667 questions
1
vote
0
answers
24
views
Freeswitch mask and unmask commands doesnt work when recording is started by lua script
I am trying to mask and unmask the call recordings which is being recorded automatically through some lua scrip using the command record_session. the lua script execut this command when the call is ...
- 149
1
vote
1
answer
30
views
conference_auto_record sometimes not working when use api_on_answer uuid_transfer to conference
Freeswitch, conference_auto_record sometimes won't work when I transfer a connected call to conference:
set conference_auto_record=fsRecordAudioPath
bridge {api_on_answer='sched_api +1 none ...
- 13
0
votes
1
answer
48
views
Freeswitch check if message received is "INFO" in dialplan
I'm trying to check in the dialplan if an incoming message to freeswitch is an "INFO" message. If the condition is true i need to execute a local script.
Is possible to do this?. Maybe ...
- 127
0
votes
0
answers
33
views
SIP.JS + Freeswitch doesn't send Notify for PRESENCE?
I'm using SIP.js to implement WebRTC functionality. I have implemented the SUBSCRIBE method, but FreeSWITCH doesn't send any NOTIFY packet to the SIP.js client. However, when I test the same ...
- 976
0
votes
1
answer
83
views
Config load-balancer Kamailio + FreeSwitch for large conference
I'm using Kamailio and 2 FreeSwitches.
I configured the loadbalance for normal calls and it worked fine.
Use the dispatcher module:
# Dispatch requests
route[DISPATCH] {
# round robin dispatching ...
- 39
0
votes
0
answers
40
views
how to create conference for bridge call after answer
freeswitch, how to create conference for bridge call after answer, configure on dialplan. I tried all methods but no success:
<action name="api_on_answer" value="uuid_transfer ${uuid}...
0
votes
0
answers
56
views
I'm trying to use pjsip for video calling in react native. But it doesn't work
In my React native project, I want to access freeswitch data using pjsip and make video calls between two phones. Actually, one is a phone and the other is a doorbell panel with camera. Does anyone ...
0
votes
1
answer
163
views
Use custom generated call_id in the Sipp scenarion send Invite instead
I'm using Sipp to run some test cases. In my use case, the Sipp scenario tests a SIP Invite sent to a remote server and validate for 100, 180 and 200 OK finally, a basic uac.xml
The remote end point I'...
- 86
0
votes
0
answers
56
views
Expose ports dynamically to the internet inside Kubernetes
I'm working on deploying a VoIP app to Kubernetes. The app listens on the well-known port 5060/udp for SIP messages and then selects a dynamic port between 16000 and 17000 to receive RTP packets.
I ...
- 1,869
0
votes
1
answer
204
views
Freeswitch Lua API - how to get Channel State?
Using Freeswitch, I am desperately trying to get the channel Call State value in Lua.
The Lua script is executed by posting an HTTP request to Verto..
This is the Lua code I have, so far:
api = ...
- 93
0
votes
1
answer
62
views
Freeswitch basic B2BUA
I'm new to freeswitch and trying to figure it out.
I'm having issues getting a basic configuration/dialplan going in order for a new request coming in from SIPP (LEG A) to be routed to its destination ...
- 349
0
votes
0
answers
54
views
How to include require:timer in the 200 OK message after receiving an UPDATE message in freeswitch
FS ：Version 1.10.11-release 64bit
I need to reply to the IMS server when receiving an UPDATE message and add require:timer to the message.
I also used
<action application="set" data=&...
1
vote
0
answers
65
views
How to replace frame in FreeSwitch?
I tried to implement end to end encryption to freeswitch, I attempt to decrypt audio payload, after I log my decryption function, it seems that I can decrypt the payload. Now I am confused how to ...
-1
votes
2
answers
122
views
How to send dtmf and play audio stream like ivr using websocket send to freeswitch?
I'm building a call application using webrtc to freeswitch using java as sip gateway, I want to implement ivr function but don't know how to send dtmf to freeswitch via websocket on java. I cannot ...
0
votes
0
answers
149
views
Custom Conference Room in fusionPBX
I have two issues going on
I want to create a conference room that calls an external number as soon as anyone joins the room,
if the external call fails, the user should be directed to another fixed ...
