- FreeSWITCH-users - FreeSWITCH Mailing Lists

[SOLVED] Re: Chrome/VC without STUN sends 0.0.0.0 and gets INCOMPATIBLE_DESTINATION
by Richard Chan 14 Sep '17

14 Sep '17

Sorry for the noise — it was the Privacy Badger extension in Chrome hiding all the host candidates. https://github.com/EFForg/privacybadger/issues/1099 It seems to be quite an issue with the way it sets chrome.privacy.network.webRTCIPHandlingPolicy Richard

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Chrome/VC without STUN sends 0.0.0.0 and gets INCOMPATIBLE_DESTINATION
by Richard Chan 14 Sep '17

14 Sep '17

Hi, I'd like to get your opinion on whether this WebRTC failure to connect points to a Chrome/VC or FS bug. FreeSWTICH: 1.6.19 Verto Communicator: from v1.8 branch Chrome: 59-61 Reproducer: 0. Chrome and FS on a local LAN 1. In VC, disable STUN as environment is local LAN 2. Chrome sends 2017-09-14 11:48:33.658249 [DEBUG] mod_rtc.c:389 () State Change CS_NEW -> CS_INIT 2017-09-14 11:48:33.658249 [DEBUG] switch_core_session.c:615 N/A set UUID=b138714b-740a-de23-101c-c76d5676a6d4 2017-09-14 11:48:33.658249 [NOTICE] switch_channel.c:1104 New Channel verto.rtc/3000 [b138714b-740a-de23-101c-c76d5676a6d4] 2017-09-14 11:48:33.658249 [DEBUG] mod_verto.c:3661 Remote SDP verto.rtc/3000: v=0 o=- 3881351537341052870 2 IN IP4 127.0.0.1 s=- t=0 0 a=group:BUNDLE audio video a=msid-semantic: WMS Tub3STBaDw6tNMrINhAyvJBWK96IeIanerjj m=audio 9 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126 c=IN IP4 0.0.0.0 a=rtcp:9 IN IP4 0.0.0.0 a=ice-ufrag:VU1P a=ice-pwd:davKC7PLYeQ04ZkN4TNvGo9s a=ice-options:trickle a=fingerprint:sha-256 79:A5:79:AE:62:B9:7F:00:5F:0C:04:C1:18:81:C4:09:ED:47:3B:1B:0A:3C:2F:F2:7B:6F:D5:E7:3C:DA:E0:2C a=setup:actpass a=mid:audio a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=sendrecv a=rtcp-mux 3. FS replies [DEBUG] switch_core_media.c:3523 Searching for rtp candidate. [DEBUG] switch_core_media.c:3523 Searching for rtcp candidate. [DEBUG] switch_core_media.c:3567 verto.rtc/3000 no suitable candidates found. [NOTICE] switch_channel.c:3812 Hangup verto.rtc/3000 [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] [DEBUG] switch_core_session.c:2815 verto.rtc/3000 skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 4. With STUN on, it connects successfully. Chrome sends the external NAT address (STUN discovered). This seems unnecessary for a local LAN. 5. For Firefox+VC without STUN, the verto.invite contains all the local LAN addresses and it connect successfully. Any suggestions? Should Chrome+VC without be expected to work w/o STUN on a local LAN while sending c=IN IP4 0.0.0.0? -- Richard Chan

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OSLEC on FreeTDM on Sangoma A102
by Nandy Dagondon 14 Sep '17

14 Sep '17

Hello folks, I encountered echo problem with FreeTDM Wanpipe R2 with Sangoma A102 in an insert-and-drop setup. Anyone tried OSLEC (DAHDI) ? Is it working? Tks, /Nandy

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Tracking down calls with spoofed Caller ID
by Chad Phillips 13 Sep '17

