I ran the netstat command, and it doesn't appear to be listening. It doesn't appear to be listening on anything.  I do have this error in the console when starting:

2013-09-11 18:16:14.445921 [ERR] switch_nat.c:201 Error checking for PMP [general error]

AND

2013-09-11 18:09:30.233568 [CRIT] switch_loadable_module.c:1383 Error Loading module /opt/freeswitch-1.4b/mod/mod_cluechoo.so
**/opt/freeswitch-1.4b/mod/mod_cluechoo.so: undefined symbol: waddch**

I  believe I overlooked these earlier. But it's possible I'm missing a dependency.  I'm going to install the libncurses packages as described here http://jira.freeswitch.org/browse/FS-3689 and then rebuild to see if that helps.  I'm running Ubuntu 12.10.

James



James

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On Wed, Sep 11, 2013 at 11:07 AM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
Did you open all the necessary firewall ports?
Playing around on amazon as your first try complicates thing a bit for you.

You should be able to verify its listening on the port with netstat -an | grep 5066



On Wed, Sep 11, 2013 at 12:55 PM, James Mortensen <james.mortensen@synclio.com> wrote:
Here's another update to my adventures in Freeswitch WebRTC.  I assume from the getting started documentation that there are users created by default with the password 1234, so I'm trying to create the ws 5066 connection from the TryIt JsSIP demo:  http://tryit.jssip.net

Name: James
SIP URI: 1000@Y.Y.Y.Y   <--- Local IP of EC2 server
SIP password: 1234
WS URI:  ws://X.X.X.X:5066   <--- Public IP of EC2 server

Hope this helps!
James


On Wed, Sep 11, 2013 at 10:31 AM, James Mortensen <james.mortensen@synclio.com> wrote:
Hi Anthony,

I have a Chrome client behind a NAT'd network, and I have a Freeswitch 1.4b instance sitting behind a NAT'd EC2 server.  I enabled ws-binding in internal.xml. It's the only area where I see anything related to Websockets.  

I also verified my firewall is opened on port 5066. I can see requests coming in from Chrome when I try to register, but Freeswitch's logs don't even twitch.  I followed the instructions here to enable DEBUG logging: http://wiki.freeswitch.org/wiki/Troubleshooting_Freeswitch

I've also grepped the configuration files and internal.xml is the only location of "websocket", other than the htdocs.... which seem to relate to getting a GUI running.

Perhaps there's something I'm misunderstanding about logging in Freeswitch?  I verified from the console that debug logging is enabled.

Hope this helps and thank you!
James



On Wed, Sep 11, 2013 at 9:15 AM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
Are you testing by connecting to internal or external IP?
Uncommenting the param is really the only thing you have to do.




On Wed, Sep 11, 2013 at 10:28 AM, James Mortensen <james.mortensen@synclio.com> wrote:
Hi Brian,

I didn't stray too far from the default configuration.  I just uncommented the param and left the default port in place, 5066.  Is there a way to tell if the WS/HTTP server is running from within Freeswitch?  I didn't see a command for that.

I also replaced the ext-sip-ip and ext-rtp-ip with my stun server and my server's public IP in internal.xml:

    <param name="ext-rtp-ip" value="stun:stun.l.google.com:19302"/>
    <param name="ext-sip-ip" value="MY_PUBLIC_IP" />

The server starts up, but doesn't accept websocket connections.  What information can I get you so this is more helpful?

Thank you!
James 


James

"Every Call, Every Time!"


Synclio Extension:  110627
Phone:  866-707-4590
Skype: No! Call me on Synclio! :)


On Wed, Sep 11, 2013 at 6:13 AM, Brian West <brian@freeswitch.org> wrote:
What exactly did you put into the ws-binding param?

/b


On Sep 10, 2013, at 8:48 PM, James Mortensen <james.mortensen@synclio.com> wrote:

> Hello,
>
> I'm working on a one-way audio bug that may or may not be either Asterisk or Chrome WebRTC related. Details are here: https://code.google.com/p/webrtc/issues/detail?id=2347, but basically the problem is that I get about 1 in 10 calls returning one way audio where audio flows from the PSTN to Chrome but not the other way.
>
> I believe the issue may be with Asterisk 11, based on Wireshark traces and other troubleshooting, and if that's the case, I'm happy to try Freeswitch as an alternative.
> I tried the demo here:  https://webrtc.freeswitch.org/webrtc/portal.html, and the 20 test calls to the PSTN worked perfect with two way audio.
>
> So, now, onto the Freeswitch question:  Where is all the documentation for configuring the Websocket server on Freeswitch 1.4b?  Where is the documentation that explains how to get ICE enabled?  Why, when I enabled the ws-bind port in internal.xml does the client fail to establish a WS connection?
>
> Basically, I'd love to get my hands on the docs for setting this up, but the Internet is largely silent on this matter?  Is it possible for the Freeswitch team to share the configuration they used to get that JsSIP/Freeswitch demo setup?
>
> If not the configuration, where would I find the docs for Freeswitch 1.4b?
>
> Thank you!!
>
> James
>
> "Every Call, Every Time!"
>
> How did I do?
>

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_________________________________________________________________________
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_________________________________________________________________________
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http://www.freeswitchsolutions.com

FreeSWITCH-powered IP PBX: The CudaTel Communication Server
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Official FreeSWITCH Sites
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--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
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Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale@hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com
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FreeSWITCH Developer Conference
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googletalk:conf+888@conference.freeswitch.org
pstn:+19193869900

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

FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com

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

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