May 2018 - FreeSWITCH-users - FreeSWITCH Mailing Lists

file.exists in Freeswitch 1.8/1.9
by Nick Giannak III 15 May '18

15 May '18

Greetings. In Freeswitch 1.6, the file.exist module does not appear to be implemented in V8. The console shows it as "not implemented". This, however, is a function listed in the documentation. Is it implemented for Freeswitch 1.8? Or, alternatively, are we doing something wrong? Thanks, Nick

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Use GStreamer instead of FreeSwitch?
by Chandramouli P 14 May '18

14 May '18

Hello, I am using Kamailio as registration server and FreeSwitch for signalling (RTP packet handling). Can I use GStreamer instead of Freeswitch or Asterisk? Thank you. Regards, CM

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Tone Stream and G729
by Alexander Perkins 14 May '18

14 May '18

Hi All. I am wondering if somebody can point me in the right direction. I am playing back tone_stream on my call, but noticed that if the codec is in G729, the call fails because of a codec negotiation error (I am loading the G729 mod). Does anybody know how tone_stream can work with G279? Here's the line from my dialplan (NumberOfLoops depends on how long it takes us to find the user, historical average): <load module="mod_g729"/> <action application="playback" data="tone_stream://%(2000,4000,440.0,480.0);loops=${NumberOfLoops}" /> Thanks, Alex

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Is cidlookup asynchronous?
by Jim Lynch 12 May '18

12 May '18

I am calling cidlookup in my dial plan like this: <action application="set" data="effective_caller_id_number=$1" inline="true"/> And in my log I see: a6a3d389-245d-4705-9a1e-91d74d491dd1 Dialplan: sofia/external/+12297960236@4.59.156.121:5060 Action set(effective_caller_id_name=${cidlookup(${caller_id_number})}) I'm trying to use it in an lua script that is called right after the above action line but the log seems to indicate that it doesn't actually get executed until some time later. a6a3d389-245d-4705-9a1e-91d74d491dd1 Dialplan: sofia/external/+12297960236@4.59.156.121:5060 Action info() a6a3d389-245d-4705-9a1e-91d74d491dd1 Dialplan: sofia/external/+12297960236@4.59.156.121:5060 Action lua(cid.lua ${caller_id_number} ${effective_caller_id_name}) INLINE a6a3d389-245d-4705-9a1e-91d74d491dd1 EXECUTE sofia/external/+12297960236@4.59.156.121:5060 lua(cid.lua +12297960236 ) 2018-05-09 20:28:02.349228 [DEBUG] freeswitch_lua.cpp:365 DBH handle 0x7fefd0092e70 Connected. a6a3d389-245d-4705-9a1e-91d74d491dd1 2018-05-09 20:28:02.349228 [DEBUG] switch_cpp.cpp:745 CoreSession::setVariable(effective_caller_id_name, ALL LOCATION CZ) How can I get the data into my lua script? Thanks, Jim.

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(no subject)
by Jon Neuffer 11 May '18

11 May '18

http://total.tipadvisor.org Jon Neuffer

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Re: [Freeswitch-users] stream a file multicast with mod_esf
by Sluschny, Thomas 10 May '18

10 May '18

Hi Brian, i thought if i can stream from portaudio it is almost the same with streaming from file, so it should working already now. Is this not the design idea of channels and media to do so? regards, thomas PS: sry for improper formatted mail, i cant reply at the moment and have to copy mail from archive :( ________________________________ >From brian at freeswitch.org Thu Feb 12 07:07:36 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 12 Feb 2009 09:07:36 -0600 Subject: [Freeswitch-users] stream a file multicast with mod_esf Message-ID: <6F68FEDB-8976-44D4-9A67-F871EC7D5120(a)freeswitch.org <mailto:6F68FEDB-8976-44D4-9A67-F871EC7D5120@freeswitch.org> > esf is for multi cast paging... it currently won't let you play files... we would have to create a multicast playback application. /b On Feb 12, 2009, at 8:00 AM, Sluschny, Thomas wrote: > Hi, > > i want to stream a file per IP multicast with mod_esf. > > I can stream IP multicast with: > pa call stream XML > and in XML dialplan: > <extension name="stream"> > <condition field="destination_number" expression="^stream$"> > <action application="answer"/> > <action application="esf_page_group" data="224.168.168.168 34567"/> > <action application="esf_page_group" data=""/> > </condition> > </extension> > and i can also play files with 'playback' app, > > BUT: how can put these 2 things together? > > May be its trivial, but i cant get it make working with 'originate' > or 'uuid_broadcast' or 'bridge', 'transfer' and so on ... > > Thanks in advance, > Thomas

