Configuring Freeswitch 1.4b for WebRTC Peer to Peer and to the PSTN
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? [image: Happy] <https://secure.teamhively.com/welcome/rate/735/3>[image: Satisfied] <https://secure.teamhively.com/welcome/rate/735/2>[image: Unhappy] <https://secure.teamhively.com/welcome/rate/735/1> Click on a face to provide feedback on my performance!<https://secure.teamhively.com/welcome/rate/735> Synclio Extension:* 110627* Phone: * 866-707-4590* Skype: *No!* *Call me on Synclio! :)* - "*Another great thing about keeping the bug count at zero is that you can respond much faster to competition. Some programmers think of this as keeping the product ready to ship at all times." - Joel Spolsky, Author of Smart and Gets Things Done*
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?
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!"* How did I do? [image: Happy] <https://secure.teamhively.com/welcome/rate/735/3>[image: Satisfied] <https://secure.teamhively.com/welcome/rate/735/2>[image: Unhappy] <https://secure.teamhively.com/welcome/rate/735/1> Click on a face to provide feedback on my performance!<https://secure.teamhively.com/welcome/rate/735> 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?
_________________________________________________________________________ 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
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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!"*
How did I do?
[image: Happy] <https://secure.teamhively.com/welcome/rate/735/3>[image: Satisfied] <https://secure.teamhively.com/welcome/rate/735/2>[image: Unhappy] <https://secure.teamhively.com/welcome/rate/735/1>
Click on a face to provide feedback on my performance!<https://secure.teamhively.com/welcome/rate/735>
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?
_________________________________________________________________________ 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
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ 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
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
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!"*
How did I do?
[image: Happy] <https://secure.teamhively.com/welcome/rate/735/3>[image: Satisfied] <https://secure.teamhively.com/welcome/rate/735/2>[image: Unhappy] <https://secure.teamhively.com/welcome/rate/735/1>
Click on a face to provide feedback on my performance!<https://secure.teamhively.com/welcome/rate/735>
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?
_________________________________________________________________________ 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
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ 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
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org 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
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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!"*
How did I do?
[image: Happy] <https://secure.teamhively.com/welcome/rate/735/3>[image: Satisfied] <https://secure.teamhively.com/welcome/rate/735/2>[image: Unhappy] <https://secure.teamhively.com/welcome/rate/735/1>
Click on a face to provide feedback on my performance!<https://secure.teamhively.com/welcome/rate/735>
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?
_________________________________________________________________________ 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
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ 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
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org 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
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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!"*
How did I do?
[image: Happy] <https://secure.teamhively.com/welcome/rate/735/3>[image: Satisfied] <https://secure.teamhively.com/welcome/rate/735/2>[image: Unhappy] <https://secure.teamhively.com/welcome/rate/735/1>
Click on a face to provide feedback on my performance!<https://secure.teamhively.com/welcome/rate/735>
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?
_________________________________________________________________________ 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
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ 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
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org 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
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ 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
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
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 *"Every Call, Every Time!"* How did I do? [image: Happy] <https://secure.teamhively.com/welcome/rate/735/3>[image: Satisfied] <https://secure.teamhively.com/welcome/rate/735/2>[image: Unhappy] <https://secure.teamhively.com/welcome/rate/735/1> Click on a face to provide feedback on my performance!<https://secure.teamhively.com/welcome/rate/735> Synclio Extension:* 110627* Phone: * 866-707-4590* Skype: *No!* *Call me on Synclio! :)* 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!"*
How did I do?
[image: Happy] <https://secure.teamhively.com/welcome/rate/735/3>[image: Satisfied] <https://secure.teamhively.com/welcome/rate/735/2>[image: Unhappy] <https://secure.teamhively.com/welcome/rate/735/1>
Click on a face to provide feedback on my performance!<https://secure.teamhively.com/welcome/rate/735>
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? >
_________________________________________________________________________ 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
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ 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
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org 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
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ 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
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org 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
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Okay, I added in the dependencies and still get those errors. Do I need to file a bug in JIRA for this or am I just missing something? I'm happy to try a different OS if there's one that's been tried and tested. Of course, more googling reveals that mod_cluechoo is just a joke? http://wiki.freeswitch.org/wiki/Mod_cluechoo
SL (Steam Locomotive) runs across your terminal when you type "sl" as you meant to type "ls". It's just a joke command, and not usefull at all. Put the binary to /usr/local/bin.
I hope I'm not chasing a problem that has absolutely no impact on my ability to get WebRTC working here. :D Thanks for any additional help you can provide, James On Wed, Sep 11, 2013 at 11:19 AM, James Mortensen < james.mortensen@synclio.com> wrote:
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
*"Every Call, Every Time!"*
How did I do?
