October 2020 - FreeSWITCH-users - FreeSWITCH Mailing Lists

mod_conference: recommended values for comfort noise and energy level?
by Frederic Jean 21 Oct '20

21 Oct '20

Hello group, We have setup a new conference service and it works well, except for some complaints here and there about the voice quality; I was then wondering what are the recommended values for comfort noise and energy level as per your experience for conferences? My feeling is that comfort noise should be disabled but I am unsure about the energy level; is this a try and see parameter, or should it ideally simply be set to 0 or a small value for a conference call system? Presently: <profile name="confmodreception"> <param name="domain" value="$${domain}"/> <param name="rate" value="16000"/> <param name="ivr-dtmf-timeout" value="2000"/> <param name="interval" value="20"/> <param name="caller-controls" value="reception"/> <param name="moderator-controls" value="reception"/> <param name="energy-level" value="200"/> <param name="alone-sound" value="conference/conf-alone.wav"/>  <param name="comfort-noise" value="true"/> <param name="suppress-events" value="start-talking,stop-talking,energy-level,volume-level,gain-level,energy-level-member,play-file,floor-change,play-file-done,play-file-member"/> <param name="enter-sound" value="tone_stream://%(200,0,500,600,700)"/> <param name="exit-sound" value="tone_stream://%(500,0,300,200,100,50,25)"/> <param name="muted-sound" value="/usr/local/freeswitch/sounds/fr/ca/june/conference/16000/conf-muted.wav"/> <param name="unmuted-sound" value="/usr/local/freeswitch/sounds/fr/ca/june/conference/16000/conf-unmuted.wav"/> <param name="terminate-on-silence" value="120"/> <param name="locked-sound" value="/usr/local/freeswitch/sounds/fr/ca/june/conference/16000/conf-locked.wav"/> <param name="is-locked-sound" value="/usr/local/freeswitch/sounds/fr/ca/june/conference/16000/conf-is-locked.wav"/> <param name="is-unlocked-sound" value="/usr/local/freeswitch/sounds/fr/ca/june/conference/16000/conf-is-unlocked.wav"/> </profile>

