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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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error: field 'ctx' has incomplete type EVP_CIPHER_CTX

Problem: I need to install Cepstral (tts engine) into Freeswitch running Debian 8. Freeswitch is already up and running, but I needed to build it from source in order for it create the mod_cepstral module. When I run make this is the error I…
Joe
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WebRTC + IOS + Freeswitch : Can't hear audio

I'm trying to implement mod_verto on IOS (calling from iPhone to Desktop). I'm using Google's libjingle library for the RTC side, got it up and running using this excellent tutorial. When making a call from my iPhone, I get the call on the desktop…
Shlomi Schwartz
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Are there parallels to Asterisk AMI and AGI in FreeSWITCH?

Asterisk has Asterisk Manager Interface (AMI) and Asterisk Gateway Interface (AGI), using which one can trigger PHP scripts at certain events from Asterisk. Using the same PHP scripts can also instruct Asterisk what to do next to a call/conference.…
jeff musk
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How to get a list of freeswitch registered accounts

I am developing a pjsip application. I am using Freeswitch as SIP server. (this is VOIP) Is there a command in fs-cli that lists all the registered users in Freeswitch. I need something like that to know which accounts are in a conference and so…
real_yggdrasil
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TypeError: argument of type 'char const *'

I'm currently working with Freeswitch and its event socket library (through the mod event socket). For instance: from ESL import ESLconnection cmd = 'uuid_kill %s' % active_call # active_call comes from a Django db and is unicode con =…
Anto
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Freeswitch ACL configuration for remote event socket

I have a FS server running on one server and on a remote server I have a Node JS instance controlling it using node_esl (a Node JS Event Socket library for FS). Every time I'm sending a request to the server I have the following error: [WARNING]…
Stephane Paquet
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Freeswitch bgapi originate command with ignore_early_media=true

I'm trying the following scenario on freeswitch: Create a call (a-leg) Create another call (b-leg) Bridge then together The b-leg phone is a dial plan in other freeswitch is the following:
Lucas Nunes
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FreeSWITCH minimal installation and module selection

As someone who is very new to the opensource PBX projects like Asterisk and FreeSWITCH, I am grappling with some information overload. Have read the basic FreeSWITCH docs on Wiki, but still have few questions. Since I am not very familiar with the…
jay
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Enumeration (enum in lua) .Want to use enum in lua5.2.4

I have a condition where in my lua script I want to use enum like for SUCCESS I can give 1 and for FAILURE I can give 0 I am using lua version 5.2.4 Can anyone please help me on how to use enum I want to use enum elseif(cc_config_cmd == "DELETE" and…
ayush jain
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Running Freeswitch on Google Container Engine

I'm trying to migrate a voip service using freeswitch on GKE (google cloud managed kubernetes cluster) in order to make the service scalable. I have managed to migrate freeswitch to docker and get it to run. I require a high number of ports to be…
Sebastien
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Freeswitch. Error 487 Role Conflict (on REINVITE)

I'm trying to add video track to stream and then call renegotiate() from JsSip. However, when I'm doing it from caller it works fine, but when I'm doing it from callee it's not working (session terminating). I looked into the Freeswitch logs and…
morozRed
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Freeswitch ipv6 call is not working

User is connected to an ipv6 network. SIP registration is fine . But when i make call to that user from freeswitch commandline(fs_cli) it is showing error as Cannot create outgoing channel of type [error] cause: [USER_NOT_REGISTERED].it is only…
Hansika Penugonda
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FreeSwitch very slow

I've installed a default out of the box FreeSwitch instance but when I try to make an internal call (extension to extension) it take around 12 seconds before the call is established and I can hear the ring tone. When I look at the log I see the…
Amir Peivandi
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Setting up freeswitch server on Google cloud compute

I am trying setup freeswitch server on google cloud compute (ubuntu 14.04) although it work fine for me locally, I seem to get the following error when I start freeswitch server on google cloud compute.Can any one explain? 2015-06-11 05:40:32.001508…
Pandurang Waghulde
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difference between freeswitch and red5

Firstly I just want to know that what is the difference between freeSwitch and Red5? As I have very good working experience with red5 and I have made many app's that streamed video/audio using Red5. But now I am not able to understand that If Red5…
Arjun Thakur
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