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WebRTC using rtp_sdes_suites=AES_CM_128_HMAC_SHA1_80
by Jerry Chinn 09 May '18

09 May '18

Good Day, Running FS 1.6.17 on CentOS 7.4 We are running WebRTC and are required to use AEAD_AES_256_GCM_8 or AEAD_AES_128_GCM_8 for security. I have eliminated all of the options in the vars file except rtp_sdes_suites=AEAD_AES_256_GCM_8|AEAD_AES_128_GCM_8. Calls are successfully completing, however, in debug we are seeing AES_CM_128_HMAC_SHA1_80 as the sdes suite for srtp:dtls. 2018-05-04 22:38:30.429310 [INFO] switch_rtp.c:3185 Changing audio DTLS state from HANDSHAKE to SETUP 2018-05-04 22:38:30.450549 [INFO] switch_rtp.c:3094 audio Fingerprint Verified. 2018-05-04 22:38:30.450549 [INFO] switch_rtp.c:3908 Activating audio Secure RTP SEND 2018-05-04 22:38:30.450549 [DEBUG] switch_core_sqldb.c:2617 Secure Type: srtp:dtls:AES_CM_128_HMAC_SHA1_80 2018-05-04 22:38:30.450549 [INFO] switch_rtp.c:3886 Activating audio Secure RTP RECV 2018-05-04 22:38:30.450549 [INFO] switch_rtp.c:3134 Changing audio DTLS state from SETUP to READY 2018-05-04 22:38:30.450549 [DEBUG] switch_core_sqldb.c:2617 Secure Type: srtp:dtls:AES_CM_128_HMAC_SHA1_80 2018-05-04 22:38:30.450549 [DEBUG] switch_rtp.c:1885 rtcp_stats_init: audio ssrc[3910337773] base_seq[2433] Any ideas on how or where to change this to the desired encryption protocol? Jerry Chinn Telecom VoIP Specialist .

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eavesdrop
by Francesco Facco de Lagarda 09 May '18

09 May '18

I need to enable client to eavesdrop on certain calls. The fs is in a penitentiary institute and certain types of criminals (i..e mafia related) must be listened. I am trying to implement eavesdropping like so: api originate {sip_secure_media=true}user/$ext &eavesdrop($uid) (where, I hope, $ext and $uid are self-explicatory) Is there ANY way to mute the microphone of the eavesdropper? It's a bit of a give-away if the criminal can hear the background noise Thanks Francesco

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how to use single port per call with ASR unimrcp module using mrcp-v1 protocol?
by Jignesh Vasoya 08 May '18

08 May '18

Hi, How to configure and use "unimrcp" where it will use a single port(resource) with MRCP server throughout the call. Idea is to use only single port between FreeSwitch and MRCP server for each call. FreeSwitch will load/unload required grammar files during the call. I am looking to use this with mrcp-v1 protocol, is it possible? -- *Thanks,* *Jignesh*

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ESL in .Net
by Hossein Gholi 08 May '18

08 May '18

Hi I'm new in FreeSwitch ... I want to start voice call project in .Net but found no sample for it anywhere ... someone said when downloading source of freeswitch there is a sample project for .Net in libs/esl/managed but there is no folders like these ... Please if it's possible for anyone help me ... i do'nt have any sample project for it If you have send me Thank's

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