13 Sep '17

This question isn’t directly related to FreeSWITCH, and, I’m curious what others in this community have learned about dealing with incoming ‘spam’ phone calls… For many months now I’ve been getting travel-related spam phone calls to my personal cell. I figured the caller ID was spoofed, so would just hang up, but a few days ago I decided to call one of the numbers back to see what would happen. A woman answered, and I asked her to remove me from their call list, to which she angrily responded “no, you are the one who has been calling me, and I’ve contacted the police!” It was then that I realized the spammers were using caller ID numbers of other people they were spamming…as if this industry could get any slimier. I would love to bust these jerks, but I’m not sure it’s even possible to trace the origins of these calls. It seems futile to block incoming calls based on caller ID, because they change all the time. How do the telephony bad-asses in this community handle such shenanigans? Chad

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Re: [Freeswitch-users] PortAudio and Windows devices
by Clive Lansink 13 Sep '17

13 Sep '17

Thanks. That's great news. Cheers. Clive Lansink Email: Clive(a)Lansink.Co.NZ Phone: +64 9 520-4242 Mobile: +64 21 663-999 Fax: +64 21 789-150 -----Original message----- From: Michael Jerris <mike(a)jerris.com> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Subject: Re: [Freeswitch-users] PortAudio and Windows devices Reply-to: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Date: Mon, 11 Sep 2017 15:35:45 -0400 https://freeswitch.org/jira/browse/FS-10663 <https://freeswitch.org/jira/browse/FS-10663> fixed in master now. Thanks Andrey! > On Sep 9, 2017, at 9:04 PM, Clive Lansink <clive(a)lansink.co.nz> wrote: > > Hi again everyone. > > OK I appreciate that nobody has yet answered this but now I have more info that may help. > > I succeeded in building the Freeswitch executable for Windows from the source. I git cloned version 1.6.19 which I understand is the most recent stable release. I am using Visual Studio 2015 Express. > > I first built the debug version 64-bit. There were three errors but at least there was an executable so I ran that. Now the pa devlist command lists my sound devices so that works. Also the debug information indicates that it is using DirectSound. I actually haven't made a call because of another issue but at least I can confirm that mod_portaudio sees my audio devices in this situation. > > Then I cleaned the solution and built the release version. Again I got three errors but again there was an executable so I ran that. Now the pa devlist command produces no devices. > > This suggests to me that something is amiss with the process for generating the release version for Windows which also makes the Windows installer, that impacts on PortAudio. In that case, the result In my experience is that PortAudio just doesn't work because it doesn't see any audio devices. > > It's beyond me I'm afraid to figure out the difference here. I'm just not experienced with Visual Studio to understand what is going on with these two scenarios. > > Hope this helps. > > > Clive Lansink > Email: Clive(a)Lansink.Co.NZ > Phone: +64 9 520-4242 > Mobile: +64 21 663-999 > Fax: +64 21 789-150 > > -----Original message----- > From: Clive Lansink <clive(a)lansink.co.nz> > To: "Freeswitch users list" <freeswitch-users(a)lists.freeswitch.org> > Subject: [Freeswitch-users] PortAudio and Windows devices > Reply-to: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Date: Fri, 8 Sep 2017 10:03:32 +1200 > > > Hi everyone. Can someone please help with the following problem? I have spent several hours on this today, just googling around, and although I think I am not the only one with this problem, I haven't actually found the solution. > > Problem is PortAudio reports there are no input and output devices. I am running Freeswitch on Windows 10 64-bit. The Freeswitch installer is FreeSWITCH-1.6.18-x64-Release.msi which I downloaded recently from files.freeswitch.org. I should explain that I have been using Freeswitch for a number of years and PortAudio used to work, even on this Windows 10 PC. The problem seems to be with the more recent versions of Freeswitch. > > We know that Freeswitch for Windows includes PortAudio. That's good because I don't know how to build Freeswitch for Windows. I just know how to install and use it. > > I think I read somewhere that the Freeswitch PortAudio is built for DirectEx. I came across the Dxdiag tool which tells me about DirectEx devices. I have several input and output devices on this machine, but Dxdiag only lists one output device and it lists no input devices. It tells me I have DirectEx version 12. > > I do a lot of programming in Python and that includes a Python module that provides bindings for PortAudio. That code still works. > > I read that PortAudio by default is built for MME, or maybe it is PyAudio that is built for MME. Whatever the case, that code works. > > But if Freeswitch's PortAudio module is built for DirectEx, could there be a problem with DirectEx on Windows 10? I just get the impression that Microsoft is favouring newer Windows sound APIs. Issuing the pa devlist command just lists nothing so it is as if it is just not seeing any devices, but they are certainly there. > > I think this has been a problem for several years because I have Freeswitch version 1.5.15 on another Windows 10 64-bit PC and PortAudio on that PC has the same problem. I just haven't had to use it on that PC. > > By the way, I did try running Freeswitch in console mode with administrator privileges and it made no difference. > > I hope someone can suggest something to help me fix this problem. > > > Clive Lansink > Email: Clive(a)Lansink.Co.NZ > Phone: +64 9 520-4242 > Mobile: +64 21 663-999 > Fax: +64 21 789-150 > _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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log libsofia
by Sergey Safarov 13 Sep '17

