freeswitch | Software Defined Telecom Stack enabling the digital | TCP library
kandi X-RAY | freeswitch Summary
kandi X-RAY | freeswitch Summary
FreeSWITCH is a Software Defined Telecom Stack enabling the digital transformation from proprietary telecom switches to a versatile software implementation that runs on any commodity hardware. From a Raspberry PI to a multi-core server, FreeSWITCH can unlock the telecommunications potential of any device. Combined with our hosted cloud platform, SignalWire, FreeSWITCH can interconnect with the outside world and scale to any size. Visit or for more info.
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Currently covering the most popular Java, JavaScript and Python libraries. See a Sample of freeswitch
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Community Discussions
Trending Discussions on freeswitch
QUESTION
I have code that is working targeted to .NET Framework 4.0, which is no longer MS supported. You can not even install .NET Framework 4.0 SDK to newest versions of Visual Studio. I am attempting to update to a later framework, preferably 4.7.2 for widest compatibility, but 4.6.1 would be acceptable as I can at least have the SDK available in my IDE. Code compiles without any errors. 64 bit would be a nice bonus, but not a requirement.
The error happens at the first call to initialize the FreeSWITCH.Managed.dll as outlined here:
Relevant code here:
...ANSWER
Answered 2022-Feb-28 at 22:28Actually got this working now.
The answer was fairly straight forward, I updated the DLL to the latest from FreeSwitch instead of trying to use the same ancient version the old .NET 4.0 code had been using.
The newer DLL will only run as 64 bit whereas the old ones would only run in 32 bit. So keeping an archive copy of the old programs for any 32 bit users will be necessary, but can finally get to actually updating the code now.
There is an MSI installer that has all the updated libraries linked from the FreeSwitch webpage for Windows installation.
FreeSwitch Windows Install Page
Do Note: With the updated DLL, all the ref keyword on calls to the FreeSwitch libraries will need to be changed to out keyword.
QUESTION
I have installed Freeswitch-mod-python3 on my freeswitch box running version 1.10.7. I then uncommented out the python3 line in the module.conf file. Then did a reloadxml, this reloaded successfully. When I do a "module_exists mod_python3" I get false. Then I tried to load it and the error I get is below, which is weird cause python3.so is a file in that directory and has the same permissions as every other file in that dir. I have rebooted the box as well as restarted FS and no luck. What else can I try?
...ANSWER
Answered 2022-Feb-22 at 06:02This worked for me:
apt-get install freeswitch-mod-python3
Then I loaded the module by adding it to the end of my autoload_configs/modules.conf.xml
QUESTION
As the title says, I have a problem of getting dtmf digits from b-leg. When I call a queue in callcenter, the two legs can be bridged. But It doesn't have any reaction when I press '*7' on my b-leg telephone.
Here is the dialplan:
...ANSWER
Answered 2021-Nov-03 at 03:04I know how to fix it now. It's cause by freeswitch set 2833 dtmf send payload to 105 recv payload to 101, but the telephone still send 105 dtmf payload type to freeswitch. I modified 'rfc2833-pt' in sip_profiles/internal.xml to '105' and it worked for me.
QUESTION
Is it possible from FreeSWITCH CHANNEL_HANGUP event (or others) determine who has disconnected a call?
...ANSWER
Answered 2021-Oct-04 at 09:12I have resolved this by inspecting the following SIP header - 'sip_hangup_disposition'
Link to the corresponding documentation https://freeswitch.org/confluence/display/FREESWITCH/sip_hangup_disposition
QUESTION
In Amazon connect, I need to pass call flow to an external application via a socket and control the call from that application .
Something like ESL
in Freeswitch
: Event Socket Library
For those who do not know what ESL is, it pass the call to a socket in external application and get command from that application like Play
, Disconnect
And then all commands are available in ESL
library in Freeswitch
.
Does amazon connect have such ability?
Thanks
...ANSWER
Answered 2021-Sep-18 at 16:15Yes, you can start connection using websockets
As per AWS manual for the Amazon Connect Participant Service
:
SampleMethod
CreateParticipantConnection
creates the participant's connection. Note that ParticipantToken is used for invoking this API instead of ConnectionToken.The response URL for
WEBSOCKET
Type has a connect expiry timeout of 100s. Clients must manually connect to the returned websocket URL and subscribe to the desired topic.Upon websocket URL expiry, as specified in the response ConnectionExpiry parameter, clients need to call this API again to obtain a new websocket URL and perform the same steps as before.
Request Syntax
``POST /participant/connection HTTP/1.1 X-Amz-Bearer:
ParticipantToken
Content-type: application/json{ "Type": [ "
string
" ] }``URI Request Parameters
The request uses the following URI parameters.
This is a header parameter.
The Participant Token as obtained from StartChatContact API response.
Length Constraints: Minimum length of 1. Maximum length of 1000.
Required: Yes
Request BodyThe request accepts the following data in JSON format.
Type of connection information required.
Type: Array of strings
Array Members: Minimum number of 1 item.
Valid Values:
WEBSOCKET | CONNECTION_CREDENTIALS
GO
code
QUESTION
I am new to FreeSWITCH and I am trying to apply click-to-call in my NodeJs application.
This is my code when executing click-to-call, I use the 'originate command', this is my reference
...ANSWER
Answered 2021-Aug-26 at 06:06Use the CHANNEL_HANGUP_COMPLETE event for this purpose.you:
it gives following response ...
variable_billsec: 17
variable_billmsec: 16660
variable_billusec: 16660036
QUESTION
I've got a js Freeswitch script where caller has to call another number after a digit.
I've just made a bridge in this way
ANSWER
Answered 2021-May-23 at 14:40QUESTION
I'm trying to get rid of controlling audio played via session.streamFile() in Freeswitch. For this I tried the 3rd example of this documentation.
Almost everything here is working, but the DTMF 1 (seek:-500) doesn't seek back. It always starts from the beginning (like seek:0). What could be the reason here? I'm using exactly the example.
Code:
...ANSWER
Answered 2021-May-16 at 14:32Meanwhile I found the reason. It belongs to the samplerate. Freeswitch prefers wav's with a samplerate of 16000. Others are working too, but makeing trouble with seeking etc.
QUESTION
I know that Kamailio works as a SIP Proxy and I also know that Asterisk/FreeSWITCH or other similar products can do what I'm asking here, but still wondering if it's possible to use Kamailio to answer a call or originate a call?
Let's say, "User ---> Kamailio", Kamailio answer the call with "200 OK to INVITE", and then play an announcement "Thank you for using Kamailio", then BYE the call?
I can't find any related information so I guess the answer is no, but still want to double check.
Many thanks in advance.
...ANSWER
Answered 2021-Feb-27 at 22:59You can answer and play message if you use kamailio with mediaproxy.
Mediaproxy docs have some samples.
Originate is problematic and unreasonable in most cases, but there is https://www.kamailio.org/docs/modules/devel/modules/rtimer.html
Much simple method is use SIPP or perl/python sip module(external), after that rewrite call in kamailio as you wish.
QUESTION
I have installed bigbluebutton on my serverand it was working properly but suddenly microphone can not connect anymore and after a while being stucked in "echo test" i'll get 1006 error and i have tested:
...ANSWER
Answered 2020-Aug-30 at 16:52run the command:
sudo nano /etc/bigbluebutton/nginx/sip.nginx
something like this will open:
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Debian 10 Buster [Recommended]
Raspberry Pi
CentOS 7
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