0
votes
0
answers
156
views
Sending message from FusionPBX to sip agent
I am using FusionPBX as SIP server(registar). FusionPBX works fine for call and SIMPLE message routing, but I need to modify it so that I can send SIP/SIMPLE messages to registered users from ...
- 26
0
votes
1
answer
387
views
How to fix FreeSWITCH advertising private ip address in the SDP
I have the following setup: an AWS Application Load Balancer (ALB) forwarding WebRTC connections to FreeSWITCH. The ALB also acts as an SSL terminator. Given that the ALB and FreeSWITCH (EC2) are in ...
- 1,869
0
votes
1
answer
155
views
Unable to catch the CHANNEL_ANSWER event In node js script from the FREESWITCH
I hope everybody is fine. I have a question about freeswitch.
I have a very minimal experience in freeswitch and working to connect my node js script to the freeswitch server. I have successfully ...
- 1
0
votes
1
answer
138
views
GoESL with Temporal: Calls Not Originating Past Certain Point in FreeSWITCH
I'm integrating GoESL (https://github.com/0x19/goesl) with Temporal to automate dialing through FreeSWITCH. The setup allows for 1,000 concurrent channels and 50 calls per second (CPS). Each dial ...
1
vote
0
answers
186
views
Latest FreeSWITCH versions have problems with conferences
We have issues with FreeSWITCH 1.10.1 dealing with multiple participants in conferences. Randomly we get a lot of disturbing noise that disappears after a few seconds or even minutes.
We did a lot of ...
- 446
2
votes
0
answers
249
views
Adding P-Early-Media on FreeSwitch
How can i modify the dial plan / sofia profile to insert the P-Early-Media Headers on Freeswitch?
I want to integrate with 3GPP base telco core, so I want to when using pre-answering add P-Early-Media ...
- 2,953
0
votes
1
answer
104
views
Detect if user's phone ringed for outbound call in freeswitch javascript
I'm using javascript in Freeswitch. I'm finding the way to detect user's phone ringed for outbound call. Any chance to detect if session is ringing or detect callstate change from DOWN->RINGING ? . ...
- 85
0
votes
0
answers
172
views
Freeswitch Hangup issue
I run a Python script that plays iVR and bridges to the available extension (agent) in my freeswitch. It functions correctly, but when the customer or agent hangs up, the call is still in an active ...
1
vote
0
answers
230
views
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 ...
- 31
1
vote
0
answers
212
views
Freeswitch Play a sound in ring group when brigde multi call
Im using freeswitch to build our PBX.
My case when have incomming call to a ring group with strategy is rollover, i want to play just one ringtone file when call bridge from user A->B->C in ring ...
- 327
1
vote
0
answers
62
views
How to get node-esl data outside of the connection object?
I have tried and implemented the esl part for freeswitch using this.
It works fine just that I want the data outside of the connection object and I cannot get it.
This is the code:
connection.on('esl::...
- 11
0
votes
1
answer
667
views
How can I use sip.js connect freeswitch with wss
I am trying to register with the server using sipjs.It can works with ws,but when i use wss,some errors occured.I have used a self-signed certificate wss.pem to instead the old one
freeswitch error
...
- 1
0
votes
1
answer
124
views
Giving Error when try to insert CDR using mod_cdr_mongodb in freeswitch
I am trying to insert CDR into MongoDB using mod_cdr_mongodb. Without setting the username and password, I am able to load the mod_cdr_mongodb module and insert CDR after a hangup call into the ...
- 852
2
votes
0
answers
161
views
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 ...
0
votes
1
answer
577
views
WebRTC causes FreeSWITCH to not provide service
I am using two FreeSWITCH clusters, where the difference between them is that Cluster A uses TCP protocol for SIP and Cluster B uses webRTC protocol for SIP. Recently, there have been several ...
0
votes
1
answer
99
views
Escape characters in Freeswitch DSN string. (using maria_db)
I am using the maria_db module to connect freeswitch to a db backend.
This works fine, but I am now needing to connect to a database where there is an equals sign in the password.
How can I escape the ...
- 83
1
vote
0
answers
177
views
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 ...
