[Freeswitch-users] Where to go from here?

[Bote Man]

CORRECTION: 

- Please copy all requested files to 

https://pastebin.freeswitch.org/

(use the credentials in the note  above the prompt dialog box that pops up)

 

- Choose FreeSWITCH highlighting

 

- And post the link to your pastebin here to keep the thread readable.

 

Thanks.

 

It sounds like you are almost there, so don’t despair. Determine which FreeSWITCH profile the call from your SIP provider is hitting, most likely “external” which should apply the context “public” for call treatment. You control the public dialplan under conf/dialplan/public.xml and its child directory of that same name.

 

Then it’s down to the ever so fun work of reading the detailed log files to see where the call is failing. My guess is that your provider is putting the destination number in a field that FS does not look for out of the box, or you have not even configured your public dialplan yet because it is so new. It should take only a few tweaks to get it receiving calls at this point. The stock public dialplan maps a generic PSTN number to extension 1000 to get you started, but all of that can be changed to suit your needs.

 

Success to you.

 

Bote

 

 

From: Frank Ochere
Sent: Saturday, 21 February, 2015 15:17
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Where to go from here?

 

TF

 

Please attach the log files (increase verbosity, on the freeswitch cli type /log 7), the dialplan, internal & external xml files (blank out any authentication credentials)

 

Regards

 

Frank

 

On Sat, Feb 21, 2015 at 12:45 AM, T Fred Farmington <tfred31 at yahoo.com> wrote:

Pardon my naivety, but I am fairly new to FreeSWITCH and, despite having read the FreeSWITCH 1.2 book and doing MANY web searches, I cannot seem to find the correct combination of settings to get things working.

Currently I have an 'outside' SIP line number pointed to my FreeSWITCH server.
When I call that number I can see its arrival via FS_CLI console and it shows in the freeswitch.log file
        (I can include those log file lines if it would help but it is 106 lines - I don't want to 'swamp' the posting thread)
However when I attempt to establish the call the line consistently goes Busy and will not establish a connection.

I have tried various recommended settings for:
*  ..\conf\autoload_configs\acl.conf.xml
*  ..\conf\sip_profiles\external.xml    (and its ipv6 version)
*  ..\conf\sip_profiles\internal.xml    (and its ipv6 version)
*  ..\conf\vars.xml

Each time I try something, it seems like everything else quits working.
So, instead of trying to 'guess' which of the configuration changes made things worse, I Uninstall FreeSWITCH and then Re-Install it.

After that, my in-house, internal Softphones  (Phoner, XTalk, & Voiper)  all work just fine to the FreeSWITCH default extensions.
But the external SIP call does not work.

I'd seriously like to get beyond these initial configuration problems and begin working on actually configuring how the overall phone system is intended to operate.

Where/How do I get things working?

Any advice/suggestions you can offer would be greatly appreciated.

Thanks
TF


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150222/b15c4228/attachment-0001.html