[image: Happy] <https://secure.teamhively.com/welcome/rate/735/3>[image: Satisfied] <https://secure.teamhively.com/welcome/rate/735/2>[image: Unhappy] <https://secure.teamhively.com/welcome/rate/735/1>
Click on a face to provide feedback on my performance!<https://secure.teamhively.com/welcome/rate/735>
Synclio Extension:* 110627* Phone: * 866-707-4590* Skype: *No!* *Call me on Synclio! :)*
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!"*
How did I do?
[image: Happy] <https://secure.teamhively.com/welcome/rate/735/3>[image: Satisfied] <https://secure.teamhively.com/welcome/rate/735/2>[image: Unhappy] <https://secure.teamhively.com/welcome/rate/735/1>
Click on a face to provide feedback on my performance!<https://secure.teamhively.com/welcome/rate/735>
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? > > > > > _________________________________________________________________________ > 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 > > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >
_________________________________________________________________________ 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
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org 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
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ 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
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org 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
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Installing ibncursesw5 and libncursesw5-dev did resolve the issue. I can now connect to the WebSocket server. Seems the cluechoo module is just something that was added in to separate the help vampires from the people with legitimate issues. :D I got 405 Method Not Allowed errors, and I resolved them by making sure the IP address in the SIP URI matches the IP address in the WS field. At this time, I've successfully registered and am getting back 200 OK's and have now moved onto tweaking the settings so I can get audio going both ways. The candidates appear to be only showing the public IP, so I'll have to figure that out. I'll provide more updates, or questions, as I move forward. James On Wed, Sep 11, 2013 at 11:32 AM, James Mortensen < james.mortensen@synclio.com> wrote:
Okay, I added in the dependencies and still get those errors. Do I need to file a bug in JIRA for this or am I just missing something?
I'm happy to try a different OS if there's one that's been tried and tested.
Of course, more googling reveals that mod_cluechoo is just a joke? http://wiki.freeswitch.org/wiki/Mod_cluechoo
SL (Steam Locomotive) runs across your terminal when you type "sl" as you meant to type "ls". It's just a joke command, and not usefull at all. Put the binary to /usr/local/bin.
I hope I'm not chasing a problem that has absolutely no impact on my ability to get WebRTC working here. :D
Thanks for any additional help you can provide, James
On Wed, Sep 11, 2013 at 11:19 AM, James Mortensen < james.mortensen@synclio.com> wrote:
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
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
Here is my follow up to this issue. It seems something happened to the server where it stopped sending Binding responses to Chrome. This is a known issue that I've seen before that Google says was an Asterisk issue. However, the same thing happens to Freeswitch as well, indicating the problem is the network/server, not the software. I booted up another server from a snapshot, verified two way audio with Asterisk, then reinstalled Freeswitch, uncommented the ws-binding parameter to enable WS on port 5066, then registered user 1000 and 1002 from the tryit.jssip.net demo using the following configuration: Name: James SIP URI: sip:1000@54.X.X.X <-- public IP of server password: 1234 WS URI: ws://54.X.X.X:5066 <--- same public IP and I substituted 1002 in place of 1000 for another user on another network. I verified audio flows both ways with audio flowing through Freeswitch. I made no other changes to the configuration. This was a lot easier to get started with than Asterisk WebRTC. I just made it out to be a lot harder than it was by assuming I needed to manually add in my server's IP address in the configuration files. This happens automatically, even on a NAT'd server. Amazing! :) Now, for the PSTN part, I configured my BANDWIDTH.com provider as an internal context and as an outbound and inbound dialplan. I can connect a call between my cellphone and Chrome, but there's no audio flowing to/from Bandwidth. I suspect the problem has something to do with bridging AVP and SAVPF, since the carriers don't support SRTP. I've grepped the configuration files, and I don't see how I would configure the system to bridge AVP and SAVPF and do transcoding. Any ideas? Thank you!! James On Wed, Sep 11, 2013 at 12:01 PM, James Mortensen < james.mortensen@synclio.com> wrote:
Installing ibncursesw5 and libncursesw5-dev did resolve the issue. I can now connect to the WebSocket server. Seems the cluechoo module is just something that was added in to separate the help vampires from the people with legitimate issues. :D
I got 405 Method Not Allowed errors, and I resolved them by making sure the IP address in the SIP URI matches the IP address in the WS field.
At this time, I've successfully registered and am getting back 200 OK's and have now moved onto tweaking the settings so I can get audio going both ways. The candidates appear to be only showing the public IP, so I'll have to figure that out. I'll provide more updates, or questions, as I move forward.
James
On Wed, Sep 11, 2013 at 11:32 AM, James Mortensen < james.mortensen@synclio.com> wrote:
Okay, I added in the dependencies and still get those errors. Do I need to file a bug in JIRA for this or am I just missing something?
I'm happy to try a different OS if there's one that's been tried and tested.
