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<TITLE>Re: [Freeswitch-users] FreeSWITCH adds WebRTC support to new 1.4 BETA.</TITLE>
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<FONT FACE="Monaco, Courier New"><SPAN STYLE='font-size:11pt'>Sounds like you need to enable INFO dtmf or set it for liberal dtmf<BR>
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On 7/9/13 6:01 PM, "Henry Huang" <<a href="red.rain.seven@gmail.com">red.rain.seven@gmail.com</a>> wrote:<BR>
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</SPAN></FONT><BLOCKQUOTE><FONT SIZE="2"><FONT FACE="Verdana, Helvetica, Arial"><SPAN STYLE='font-size:10pt'>Anthony, <BR>
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May I ask what version of FreeSWITCH is running on webrtc.freeswitch.org <<a href="http://webrtc.freeswitch.org">http://webrtc.freeswitch.org</a>> ? I am getting different result when pressing dtmf from webrtc client over my FreeSWITCH (Version 1.5.3b git e0210d7 2013-07-03 20:01:03Z) with default sip_profile configuration.<BR>
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</SPAN></FONT></FONT><FONT FACE="Verdana, Helvetica, Arial"><FONT COLOR="#FF00FF"><SPAN STYLE='font-size:11pt'>2013-07-09 22:58:57.724523 [WARNING] sofia.c:7598 IGNORE INFO DTMF(1) (This channel was not configured to use INFO DTMF!)<BR>
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Thanks,<BR>
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Henry<BR>
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On Wed, Jul 3, 2013 at 9:48 PM, Anthony Minessale <<a href="anthony.minessale@gmail.com">anthony.minessale@gmail.com</a>> wrote:<BR>
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The liberal dtmf option will allow all types of dtmf.<BR>
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On Jul 3, 2013 10:03 PM, "Henry Huang" <<a href="red.rain.seven@gmail.com">red.rain.seven@gmail.com</a>> wrote:<BR>
</FONT></SPAN><BLOCKQUOTE><FONT SIZE="2"><FONT FACE="Verdana, Helvetica, Arial"><SPAN STYLE='font-size:10pt'>I don't know what it is, but I am getting caught in more issues. I really want to say thank you and it seems like issue on my end since your demo site is working fine. <BR>
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I don't think the vanilla config is using dtmf-type "info" by default. I noticed that because my dtmf info start to break. I have from the beginning used dtmf-type info in my internal sip profile and it has been working since April when I dial into 5000 IVR with JsSIP. But since I upgraded to the latest Git last week, I am getting log message saying my channel doesn't support dtmf info, therefore I can't try anything after dialing into the 5000 IVR. <BR>
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I am on the verge to reinstall everything and start over. But before I do that, is there anything I am not doing correctly here? Is it Ubuntu? and why would the dtmf info stops working?<BR>
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Thanks,<BR>
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Henry<BR>
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On Wed, Jul 3, 2013 at 7:04 PM, Anthony Minessale <<a href="anthony.minessale@gmail.com">anthony.minessale@gmail.com</a>> wrote:<BR>
</SPAN></FONT><BLOCKQUOTE><FONT FACE="Monaco, Courier New"><SPAN STYLE='font-size:11pt'>Its the defaults from the vanilla config with the webrtc opened up.<BR>
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On Wed, Jul 3, 2013 at 6:09 PM, Henry Huang <<a href="red.rain.seven@gmail.com">red.rain.seven@gmail.com</a>> wrote:<BR>
</SPAN></FONT><BLOCKQUOTE><FONT SIZE="2"><FONT FACE="Verdana, Helvetica, Arial"><SPAN STYLE='font-size:10pt'>Thanks, the demo site does work. Is it possible that you can share the sip_profile configuration for the demo site here?<BR>
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On Wed, Jul 3, 2013 at 2:15 PM, Anthony Minessale <<a href="anthony.minessale@gmail.com">anthony.minessale@gmail.com</a>> wrote:<BR>
