I have installed and configured jigasi. But i can’t make sip call from jitsi meet.
I have followed the procedure described here.
https://github.com/jitsi/jigasi
After installing this jigasi module in jitsi meet i have new button (+ button) in lower right corner of my meet room. using that button i tried to call invite people to join through gsm call.
After pressing this button it asks a number to call. i put some number and there and try to dial it says “Failed to add participant”
if i inspect the page it shows an error like below
Logger.js:154 2020-04-22T05:41:49.913Z [features/invite] Error inviting phone number:
The sip configuration of my jigasi is something like this
#Sample config with one XMPP and one SIP account configured
#Replace {sip-pass-hash} with SIP user password hash
#as well as other account properties
#Name of default JVB room that will be joined if no special header is included
#in SIP invite
org.jitsi.jigasi.DEFAULT_JVB_ROOM_NAME=siptest
net.java.sip.communicator.impl.protocol.SingleCallInProgressPolicy.enabled=false
#Should be enabled when using translator mode
#net.java.sip.communicator.impl.neomedia.audioSystem.audiosilence.captureDevice_list=[“AudioSilenceCaptureDevice:noTransferData”]
#Adjust opus encoder complexity
net.java.sip.communicator.impl.neomedia.codec.audio.opus.encoder.COMPLEXITY=10
#Disables packet logging
net.java.sip.communicator.packetlogging.PACKET_LOGGING_ENABLED=true
net.java.sip.communicator.impl.protocol.sip.acc1403273890647=acc1403273890647
net.java.sip.communicator.impl.protocol.sip.acc1403273890647.ACCOUNT_UID=SIP:user@sipswitch.mydomain.com:2460
net.java.sip.communicator.impl.protocol.sip.acc1403273890647.PASSWORD=BASE64PASS
net.java.sip.communicator.impl.protocol.sip.accl403273890647.authorization_name=user@sipswitch.mydomain.com:2460
net.java.sip.communicator.impl.protocol.sip.acc1403273890647.PROTOCOL_NAME=SIP
net.java.sip.communicator.impl.protocol.sip.acc1403273890647.SERVER_ADDRESS=sipswitch.mydomain.com:2460
net.java.sip.communicator.impl.protocol.sip.acc1403273890647.USER_ID=user@sipswitch.mydomain.com:2460
net.java.sip.communicator.impl.protocol.sip.acc1403273890647.KEEP_ALIVE_INTERVAL=25
net.java.sip.communicator.impl.protocol.sip.acc1403273890647.KEEP_ALIVE_METHOD=OPTIONS
net.java.sip.communicator.impl.protocol.sip.acc1403273890647.VOICEMAIL_ENABLED=false
net.java.sip.communicator.impl.protocol.sip.acc1403273890647.Encodings.AMR-WB/16000=750
net.java.sip.communicator.impl.protocol.sip.acc1403273890647.Encodings.G722/8000=700
net.java.sip.communicator.impl.protocol.sip.acc1403273890647.Encodings.GSM/8000=0
net.java.sip.communicator.impl.protocol.sip.acc1403273890647.Encodings.H263-1998/90000=0
net.java.sip.communicator.impl.protocol.sip.acc1403273890647.Encodings.H264/90000=0
net.java.sip.communicator.impl.protocol.sip.acc1403273890647.Encodings.PCMA/8000=600
net.java.sip.communicator.impl.protocol.sip.acc1403273890647.Encodings.PCMU/8000=650
net.java.sip.communicator.impl.protocol.sip.acc1403273890647.Encodings.SILK/12000=0
net.java.sip.communicator.impl.protocol.sip.acc1403273890647.Encodings.SILK/16000=0
net.java.sip.communicator.impl.protocol.sip.acc1403273890647.Encodings.SILK/24000=0
net.java.sip.communicator.impl.protocol.sip.acc1403273890647.Encodings.SILK/8000=0
net.java.sip.communicator.impl.protocol.sip.acc1403273890647.Encodings.VP8/90000=0
net.java.sip.communicator.impl.protocol.sip.acc1403273890647.Encodings.iLBC/8000=10
net.java.sip.communicator.impl.protocol.sip.acc1403273890647.Encodings.opus/48000=1000
net.java.sip.communicator.impl.protocol.sip.acc1403273890647.Encodings.red/90000=0
net.java.sip.communicator.impl.protocol.sip.acc1403273890647.Encodings.speex/16000=0
net.java.sip.communicator.impl.protocol.sip.acc1403273890647.Encodings.speex/32000=0
net.java.sip.communicator.impl.protocol.sip.acc1403273890647.Encodings.speex/8000=0
net.java.sip.communicator.impl.protocol.sip.acc1403273890647.Encodings.telephone-event/8000=1
net.java.sip.communicator.impl.protocol.sip.acc1403273890647.Encodings.ulpfec/90000=0
net.java.sip.communicator.impl.protocol.sip.acc1403273890647.OVERRIDE_ENCODINGS=true
net.java.sip.communicator.impl.protocol.sip.acc1403273890647.DEFAULT_ENCRYPTION=false
#If an authenticated (hidden) domain is used to connect to a conference,
#PREVENT_AUTH_LOGIN will prevent the SIP participant from being seen as a
#hidden participant in the conference
#net.java.sip.communicator.impl.protocol.sip.acc1403273890647.PREVENT_AUTH_LOGIN=FALSE
#Used when incoming calls are used in multidomain environment, used to detect subdomains
#used for constructing callResource and eventually contacting jicofo
net.java.sip.communicator.impl.protocol.sip.acc1403273890647.DOMAIN_BASE=<<DOMAIN_BASE>>
#the pattern to be used as bosh url when using bosh in multidomain environment
#net.java.sip.communicator.impl.protocol.sip.acc1403273890647.BOSH_URL_PATTERN=https://{host}{subdomain}/http-bind?room={roomName}
#can be enabled to disable audio mixing and use translator, jigasi will act as jvb, just forward every ssrc stream it receives.