Of course, more googling reveals that mod_cluechoo is just a joke? http://wiki.freeswitch.org/wiki/Mod_cluechoo
SL (Steam Locomotive) runs across your terminal when you type "sl" as you meant to type "ls". It's just a joke command, and not usefull at all. Put the binary to /usr/local/bin.
I hope I'm not chasing a problem that has absolutely no impact on my ability to get WebRTC working here. :D
Thanks for any additional help you can provide, James
On Wed, Sep 11, 2013 at 11:19 AM, James Mortensen < james.mortensen@synclio.com> wrote:
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
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
That should not matter. It will be taken care of. Verify you can just use bandwidth to call the server and run an echo test or something. Then try a sip phone registered. Then on to the webrtc instance. You probably have some nat problems to bandwidth regardless of WebRTC. On Fri, Sep 13, 2013 at 1:42 PM, James Mortensen < james.mortensen@synclio.com> wrote:
Here is my follow up to this issue. It seems something happened to the server where it stopped sending Binding responses to Chrome. This is a known issue that I've seen before that Google says was an Asterisk issue. However, the same thing happens to Freeswitch as well, indicating the problem is the network/server, not the software.
I booted up another server from a snapshot, verified two way audio with Asterisk, then reinstalled Freeswitch, uncommented the ws-binding parameter to enable WS on port 5066, then registered user 1000 and 1002 from the tryit.jssip.net demo using the following configuration:
Name: James SIP URI: sip:1000@54.X.X.X <-- public IP of server password: 1234 WS URI: ws://54.X.X.X:5066 <--- same public IP
and I substituted 1002 in place of 1000 for another user on another network. I verified audio flows both ways with audio flowing through Freeswitch.
I made no other changes to the configuration. This was a lot easier to get started with than Asterisk WebRTC. I just made it out to be a lot harder than it was by assuming I needed to manually add in my server's IP address in the configuration files. This happens automatically, even on a NAT'd server. Amazing! :)
Now, for the PSTN part, I configured my BANDWIDTH.com provider as an internal context and as an outbound and inbound dialplan. I can connect a call between my cellphone and Chrome, but there's no audio flowing to/from Bandwidth. I suspect the problem has something to do with bridging AVP and SAVPF, since the carriers don't support SRTP.
I've grepped the configuration files, and I don't see how I would configure the system to bridge AVP and SAVPF and do transcoding. Any ideas?
Thank you!!
James
On Wed, Sep 11, 2013 at 12:01 PM, James Mortensen < james.mortensen@synclio.com> wrote:
Installing ibncursesw5 and libncursesw5-dev did resolve the issue. I can now connect to the WebSocket server. Seems the cluechoo module is just something that was added in to separate the help vampires from the people with legitimate issues. :D
I got 405 Method Not Allowed errors, and I resolved them by making sure the IP address in the SIP URI matches the IP address in the WS field.
At this time, I've successfully registered and am getting back 200 OK's and have now moved onto tweaking the settings so I can get audio going both ways. The candidates appear to be only showing the public IP, so I'll have to figure that out. I'll provide more updates, or questions, as I move forward.
James
On Wed, Sep 11, 2013 at 11:32 AM, James Mortensen < james.mortensen@synclio.com> wrote:
Okay, I added in the dependencies and still get those errors. Do I need to file a bug in JIRA for this or am I just missing something?
I'm happy to try a different OS if there's one that's been tried and tested.
Of course, more googling reveals that mod_cluechoo is just a joke? http://wiki.freeswitch.org/wiki/Mod_cluechoo
SL (Steam Locomotive) runs across your terminal when you type "sl" as you meant to type "ls". It's just a joke command, and not usefull at all. Put the binary to /usr/local/bin.