</SPAN></FONT><BLOCKQUOTE><FONT FACE="Monaco, Courier New"><SPAN STYLE='font-size:11pt'>you should use ext up in 9xxx for chrome too, those lower ext spaces are reserved for other stuff.<BR>
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On Wed, Jul 3, 2013 at 3:45 PM, Henry Huang <<a href="red.rain.seven@gmail.com">red.rain.seven@gmail.com</a>> wrote:<BR>
</SPAN></FONT><BLOCKQUOTE><FONT SIZE="2"><FONT FACE="Verdana, Helvetica, Arial"><SPAN STYLE='font-size:10pt'>Anthony,<BR>
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Thanks, I am able to register a regular SIP to extension 9001 on webrtc.freeswitch.org <<a href="http://webrtc.freeswitch.org">http://webrtc.freeswitch.org</a>> and I have a user: 4000 on Chrome. When I dial from 9001 to 4000, I get a voice prompt asking for "please enter your ID followed by #" and I don't know how to get pass that.<BR>
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Thanks,<BR>
<BR>
Henry<BR>
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On Wed, Jul 3, 2013 at 10:02 AM, Anthony Minessale <<a href="anthony.minessale@gmail.com">anthony.minessale@gmail.com</a>> wrote:<BR>
</SPAN></FONT></BLOCKQUOTE></BLOCKQUOTE></BLOCKQUOTE></BLOCKQUOTE></BLOCKQUOTE></BLOCKQUOTE></BLOCKQUOTE><FONT FACE="Monaco, Courier New"><SPAN STYLE='font-size:11pt'>You can register normal sip clients to that demo too, it has blind reg enabled.<BR>
Just pick some unique ext numbers up in the 9xxx range <BR>
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On Wed, Jul 3, 2013 at 11:45 AM, Henry Huang <<a href="red.rain.seven@gmail.com">red.rain.seven@gmail.com</a>> wrote:<BR>
</SPAN></FONT><FONT SIZE="2"><FONT FACE="Verdana, Helvetica, Arial"><SPAN STYLE='font-size:10pt'>Michael,<BR>
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I understand that webrtc.freeswitch.org <<a href="http://webrtc.freeswitch.org">http://webrtc.freeswitch.org</a>> works and so is my test server. The one thing I can't verify from webrtc.freeswitch.org <<a href="http://webrtc.freeswitch.org">http://webrtc.freeswitch.org</a>> is calling from regular SIP client back to webrtc client, and this happened to be the only thing I am having issue with. Everything you see and do on webrtc.freeswitch.org <<a href="http://webrtc.freeswitch.org">http://webrtc.freeswitch.org</a>> works fine on my test environment. <BR>
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So again, is there anyone who can call from pstn/sip to webrtc without problems? <BR>
<FONT COLOR="#888888"><BR>
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Henry<BR>
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On Wed, Jul 3, 2013 at 12:58 AM, Michael Jerris <<a href="mike@jerris.com">mike@jerris.com</a>> wrote:<BR>
yes. you can see it working on the demo at <a href="http://webrtc.freeswitch.org">http://webrtc.freeswitch.org</a>.<BR>
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On Jul 3, 2013, at 12:53 AM, Henry Huang <<a href="red.rain.seven@gmail.com">red.rain.seven@gmail.com</a>> wrote:<BR>
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Anyone getting audio working bridging call to webrtc client without bypassing media?<BR>
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On Jul 2, 2013 6:45 PM, "João Mesquita" <<a href="jmesquita@freeswitch.org">jmesquita@freeswitch.org</a>> wrote:<BR>
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I think you might need to re bootstrap maybe?<BR>
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On Jul 2, 2013 8:54 PM, "Henry Huang" <<a href="red.rain.seven@gmail.com">red.rain.seven@gmail.com</a>> wrote:<BR>
</SPAN></FONT><FONT SIZE="2"><FONT FACE="Verdana, Helvetica, Arial"><SPAN STYLE='font-size:10pt'>I did, and just in case I ran it again. <BR>
./configure<BR>
make<BR>
make install<BR>
restart freeswitch<BR>
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This is what I am still seeing. <BR>
