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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 connect to ESL from a remote server?

I would like to make a web interface in PHP to see the FreeSWITCH activities (calls, etc), possibly hosted on a different server than the one where FS is running. I've seen the server status on the FS server using command line (php…
DouzeBri DouzeBra
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How to set agent status to 'Logged out' when unregister

Is there any way to change Agent status to something else when he unregister? For reference https://wiki.freeswitch.org/wiki/Mod_callcenter#status
Kasinath Kottukkal
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Getting transport Error while trying to connect to Freeswitch using SipML5 api and code

I am trying to call the user 1001 registered on Twinkle using the webpage from chrome. But I am getting Terminated_X_transportError error. JavaScript code is this
Anurag Rana
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Access the name of agent who received the call in a Queue

Is there any way by which i can receive the information about which agent received the call in queue and what is the duration of call and other information. I read about the variables that hold this information like cc_agent,…
Satyajeet
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Counting duration of a freeswitch call, if greater then 10 second nobody joined then play IVR

How to count the duration of call? When a user join 3200 and wait for 10 seconds and nobody yet joined, 3200 then i want to play audio file. But how do i count the duration any idea please? I have tried following but its not working cause it only…
user285594
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FreeSwitch - I can't hear sound on Amazon EC2 server

I've successfully installed FreeSwitch on Local Linux Server (CentOS 6.5), and it worked fine. I could make a call and hear sound from other device. I followed exactly same process on Amazon EC2 CentOS server.…
Jake
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Freeswitch not able to run the SampleClient

I have got hold of the freeswitch sample client for using Freeswitch.Managed.dll from here. Just was able to build it properly but when i run it i get: The type initializer for 'FreeSWITCH.Native.freeswitch' threw an exception. on clicking the…
Nezam
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FreeSwitch Create Extension/SIP User using API

I have been following this http://wiki.freeswitch.org/wiki/Webapi http://wiki.freeswitch.org/wiki/Mod_commands For using the webapi to communicate to the Freeswitch server to execute commands. Isnt there a command to create extensions? If their…
Nezam
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Freeswitch can't connect to external sip profile on server

I am using port 5060 set on internal profile while the external has 5080.I have created extensions on the server. I want user to connect sip server using the external profile defined in external.xml i.e ext_no@XX.YY.ZZ.PP:5080.It says 2013-12-06…
Nezam
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How to record in async mode in freeswitch

I am trying to execute record command in async mode from Java esl, the reason is I have to play music on hold when ever request is processing, and play the wave file to theuser, stopping the moh, I have tried in sync mode but it did not work, I have…
Ravikiran Reddy
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How to send and Receive DTMF in Freeswitch ESL client

I am working with Freeswitch ESL client, I worked on originating call and establishing connection between two applications and making them communicate with each other. I have tried playing sound at one end and recording at the other, It is working…
Ravikiran Reddy
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Writing a Lua script using FreeSWITCH's native PostgreSQL support?

I am learning how to use FreeSWITCH using the FreeSWITCH 1.2 book written by the authors of FreeSWITCH. In Chapter 7, it is explained how to use a Lua script along with connecting to a database. However, I have a feeling this book was written before…
springloaded
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Automatic Conference Freeswitch calling to multiple callers depending on caller

I am new to Freeswitch. Is it possible that a specific user calls on a number, which results to a conference call. And system adds multiple people to this conference call automatically. e.g. User "A" calls at 5656. And whenever user A calls at this…
Saghar
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Location of Config files in Adhearsion and FreeSWITCH

I'm getting on well, hooking up the ruby engine Adhearsion with the telephony engine FreeSwitch. However, the instructions tell me to give some config files a once over. Specifically config.punchblock.platform and the permissions set on the…
Starkers
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Outbound calls using freeswitch

This is my external sip_profile:
Amit
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