I hope I'm not chasing a problem that has absolutely no impact on my ability to get WebRTC working here. :D
Thanks for any additional help you can provide, James
On Wed, Sep 11, 2013 at 11:19 AM, James Mortensen < james.mortensen@synclio.com> wrote:
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
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
_________________________________________________________________________ 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
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
Hi Anothony, Inbound calling is proving to be more challenging. Here's what I've tried from https://wiki.freeswitch.org/wiki/Auto_NAT: 1. Verified the following params were set in my sip profiles (these were already set by default): <param name="ext-rtp-ip" value="auto-nat"/> <param name="ext-sip-ip" value="auto-nat"/> 2. Ran sofia status, which shows this: Name Type Data State ================================================================================================= 10.166.245.111 alias internal ALIASED internal-ipv6 profile sip:mod_sofia@[::1]:5060 RUNNING (0) external profile sip:mod_sofia@10.166.245.111:5080 RUNNING (0) external::example.com gateway sip:joeuser@example.com NOREG internal profile sip:mod_sofia@10.166.245.111:5060 RUNNING (0) internal::bandwidth.com gateway sip:your user name@67.231.8.195 NOREG ================================================================================================= 3 profiles 1 alias 3. Ran *sofia status profile internal* expecting to see ext-rtp-ip and ext-sip-ip set, but they're missing: Name internal Domain Name N/A Auto-NAT false DBName sofia_reg_internal Pres Hosts 10.166.245.111,10.166.245.111 Dialplan XML Context public Challenge Realm auto_from RTP-IP 10.166.245.111 SIP-IP 10.166.245.111 URL sip:mod_sofia@10.166.245.111:5060 BIND-URL sip:mod_sofia@10.166.245.111:5060;transport=udp,tcp HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS IN G722,PCMU,PCMA,GSM CODECS OUT G722,PCMU,PCMA,GSM TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG true PROXY-MEDIA false ZRTP-PASSTHRU true AGGRESSIVENAT false CALLS-IN 86 FAILED-CALLS-IN 86 CALLS-OUT 0 FAILED-CALLS-OUT 0 REGISTRATIONS 1 So regarding NAT issues, it seems Freeswitch isn't able to get the server's public IP perhaps, despite auto-nat being the correct setting? This is just an EC2 server behind Amazon NAT. Also, I ran: nat_map status Nat Type: UNKNOWN, ExtIP: NAT port mapping enabled. 0 total. I did perform an echo test from user 1000 to 9196 from Chrome to Freeswitch, which works as expected, but getting Bandwidth incoming calls to be processed correctly is a challenge. It just loops over and over again with this in the DEBUG console: 2013-09-13 20:27:31.255055 [NOTICE] switch_channel.c:1030 New Channel sofia/internal/+19712383780@192.168.37.68[80196f26-febf-4363-b48c-ee88022c6b9c] 2013-09-13 20:27:31.255055 [DEBUG] switch_core_session.c:1006 Send signal sofia/internal/+19712383780@192.168.37.68 [BREAK] 2013-09-13 20:27:31.255055 [DEBUG] switch_core_session.c:1006 Send signal sofia/internal/+19712383780@192.168.37.68 [BREAK] 2013-09-13 20:27:31.255055 [DEBUG] switch_core_state_machine.c:418 (sofia/internal/+19712383780@192.168.37.68) Running State Change CS_NEW 2013-09-13 20:27:31.255055 [DEBUG] switch_core_state_machine.c:436 (sofia/internal/+19712383780@192.168.37.68) State NEW 2013-09-13 20:27:31.275057 [DEBUG] sofia.c:8003 IP 67.231.4.195 Rejected by acl "domains". Falling back to Digest auth. 2013-09-13 20:27:31.275057 [DEBUG] switch_core_session.c:1006 Send signal sofia/internal/+19712383780@192.168.37.68 [BREAK] 2013-09-13 20:27:31.275057 [DEBUG] sofia.c:1787 detaching session 80196f26-febf-4363-b48c-ee88022c6b9c So in summary, the WebRTC seems good but not the PSTN. I can do Chrome to FS to Chrome, Chrome to FS Echo, I can call from Chrome to FS to Bandwidth (NO AUDIO) but calling inbound loops. I'm assuming all of the routing is done using the Bandwidth DID I've pointed to the server, not the 971xxxxxxx number that I'm calling from on Verizon/Google Voice. Thank you for your help! James On Fri, Sep 13, 2013 at 11:51 AM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
That should not matter. It will be taken care of.
Verify you can just use bandwidth to call the server and run an echo test or something. Then try a sip phone registered. Then on to the webrtc instance.
You probably have some nat problems to bandwidth regardless of WebRTC.
On Fri, Sep 13, 2013 at 1:42 PM, James Mortensen < james.mortensen@synclio.com> wrote:
Here is my follow up to this issue. It seems something happened to the server where it stopped sending Binding responses to Chrome. This is a known issue that I've seen before that Google says was an Asterisk issue. However, the same thing happens to Freeswitch as well, indicating the problem is the network/server, not the software.
I booted up another server from a snapshot, verified two way audio with Asterisk, then reinstalled Freeswitch, uncommented the ws-binding parameter to enable WS on port 5066, then registered user 1000 and 1002 from the tryit.jssip.net demo using the following configuration:
Name: James SIP URI: sip:1000@54.X.X.X <-- public IP of server password: 1234 WS URI: ws://54.X.X.X:5066 <--- same public IP
and I substituted 1002 in place of 1000 for another user on another network. I verified audio flows both ways with audio flowing through Freeswitch.
I made no other changes to the configuration. This was a lot easier to get started with than Asterisk WebRTC. I just made it out to be a lot harder than it was by assuming I needed to manually add in my server's IP address in the configuration files. This happens automatically, even on a NAT'd server. Amazing! :)
Now, for the PSTN part, I configured my BANDWIDTH.com provider as an internal context and as an outbound and inbound dialplan. I can connect a call between my cellphone and Chrome, but there's no audio flowing to/from Bandwidth. I suspect the problem has something to do with bridging AVP and SAVPF, since the carriers don't support SRTP.
