Stack Overflow
Stack Overflow

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

674 questions
0
votes
1 answer

Freeswitch SIP User creation issue

I am running some tests on the freeswitch PBX on Ubuntu Precise Pangoline server. The version of the Freeswitch I am using is the one from the git repo. I added a SIP user and from a SIP phone to access the Freeswitch I keep on getting the following…
Arsene
  • 1,037
  • 4
  • 20
  • 47
0
votes
1 answer

How do I get FreeSWITCH to workout the sofia profile for me from Originate

I have some code using the FreeSWITCH api over sockets. I am originating calls. When I use my SIP phone to test things I can simply enter a number and mod_dialplan_xml kicks in on FreeSWITCH to work out the necessary sofia profiles. When I want to…
C Hall
  • 175
  • 2
  • 15
0
votes
1 answer

Change to three way calling after monitoring two way conversation using ChanSpy ---Asterisk---

I have a scenario that i'd like to implement using Asterisk. ↓↓ I'd like to have 3 participants on a phone call: A, B and C. A and B should be able to talk to and hear each other. C should be able to hear conversation between A and B, and also able…
Alpha3Omega
  • 91
  • 1
  • 1
  • 3
0
votes
0 answers

Java API like smack using XMPP to retrieve online GTalk-chat friends can receive audio or video call

I am working on a module which is using inbuilt Freeswitch utility to call An Online GtalkUser. I am in search of quick API interface to verify whether user's browser is enabled for Voice or Video Call( Browser requires gtalk-audio/video plugin, if…
pkm1986
  • 195
  • 2
  • 6
0
votes
1 answer

Freeswitch telcentris

I've been trying to configure FreeSWITCH with telcentris but not working. FreeSWITCH Currently it works perfectly with other providers. I tried using the XML configuration that is in the official website (here the link:…
Sansa
  • 11
  • 1
  • 3
0
votes
1 answer

converting freeswitch to asterisk

I have the following code in freeswitch. We have decided to use asterisk instead. I've setup so that if you dial 8XXXX you will dial the other server. sip1:/usr/local/freeswitch/conf/autoload_configs/acl.conf.xml
liv2hak
  • 14,472
  • 53
  • 157
  • 270
0
votes
2 answers

Setting duration of a freeswitch call

I'm a bit new on freeswitch but I'd like to call a number, play a sound and hang up after a certain amount of time. It could be that the call lasts longer or shorter then the sound file. I was hoping to do it with Javascript and I got as far as…
Tomcomm
  • 233
  • 2
  • 6
  • 16
0
votes
2 answers

Freeswitch with Skype Connect external profile keeps timing out and disconnecting

I have a Skype Connect SIP profile set up on a small Freeswitch setup. All seems fine - I can make incoming and outgoing calls. Problem is - if there is no activity for a while, incoming calls time out and drop. That is, the caller hears a long…
user1731782
  • 322
  • 2
  • 12
0
votes
1 answer

Logging ISDN events with dahdi and freeswitch

I am using freeswitch with dahdi. I have a need to log the ISDN events for all the calls that land on the PBX (freeswitch). Is there any way to do this?. I thank you kindly for your help. Thanks in advance.
karthi_ms
  • 5,568
  • 11
  • 38
  • 39
0
votes
2 answers

How can I get php_ESL.dll and load it from php in XAMPP to use it with Mod event socket for freeswitch

I have freeswitch running on a Centos 6 remote server and I have XAMPP installed on the local machine ( windows 7 x64 PHP Version 5.3.8 ) for testing. I am trying to use Mod event socket ( http://wiki.freeswitch.org/wiki/Event_Socket ) to connect to…
Berry
  • 59
  • 1
  • 10
0
votes
1 answer

Why SDP not match when a msrp invite coming to the freeswitch

When i make a call that from 1016 to 1015 ,and my voip phone support msrp,so i want to send text-message form one leg to the other. I find that FreeSwitch handle the incoming message as a normal sip invite and will generate a session (also a…
argpunk
  • 1
0
votes
1 answer

making multiple incoming calls from PSTN

I have VoIP test setup that uses a media gateway and a softswitch (YATE). The media gateway converts incoming calls from PSTN on an E1 PRI to SIP INVITEs and send them to the softswitch which forwards the INVITEs to the clients that have registered…
John Qualis
  • 1,663
  • 4
  • 33
  • 52
-1
votes
0 answers

FreeSWITCH on MacOS Machin : - error libfreeswitch.la. linker command failed with exit code 1 (use -v to see invocation) make Command fire

I try to set up FreeSWITCH on MacOS Machin. I am following the step to setup FreeSWITCH - text Still, everything going properly way but I am stuck hitting the "make" command to get the linker error linker command failed with exit code 1 (use -v to…
Vivek Shukla
  • 7
  • 1
  • 3
-1
votes
2 answers

Get CallerID while calling out of a Dinstar GSM/SIP gateway

I am using freeswitch to dial calls through a dinstar GSM gateway with 32 sim cards. Calls works fine, and it goes out through any one of the free channels. However, the customer wants to know the cell number of the SIM card through which each call…
Sharath
  • 1,627
  • 2
  • 18
  • 34
-1
votes
1 answer

freeswitch.Dbh cause freeswitch auto shudtown

When I freeswitch test lua odbc mysql in my docker, when exec to freeswitch.Dbh("db","user","password"), the freeswitch auto shutdown. Anybody met this issue, can give me some advice? Thank you.
JACK
  • 1
1 2 3
…
44 45