Jigasi - not connecting to siptest room

Hi all,

I need to create a conference room (only audio) to which several hosts can connect. The hosts are equipped with SIP clients (Ekiga on Linux)
In my application I don’t want to use web browser but automatically connect to the conference as soon as the host boots up.

So I created a VM with Ubuntu 18.04 and installed jitsi-meet following the quick guide on github (the installation has been performed using the static IP of the VM, not domain).
I tested jitsi with other machines via web browser and everything works fine.

Then I installed on the same VM MariaDB and Kamailio as a SIP server, creating a specific user for Jigasi.

I tested SIP with Ekiga on the two hosts and SIP works fine (i register and i can make calls).

Then I installed Jigasi giving the proper credentials. After a little tweak on the sip configuration file, I see Jigasi registering on the SIP Server, but when I try to call the “room” from ekiga istance it does not connect. If I open a web browser and access the room “siptest”, when the ekiga instance calls, a disconnection on the web browser occurs, to be restablished when the SIP call attempted by the host fails.

Here follows the Jigasi log:

2020-04-15 10:55:48.963 INFO: [466] org.jitsi.jigasi.SipGateway.incomingCallReceived().196 [ctx=15869409489631044449073] Incoming call received...
2020-04-15 10:55:50.000 INFO: [467] org.jitsi.jigasi.SipGatewaySession.run().1464 [ctx=15869409489631044449073]Using default JVB room name property siptest
2020-04-15 10:55:50.001 INFO: [467] org.jitsi.jigasi.JvbConference.start().422 [ctx=15869409489631044449073] Starting JVB conference room: siptest
2020-04-15 10:55:50.004 INFO: [467] org.jitsi.jigasi.JvbConference.setXmppProvider().553 [ctx=15869409489631044449073] Using ProtocolProviderServiceJabberImpl(Jabber:20624e79@192.168.1.155/20624e79)
2020-04-15 10:55:50.127 INFO: [470] impl.protocol.jabber.OperationSetBasicTelephonyJabberImpl.registrationStateChanged().127 Jingle : ON 
2020-04-15 10:55:50.129 INFO: [470] org.jitsi.jigasi.JvbConference.registrationStateChanged().606 [ctx=15869409489631044449073] Registering XMPP.
2020-04-15 10:55:50.145 INFO: [470] impl.protocol.jabber.ProtocolProviderServiceJabberImpl.authenticated().2535 Authenticated: false
2020-04-15 10:55:50.152 INFO: [470] org.jitsi.jigasi.JvbConference.joinConferenceRoom().685 [ctx=15869409489631044449073] Joining JVB conference room: siptest
2020-04-15 10:55:50.292 INFO: [474] impl.protocol.jabber.ChatRoomJabberImpl.joined().1256 siptest@conference.192.168.1.155/focus has joined the siptest@conference.192.168.1.155 chat room.
2020-04-15 10:56:20.268 SEVERE: [476] org.jitsi.jigasi.JvbConference.run().1541 [ctx=15869409489631044449073] Did not received session invite (30000 ms)
2020-04-15 10:56:20.274 INFO: [476] org.jitsi.jigasi.JvbConference.memberPresenceChanged().964 [ctx=15869409489631044449073] Member left : OWNER focus@auth.192.168.1.155/focus43395020017
2020-04-15 10:56:20.275 INFO: [476] org.jitsi.jigasi.JvbConference.memberPresenceChanged().972 [ctx=15869409489631044449073] Focus left! - stopping
2020-04-15 10:56:20.275 SEVERE: [476] org.jitsi.jigasi.JvbConference.stop().483 [ctx=15869409489631044449073] Already stopped !
2020-04-15 10:56:20.276 INFO: [476] org.jitsi.jigasi.JvbConference.memberPresenceChanged().964 [ctx=15869409489631044449073] Member left : GUEST 20624e79@192.168.1.155
2020-04-15 10:56:20.276 INFO: [476] org.jitsi.jigasi.JvbConference.stop().513 [ctx=15869409489631044449073] Removing account Jabber:20624e79@192.168.1.155/20624e79
2020-04-15 10:56:20.330 INFO: [476] impl.protocol.jabber.OperationSetBasicTelephonyJabberImpl.registrationStateChanged().134 Jingle : OFF 
2020-04-15 10:56:20.347 INFO: [87] org.jitsi.jigasi.SipGatewaySession.handleCallState().1318 [ctx=15869409489631044449073] SIP call ended: CallPeerChangeEvent: type=CallPeerStatusChange oldV=net.java.sip.communicator.service.protocol.CallPeerState:Incoming Call newV=net.java.sip.communicator.service.protocol.CallPeerState:Failed for peer=test3 <test3@192.168.1.155>;status=Failed
2020-04-15 10:56:20.348 INFO: [87] org.jitsi.jigasi.SipGatewaySession.sipCallEnded().585 [ctx=15869409489631044449073] Sip call ended: Call: id=1586940948962884567105 peers=0
2020-04-15 10:56:20.349 INFO: [87] org.jitsi.jigasi.AbstractGateway.notifyCallEnded().128 [ctx=15869409489631044449073] Removed session for call. Sessions:0
2020-04-15 10:56:20.350 INFO: [87] org.jitsi.jigasi.SipGatewaySession.peerStateChanged().1384 null SIP peer state: Failed

Here follows the sip.communication file:

#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\:c2room
net.java.sip.communicator.impl.protocol.sip.acc1403273890647.PASSWORD=RWxldHRyb25pY2Ew
net.java.sip.communicator.impl.protocol.sip.acc1403273890647.PROTOCOL_NAME=SIP
net.java.sip.communicator.impl.protocol.sip.acc1403273890647.SERVER_ADDRESS=192.168.1.155
net.java.sip.communicator.impl.protocol.sip.acc1403273890647.USER_ID=c2room
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](http://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=192.168.1.155

# 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=192.168.1.155
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=192.168.1.155
#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

# 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

Any help will be appreciated.

Edit:

After a downgrade to Jigasi 1.0-235 the problem is solved.

See here.