1 0

exired lettencript cert
by Sergey Safarov 20 Oct '20

20 Oct '20

some of the "files.freeswitch.org" servers have expired certificates. Please update the certificate on all servers. Details [safarov@safarov-dell esrp-packer]$ openssl s_client -showcerts -connect files.freeswitch.org:443 CONNECTED(00000003) depth=2 O = Digital Signature Trust Co., CN = DST Root CA X3 verify return:1 depth=1 C = US, O = Let's Encrypt, CN = Let's Encrypt Authority X3 verify return:1 depth=0 CN = files.freeswitch.org verify error:num=10:certificate has expired notAfter=Oct 17 04:58:08 2020 GMT verify return:1 depth=0 CN = files.freeswitch.org notAfter=Oct 17 04:58:08 2020 GMT verify return:1 --- Certificate chain 0 s:CN = files.freeswitch.org i:C = US, O = Let's Encrypt, CN = Let's Encrypt Authority X3 -----BEGIN CERTIFICATE----- MIIFejCCBGKgAwIBAgISA6WIL9TQrycry+w8B8l5HdFkMA0GCSqGSIb3DQEBCwUA MEoxCzAJBgNVBAYTAlVTMRYwFAYDVQQKEw1MZXQncyBFbmNyeXB0MSMwIQYDVQQD ExpMZXQncyBFbmNyeXB0IEF1dGhvcml0eSBYMzAeFw0yMDA3MTkwNDU4MDhaFw0y MDEwMTcwNDU4MDhaMB8xHTAbBgNVBAMTFGZpbGVzLmZyZWVzd2l0Y2gub3JnMIIB IjANBgkqhkiG9w0BAQEFAAOCAQ8AMIIBCgKCAQEA2kvIqeeTRraRcWEWGcS+Ozzw DEpR68LbqVDsAu4GUmkyTzCrNZGQcA1vl+iyRLlA6aVp2HDPaG35iSQUSoTWD/w+ kdEUnIlz1EWMmkbBGqy+HNo6R9atlScIABZRUzY6KWDkuaxxk84Fuu+ZsfXIeJUl CEURSOjKxpKDiAOIZiCQYj43km14a0JBObIO2hHqdYw0QHn+sIvWVa7kDGG5WqyQ WSNUrxKQE3D5SufucKrOEx0Ktgs9NnboMgRjn+9bJefaT/vLPXqTvnKy3kNBwAZo E5Na73es8FDtTiDIrGAkWZQNqrD5DHM08ERYHwBzqqtsgcGn5mdBqfvOUSJH3wID AQABo4ICgzCCAn8wDgYDVR0PAQH/BAQDAgWgMB0GA1UdJQQWMBQGCCsGAQUFBwMB BggrBgEFBQcDAjAMBgNVHRMBAf8EAjAAMB0GA1UdDgQWBBTWzYsuNS1t+dCXV+s7 xpqtK8k/OTAfBgNVHSMEGDAWgBSoSmpjBH3duubRObemRWXv86jsoTBvBggrBgEF BQcBAQRjMGEwLgYIKwYBBQUHMAGGImh0dHA6Ly9vY3NwLmludC14My5sZXRzZW5j cnlwdC5vcmcwLwYIKwYBBQUHMAKGI2h0dHA6Ly9jZXJ0LmludC14My5sZXRzZW5j cnlwdC5vcmcvMDoGA1UdEQQzMDGCGWZpbGVzLXN5bmMuZnJlZXN3aXRjaC5vcmeC FGZpbGVzLmZyZWVzd2l0Y2gub3JnMEwGA1UdIARFMEMwCAYGZ4EMAQIBMDcGCysG AQQBgt8TAQEBMCgwJgYIKwYBBQUHAgEWGmh0dHA6Ly9jcHMubGV0c2VuY3J5cHQu b3JnMIIBAwYKKwYBBAHWeQIEAgSB9ASB8QDvAHYAsh4FzIuizYogTodm+Su5iiUg Z2va+nDnsklTLe+LkF4AAAFzZaatSwAABAMARzBFAiEAhSZUY/ZugqtLtgSnR8Q6 2d4H77VHpOp8P3mLsf1pctECIFFh0zMgkbRrC0Vt2DVaprFHs1gWwAk1IbnRJuct XTRhAHUAb1N2rDHwMRnYmQCkURX/dxUcEdkCwQApBo2yCJo32RMAAAFzZaatdwAA BAMARjBEAiATkNcbjpzgXBhIIMg8F5SuR99wBY9C9mPkoMFVVGnH1QIgEuau5oTS UfIlnngJHyZdt9DIyMDywUbejQVjFtvuuewwDQYJKoZIhvcNAQELBQADggEBAIAN