I've grepped the configuration files, and I don't see how I would configure the system to bridge AVP and SAVPF and do transcoding. Any ideas?
Thank you!!
James
On Wed, Sep 11, 2013 at 12:01 PM, James Mortensen < james.mortensen@synclio.com> wrote:
Installing ibncursesw5 and libncursesw5-dev did resolve the issue. I can now connect to the WebSocket server. Seems the cluechoo module is just something that was added in to separate the help vampires from the people with legitimate issues. :D
I got 405 Method Not Allowed errors, and I resolved them by making sure the IP address in the SIP URI matches the IP address in the WS field.
At this time, I've successfully registered and am getting back 200 OK's and have now moved onto tweaking the settings so I can get audio going both ways. The candidates appear to be only showing the public IP, so I'll have to figure that out. I'll provide more updates, or questions, as I move forward.
James
On Wed, Sep 11, 2013 at 11:32 AM, James Mortensen < james.mortensen@synclio.com> wrote:
Okay, I added in the dependencies and still get those errors. Do I need to file a bug in JIRA for this or am I just missing something?
I'm happy to try a different OS if there's one that's been tried and tested.
Of course, more googling reveals that mod_cluechoo is just a joke? http://wiki.freeswitch.org/wiki/Mod_cluechoo
SL (Steam Locomotive) runs across your terminal when you type "sl" as you meant to type "ls". It's just a joke command, and not usefull at all. Put the binary to /usr/local/bin.
I hope I'm not chasing a problem that has absolutely no impact on my ability to get WebRTC working here. :D
Thanks for any additional help you can provide, James
On Wed, Sep 11, 2013 at 11:19 AM, James Mortensen < james.mortensen@synclio.com> wrote:
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
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 > > > >
_________________________________________________________________________ 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
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org 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
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
did you do all the things in https://wiki.freeswitch.org/wiki/Amazon_EC2 Also are you using the external or internal profile (port 5065 is the default unauthenticated port) if you want to use internal you have to comment out the auth-calls and the inbound acl lines. Most of this should be available in our various books and wiki articles. On Fri, Sep 13, 2013 at 3:31 PM, James Mortensen < james.mortensen@synclio.com> wrote:
Hi Anothony,
Inbound calling is proving to be more challenging. Here's what I've tried from https://wiki.freeswitch.org/wiki/Auto_NAT:
1. Verified the following params were set in my sip profiles (these were already set by default):
<param name="ext-rtp-ip" value="auto-nat"/> <param name="ext-sip-ip" value="auto-nat"/>
2. Ran sofia status, which shows this:
Name Type Data State
================================================================================================= 10.166.245.111 alias internal ALIASED internal-ipv6 profile sip:mod_sofia@ [::1]:5060 RUNNING (0) external profile sip:mod_sofia@10.166.245.111:5080 RUNNING (0) external::example.com gateway sip:joeuser@example.com NOREG internal profile sip:mod_sofia@10.166.245.111:5060 RUNNING (0) internal::bandwidth.com gateway sip:your user name@67.231.8.195 NOREG
================================================================================================= 3 profiles 1 alias
3. Ran *sofia status profile internal* expecting to see ext-rtp-ip and ext-sip-ip set, but they're missing:
Name internal Domain Name N/A Auto-NAT false DBName sofia_reg_internal Pres Hosts 10.166.245.111,10.166.245.111 Dialplan XML Context public Challenge Realm auto_from RTP-IP 10.166.245.111 SIP-IP 10.166.245.111 URL sip:mod_sofia@10.166.245.111:5060 BIND-URL sip:mod_sofia@10.166.245.111:5060;transport=udp,tcp HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS IN G722,PCMU,PCMA,GSM CODECS OUT G722,PCMU,PCMA,GSM TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG true PROXY-MEDIA false ZRTP-PASSTHRU true AGGRESSIVENAT false CALLS-IN 86 FAILED-CALLS-IN 86 CALLS-OUT 0 FAILED-CALLS-OUT 0 REGISTRATIONS 1
So regarding NAT issues, it seems Freeswitch isn't able to get the server's public IP perhaps, despite auto-nat being the correct setting? This is just an EC2 server behind Amazon NAT.
Also, I ran: nat_map status
Nat Type: UNKNOWN, ExtIP: NAT port mapping enabled.
0 total.
