Multiple Jitsi Instances One Sip Login

Is there a way to allow multiple jitsi instances to use the same sip user? Whenever I add a second jigasi instance that uses the same sip user, I can no longer call into the conference and my SIP system says the user is busy. Here is my jigasi SIP config:

#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@conference.jitsiinstance.domain.com

org.jitsi.jigasi.MUC_SERVICE_ADDRESS=conference.jitsiinstance.domain.com

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"]
org.jitsi.impl.neomedia.transform.csrc.CsrcTransformEngine.DISCARD_CONTRIBUTING_SOURCES=true

# 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=false

net.java.sip.communicator.impl.protocol.sip.acc1403273890647=acc1403273890647
net.java.sip.communicator.impl.protocol.sip.acc1403273890647.ACCOUNT_UID=SIP\:doctogether1@10.10.30.41
net.java.sip.communicator.impl.protocol.sip.acc1403273890647.PASSWORD=pass
net.java.sip.communicator.impl.protocol.sip.acc1403273890647.PROTOCOL_NAME=SIP
net.java.sip.communicator.impl.protocol.sip.acc1403273890647.SERVER_ADDRESS=10.10.30.41
net.java.sip.communicator.impl.protocol.sip.acc1403273890647.USER_ID=doctogether1@10.10.30.41
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=1net.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


#org.jitsi.jigasi.xmpp.acc.USE_TRANSLATOR_IN_CONFERENCE=true
net.java.sip.communicator.service.gui.ALWAYS_TRUST_MODE_ENABLED=true

# Freeswitch
# Use standard X- header 
#net.java.sip.communicator.impl.protocol.sip.acc1.JITSI_MEET_ROOM_HEADER_NAME=X-Room-Name
net.java.sip.communicator.impl.protocol.sip.acc1403273890647.JITSI_MEET_ROOM_HEADER_NAME=X-Room-Name
# don’t overload Jigasi with focus changes
net.java.sip.communicator.impl.protocol.sip.SKIP_REINVITE_ON_FOCUS_CHANGE_PROP=true

# 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.1403273890647.USE_TRANSLATOR_IN_CONFERENCE=false
#org.jitsi.jigasi.xmpp.acc.USE_TRANSLATOR_IN_CONFERENCE=false
# allow a conference to start muted
#org.jitsi.jigasi.ENABLE_SIP_STARTMUTED=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=jitsiinstance.domain.com

# 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

# The following account is used only for dial-out from a meeting, it connects and join the brewery room used
# by Jicofo to discover available jigasi instances and use them (they can be used for outgoing calls or transcriptions if enabled or both)
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1=acc-xmpp-1
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.ACCOUNT_UID=Jabber:jigasi@auth.jitsiinstance.domain.com
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.USER_ID=jigasi@auth.jitsiinstance.domain.com
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.IS_SERVER_OVERRIDDEN=true
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.SERVER_ADDRESS=127.0.0.1
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.SERVER_PORT=5222
#net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.BOSH_URL=https://127.0.0.1/http-bind
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.ALLOW_NON_SECURE=true
#base64 AES keyLength:256 or 128
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.PASSWORD=*****==

#net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.RESOURCE=jigasi
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.AUTO_GENERATE_RESOURCE=true
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.RESOURCE_PRIORITY=30

net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.KEEP_ALIVE_METHOD=XEP-0199
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.KEEP_ALIVE_INTERVAL=30
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.CALLING_DISABLED=true
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.JINGLE_NODES_ENABLED=false
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.IS_CARBON_DISABLED=true
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.DEFAULT_ENCRYPTION=true
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.IS_USE_ICE=true
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.IS_ACCOUNT_DISABLED=false
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.IS_PREFERRED_PROTOCOL=false
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.AUTO_DISCOVER_JINGLE_NODES=false
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.PROTOCOL=Jabber
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.IS_USE_UPNP=false
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.IM_DISABLED=true
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.SERVER_STORED_INFO_DISABLED=true
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.IS_FILE_TRANSFER_DISABLED=true
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.USE_DEFAULT_STUN_SERVER=true
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.ENCRYPTION_PROTOCOL.DTLS-SRTP=0
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.ENCRYPTION_PROTOCOL_STATUS.DTLS-SRTP=true
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.OVERRIDE_ENCODINGS=true
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.Encodings.G722/8000=0
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.Encodings.GSM/8000=0
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.Encodings.H263-1998/90000=0
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.Encodings.H264/90000=0
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.Encodings.PCMA/8000=0
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.Encodings.PCMU/8000=0
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.Encodings.SILK/12000=0
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.Encodings.SILK/16000=0
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.Encodings.SILK/24000=0
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.Encodings.SILK/8000=0
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.Encodings.VP8/90000=0
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.Encodings.iLBC/8000=0
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.Encodings.opus/48000=0
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.Encodings.speex/16000=0
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.Encodings.speex/32000=0
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.Encodings.speex/8000=0
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.BREWERY=JigasiBrewery@internal.auth.jitsiinstance.domain.com
net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.BOSH_URL_PATTERN=https://{host}{subdomain}/http-bind?room={roomName}

net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.DOMAIN_BASE=jitsiinstance.domain.com

#net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.CallStats.appId=
#net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.CallStats.keyId=
#net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.CallStats.keyPath=/etc/jitsi/jigasi/ecpriv.jwk
#net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.CallStats.conferenceIDPrefix=jitsiinstance.domain.com
#net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.CallStats.jigasiId=jigasi-1
#net.java.sip.communicator.impl.protocol.jabber.acc-xmpp-1.CallStats.STATISTICS_INTERVAL=60000

# we can receive dial/hangup only from the control muc
org.jitsi.jigasi.ALLOWED_JID=JigasiBrewery@internal.auth.jitsiinstance.domain.com

org.jitsi.jigasi.BREWERY_ENABLED=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=jitsiinstance.domain.com
#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

# Vosk server
# org.jitsi.jigasi.transcription.customService=org.jitsi.jigasi.transcription.VoskTranscriptionService
# org.jitsi.jigasi.transcription.vosk.websocket_url=ws://localhost:2700

# 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

# Enabled or disable the notification when max occupants limit is reached
# org.jitsi.jigasi.NOTIFY_MAX_OCCUPANTS=false

How do you add two sip users, please elaborate on that?

I installed jigasi on two different Jitsi meet servers. When I first installed jigasi, I used the same sip credentials for each instance. When I added the second jigasi, my calls were no longer being answered by either instance. To fix that problem, I requested a new phone number from my provider, and configured my Freeswitch server to route calls to that new phone number to a new SIP user I created in Freeswitch. On my second instance of Jitsi Meet/Jigasi, I configured Jigasi to use the credentials for the second SIP user I created. I was hoping that I could use the same SIP user and Jigasi instances would all be able to use the same user and answer the calls based on the room name headers but that did not seem to be the case even though the room names contain a FQDN.