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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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Connecting freeswitch remotely using node esl fails

Im trying to connect remotely to FreeSwitch service using ESL. Connecting machine and FS Service both are in my local VM's, 2 different VM's But i get below error [WARNING] mod_event_socket.c:2639 IP 10.95.38.254 Rejected by acl…
sravis
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How can free switch initiate a conference by a scheduled time

I've tried to set up conference call using asterisk & free switch as well where my SIP soft phone is XLite. I'm able to do conference using both asterisk & free switch with XLite. Now i'm trying the reverse way that instead of endpoints start the…
abhisek
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Adding P-Asserted-Identity / P-Preferred-Identity on FreeSwitch

How can i modify the dial plan / sofia profile to insert the P-Asserted-Identity or the P-Preferred-Identity Headers on Freeswitch? I have the information in FROM Header and like to anonymize it and provide it in one of the P-Headers.
Rajesh
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Call Recording in Freeswitch

I have a freeswitch working on one server and call is working fine. But now i want to record each and every call to some specific format like .wav OR .gsm I already tried with "record_session" application.Record session application Is it right…
Bhavik Patel
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How to connect FreeSwitch to FreeSwitch?

i have a problem with FreeSwitch. I tried for hours to connect a FreeSwitch server on my system with a FreeSwitch server on another system. However, what i want is to call with user "abc@myip" the user "123@buddysIp". What i tried is to add a new…
Zero
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freeswitch stream remote audio file

i'm saving the remote audio files on google cloud storage. i want to play these files in freeswitch. when i use: mediaLink = "http://storage.googleapis.com/myBucket/file.wav"; session:streamFile(mediaLink); it works great. But when i use signed…
snabel
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How to configure Supervisord for FreeSWITCH?

I am trying to configure a Supervisor for controlling the FreeSWITCH. Following is the configuration at this moment present in supervisord.conf. [program:freeswitch] command=/usr/local/freeswitch/bin/freeswitch -nc -u root -g root…
Nikhil N
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Which is more appropriate for pbx events, TIdTCPServer or TIdTCPClient?

I am trying out some IP PBX systems, Asterisk, Freeswitch, and Yate,to register for events in the PBX, and I want to know which of these components is the better one. The component is supposed to register with the PBX for events, receive them, send…
vfclists
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What does the for(;;) (2 semi colons) loop mean?

I'm working on a script for the VoIP software, Freeswitch. The script will run as an instance listening for inbound messages to a socket. I used an example script provided with Freeswitch, to start with, and everything works fine. However, one bit…
BIGMOOSE
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FreeSwitch - using mod_xml_curl for Dialplan

I am using mod_xml_curl to register SIP users on FreeSwitch server. I could get user information from My-SQL database, and now I have a problem with FreeSwitch dialplan. When I make a call to another registered user, it makes hold_music sound. It…
Jake
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SIP Redirect via Proxy (SIP.js)

I'm trying to create a minimal SIP Proxy that serves one purpose: redirects requests to another domain. The catch is the domain I'm redirecting to requires authorization so I assume I need to rewrite some SIP attributes since SIP authorization is…
Dan
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Call recordings in Asterisk using MixMonitor

I'm just starting out with Asterisk and following the O'Reilly Guide to set up a test Asterisk server. I have set up a VM with CentOS 6.4, which has 1GB RAM and 50 GB HDD. After installation, I set up soft phones successfully on 2 PCs which were…
rahuL
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FusionPBX External Sip Profile Conf INVALID

I was having the exact same problem as this guy.I followed the answer but that gives me INVALID Profile I have only tried to configure external.xml file to my External IP.But everytime i reloadxml it tells me my sip profile conf is invalid What i…
Nezam
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MJSIP: Register android client with server: onUaRegistrationFailure; Wireshark 400/Bad Request

'Hello I try to develop a softphone with MJSIP for android. I have a simple test setup: 1 PC (Win7) with a sip phone (number 1000) 1 VM (Win7) with a sip phone (number 1001) and Freeswitch installed sip phone #1000 can call #1001 and…
B770
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Easiest way to make automated SIP phone calls from a web app?

I have a client company with a simple web application (Python Flask) and I need to add a phone notification functionality to it. The main requirement is that the app should call users, play a certain sound file and accept some tone input ("Hello!…
skanatek
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