I did perform an echo test from user 1000 to 9196 from Chrome to Freeswitch, which works as expected, but getting Bandwidth incoming calls to be processed correctly is a challenge. It just loops over and over again with this in the DEBUG console:
2013-09-13 20:27:31.255055 [NOTICE] switch_channel.c:1030 New Channel sofia/internal/+19712383780@192.168.37.68[80196f26-febf-4363-b48c-ee88022c6b9c] 2013-09-13 20:27:31.255055 [DEBUG] switch_core_session.c:1006 Send signal sofia/internal/+19712383780@192.168.37.68 [BREAK] 2013-09-13 20:27:31.255055 [DEBUG] switch_core_session.c:1006 Send signal sofia/internal/+19712383780@192.168.37.68 [BREAK] 2013-09-13 20:27:31.255055 [DEBUG] switch_core_state_machine.c:418 (sofia/internal/+19712383780@192.168.37.68) Running State Change CS_NEW 2013-09-13 20:27:31.255055 [DEBUG] switch_core_state_machine.c:436 (sofia/internal/+19712383780@192.168.37.68) State NEW 2013-09-13 20:27:31.275057 [DEBUG] sofia.c:8003 IP 67.231.4.195 Rejected by acl "domains". Falling back to Digest auth. 2013-09-13 20:27:31.275057 [DEBUG] switch_core_session.c:1006 Send signal sofia/internal/+19712383780@192.168.37.68 [BREAK] 2013-09-13 20:27:31.275057 [DEBUG] sofia.c:1787 detaching session 80196f26-febf-4363-b48c-ee88022c6b9c
So in summary, the WebRTC seems good but not the PSTN. I can do Chrome to FS to Chrome, Chrome to FS Echo, I can call from Chrome to FS to Bandwidth (NO AUDIO) but calling inbound loops. I'm assuming all of the routing is done using the Bandwidth DID I've pointed to the server, not the 971xxxxxxx number that I'm calling from on Verizon/Google Voice.
Thank you for your help! James
On Fri, Sep 13, 2013 at 11:51 AM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
That should not matter. It will be taken care of.
Verify you can just use bandwidth to call the server and run an echo test or something. Then try a sip phone registered. Then on to the webrtc instance.
You probably have some nat problems to bandwidth regardless of WebRTC.
On Fri, Sep 13, 2013 at 1:42 PM, James Mortensen < james.mortensen@synclio.com> wrote:
Here is my follow up to this issue. It seems something happened to the server where it stopped sending Binding responses to Chrome. This is a known issue that I've seen before that Google says was an Asterisk issue. However, the same thing happens to Freeswitch as well, indicating the problem is the network/server, not the software.
I booted up another server from a snapshot, verified two way audio with Asterisk, then reinstalled Freeswitch, uncommented the ws-binding parameter to enable WS on port 5066, then registered user 1000 and 1002 from the tryit.jssip.net demo using the following configuration:
Name: James SIP URI: sip:1000@54.X.X.X <-- public IP of server password: 1234 WS URI: ws://54.X.X.X:5066 <--- same public IP
and I substituted 1002 in place of 1000 for another user on another network. I verified audio flows both ways with audio flowing through Freeswitch.
I made no other changes to the configuration. This was a lot easier to get started with than Asterisk WebRTC. I just made it out to be a lot harder than it was by assuming I needed to manually add in my server's IP address in the configuration files. This happens automatically, even on a NAT'd server. Amazing! :)
Now, for the PSTN part, I configured my BANDWIDTH.com provider as an internal context and as an outbound and inbound dialplan. I can connect a call between my cellphone and Chrome, but there's no audio flowing to/from Bandwidth. I suspect the problem has something to do with bridging AVP and SAVPF, since the carriers don't support SRTP.
I've grepped the configuration files, and I don't see how I would configure the system to bridge AVP and SAVPF and do transcoding. Any ideas?
Thank you!!
James
On Wed, Sep 11, 2013 at 12:01 PM, James Mortensen < james.mortensen@synclio.com> wrote:
Installing ibncursesw5 and libncursesw5-dev did resolve the issue. I can now connect to the WebSocket server. Seems the cluechoo module is just something that was added in to separate the help vampires from the people with legitimate issues. :D
I got 405 Method Not Allowed errors, and I resolved them by making sure the IP address in the SIP URI matches the IP address in the WS field.
At this time, I've successfully registered and am getting back 200 OK's and have now moved onto tweaking the settings so I can get audio going both ways. The candidates appear to be only showing the public IP, so I'll have to figure that out. I'll provide more updates, or questions, as I move forward.
James
On Wed, Sep 11, 2013 at 11:32 AM, James Mortensen < james.mortensen@synclio.com> wrote:
Okay, I added in the dependencies and still get those errors. Do I need to file a bug in JIRA for this or am I just missing something?
I'm happy to try a different OS if there's one that's been tried and tested.
Of course, more googling reveals that mod_cluechoo is just a joke? http://wiki.freeswitch.org/wiki/Mod_cluechoo
SL (Steam Locomotive) runs across your terminal when you type "sl" as you meant to type "ls". It's just a joke command, and not usefull at all. Put the binary to /usr/local/bin.
