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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Cross Compiling freeswitch for ARM on Centos 6.3 using ELDK 5.3 APR.H ssize_t error

I am trying to cross compile freeswitch for ARM using ELDK 5.3 on CentOS 6.3. Found this error on make: ./include/apr.h:347:2: error: #error Can not determine the proper size for ssize_t I already done declaring the following declare -x…
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How do I call a local softphone on freeswitch?

I've set up a local softphone on freeswitch with the extension 1000. It connects and I can play the tetris theme etc. I would like to call this softphone using a freeswitch command. Can anyone help? I know it's to do with sofia and originate…
Tomcomm
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How to work with freeswitch

I should setup a VOIP network. I installed Virtualbox and install ubuntu server on a virtual machine and installed freeswitch on it. I also installed Ekiga softphone on my ubuntu desktop. Now , I need a manual or tutorial to help me how to work with…
Fery
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How to integrate Sphinx 4 with Freeswitch, receive call audio and do some recognition

I have a working knowledge of setting IVR on Freeswitch. I have installed Sphinx 4 and it's working good for Speech captured from microphone. However I want to integrate FS with Sphinx 4. I read somewhere it says that it requires some MRCP server…
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is SDP sendonly means to open one RTP Audio stream in this case?

I am a newbie to sip/sdp world. From my understanding of SDP protocol, if we define a=sendonly from sip server to client softphone, the softphone should open one RTP session for listening, but it should not send any RTP packets to destination. Am I…
xijing dai
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freeswitch configuration for ipkall

i have setup my freeswitch with plivo but now i m unable to configure ipkall number for my sip user 1005@ipadress since the configuration of freeswitch wiki for ipkall is not clear. Please can anybody give me steps to configure? Thanks in advance.
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How to deduct customer balance when has more than one call connected at a time

we do allow him to call based on his current balance, the call is scheduled for 10 min, next call comes in it is also scheduled for 10 min because the customer balance was not updated as his first call is still in process. we only update the balance…
Khawer Zeshan
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Kamailio as dispatcher in front of FreeSwitch

I'm struggling with Kamailio as a simple dispatcher for FreeSwitch. This is my configuration so far: http://pastebin.com/nBPSpe6S Connecting an iPhone and an Android makes the calls between them timeout. Connecting one of the phones and my laptops…
user809829
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Microsoft teams connect with freeswitch

I have configure the fqdn domain for the direct routing with the freeswitch but in that my tls connectivity status is not activating and also error showing in network effectiveness so how can connect the my domain with tls. This status should be…
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How to get node-esl data outside of the connection object?

I have tried and implemented the esl part for freeswitch using this https://github.com/englercj/node-esl/tree/master It works fine just that I want the data outside of the connection object and I cannot get it. This is the…
Ritam
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How to connect Free Swtich PBX Server through sip_ua and webrtc packages in flutter

I need your support that guide step by step. Cause there is no explanation in it's documentation. I have worked with Webrtc in web through following classes such as (RTCpeerconnection) but, in flutter i don't know how to use it.
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How to send external events to FreeSwitch ESL

I have a requirement where User needs to listen to hold music, until a operator approves his call, then ESL can bridge the call, otherwise hangup. Question is how do I tell ESL that operator has approved the call. After reading bit more, what I need…
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FreeSWITCH can not call hotspot user or wifi user

I met a strange question in FreeSWITCH. FreeSWITCH version: 1.10.7, OS: CentOS 7.9 Registered two users A and B, A is hotspot or wifi, B is 4G, B make call A, it is successful, voice and video is ok between A and B, But I call user B by…
bookyao
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How to avoid caching in freeswitch mod_python3?

developing python applications for freeswitch I'm currently struggling with the python cache feature. In my dailplan I'm calling a python script using mod_python3. That script imports further modules which I want to modify and test with the next…
Max
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Got an error trying to install mod_v8 for FreeSwitch 10 in Debian 11: You need to either install libv8-6.1-dev, ibv8fs-dev

On a "clean" Debian 11, I deployed all the necessary packages and began to build FreeSWITCH 10 with the mod_v8 module enabled. When executing the ./configure command, I get the message: checking for v8-6.1_static >= 6.1.298... checking for…
Sly Fox
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