Error with SIP + Asterisk

[lfdominguez]

Hello i configured Asterisk + pjSIP with an account for jigasi, but when I run jigasi this is the errors in log:

May 09, 2018 3:13:30 PM net.java.sip.communicator.util.Logger error
SEVERE: couldn't find a ProtocolProviderServiceSipImpl to dispatch to
May 09, 2018 3:13:33 PM net.java.sip.communicator.util.Logger info
INFO: Info from the JAIN-SIP stack: Setting SIPMessage peerPacketSource to: /10.11.0.35:5061
May 09, 2018 3:13:33 PM net.java.sip.communicator.util.Logger error
SEVERE: no listeners
May 09, 2018 3:13:33 PM net.java.sip.communicator.util.Logger error
SEVERE: couldn't find a ProtocolProviderServiceSipImpl to dispatch to
May 09, 2018 3:13:34 PM net.java.sip.communicator.util.Logger info
INFO: Info from the JAIN-SIP stack: Setting SIPMessage peerPacketSource to: /10.11.0.35:5061
May 09, 2018 3:13:34 PM net.java.sip.communicator.util.Logger error
SEVERE: no listeners
May 09, 2018 3:13:34 PM net.java.sip.communicator.util.Logger error
SEVERE: couldn't find a ProtocolProviderServiceSipImpl to dispatch to

And Asterisk register and unregister the user

 == Contact jitsi/sip:jitsi@10.11.0.11:5060;transport=udp;registering_acc=sip_mtz_desoft_cu has been deleted
    -- Added contact 'sip:jitsi@10.11.0.11:5060;transport=udp;registering_acc=sip_mtz_desoft_cu' to AOR 'jitsi' with expiration of 600 seconds
  == Contact jitsi/sip:jitsi@10.11.0.11:5060;transport=udp;registering_acc=sip_mtz_desoft_cu has been created
  == Endpoint jitsi is now Reachable
    -- Contact jitsi/sip:jitsi@10.11.0.11:5060;transport=udp;registering_acc=sip_mtz_desoft_cu is now Unreachable.  RTT: 0.000 msec
  == Endpoint jitsi is now Unreachable

[damencho]

Hey, @lfdominguez can you share your configuration of jigasi, the sip-communicator.properties file? I've been testing jigasi these days and have no problem registering to asterisk and making calls.

[lfdominguez]

ohh thanks for respond, now i will send you the configuration

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=test7

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=false

net.java.sip.communicator.impl.protocol.sip.acc1403273890647=acc1403273890647 net.java.sip.communicator.impl.protocol.sip.acc1403273890647.ACCOUNT_UID=SIP\: jitsi@sip.mtz.desoft.cu 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= sip.mtz.desoft.cu net.java.sip.communicator.impl.protocol.sip.acc1403273890647.USER_ID=jitsi 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

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= mtz.desoft.cu

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=jabber.mtz.desoft.cu

org.jitsi.jigasi.xmpp.acc.VIDEO_CALLING_DISABLED=false

Or you can use bosh for the connection establishment by specifing the URL

to use. org.jitsi.jigasi.xmpp.acc.BOSH_URL_PATTERN= https://jabber.mtz.desoft.cu:7443/http-bind?room={roomName}

Used when outgoing calls are used in multidomain environment, used to

detect subdomains

org.jitsi.jigasi.xmpp.acc.DOMAIN_BASE=reuniones.mtz.desoft.cu

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=focus@mtz.desoft.cu org.jitsi.jigasi.xmpp.acc.PASS=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

delivering final transcript

org.jitsi.jigasi.transcription.DIRECTORY=/var/lib/jigasi/transcripts

org.jitsi.jigasi.transcription.BASE_URL=http://localhost/

org.jitsi.jigasi.transcription.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

El mié., 16 may. 2018 a las 11:10, Дамян Минков (notifications@github.com) escribió:

Hey, @lfdominguez https://github.com/lfdominguez can you share your configuration of jigasi, the sip-communicator.properties file? I've been testing jigasi these days and have no problem registering to asterisk and making calls.

— You are receiving this because you were mentioned. Reply to this email directly, view it on GitHub https://github.com/jitsi/jigasi/issues/100#issuecomment-389554393, or mute the thread https://github.com/notifications/unsubscribe-auth/AFWDI8dCjXyRIMTqup7A2TY7i8rbZPPQks5tzEFegaJpZM4T4gTa .

[damencho]

You cannot use focus user for jigas, this can lead to unpredictable results, change that and try again.

[lfdominguez]

ummm .. i need to create a new user for jigasi??

El mié., 16 may. 2018 a las 15:06, Дамян Минков (notifications@github.com) escribió:

You cannot use focus user for jigas, this can lead to unpredictable results, change that and try again.

— You are receiving this because you were mentioned. Reply to this email directly, view it on GitHub https://github.com/jitsi/jigasi/issues/100#issuecomment-389631370, or mute the thread https://github.com/notifications/unsubscribe-auth/AFWDI67aQrdTTEXhDLMOSp9YNP-3FmR-ks5tzHiogaJpZM4T4gTa .

[damencho]

Donyou actually need one? Have you enabled authentication ob your deployment? This is needed if you have authentication enabled.

[lfdominguez]

yeap i need authentication because is the main OpenFire of my network

El mié., 16 may. 2018 a las 15:21, Дамян Минков (notifications@github.com) escribió:

Donyou actually need one? Have you enabled authentication ob your deployment? This is needed if you have authentication enabled.

— You are receiving this because you were mentioned. Reply to this email directly, view it on GitHub https://github.com/jitsi/jigasi/issues/100#issuecomment-389635705, or mute the thread https://github.com/notifications/unsubscribe-auth/AFWDI3TTS0dgyXAh61ncpdSdD887Lnolks5tzHwogaJpZM4T4gTa .

[damencho]

So you need another user for jigasi.

[lfdominguez]

i will test that

[lfdominguez]

Same error...... asterisk.log jigasi.log

[damencho]

It seems to me the incoming sip messages cannot be matched to the sip protocol provider. This is where it is matched: https://github.com/jitsi/jitsi/blob/288c1a135b3391585f8f6fe56306487d4b55f3ae/src/net/java/sip/communicator/impl/protocol/sip/SipStackSharing.java#L903 In your asterisk config are you using the account peer that is registered to send the calls or you are just sending the sip to the jigasi machine using its ip and port, the second approach will not work?

[lfdominguez]

Well, my asterisk has 2 interfaces, one public and one privated, i configure static that PJSIP listen only in the public iface (that one connect to jigasi) and now works!!!!!!! reading the asterisk logs, sems that asterisk catch de private IP as public and when jigasi try connect, asterisk respond with a wrong public IP, i dont know why.....

Thanks for let me see the problem.....

[fuqiangleon]

@damencho hope to ask you a question: how can i config the SIP port if we used a different port ,eg:6060 .

tks.

[damencho]

@fuqiangleon Can you ask your questions on the community forum and not post on random issues that has nothing to do with your question. Thank you.