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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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Capture SIP "200 OK" in Freeswitch python script

I'm trying to run a Python script using "mod_python" in my freeswitch server in order to check a local REDIS db. So far i was able to run the script from the dialplan like this:
Ricardo
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Freeswitch where command history location

I see the freeswitch have socket mod and they have function: save_history (mod_command) Usage: fsctl save_history Write out the command history in anticipation of executing a configuration that might crash FS. This is useful when debugging a…
QViet
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Freeswitch 1.10 on CentOS 7: Error while starting the server and no user registration

We used to have FS 1.4 working on CentOS 6. But that server had to be migrated to CentOS 7. When we take Freeswitch 1.10 and try to run with default configuration, we get below error - [ERR] sofia.c:3254 Error Creating SIP UA for profile: external…
Krishnan V S
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wss connection failed: Error in connection establishment: net::ERR_CERT_COMMON_NAME_INVALID

I've build up a working FreeSwitch box and can make internal calls on desktops using ws connections. Later on, I've configured the box to use SSL with certificates issued by Letsencrypt. SSL certificates are validated all good by…
Kai
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Freeswitch Guide how to get CDR UUID from client side

My system contain: - Freeswitch server - Sip Client: Web using sipjs , mobile react-native using https://github.com/datso/react-native-pjsip to receive call. My problem is when call done i need to know the uuid of CDR recently add to Postgres DB of…
QViet
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Use Freeswitch Modules in Asterisk 1.6

I need to run Freeswitch modules in Asterisk, Voice Recognition in particular, because Pocketsphinx can be used there directly without the bottleneck of UniMRCP (which is slow and not really open source anymore). For Freeswitch is based on Asterisk…
scumpidou
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dial an extension after call is setup

I need to dial an extension within an IVR. So my freeswitch application will test PBX extensions. The application will dial an IVR (Could be running any PBX vendor), when the PBX picks up it will dial an extension and run several tests including…
user2236794
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FusionPBX installation 502 Bad Gateway

I am trying to install FusionPBX on Centos 7 operating system using this below link. https://www.fusionpbx.com/download.php After successful installation, I am getting an error 502 Bad Gateway nginx/1.12.2 FusionPBX Error Screenshot Please, can…
Kavirajan ST
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freeswitch sip trunk not receiving inbound calls

my sip trunk provider has given me a user name sip123456 when I configure that siip trunk as a gateway, I can make calls out no problem but I cannot receive any inbound calls! Now I did a lot of investigation and I found out that the user name has…
Solutel WW
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freeswitch recordings during IVR

in last version of freeswitch i got a problem - on stereo records during ivr some noice presnt in b-channel. dialplan:
Дмитрий Макаров
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pass variable in to command in python

i'm writing a python script to execute shell command, and i'm taking arguments and i want to pass the value of that to the command #!/usr/bin/env python import commands import subprocess import sys command = commands.getoutput('fs_cli -x "sofia…
frodo
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How to implement BFCP for screensharing using SIP

I need to implement a screen sharing application using BFCP but not able to find much, can some one please describe or explain in brief how this can be achieved. There is very little information about this on the internet now sure why. SO doesnot…
Akshay Kasar
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Is it possible to detect if the call is being transferred on FreeSWITCH?

I want to detect if the call is being transferred in FreeSWITCH. For e.g If I am now calling someone and that someone transfers me to an other person, I want to know during the call if a transfer happens. I wonder if there is in Freeswitch an event…
Anis Bedhiafi
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Video Conferencing in Freeswitch

I am working on Calling Solution which uses FreeSwitch for Audio/Video Calling. I am stuck with Video Conferencing system that will be run by iOS and Andriod client devices. When I create a Video Conference Call, every user can see Video of only…
Azhar Nawaz
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Freeswitch originate

I'm using Freeswitch 1.6 ESL and when I place a call using API and remote IP Address I get: NO_ROUTE_DESTINATION 2016-07-08 06:24:13.381491 [DEBUG] switch_core_state_machine.c:296 No Dialplan on answered channel, changing state to HANGUP It works…
gogasca
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