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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 retry when returning 486 sipp scenarios

I want to complete this scenarios in sipp. When I make a call, if it return 486 status(busy), It could call pause a while, then continue recalling.
user1369887
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How to share common id for two channels in a call

I am working on communication between two applications in freeswitch, I have done following from a java program, ProcessBuilder processBuilder = new ProcessBuilder( "/bin/bash", "-c", "cd /usr/local/freeswitch/bin && ./fs_cli -x \"originate…
Ravikiran Reddy
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Is it possible to exclude (filter out) events from the Mod Event Socket telnet client output?

Mod Event Socket provides a useful telnet client. I can filter specific events and keep track of just these, but I can't find how to exclude events I don't want to monitor (typically HEARTBEAT and RE_SCHEDULE). Is it possible?
Anto
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BadImageFormat exception at runtime

I am trying to build the FreeSWITCH .NET ESL client library, as per the instructions here on a 64-bit Windows 7 machine, with VS 2012 targeting .NET 4.5, and reference the built DLLs in my own project. The build of both libraries is successful, as…
Zev Spitz
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FreeSWITCH Audio Stream Initialiase

I am working with FreeSWITCH and trying to understand how audio streams are initialised. I am working with ESL (inbound and outbound) and am trying to record some audio. Through testing i have proven that audio is working correctly for bridged…
puppyFlo
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invalid input-output audio device freeswitch

i am using freeswitch for a VOIP application. after loading mod_portaudio, pa devlist results blank , pa call 9999 and pa call 9996 reasult mod_portaudio.c:2453 Error invalid output audio device. . Can anyone please help me why these errors are…
user1234
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configurations to communicate local freeswitch with freeswitch on cloud

I want to make my local freeswitch communicate with the cloud freeswitch server. I am running recording application on local freeswitch and playing application on cloud. Both of them are getting invoked by origination command but my recording app…
Ravikiran Reddy
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how to originate two applications from fs_cli

Please answer this interesting question, I want to make two applications communicate with each other in free switch without a sip user, I have tried following things on fs_cli originate user/1001 &bridge(Sofia/internal/1789) user is able to…
Ravikiran Reddy
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FreeSwitch - how do i parse and take action using python?

I have this dial plan and i want to handle the dialed number using python. But its not working any idea? default.xml (dialplan):
user285594
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freeswitch python scripts errno 10 no child processes

I' ve got an issue when running freeswitch with some python scripts inside dialplan using django.db models. Whenever it starts it causes errors: freeswitch@ubuntu> 2013-08-15 06:56:08.094348 [ERR] mod_python.c:231 Error importing module 2013-08-15…
Piotrek Janus
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Freeswitch, fileaccess with javascript?

I have a FreeSwtich solution running on Linux with quite a lot of configuration scripts written in javascript. The problem is that we need to write and read files; which javascript normally doesn't support. I tried the SpiderMonkey File Object but…
Viking
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some issue about sipp

I wrote a program to run sipp. But It cannot auto exit after It call a large total numbers, or It has some other ways to know the sipp has completed, And second issues: When it call after count number, The call become very slow!
user1369887
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Freeswitch codec G729

I'm working with freeswitch and I made the connection between my server and another one, for hearing each other I used the codec G729. The issue is the next: I call them, the call is established and I can hear the other part perfectly but they can't…
Anna
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Buddy online status using FreeSWITCH's mod_skypopen and PHP

I am new to FreeSWITCH/Skypopen. I need to get the the status of a buddy using the skypopen module and PHP. I am using the following PHP script to get the status.
Abani Meher
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Freeswitch Event Socket

I am building a VoIP application using the soft PBX Freeswitch. So far I am able to use event socket library smoothly. However I would like to know how to get the data of a paricular made from code. For instance assuming I have configured an…
Arsene
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