#net.java.sip.communicator.impl.protocol.sip.acc1403273890647.USE_TRANSLATOR_IN_CONFERENCE=true
#We can use the prefix org.jitsi.jigasi.xmpp.acc to override any of the
#properties that will be used for creating xmpp account for communication.
#The following two props assume we are using jigasi on the same machine as
#the xmpp server.
org.jitsi.jigasi.xmpp.acc.IS_SERVER_OVERRIDDEN=true
org.jitsi.jigasi.xmpp.acc.SERVER_ADDRESS=127.0.0.1
org.jitsi.jigasi.xmpp.acc.VIDEO_CALLING_DISABLED=true
org.jitsi.jigasi.xmpp.acc.JINGLE_NODES_ENABLED=false
org.jitsi.jigasi.xmpp.acc.AUTO_DISCOVER_STUN=false
org.jitsi.jigasi.xmpp.acc.IM_DISABLED=true
org.jitsi.jigasi.xmpp.acc.SERVER_STORED_INFO_DISABLED=true
org.jitsi.jigasi.xmpp.acc.IS_FILE_TRANSFER_DISABLED=true
#Or you can use bosh for the connection establishment by specifing the URL to use.
#org.jitsi.jigasi.xmpp.acc.BOSH_URL_PATTERN=https://server.com/http-bind?room={roomName}
#Used when outgoing calls are used in multidomain environment, used to detect subdomains
#org.jitsi.jigasi.xmpp.acc.DOMAIN_BASE=<<DOMAIN_BASE>>
#org.jitsi.jigasi.xmpp.acc.BOSH_URL_PATTERN=https://{host}{subdomain}/http-bind?room={roomName}
#can be enabled to disable audio mixing and use translator, jigasi will act as jvb, just forward every ssrc stream it receives.
#org.jitsi.jigasi.xmpp.acc.USE_TRANSLATOR_IN_CONFERENCE=true
#If you want jigasi to perform authenticated login instead of anonymous login
#to the XMPP server, you can set the following properties.
#org.jitsi.jigasi.xmpp.acc.USER_ID=SOME_USER@SOME_DOMAIN
#org.jitsi.jigasi.xmpp.acc.PASS=SOME_PASS
#org.jitsi.jigasi.xmpp.acc.ANONYMOUS_AUTH=false
#If you want to use the SIP user part of the incoming/outgoing call SIP URI
#you can set the following property to true.
org.jitsi.jigasi.USE_SIP_USER_AS_XMPP_RESOURCE=true
#Activate this property if you are using self-signed certificates or other
#type of non-trusted certicates. In this mode your service trust in the
#remote certificates always.
net.java.sip.communicator.service.gui.ALWAYS_TRUST_MODE_ENABLED=true
#Enable this property to be able to shutdown gracefully jigasi using
#a rest command
#org.jitsi.jigasi.ENABLE_REST_SHUTDOWN=true
#Options regarding Transcription. Read the README for a detailed description
#about each property
#org.jitsi.jigasi.ENABLE_TRANSCRIPTION=false
org.jitsi.jigasi.ENABLE_SIP=true
#whether to use the more expensive, but better performing
#“video” model when doing transcription
#org.jitsi.jigasi.transcription.USE_VIDEO_MODEL = false
#delivering final transcript
#org.jitsi.jigasi.transcription.DIRECTORY=/var/lib/jigasi/transcripts
#org.jitsi.jigasi.transcription.BASE_URL=http://localhost/
#org.jitsi.jigasi.transcription.jetty.port=-1
#org.jitsi.jigasi.transcription.ADVERTISE_URL=false
#save formats
#org.jitsi.jigasi.transcription.SAVE_JSON=false
#org.jitsi.jigasi.transcription.SAVE_TXT=true
#send formats
#org.jitsi.jigasi.transcription.SEND_JSON=true
#org.jitsi.jigasi.transcription.SEND_TXT=false
#translation
#org.jitsi.jigasi.transcription.ENABLE_TRANSLATION=false
#record audio. Currently only wav format is supported
#org.jitsi.jigasi.transcription.RECORD_AUDIO=false
#org.jitsi.jigasi.transcription.RECORD_AUDIO_FORMAT=wav
#execute one or more scripts when a transcript or recording is saved
#org.jitsi.jigasi.transcription.EXECUTE_SCRIPTS=true
#org.jitsi.jigasi.transcription.SCRIPTS_TO_EXECUTE_LIST_SEPARATOR=","
#org.jitsi.jigasi.transcription.SCRIPTS_TO_EXECUTE_LIST=script/example_handle_transcript_directory.sh
#filter out silent audio
#org.jitsi.jigasi.transcription.FILTER_SILENCE = false
#properties for optionally sending statistics to a DataDog server
#org.jitsi.ddclient.prefix=jitsi.jigasi
#org.jitsi.ddclient.host=localhost
#org.jitsi.ddclient.port=8125
#sip health checking
#Enables sip health checking by specifying a number/uri to call
#the target just needs to auto-connect the call play some audio,
#the call must be established for less than 10 seconds
org.jitsi.jigasi.HEALTH_CHECK_SIP_URI=healthcheck
#The interval between healthcheck calls, by default is 5 minutes
org.jitsi.jigasi.HEALTH_CHECK_INTERVAL=300000
#The timeout of healthcheck, if there was no successful health check for
#10 minutes (default value) we consider jigasi unhealthy
org.jitsi.jigasi.HEALTH_CHECK_TIMEOUT=600000
can you please help me detecting what did i do wrong ?