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Questions tagged [freeswitch]

FreeSWITCH is a scalable open source cross-platform telephony platform designed to route and interconnect popular communication protocols using audio, video, text or any other form of media.

FreeSWITCH is a scalable open source cross-platform telephony platform designed to route and interconnect popular communication protocols using audio, video, text or any other form of media. It was created in 2006 to fill the void left by proprietary commercial solutions. FreeSWITCH also provides a stable telephony platform on which many telephony applications can be developed using a wide range of free tools.

FreeSWITCH was originally designed and implemented by Anthony Minessale with the help of Brian West and Michael Jerris. All 3 are former developers of the popular Asterisk open source PBX. The project was initiated to focus on several design goals including modularity, cross-platform support, scalability and stability. Today, many more developers and users contribute to the project on a daily basis.

We support various communication technologies such as Skype, SIP, H.323 and GoogleTalk making it easy to interface with other open source PBX systems such as sipXecs, Call Weaver, Bayonne, YATE or Asterisk.

FreeSWITCH supports many advanced SIP features such as presence/BLF/SLA as well as TCP TLS and sRTP. It also can be used as a transparent proxy with and without media in the path to act as a SBC (session border controller) and proxy T.38 and other end to end protocols.

FreeSWITCH supports both wide and narrow band codecs making it an ideal solution to bridge legacy devices to the future. The voice channels and the conference bridge module all can operate at 8, 12, 16, 24, 32 or 48 kilohertz and can bridge channels of different rates. The G.729 codec is also available under a commercial license.

FreeSWITCH builds natively and runs standalone on several operating systems including Windows, Max OS X, Linux, BSD and Solaris on both 32 and 64 bit platforms.

FreeSWITCH supports FAX, both over audio and T.38, and can gateway between the two.

Our developers are heavily involved in open source and have donated code and other resources to other telephony projects including openSER, sipXecs, The Asterisk Open Source PBX and Call Weaver.

Resources

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Can we get frreeSWITCH outbound call duration from an API

I am trying to find an API or a method that can get the call duration from FreeSWITCH. As my search, we can get call duration from fs_cdr table or calculate the time between CHANNEL_HANGUP - CHANNEL_ANSWER events. Currently, we are having a problem…
tandathuynh148
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How to connect to freeswitch remote database?

I have been trying to figure out if there is any way to connect to remote Database with FreeSwitch API. FreeSwitch Lua API accepts only: Database name, username & password. So it establishes the connection to localhost only. local dbh =…
Abhishek Deshkars
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Why FreeSwitch api command does not work in CLI?

From this guide, I know that FreeSwitch (FS) will load modules, each module would provides some commands and all these commands can be accessed from FS_CLI or ESL. Here is my test: 1/ Show available commands from FS_CLI: freeswitch@my-pc> show…
Dev
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Freeswitch: Could not open curl.conf

I got the following error while using freeswitch switch_xml_config.c:78 Could not open curl.conf. This is very interesting, because the offical docs are saying "There is no separate config file for mod_curl.". So is this file needed, yes or no? I…
zimmerya
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How to combine multiple video webRTC mediaStreamTrack at ffmpeg / ffmpegwasm

I'm attempting to combine or mix multiple video tracks into a single mediaStreamTrack, or a mediaStream, I don't want to utilize Canvas as it's not readily available across multiple platforms [mobile devices, react native, etc] Is there such a…
junaiddarajat
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cant install mod_python3 on freeswitch

I have installed Freeswitch-mod-python3 on my freeswitch box running version 1.10.7. I then uncommented out the python3 line in the module.conf file. Then did a reloadxml, this reloaded successfully. When I do a "module_exists mod_python3" I get…
Marvin The Martian
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video conferece mix record in freeswitch

I was configuring Freeswitch to make a video conference. It was normal for everyone to enter the conference ,can watch each other and screen share. but It still existed these problems: 1: conference video record was splited several files that every…
yankunliu
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play_and_get_digits no input vs wrong input in Freeswitch

I am trying to implement the play_and_get_digits using ESL in nodejs. The code is quite simple: conn.execute('play_and_get_digits', ${minValue} ${maxValue} ${tries} ${timeout} ${terminator} ${soundFile} ${invalidFile} ${var_name} ${regex}…
venalyn sudaria
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Reduce Freeswitch video conference latency

We're experimenting with a Freeswitch based multiparty video conferencing solution (Zoom like). The users are connecting via WebRTC (Verto clients) and the streams are all muxed and displayed on the canvas (mod_conference in mux mode). It works OK,…
felixx
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Unable to retrieve Duration from freeswitch using ESL

I am new to FreeSWITCH and I am trying to apply click-to-call in my NodeJs application. This is my code when executing click-to-call, I use the 'originate command', this is my reference let app_args = `sofia/gateway/fs-test1/${phoneNumberFrom}`; …
venalyn sudaria
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Registration failed on freeswitch

I've got a registration problem in my local freeswitch. I've installed in a WSL with Ubuntu 20.04 and freeswitch 1.10.3 When I'm trying to register my user (3333) with a SIP client (ex. Microsip) but if I use the local IP address I receive an error…
Andrea Mason
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Freeswitch doesn't seek back

I'm trying to get rid of controlling audio played via session.streamFile() in Freeswitch. For this I tried the 3rd example of this documentation. Almost everything here is working, but the DTMF 1 (seek:-500) doesn't seek back. It always starts from…
Niklas Droste
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Problem with establishing voice communication Sip.js

I'm trying to make a call from client side(browser) to FreeSWITCH server using this libs: Sip.js Sip.js framework And also I use React. When I make a call I successfully invite destination URI. After click button and call function callSip our…
Elli Zorro
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Freeswitch v1.10 on CentOS 7 - "mod_event_socket" and fs_cli error in fresh installation

By referring to Freeswitch page (https://freeswitch.org/confluence/display/FREESWITCH/CentOS+7+and+RHEL+7), we did a fresh installation on CentOS 7. Built the server from source code, did not change any configuration in order to test if the server…
Krishnan V S
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Using Freeswitch ESL Api answer a call from one sip remote softphone to another sip remote softphone using java

I want to answer a call ringing from one sip remote softphone to another sip remote softphone using Freeswitch ESL API in java. I tried, using the uuid_answer API command to answer a call while getting "Unique-ID" form CHANNEL_CALLSTATE…
Hasan Syed
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