sFwzLeERpXDtzbID2UJ00BvopM8skXSRN1tQ5wu07IAfNdghLnT+DxuS81A5KUPt nkFscKwv1++EnX6+iueZlEnA6er1EeFySbCCrk3024RN8UoAwaCaqRfLA/gdxhPo 8e+IJAoL5W5lakXwSPPQhmYch1yk7SJKiWd/svRmK67BK7WM1qy0nFxxBTXscJU1 Y05rApNIdVm+4E1y8AyGWfiZUuFumitkHjgd46nre+MwrMpF0bIykAjbn3CyzgI5 8o+XhbTMTudRORPD3/aK2NtERzZnE24pWuSraDeVtSlkD7+lfBHQvcXrwteTWcpF /uSqRuA4JSv+SiLqLJU= -----END CERTIFICATE----- 1 s:C = US, O = Let's Encrypt, CN = Let's Encrypt Authority X3 i:O = Digital Signature Trust Co., CN = DST Root CA X3 -----BEGIN CERTIFICATE----- MIIEkjCCA3qgAwIBAgIQCgFBQgAAAVOFc2oLheynCDANBgkqhkiG9w0BAQsFADA/ MSQwIgYDVQQKExtEaWdpdGFsIFNpZ25hdHVyZSBUcnVzdCBDby4xFzAVBgNVBAMT DkRTVCBSb290IENBIFgzMB4XDTE2MDMxNzE2NDA0NloXDTIxMDMxNzE2NDA0Nlow SjELMAkGA1UEBhMCVVMxFjAUBgNVBAoTDUxldCdzIEVuY3J5cHQxIzAhBgNVBAMT GkxldCdzIEVuY3J5cHQgQXV0aG9yaXR5IFgzMIIBIjANBgkqhkiG9w0BAQEFAAOC AQ8AMIIBCgKCAQEAnNMM8FrlLke3cl03g7NoYzDq1zUmGSXhvb418XCSL7e4S0EF q6meNQhY7LEqxGiHC6PjdeTm86dicbp5gWAf15Gan/PQeGdxyGkOlZHP/uaZ6WA8 SMx+yk13EiSdRxta67nsHjcAHJyse6cF6s5K671B5TaYucv9bTyWaN8jKkKQDIZ0 Z8h/pZq4UmEUEz9l6YKHy9v6Dlb2honzhT+Xhq+w3Brvaw2VFn3EK6BlspkENnWA a6xK8xuQSXgvopZPKiAlKQTGdMDQMc2PMTiVFrqoM7hD8bEfwzB/onkxEz0tNvjj /PIzark5McWvxI0NHWQWM6r6hCm21AvA2H3DkwIDAQABo4IBfTCCAXkwEgYDVR0T AQH/BAgwBgEB/wIBADAOBgNVHQ8BAf8EBAMCAYYwfwYIKwYBBQUHAQEEczBxMDIG CCsGAQUFBzABhiZodHRwOi8vaXNyZy50cnVzdGlkLm9jc3AuaWRlbnRydXN0LmNv bTA7BggrBgEFBQcwAoYvaHR0cDovL2FwcHMuaWRlbnRydXN0LmNvbS9yb290cy9k c3Ryb290Y2F4My5wN2MwHwYDVR0jBBgwFoAUxKexpHsscfrb4UuQdf/EFWCFiRAw VAYDVR0gBE0wSzAIBgZngQwBAgEwPwYLKwYBBAGC3xMBAQEwMDAuBggrBgEFBQcC ARYiaHR0cDovL2Nwcy5yb290LXgxLmxldHNlbmNyeXB0Lm9yZzA8BgNVHR8ENTAz MDGgL6AthitodHRwOi8vY3JsLmlkZW50cnVzdC5jb20vRFNUUk9PVENBWDNDUkwu Y3JsMB0GA1UdDgQWBBSoSmpjBH3duubRObemRWXv86jsoTANBgkqhkiG9w0BAQsF AAOCAQEA3TPXEfNjWDjdGBX7CVW+dla5cEilaUcne8IkCJLxWh9KEik3JHRRHGJo uM2VcGfl96S8TihRzZvoroed6ti6WqEBmtzw3Wodatg+VyOeph4EYpr/1wXKtx8/ wApIvJSwtmVi4MFU5aMqrSDE6ea73Mj2tcMyo5jMd6jmeWUHK8so/joWUoHOUgwu X4Po1QYz+3dszkDqMp4fklxBwXRsW10KXzPMTZ+sOPAveyxindmjkW8lGy+QsRlG PfZ+G6Z6h7mjem0Y+iWlkYcV4PIWL1iwBi8saCbGS5jN2p8M+X+Q7UNKEkROb3N6 KOqkqm57TH2H3eDJAkSnh6/DNFu0Qg== -----END CERTIFICATE----- --- Server certificate subject=CN = files.freeswitch.org issuer=C = US, O = Let's Encrypt, CN = Let's Encrypt Authority X3 --- No client certificate CA names sent Peer signing digest: SHA256 Peer signature type: RSA Server Temp