I hope I'm not chasing a problem that has absolutely no impact on my ability to get WebRTC working here. :D
Thanks for any additional help you can provide, James
On Wed, Sep 11, 2013 at 11:19 AM, James Mortensen < james.mortensen@synclio.com> wrote:
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
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 >> >> >> >>
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ 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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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
Hi Anthony, All the ports were open, but I didn't have the external IP and the settings listed in the Amazon EC2 documentation set. After setting them and restarting Freeswitch, I made 20 successful calls in a row with 2 way audio from Chrome to the PSTN! Thank you for your help! James James *"Every Call, Every Time!"* How did I do? [image: Happy] <https://secure.teamhively.com/welcome/rate/735/3>[image: Satisfied] <https://secure.teamhively.com/welcome/rate/735/2>[image: Unhappy] <https://secure.teamhively.com/welcome/rate/735/1> Click on a face to provide feedback on my performance!<https://secure.teamhively.com/welcome/rate/735> Synclio Extension:* 110627* Phone: * 866-707-4590* Skype: *No!* *Call me on Synclio! :)* On Fri, Sep 13, 2013 at 1:42 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
did you do all the things in https://wiki.freeswitch.org/wiki/Amazon_EC2
Also are you using the external or internal profile (port 5065 is the default unauthenticated port) if you want to use internal you have to comment out the auth-calls and the inbound acl lines.
Most of this should be available in our various books and wiki articles.
On Fri, Sep 13, 2013 at 3:31 PM, James Mortensen < james.mortensen@synclio.com> wrote:
Hi Anothony,
Inbound calling is proving to be more challenging. Here's what I've tried from https://wiki.freeswitch.org/wiki/Auto_NAT:
1. Verified the following params were set in my sip profiles (these were already set by default):
<param name="ext-rtp-ip" value="auto-nat"/> <param name="ext-sip-ip" value="auto-nat"/>
2. Ran sofia status, which shows this:
Name Type Data State
================================================================================================= 10.166.245.111 alias internal ALIASED internal-ipv6 profile sip:mod_sofia@ [::1]:5060 RUNNING (0) external profile sip:mod_sofia@10.166.245.111:5080 RUNNING (0) external::example.com gateway sip:joeuser@example.com NOREG internal profile sip:mod_sofia@10.166.245.111:5060 RUNNING (0) internal::bandwidth.com gateway sip:your user name@67.231.8.195 NOREG
================================================================================================= 3 profiles 1 alias
3. Ran *sofia status profile internal* expecting to see ext-rtp-ip and ext-sip-ip set, but they're missing:
Name internal Domain Name N/A Auto-NAT false DBName sofia_reg_internal Pres Hosts 10.166.245.111,10.166.245.111 Dialplan XML Context public Challenge Realm auto_from RTP-IP 10.166.245.111 SIP-IP 10.166.245.111 URL sip:mod_sofia@10.166.245.111:5060 BIND-URL sip:mod_sofia@10.166.245.111:5060;transport=udp,tcp HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS IN G722,PCMU,PCMA,GSM CODECS OUT G722,PCMU,PCMA,GSM TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG true PROXY-MEDIA false ZRTP-PASSTHRU true AGGRESSIVENAT false CALLS-IN 86 FAILED-CALLS-IN 86 CALLS-OUT 0 FAILED-CALLS-OUT 0 REGISTRATIONS 1
So regarding NAT issues, it seems Freeswitch isn't able to get the server's public IP perhaps, despite auto-nat being the correct setting? This is just an EC2 server behind Amazon NAT.
Also, I ran: nat_map status
Nat Type: UNKNOWN, ExtIP: NAT port mapping enabled.
0 total.
I did perform an echo test from user 1000 to 9196 from Chrome to Freeswitch, which works as expected, but getting Bandwidth incoming calls to be processed correctly is a challenge. It just loops over and over again with this in the DEBUG console:
2013-09-13 20:27:31.255055 [NOTICE] switch_channel.c:1030 New Channel sofia/internal/+19712383780@192.168.37.68[80196f26-febf-4363-b48c-ee88022c6b9c] 2013-09-13 20:27:31.255055 [DEBUG] switch_core_session.c:1006 Send signal sofia/internal/+19712383780@192.168.37.68 [BREAK] 2013-09-13 20:27:31.255055 [DEBUG] switch_core_session.c:1006 Send signal sofia/internal/+19712383780@192.168.37.68 [BREAK] 2013-09-13 20:27:31.255055 [DEBUG] switch_core_state_machine.c:418 (sofia/internal/+19712383780@192.168.37.68) Running State Change CS_NEW 2013-09-13 20:27:31.255055 [DEBUG] switch_core_state_machine.c:436 (sofia/internal/+19712383780@192.168.37.68) State NEW 2013-09-13 20:27:31.275057 [DEBUG] sofia.c:8003 IP 67.231.4.195 Rejected by acl "domains". Falling back to Digest auth. 2013-09-13 20:27:31.275057 [DEBUG] switch_core_session.c:1006 Send signal sofia/internal/+19712383780@192.168.37.68 [BREAK] 2013-09-13 20:27:31.275057 [DEBUG] sofia.c:1787 detaching session 80196f26-febf-4363-b48c-ee88022c6b9c
So in summary, the WebRTC seems good but not the PSTN. I can do Chrome to FS to Chrome, Chrome to FS Echo, I can call from Chrome to FS to Bandwidth (NO AUDIO) but calling inbound loops. I'm assuming all of the routing is done using the Bandwidth DID I've pointed to the server, not the 971xxxxxxx number that I'm calling from on Verizon/Google Voice.