13 Sep '17

Is possible write to log file info produced by by libsofia tport? I can see this info using "sofia loglevel tport 9", but how to write this info to file?

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Implement PushKit or CallKit to call iOS app closed
by Jose David Jurado Alonso 13 Sep '17

13 Sep '17

Hi, I can't find any information over how to implement PushKit or Callkit to call an iOS mobile phone that has the SIP client application closed or in the background. When the application is closed it is obviously that SIP account isn't registered and the FS server no send the call. Has anyone worked with this or know how to implement it? I tried with "<variable name="sip-force-contact" value="NDLB-connectile-dysfunction"/>" option and this temporally solved when the application is alive in background. Thanks,

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Fwd: Verto
by Rick Jarvis 13 Sep '17

13 Sep '17

Resending with link to image (sorry I know images aren’t ideal for mailing lists, but the format goes if I copy&paste)... From: Rick Jarvis <rick(a)magicmail.mooo.com <mailto:rick@magicmail.mooo.com>> Subject: Re: [Freeswitch-users] Verto Date: 12 September 2017 at 13:53:10 BST To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org>> Definitely making progress... Can anyone tell me what’s going on here, this is when I try and make a call (it’s an internal call, wondered if it’s hitting the wrong context or something, but the only logging I’m getting is from mod_verto, so I don’t think it’s getting that far): Link to Chrome console output: https://pasteboard.co/GK2exjV.png <https://pasteboard.co/GK2exjV.png> Verto console output: "jsonrpc": "2.0", "id": 5, "error": { "code": -32601, "message": "Invalid Method, Missing Method or Permission Denied" }

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Anyone has experience of using httapi record?
by Lin, Xiaoping 13 Sep '17