Key: ECDH, P-256, 256 bits --- SSL handshake has read 3289 bytes and written 458 bytes Verification error: certificate has expired --- New, TLSv1.2, Cipher is ECDHE-RSA-AES256-GCM-SHA384 Server public key is 2048 bit Secure Renegotiation IS supported Compression: NONE Expansion: NONE No ALPN negotiated SSL-Session: Protocol : TLSv1.2 Cipher : ECDHE-RSA-AES256-GCM-SHA384 Session-ID: 52162023FF88BE2656B82283B18706FEB2A75670892C7E4FBF2402FB5FB6C95D Session-ID-ctx: Master-Key: E19ADB76D51F6EDBA1207396E41F0C96615F1C064B567385429A62A362B7AD21DBAC1BDF5675B8F9AAD56D5BA32705F0 PSK identity: None PSK identity hint: None SRP username: None TLS session ticket lifetime hint: 300 (seconds) TLS session ticket: 0000 - 02 86 95 bd 73 2e a3 99-ef b7 a8 d1 4d 56 c7 ea ....s.......MV.. 0010 - b0 99 68 a5 22 42 7f 11-4c e3 93 ea de 82 3a 4d ..h."B..L.....:M 0020 - 4a 36 94 65 f5 78 e8 8d-e4 0a 59 70 b5 77 2c 2f J6.e.x....Yp.w,/ 0030 - 10 3c b1 73 c8 f1 0f 5c-1d a3 17 69 70 94 89 e4 .<.s...\...ip... 0040 - 3c 70 4b f3 64 42 fb 5a-b7 99 06 1a e5 2c e3 f8 <pK.dB.Z.....,.. 0050 - 2f 5d 66 34 df ff 04 e1-e6 b7 e9 3f 87 71 4c 28 /]f4.......?.qL( 0060 - 48 27 e4 07 0a 95 6f c5-50 54 e1 b3 08 bd ef b5 H'....o.PT...... 0070 - 64 f5 63 a4 ed d0 9b f2-dc f8 9a e0 77 e5 6c 45 d.c.........w.lE 0080 - 34 3b 76 1c ca ca df 6f-df 6d 3b a7 e6 84 e9 53 4;v....o.m;....S 0090 - 14 d1 ba 11 24 40 a7 5c-e4 ec 9d e5 79 95 39 7b ....$@.\....y.9{ 00a0 - 76 2c 2f 11 5e 6f 8f b2-e0 59 ac 94 07 45 0f 8d v,/.^o...Y...E.. 00b0 - 18 6e 0b 5c 3b 04 d6 95-f7 33 d4 ed 66 b3 08 34 .n.\;....3..f..4 00c0 - 7f 51 9d 8b 52 a6 c1 7e-67 45 00 00 47 59 80 c4 .Q..R..~gE..GY.. Start Time: 1603174347 Timeout : 7200 (sec) Verify return code: 10 (certificate has expired) Extended master secret: no --- HTTP/1.1 400 Bad Request Date: Tue, 20 Oct 2020 06:12:37 GMT Server: Apache/2.4.25 (Debian) Strict-Transport-Security: max-age=15552000; includeSubDomains; preload Content-Length: 313 Connection: close Content-Type: text/html; charset=iso-8859-1 <!DOCTYPE HTML PUBLIC "-//IETF//DTD HTML 2.0//EN"> <html><head> <title>400 Bad Request</title> </head><body> <h1>Bad Request</h1> <p>Your browser sent a request that this server could not understand.<br /> </p> <hr> <address>Apache/2.4.25 (Debian) Server at files.freeswitch.org Port 443</address> </body></html> closed [safarov@safarov-dell esrp-packer]$ Sergey