0
votes
1
answer
247
views
ERROR : "No Route, Aborting" when handling incoming call
I am trying my custom configuration to handle freeswitch call using Elastic SIP Trunking. I am not using mod_Signalwire. I want to handle route custom with multi-tenant system.
Below is my ...
- 609
0
votes
0
answers
70
views
New C file creation in freeswitch
I am creating new .c file in src/ where other in-built files like switch_core_hash.c and etc. are located. Also I am creating relevant header file in src/include where other in-built .h files are ...
3
votes
0
answers
446
views
Subscribe user and get Notify when user register or unregister
I want to subscribe presence and dialog of all sip user and get notify when user on call , available, on ringing and unregister.
I am using sip.js with reactjs.
SIPJS : "sip.js": "^0.20....
- 239
0
votes
1
answer
231
views
Unable to transfer a call from one conference room to another using fs_cli
I want to transfer a call that is present in one conference room to another conference room that is active. Currently, I'm using the uuid_transfer command in the fs_cli to transfer the call but the ...
1
vote
0
answers
107
views
freeswitch mod_xml_curl.c:468 Binding has no url
I am new to freeswitch and want to have a dynamic user directory. For this I want to use mod_xml_curl but when I load it I always get this error.
For information my freeswitch runs in a docker. I ...
- 594
1
vote
1
answer
197
views
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 ...
- 13
1
vote
0
answers
148
views
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 ...
- 11
0
votes
2
answers
201
views
Unable to access django server on debian running FusionPBX
I have a Debian server running fusionPBX, I installed it using this official script. It uses nginx to host the application, I tried adding another server (Django) in the config file of nginx to ...
0
votes
1
answer
562
views
ACK message not routes when running opensips as mid_resistrar for freeswitch
This is my first time using opensips, so am probably missing something obvious.
I am trying to use opensips and mid_registrar in front of freeswitch. The opensips is behind a NAT so I am setting the ...
- 57
-1
votes
2
answers
553
views
Get CallerID while calling out of a Dinstar GSM/SIP gateway [closed]
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 ...
- 1,775
0
votes
0
answers
236
views
Contradictory settings in FreeSWITCH SIP profile
I am working with a simple FreeSWITCH installation, with the vanilla demo configuration. I see that the Internal SIP profile contains:
<param name="apply-inbound-acl" value="...
- 4,584
0
votes
1
answer
261
views
Basic Lua/ FREESWITCH array command
I am new to Lua, and this is an elementary question.
I a Lua script, I am querying the Postgress DB for two records via Freeswitch dbh.
I am able to set the first value as a local variable.
I am stuck ...
- 1
1
vote
3
answers
774
views
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:
...
- 21
0
votes
2
answers
909
views
[RTPEngine]Failed to init DTLS connection: key values mismatch
i got the connection like this
sip flow
freeswitch server --(sip)-- > opensips ----(wss)---> sip client in chrome with jssip/webrtc
the rtp flow
freeswitch server ---- > rtpengine----...
- 21
0
votes
1
answer
692
views
(FreeSWITCH) [ERR] mod_event_socket.c:2992 Socket Error! Could not listen on 127.0.0.1:8021
I'm getting following error in freeswitch.log file while connecting microphone in BigBlueButton 2.4.9 meeting.
[ERR] mod_event_socket.c:2992 Socket Error! Could not listen on 127.0.0.1:8021
I'm not ...
- 1
0
votes
1
answer
1k
views
FreeSWITCH may not be responding to requests on port 8021
I'm getting below error while restarting the BigBlueButton server:
Starting BigBlueButton
Job for freeswitch.service failed because the control process exited with error code.
See "systemctl ...
- 1
1
vote
0
answers
112
views
Can we get frreeSWITCH outbound call duration from an API
I am trying to find an API or a method that can get the call duration from FreeSWITCH. As my search, we can get call duration from fs_cdr table or calculate the time between CHANNEL_HANGUP - ...
- 308
0
votes
1
answer
577
views
freeswitch contact header (null) user
i have a freeswitch server and am using external profile to register extension 1000 from SIP.js
the connection goes well and i am able to see auth etc. passing through, however the response of ...
- 115