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</o:shapelayout></xml><![endif]--></head><body lang=EN-US link=blue vlink=purple><div class=WordSection1><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Did you open the websocket ports on your firewall? <a href="https://wiki.freeswitch.org/wiki/Firewall">https://wiki.freeswitch.org/wiki/Firewall</a> <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:7.5pt;font-family:"Verdana","sans-serif";color:#333399'>Thanks,<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:7.5pt;font-family:"Verdana","sans-serif";color:#333399'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:7.5pt;font-family:"Verdana","sans-serif";color:#333399'>Moishe Grunstein<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:7.5pt;font-family:"Verdana","sans-serif";color:#333399'>Tornado Computer Systems, Inc.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:7.5pt;font-family:"Verdana","sans-serif";color:#333399'>212.400.7650 888.IPPBX.US<br></span><b><span style='font-size:7.5pt;font-family:"Verdana","sans-serif";color:#76923C'>Service Request Email: <a href="mailto:support@nysolutions.com"><span style='color:blue'>support@nysolutions.com</span></a> </span></b><span style='font-size:10.0pt;font-family:"Arial","sans-serif";color:#333399'><o:p></o:p></span></p><p class=MsoNormal><span style='font-size:7.5pt;font-family:"Verdana","sans-serif";color:#333399'>Polycom Certified VAR<br>Microsoft Small Business Specialist, Cisco SMB Select Certified</span><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#333399'><o:p></o:p></span></p><p class=MsoNormal><a href="http://www.nysolutions.com/"><span style='font-size:10.0pt;font-family:"Arial","sans-serif";color:#1F497D;text-decoration:none'><img border=0 width=147 height=68 id="Picture_x0020_1" src="cid:image001.jpg@01CEC080.7F4FB4C0" alt="cid:image001.jpg@01C72F94.9EE45D60"></span></a><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p></o:p></span></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-size:8.0pt;font-family:"Calibri","sans-serif";color:#C0504D'>Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS</span><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] <b>On Behalf Of </b>Iwan Budi Kusnanto<br><b>Sent:</b> Thursday, October 03, 2013 9:24 PM<br><b>To:</b> FreeSWITCH Users Help<br><b>Subject:</b> Re: [Freeswitch-users] FreeSwitch + WebRTC + JsSIP + Chrome no audio<o:p></o:p></span></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal><br><br>On Friday, October 4, 2013, Rafael Santana wrote:<o:p></o:p></p><div><p class=MsoNormal>Hi,<o:p></o:p></p><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>I'm new to telephony and FreeSwitch's world, so I apologize in advance for any nonsense I speak here. <o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>I've been trying to setup an environment where It can be possible to make a call through Google Chrome Browser using JsSIP to a standard phone device on PSTN. <o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>In my network my "PSTN gateway" is an Asterisk 1.4 instance (No, I can't chance it today). To communicate with Chrome I have a FreeSwitch 1.5.5 instance and to get access to PSTN via this instance I had to register my Asterisk instance as a gateway on my Sofia's external profile. This part of my scenario works fine. I'm able to make calls using a softphone registered on FreeSwitch to standard phones on PSTN with no problems. What I wasn't able to do until now was the JsSIP + FreeSwitch integration. <o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>To setup FreeSwitch to comunicate with JsSIP, the only thing I did was uncomment the line below on sip_profiles/internal.xml.<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal><param name="ws-binding" value=":5066"/><o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>I really don't know if just this is sufficient. Am I missing something important?<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>To connect on my FreeSwitch instance from Chrome, I'm using the Tryit JsSIP demo. Today, I'm able to register on FS from Tryit demo and perform a call to a PSTN phone. The connection is established but I don't get any audio in both endpoints. The same happens when I try to call the 5000 ivr extension or an user on a softphone at the same network from my Chrome browser.<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>Assuming that all the services I've mentioned here are running on the same network, do you have any idea why I can't get audio in both endpoints of my experiment?<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>Additional information:<o:p></o:p></p></div><div><p class=MsoNormal>Ubuntu 12.04 64 bits<o:p></o:p></p></div><div><div><p class=MsoNormal>FreeSwitch version 1.5.5 default install configuration<o:p></o:p></p></div><div><p class=MsoNormal>Tryit JsSIP Demo with jssip-0.3.0.js<o:p></o:p></p></div></div></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>Try to add --with-openssl when doing ./configure<o:p></o:p></p></div><blockquote style='border:none;border-left:solid #CCCCCC 1.0pt;padding:0in 0in 0in 6.0pt;margin-left:4.8pt;margin-right:0in'><div><div><div><p class=MsoNormal><o:p> </o:p></p></div><p class=MsoNormal>Thanks in advance,<o:p></o:p></p></div><div><p class=MsoNormal>Rafael.<o:p></o:p></p></div></div></blockquote><p class=MsoNormal><br><br>-- <br>Iwan Budi Kusnanto<o:p></o:p></p></div></body></html>