2 1

Configure Freeswith with ovh SIP TRUNK
by Mourad Hedfi 20 Oct '20

20 Oct '20

Hello everyone, We have installed FreeSwitch with BigBlueButton to allow users to access BBB meetings with their phone. Alos, we bought a SIP TRUNK (5 Channels) from OVH and we have configured the system through this https://docs.ovh.com/fr/voip/freeswitch-configuration-et-utilisation/ But, only one person can access the meeting. When we add the extensions, we can't access the meeting and the dial fails. For that, can anyone tell me if I must have distinct phone numbers of the SIP TRUNK number to create the extensions ? What's the role of those extension and are they obligatory ? We really need your help. Many thanks in advance. Best regards.

1 0

ring_ready on answered call to play message
by kokoska.rokoska 20 Oct '20

20 Oct '20

Hello, I need to hang up the b-leg of the answered call, play the message to a-leg, and dial a new b-leg after a while when message is still playing. How to achieve this, please? I tried to use ring_ready, but it won't start playing until the new b-leg starts ringing. Any recommendations so I don't have to manage sessions manually? Everything will be clearer from the following dialplan. I need the message to start playing right at ring_ready and not when the callee2 starts ringing: <action application='set' data='hangup_after_bridge=false'/> <action application='set' data='execute_on_answer=sched_hangup +10 normal_clearing bleg'/> <action application='bridge' data='sofia/gateway/server/callee1'/> <action application='set' data='ringback=local_stream://message'/> <action application='ring_ready'/> <action application='sleep' data='10000'/> <action application='set' data='ignore_early_media=true'/> <action application='bridge' data='sofia/gateway/server/callee2'/> <action application='hangup'/> Best regards, kokoska.rokoska

1 0

No audio on SRTP/ZRTP
by Rachel Roch 19 Oct '20

19 Oct '20

Hi, I have achieved a functional Freeswitch configuration in that I can call both Freeswitch hosted services (e.g. talking clock) and make outbound VoIP calls through upstream provider. However this is only achievable using Linphone in Transport TLS + Encryption None config. The moment I switch Linphone Encryption to TLS or ZRTP, I get complete loss of audio (for both Freeswitch hosted and outbound VoIP). I have tried expanding the list of ciphers in "sip_tls_ciphers" and "rtp_sdes_suites" (both in vars.xml) but that makes no difference. Evidently SSL certs are correctly setup because otherwise Transport TLS would not work. I'm not quite sure where to go from here ? Laura

1 0

Port Required to communicate with Gateway
by Harshita Pandey 19 Oct '20

19 Oct '20

Hi, I have installed freeswitch on a Virtual Machine, and I am not able to configure the gateway for some reason. Can someone tell me which ports are required and need to be opened on the virtual machine to communicate with Gateway in freeswitch? Since all the ports are not opened in the Virtual Machine, this could help me open up those ports and configure Gateway for freeswitch Much help is needed. It's Urgent Thanks in advance.

5 12

sofia.c:3901 ERROR: unsupported transport
by Laura Smith 18 Oct '20

18 Oct '20

Hi, Any ideas what this obscure message means ? sofia.c:3901 ERROR: unsupported transport. It shows up when running a "sofia profile external rescan" (and originating calls doesn't work either). My provider sip_profile was copied over from a known good other machine. So I don't know what sofia is complaining about ? <include> <gateway name="foobar"> <param name="auth-username" value="none"/> <param name="password" value="none"/> <param name="extension" value="default"/> <param name="proxy" value="foobar.example.com"/> <param name="expire-seconds" value="300"/> <param name="register" value="false" /> <param name="register-transport" value="tls"/> <param name="ping" value="false" /> </gateway> </include>

1 0

Verto Communicator error
by Social Boh 17 Oct '20

17 Oct '20

Hello, this is the problem: bower bootstrap-material-design#~0.3.0 not-cached https://github.com/FezVrasta/bootstrap-material-design.git#~0.3.0 bower bootstrap-material-design#~0.3.0 resolve https://github.com/FezVrasta/bootstrap-material-design.git#~0.3.0 bower bootstrap-material-design#~0.3.0 ENORESTARGET No tag found that was able to satisfy ~0.3.0 the new location for this tool is: https://github.com/mdbootstrap/bootstrap-material-design Where can i change the new link? Thank you Regards -- --- I'm SoCIaL, MayBe