Thank you for your help! James
On Fri, Sep 13, 2013 at 11:51 AM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
That should not matter. It will be taken care of.
Verify you can just use bandwidth to call the server and run an echo test or something. Then try a sip phone registered. Then on to the webrtc instance.
You probably have some nat problems to bandwidth regardless of WebRTC.
On Fri, Sep 13, 2013 at 1:42 PM, James Mortensen < james.mortensen@synclio.com> wrote:
Here is my follow up to this issue. It seems something happened to the server where it stopped sending Binding responses to Chrome. This is a known issue that I've seen before that Google says was an Asterisk issue. However, the same thing happens to Freeswitch as well, indicating the problem is the network/server, not the software.
I booted up another server from a snapshot, verified two way audio with Asterisk, then reinstalled Freeswitch, uncommented the ws-binding parameter to enable WS on port 5066, then registered user 1000 and 1002 from the tryit.jssip.net demo using the following configuration:
Name: James SIP URI: sip:1000@54.X.X.X <-- public IP of server password: 1234 WS URI: ws://54.X.X.X:5066 <--- same public IP
and I substituted 1002 in place of 1000 for another user on another network. I verified audio flows both ways with audio flowing through Freeswitch.
I made no other changes to the configuration. This was a lot easier to get started with than Asterisk WebRTC. I just made it out to be a lot harder than it was by assuming I needed to manually add in my server's IP address in the configuration files. This happens automatically, even on a NAT'd server. Amazing! :)
Now, for the PSTN part, I configured my BANDWIDTH.com provider as an internal context and as an outbound and inbound dialplan. I can connect a call between my cellphone and Chrome, but there's no audio flowing to/from Bandwidth. I suspect the problem has something to do with bridging AVP and SAVPF, since the carriers don't support SRTP.
I've grepped the configuration files, and I don't see how I would configure the system to bridge AVP and SAVPF and do transcoding. Any ideas?
Thank you!!
James
On Wed, Sep 11, 2013 at 12:01 PM, James Mortensen < james.mortensen@synclio.com> wrote:
Installing ibncursesw5 and libncursesw5-dev did resolve the issue. I can now connect to the WebSocket server. Seems the cluechoo module is just something that was added in to separate the help vampires from the people with legitimate issues. :D
I got 405 Method Not Allowed errors, and I resolved them by making sure the IP address in the SIP URI matches the IP address in the WS field.
At this time, I've successfully registered and am getting back 200 OK's and have now moved onto tweaking the settings so I can get audio going both ways. The candidates appear to be only showing the public IP, so I'll have to figure that out. I'll provide more updates, or questions, as I move forward.
James
On Wed, Sep 11, 2013 at 11:32 AM, James Mortensen < james.mortensen@synclio.com> wrote:
Okay, I added in the dependencies and still get those errors. Do I need to file a bug in JIRA for this or am I just missing something?
I'm happy to try a different OS if there's one that's been tried and tested.
Of course, more googling reveals that mod_cluechoo is just a joke? http://wiki.freeswitch.org/wiki/Mod_cluechoo
> SL (Steam Locomotive) runs across your terminal when you type "sl" as you meant to type "ls". It's just a joke command, and not usefull at all. Put the binary to /usr/local/bin.
I hope I'm not chasing a problem that has absolutely no impact on my ability to get WebRTC working here. :D
Thanks for any additional help you can provide, James
On Wed, Sep 11, 2013 at 11:19 AM, James Mortensen < james.mortensen@synclio.com> wrote:
> 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 > > > > 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 >>> >>> >>> >>> >
_________________________________________________________________________ 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
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org 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
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ 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
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org 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
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Hi Guys, May you can help me, too =) I have two Chrome (SipML5) clients and want to make a call between them, but i can't to do a call, because every time i received error (USER_NOT_REGISTERED) I tried to check registration exists via "sofia status profile EXT reg" and i see registered users, but when i tried to check it via "list_users" i can't see them! With Great Regards -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Configuring-Freeswitch-1-4b-fo... Sent from the freeswitch-users mailing list archive at Nabble.com.
participants (4)
-
Alex S
-
Anthony Minessale
-
Brian West
-
James Mortensen