the paying customer have no say in
the matter so the easiest path is to conform to their method. (Is this
really the way it's supposed to be?) Fortunately, FreeSWITCH makes this
easy.

Your job, should you decide to accept it, is to determine what information
is being sent to your FS installation and in which fields it appears, then
edit the dialplan that services that inbound call to recognize the number
and do the right thing with it, another choice that is entirely yours. I'm
assuming that it should ring one or more extensions in which case you might
not even need to touch the "transfer" line in the dialplan since it
transfers to extension 1000. 

The vanilla public dialplan file that ships with FS looks for  
destination_number = 5551212
which I'm just guessing is not a number that they will send to you, so it
doesn't match the one and only condition test in the dialplan and terminates
the call, so your SIP provider returns a busy to the caller.

For instance, in my case I discovered by reading the FS logs that
CallCentric sends my DID/DNIS in the sip_to_user field (at least that's the
variable in which FS stuffs it), so I modified my public dialplan in
conf/dialplan/public/callcentric.xml to test that field against my number:

 <condition field="${sip_to_user}" expression="^(12024561000)$">

I have no idea what will work in your case which is why you need to get your
hands dirty looking at the logs. Once you get inbound calls transferring to
your desired extension(s) I'm sure you will be eager to delve into it more.

Bote


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of T Fred
Farmington
Sent: Tuesday, 24 February, 2015 12:58
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Where to go from here?

It appears to me that the vanilla configuration files that come with
FreeSWITCH allow you to run in a very limited mode.

As I indicated before I can get my in-house softphones to successfully call
into the FreeSWITCH and reach one of the default extensions.   
When I do this I hear the associated FreeSWITCH default voice recordings

But when I try to call in via my outside SIP line, I cannot get the
connection.   
I see the call arrive in the FS_CLI  and in the log files, but I only get a
Busy signal and no connection is made.

Editing my sip_profiles - internal & external xml files did not change that
behavior.
Either:
1.  I made the wrong edits in those files 2.  I did not edit enough files 3.
or I have no idea what I am doing.

I would rather understand what should be done than just 'groping in the
dark' making edits here and there based on a partial understanding (or lack
of understanding) of some web reference.   

I thought that the FreeSWITCH 1.2. book would help, but nowhere does it say:
If you want to receive 'outside' calls, do this......
My current SIP line is configured very simply.
*  No SIP authentication required
*  A single IP address on Port 5060 upd
*  DTMF type is set to RFC2833
*  Rtp will be from 208.85.184.41 udp ports 10,000 to 59,000  (although we
are doing voice only at this time)

I have a number of things that I would like the FreeSWITCH phone switch to
do, but I feel that I need to take 'baby steps'.
So my First goal is to just successfully receive an 'outside' call to one of
the existing default extensions - similar to how the softphones work now.

My second 'baby step' goal would be to to use one of my softphones to place
a call OUT on the same SIP line.

After that I can look at beginning to understand how to configure the
various internal extensions to do various tasks.

Thanks for your continued patience and support.





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