1 0

$500 bounty RTMP to conference room
by Nathan Stratton 16 Oct '20

16 Oct '20

Trying to broadcast from vlc or a sling box with rtmp to freeswitch conference room. I have mod_rtmp installed and why I try to steam from vlc I get: 2020-10-14 11:56:44.516563 [NOTICE] mod_rtmp.c:905 New RTMP session [88e00c1b-6b3a-4bac-83f5-bc12b879884b] 2020-10-14 11:56:44.516563 [INFO] rtmp_tcp.c:234 Rtmp connection from 50.210.153.253:36864 2020-10-14 11:56:44.516563 [DEBUG] rtmp.c:882 Sent handshake response 2020-10-14 11:56:44.536531 [DEBUG] rtmp.c:907 Done with handshake 2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:1081 [chunk_stream=3 type=0x14 ts=0 stream_id=0x0] len=155 2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:192 [amfnumber=3] Got INVOKE for connect 2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:656 [amfnumber=2 type=0x1 stream_id=0x0] len=4 2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:656 [amfnumber=2 type=0x5 stream_id=0x0] len=4 2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:656 [amfnumber=2 type=0x6 stream_id=0x0] len=5 2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:656 [amfnumber=2 type=0x4 stream_id=0x0] len=6 2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:656 [amfnumber=3 type=0x14 stream_id=0x0] len=201 2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:656 [amfnumber=3 type=0x14 stream_id=0x0] len=61 2020-10-14 11:56:44.556571 [NOTICE] rtmp_sig.c:122 Sent connect reply 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:1081 [chunk_stream=2 type=0x1 ts=0 stream_id=0x0] len=4 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:1089 SET CHUNKSIZE=512 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:1081 [chunk_stream=3 type=0x14 ts=0 stream_id=0x0] len=29 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:192 [amfnumber=3] Got INVOKE for releaseStream 2020-10-14 11:56:44.636529 [WARNING] rtmp.c:198 Unhandled invoke for "releaseStream" 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:1081 [chunk_stream=3 type=0x14 ts=0 stream_id=0x0] len=25 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:192 [amfnumber=3] Got INVOKE for FCPublish 2020-10-14 11:56:44.636529 [WARNING] rtmp.c:198 Unhandled invoke for "FCPublish" 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:1081 [chunk_stream=3 type=0x14 ts=0 stream_id=0x0] len=25 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:192 [amfnumber=3] Got INVOKE for createStream 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:656 [amfnumber=3 type=0x14 stream_id=0x0] len=29 2020-10-14 11:56:44.636529 [INFO] rtmp_sig.c:137 Replied to createStream (0) 2020-10-14 11:56:44.696580 [DEBUG] rtmp.c:1081 [chunk_stream=8 type=0x14 ts=0 stream_id=0x1] len=30 2020-10-14 11:56:44.696580 [DEBUG] rtmp.c:192 [amfnumber=8] Got INVOKE for publish 2020-10-14 11:56:44.696580 [DEBUG] rtmp.c:656 [amfnumber=2 type=0x4 stream_id=0x0] len=6 2020-10-14 11:56:44.696580 [DEBUG] rtmp.c:656 [amfnumber=5 type=0x14 stream_id=0x1] len=138 2020-10-14 11:56:44.696580 [INFO] rtmp_sig.c:290 Got publish on stream 1. 2020-10-14 11:56:45.216562 [DEBUG] rtmp.c:1081 [chunk_stream=4 type=0x12 ts=0 stream_id=0x1] len=319 2020-10-14 11:56:45.656516 [DEBUG] rtmp_tcp.c:243 Closing socket 2020-10-14 11:56:51.796530 [NOTICE] mod_rtmp.c:1027 RTMP session ended [88e00c1b-6b3a-4bac-83f5-bc12b879884b] Anyone able to help me? $500 via Paypal. ><> nathan stratton

7 12

Multiple trunks failover schema setting up !
by CpServiceSPb 16 Oct '20

16 Oct '20

Hi. I have 3 external SIP trunks for making external calls, provided by SIP telephony carriers: SIP carrier1, SIP carrier2, SIP carrier3. If SIP carrier1 trunk is ready to making callings then the outboud calling outside of FS is making using the trunk, otherwise, for example, if the trunk state is unknown/invalid or disabled by carrier that is unavailable or already ringing/busy then attempting of calling will be made using SIP carrier2 and if the same situation with SIP carrier2 trunk then attempting of calling will be made using SIP carrier3 trunk and if SIP carrier3 trunk has state which do not allow the calling voice message to caller will be send back What is the best way to make the following schema ?

1 0