13 Sep '17

HI! Anyone has experience with using freeswitch record? I am using it in my web app. It seems not working and I did not get any error message in the log. It did not generate the recording file and I do not know if anything wrong. Thanks! * Xiaoping I am using freeswitch-application-httapi-1.6.8-1.el7.centos.x86_64 The document the webapp sends over to freeswitch looks like: On the console: 2017-09-12 17:22:01.000062 [DEBUG] switch_channel.c:3770 (sofia/external/5022089400(a)10.2.11.207) Callstate Change EARLY -> ACTIVE EXECUTE sofia/external/5022089400(a)10.2.11.207 set_audio_level(read 2) 2017-09-12 17:22:01.000062 [DEBUG] switch_core_media_bug.c:828 Attaching BUG to sofia/external/5022089400(a)10.2.11.207 EXECUTE sofia/external/5022089400(a)10.2.11.207 httapi({method=POST,url=http://10.2.11.207:8690/VoiceApp/servlet/VoiceAppServlet?Context=ASRTestApp&Debug=true}) 2017-09-12 17:22:01.140851 [CRIT] mod_httapi.c:1153 Debugging Return Data: <document type="text/freeswitch-httapi"> <work> <playback file="http://10.2.11.207:8690//CA/servlet/CA/ASRTestApp/Default/en_US/AgainNumNot…" action="http://10.2.11.207:8690/VoiceApp/servlet/VoiceAppServlet;jsessionid=F47B812…" /> <playback file="http://10.2.11.207:8690//CA/servlet/CA/ASRTestApp/Default/en_US/WhatBusines…" action="http://10.2.11.207:8690/VoiceApp/servlet/VoiceAppServlet;jsessionid=F47B812…" /> <execute application="flush_dtmf"/> <record action="http://10.2.11.207:8690/VoiceApp/servlet/VoiceAppServlet;jsessionid=F47B812…" name="playback-user-input" file="/var/lib/freeswitch/storage/temp/recording_0.wav" limit="30" terminators="" digit-timeout="10000"> </record> </work> </document> 2017-09-12 17:22:01.140851 [DEBUG] mod_httapi.c:1222 Process Tag: [playback] 2017-09-12 17:22:01.140851 [DEBUG] sofia.c:6858 Channel sofia/external/5022089400(a)10.2.11.207 entering state [completed][200] 2017-09-12 17:22:01.140851 [DEBUG] switch_ivr_play_say.c:1467 Codec Activated L16@8000hz 1 channels 20ms 2017-09-12 17:22:01.168443 [DEBUG] switch_rtp.c:6707 Correct audio ip/port confirmed. 2017-09-12 17:22:01.168443 [DEBUG] switch_core_io.c:448 Setting BUG Codec PCMU:0 2017-09-12 17:22:01.220055 [DEBUG] sofia.c:6858 Channel sofia/external/5022089400(a)10.2.11.207 entering state [ready][200] 2017-09-12 17:22:04.700049 [DEBUG] switch_ivr_play_say.c:1910 done playing file http://10.2.11.207:8690//CA/servlet/CA/ASRTestApp/Default/en_US/AgainNumNot… 2017-09-12 17:22:04.700049 [DEBUG] mod_httapi.c:1222 Process Tag: [playback] 2017-09-12 17:22:04.700049 [DEBUG] switch_ivr_play_say.c:1467 Codec Activated L16@8000hz 1 channels 20ms 2017-09-12 17:22:07.660037 [DEBUG] switch_ivr_play_say.c:1910 done playing file http://10.2.11.207:8690//CA/servlet/CA/ASRTestApp/Default/en_US/WhatBusines… 2017-09-12 17:22:07.660037 [DEBUG] mod_httapi.c:1222 Process Tag: [execute] EXECUTE sofia/external/5022089400(a)10.2.11.207 flush_dtmf() 2017-09-12 17:22:07.660037 [DEBUG] mod_httapi.c:1222 Process Tag: [record] 2017-09-12 17:22:07.660037 [DEBUG] switch_ivr_play_say.c:673 Raw Codec Activated 2017-09-12 17:22:07.660037 [DEBUG] switch_core_codec.c:221 sofia/external/5022089400(a)10.2.11.207 Push codec L16:100 2017-09-12 17:22:13.480059 [DEBUG] switch_core_codec.c:246 sofia/external/5022089400(a)10.2.11.207 Restore previous codec PCMU:0. 2017-09-12 17:22:13.480059 [CRIT] mod_httapi.c:1153 Debugging Return Data: <document type="text/freeswitch-httapi"> <work> <hangup action="http://10.2.11.207:8690/VoiceApp/servlet/VoiceAppServlet;jsessionid=F47B812…" cause="Normal" /> </work> </document> 2017-09-12 17:22:13.480059 [DEBUG] mod_httapi.c:1222 Process Tag: [hangup] 2017-09-12 17:22:13.480059 [NOTICE] mod_httapi.c:880 Hangup sofia/external/5022089400(a)10.2.11.207 [CS_EXECUTE] [NORMAL_CLEARING]

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how to get event of failed call
by Sergey Safarov 13 Sep '17

13 Sep '17

FS originates call to carrier. FS receives "200 OK" from carrier and send ACK but for some reasons ACK not received by carrier. Then carrier retransmits "200 OK" messages and all retransmits also will failed. Carrier send BYE message after timeout as call not established correctly. Is possible to get event of failed call on FS side. As example - we got several "200 OK" messages and "BYE" message is received after 10 sec of last "20 OK"?

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