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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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How to detect hangup event in Freeswitch?

I'm new to Freeswitch and looking for help from experts. My problem is below. I'm trying to do below scenario in Perl: When I'm getting an incoming call to script (test.pl) I play a file to it and then put inbound session to on-hold. Then I try to…
Yasiru G
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Transfer a call programmatically inside a FreeSWITCH node-esl server when it is already bridged to another endpoint

I am trying to transfer the call that is connected to node-esl. I have successfully bridged that call to another endpoint. Now, I want to programatically transfer that call over to another extension or another number without the call flow being cut.…
Ashwin Chandar
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How to cancel a hangup scheduled with `sched_hangup`?

The sched_hangup doc doesn't mention how to do this, and sched_cancel only works for sched_transfer and sched_broadcast. sched_cancel needs a task or group ID, and only the latter two set the ID in the session, I tested. Or is there another way to…
toraritte
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Send SMS Freeswitch via web interface

I installed the freeswitch and bought a Khomp device to send SMS trought it! I'm able to send sms with the command line inside the freeswitch client(fs_cli). I enabled the XML-RPC module to send it via web, but I'm not able to start an application…
digoferra
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Failed to start Freeswitch in centos?

I have just installed Freeswitch in Centos7 . when i check my Freeswitch status , its failing to start with following error. Any help would be highly appreciated. I have attached copy of both command and the logs file showing errors as…
delek tashi
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Keeping b-leg on line after a-leg hangs up

I have a situation where I need to keep the b-leg on the line after the a-leg hangs up so that the b-leg can be presented with an IVR menu. Is this possible to do? Can it be done purely in the dialplan XML or is a different approach, e.g. a Lua…
Graham Mayer
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Getting black screen for remote video: WebRTC

I'm developing video calling functionality with WebRTC and facing a very strange problem. When I make a call everything is fine and I'm getting a remote video stream, but when I receive a call, I get a black screen with no remote video. The strange…
ishan shah
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VoIp recording solutions (Asterisk or/and FreeSwitch)

currently we're looking for voip recording solution - it must record all incoming/outgoing/internal/conference calls within the company. later on we going to develop applications that let us find/analyze recordings. the main problem as I see at the…
Leonid
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Freeswitch channel exception

I use official java client library for esl. I get the following exception java.nio.channels.ClosedChannelException when I try to connect to a socket from Freeswitch after some time of normal work. Please help me to resolve it.
carapuz
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How can I fix "Cannot read property 'srcObject' of undefined" in Freeswitch Verto

I'm trying to get verto to work on in Chrome 57.0.2987.133 (64-bit). After following https://dopensource.com/2017/01/21/setting-up-freeswitch-webrtc-functionality/, I was able to get webrtc/verto working on the server. I was able to get the demo app…
Dayo
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Catch audio stream in freeswitch

Given a sip call between two persons using freeswitch as my telephony engine ,how to catch audio stream of each person separately and process it before it's sent to the other end. Thanks for your help in advance.
Hataki
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Regular expression how to from value agent and username and number?

I have a regular expression requirement for a telephone dialpad, from where i need to parse agent name, username, a telephone number. Example random user inputs like…
user285594
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Freeswitch command to add sip users

Is there a command to add sip users in freeswitch as there is one in kamailio, i.e. like kamctl add [user] [password]?
rim
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FreeSWITCH: Unable to connect from browser(WebRTC) behind enterprise firewall

I am running a FreeSWITCH server on a public domain. I am able to make call from my browser(WebRTC with sipjs) to the FreeSWITCH server from a direct Internet connection. I am testing it on a Firewall that allows only TCP on port 443. I am running a…
Senthil Ganesh
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Freeswitch 1.6 installation for ESL python

I'm running Freeswitch 1.6 and Mac 10.10.5 My Freeswitch server will be in AWS and need to install Python ESL module in my local environment to start development using ESL. I havent been able to find ESL module. I got the whole source and tried to…
gogasca
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