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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Freeswitch verto performance is low

This is my first question, thanks in advance for all your help! I setup a freeswitch server, and call it from a web page, I found freeswitch server costs more than 13 percent cpu, when it's idle it costs less than 1 percent, if I use a sip client…
Mark
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I am writing an IVR using Lua and want to run xml script in between

This is the Lua code which runs IVR i = 0 j = 0 while (session:ready() == true) do session:setAutoHangup(false); repeat digits = session:playAndGetDigits(1, 3, 3, 3000, "#",…
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bigbluebutton ...

Hello i have a BigBlueButton (0.9.0-beta (622)) installation on Debian Wheezy (7.8) all is ok except archiving recordings... in the log (/var/log/bigbluebutton/archive-488052dc7c095c74bf8992ec51a66298db04b765-1425642166675.log) i have always same…
signo
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Block unknown SIP request in pfsense

I have installed freeswitch on a machine which is kept in the LAN behind the firewall named PFSense.I was looking in the freeswitch logs and I found that there are some unknown sip request coming from the unknown IP from outside of the network. I…
Vinay Saini
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Change tcp/udp port for user

I am using the Linksys SPA3102 with my freeswitch raspberry pi. The PSTN line uses port 5061. I have created a user with ID "23" for the PSTN line however I believe I need to have that user use port 5061. Is there a way to set the port for a…
MikeKulls
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Freeswitch Issues incoming call Abandoned

I have a server with FS, my problem is making a call to FreeSWITCH from my personal phone. The incoming call is detailed by FS, but it doesn’t process throw DialPlan. Mi server IP is 166.227.33.23 and the incoming call is to throw 65.34.236.149 as…
Raimundo
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WebRTC websocket SSL certs use to work but not any more? ERR_INSECURE_RESPONSE

I am using sipml5 to register sip endpoints and using sipml5 as my client and FreeSwitch as my SIP server. I use to be able to register my sip endpoint and establish a connection, but recently I been getting this error in my chrome and Firefox dev…
ToyeO
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How are C files included in freeSwitch (kind of framework for calls)?

Today I downloaded Freeswitch that (based on what I read) allows me to have a messaging (audio and video calls) server in my computer. I am not very good in programming. When analysing the files contained in its .zip file, I noticed that almost all…
youssouf
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mobicent diamater with asterik , freepbx

I have already used Ericsson Ericsson Diameter SDK to check the charging system from below example: http://www.mobicents.org/diameter_event_charging.html Can any one guide how how i put mobicent dimameter with my freepbx, asterik to charge a call?
new developer
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Importing CDR from XML files in MySQL

Is there any script available which can be used to load xml cdr from log dir into mysql db? I found some scripts in /contrib but they are not loading it in db. I would really appreciate if someone can share their script.
user3310052
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How to install CDR-STATS on Exiting CentOS FreeSWITCH Server Manually without Bash Script

I am trying to install CDR-STATS on Exiting CentOS 6.4 server where FreeSWITCH is already running well, i read all available documentation and mailing list but i didn;t find any notes where can i install step by step. Please any one know help me to…
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Using variables in exec.Command Golang

I am trying to use variables within the string i have tried the following to no avail :( astatus, _ := exec.Command("/bin/sh", "-c", "fs_cli -x\"callcenter_config agent get status\"", agent_uuid).Output() astatus, _ := exec.Command("/bin/sh", "-c",…
MikeyR
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How to resolve "LUA script parse/execute error" during Freeswitch installation?

2016-10-24 02:37:33.733197 [INFO] mod_dialplan_xml.c:637 Processing 1985922714 <1985922714>->4595609013 in context default 2016-10-24 02:37:33.753200 [ERR] switch_odbc.c:368 STATE: 01000 CODE 0 ERROR: [unixODBC][Driver Manager]Can't open lib…
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looking for a way to determine if the phone number i'm calling is a cell phone or not

Im looking for a way to determine if the phone number I'm calling is a cell/mobile phone or not since it is not allowed to automatically dial those type of numbers.
Summers
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