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</SPAN></FONT></FONT><FONT FACE="Verdana, Helvetica, Arial"><FONT COLOR="#333333"><SPAN STYLE='font-size:11pt'>2013-07-02 16:40:51.135302 [ALERT] switch_core_media.c:1310 Dispatched RTCP event<BR>
</SPAN></FONT><SPAN STYLE='font-size:11pt'><FONT COLOR="#FF0000">2013-07-02 16:40:51.175310 [ALERT] switch_rtp.c:3883 Drop audio packet 70 bytes (dtls not ready!) b=4294967168<BR>
</FONT><FONT COLOR="#333333">2013-07-02 16:40:52.255306 [ALERT] switch_rtp.c:1510 Setting RTCP src-1 to 10.128.7.103<BR>
2013-07-02 16:40:52.255306 [ALERT] switch_rtp.c:1515 Setting RTCP src-1 LENGTH to 12 (1280, 10.128.7.103)<BR>
2013-07-02 16:40:52.255306 [ALERT] switch_rtp.c:1516 Setting msw = -1267375797, lsw = -601115572<BR>
2013-07-02 16:40:52.255306 [ALERT] switch_rtp.c:1517 now = 0, now lo = 0, now hi = 0<BR>
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Henry<BR>
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On Tue, Jul 2, 2013 at 3:13 PM, Ken Rice <<a href="krice@freeswitch.org">krice@freeswitch.org</a>> wrote:<BR>
Did you reconfigure? <BR>
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On 7/2/13 4:29 PM, "Henry Huang" <<a href="red.rain.seven@gmail.com">red.rain.seven@gmail.com</a> <<a href="http://red.rain.seven@gmail.com/">http://red.rain.seven@gmail.com/</a>> > wrote:<BR>
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</FONT></SPAN><FONT SIZE="2"><FONT FACE="Verdana, Helvetica, Arial"><SPAN STYLE='font-size:10pt'>Ok, I have upgraded openssl to 1.0.1e, recompiled FreeSWITCH, but I am still getting the "dtls not ready!" alerts and no audio<BR>
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</SPAN></FONT></FONT><FONT FACE="Verdana, Helvetica, Arial"><FONT COLOR="#FF0000"><SPAN STYLE='font-size:11pt'>2013-07-02 14:26:47.695312 [ALERT] switch_rtp.c:4408 <a href="sofia/internal/1004@10.128.7.103">sofia/internal/1004@10.128.7.103</a> <<a href="http://sofia/internal/1004@10.128.7.103">http://sofia/internal/1004@10.128.7.103</a>> Hot Hit 2<BR>
2013-07-02 14:26:47.695312 [ALERT] switch_rtp.c:4425 <a href="sofia/internal/1004@10.128.7.103">sofia/internal/1004@10.128.7.103</a> <<a href="http://sofia/internal/1004@10.128.7.103">http://sofia/internal/1004@10.128.7.103</a>> timer while HOT<BR>
2013-07-02 14:26:47.715313 [ALERT] switch_rtp.c:5443 Skip sending audio packet 172 bytes (dtls not ready!)<BR>
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</SPAN></FONT></FONT><SPAN STYLE='font-size:11pt'><FONT FACE="Monaco, Courier New">Here is my openssl version:<BR>
root@fusionpbx:/usr/src# openssl version<BR>
OpenSSL 1.0.1e 11 Feb 2013<BR>
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On Tue, Jul 2, 2013 at 9:35 AM, Michael Collins <<a href="msc@freeswitch.org">msc@freeswitch.org</a> <<a href="http://msc@freeswitch.org/">http://msc@freeswitch.org/</a>> > wrote:<BR>
You'll need to make sure you have OpenSSL 1.0.1e which was released in Feb of this year. I suspect the version in Ubuntu might not be older.<BR>
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-MC<BR>
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On Tue, Jul 2, 2013 at 9:02 AM, Henry Huang <<a href="red.rain.seven@gmail.com">red.rain.seven@gmail.com</a> <<a href="http://red.rain.seven@gmail.com/">http://red.rain.seven@gmail.com/</a>> > wrote:<BR>
</FONT></SPAN><FONT SIZE="2"><FONT FACE="Verdana, Helvetica, Arial"><SPAN STYLE='font-size:10pt'>Michael, <BR>
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I am just using whatever comes with Ubuntu 12.04 64bit. <BR>
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Package: openssl<BR>
Version: 1.0.1-4ubuntu5.8<BR>
Depends: libc6 (>= 2.15), libssl1.0.0 (>= 1.0.1)<BR>
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</FONT></SPAN><FONT SIZE="2"><FONT FACE="Verdana, Helvetica, Arial"><SPAN STYLE='font-size:10pt'>Thanks,<BR>
<BR>
Henry<BR>
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