[asterisk-bugs] [JIRA] (ASTERISK-30312) Not able to Call

[Ajaykrishna P (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-30312?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Ajaykrishna P updated ASTERISK-30312:
-------------------------------------

    Attachment: Screenshot from 2022-11-14 15-29-59.png
                Screenshot from 2022-11-14 15-31-53.png
                Screenshot from 2022-11-14 15-32-06.png
                Screenshot from 2022-11-14 15-33-29.png
                Screenshot from 2022-11-14 15-33-56.png

All configuration file

> Not able to Call
> ----------------
>
>                 Key: ASTERISK-30312
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-30312
>             Project: Asterisk
>          Issue Type: Information Request
>      Security Level: None
>          Components: . I did not set the category correctly.
>    Affects Versions: 16.29.0
>         Environment: Ubuntu
>            Reporter: Ajaykrishna P
>            Severity: Critical
>              Labels: webrtc
>         Attachments: Screenshot from 2022-11-14 15-29-59.png, Screenshot from 2022-11-14 15-31-53.png, Screenshot from 2022-11-14 15-32-06.png, Screenshot from 2022-11-14 15-33-29.png, Screenshot from 2022-11-14 15-33-56.png
>
>
> Couldn't make the calls with asterisk
> I want to connect to asterisk and make calls from my odoo server.I did the configurations as per the asterisk document. But not able to make the call.
> sip.conf--->
> [1060] ; This will be WebRTC client
> type=friend
> username=1060 ; The Auth user for SIP.js
> host=dynamic ; Allows any host to register
> secret=password ; The SIP Password for SIP.js
> encryption=yes ; Tell Asterisk to use encryption for this peer
> servername=Asterisk
> enabled=yes
> tlsenable=yes
> bindaddr=0.0.0.0
> bindport=8088
> tlsbindaddr=0.0.0.0:8089
> tlscipher=ALL
> tlsclientmethod=tlsv1, sslv3, sslv2
> prefix=asterisk
> tlscertfile=/etc/asterisk/keys/asterisk.crt
> tlsprivatekey=/etc/asterisk/keys/asterisk.key
> avpf=yes ; Tell Asterisk to use AVPF for this peer
> icesupport=yes ; Tell Asterisk to use ICE for this peer
> context=default ; Tell Asterisk which context to use when this peer is dialing
> directmedia=no ; Asterisk will relay media for this peer
> transport= tls,ws,wss ; Asterisk will allow this peer to register on UDP or WebSockets
> force_avp=yes ; Force Asterisk to use avp. Introduced in Asterisk 11.11
> rtcp_mux=yes ; Tell Asterisk to do RTCP mux
> stunaddr = stun.l.google.com:19302
> dtlsenable=yes ; Tell Asterisk to enable DTLS for this peer
> dtlsverify=no ; Tell Asterisk to not verify your DTLS certs
> dtlscertfile=/etc/asterisk/keys/asterisk.crt ; Tell Asterisk where your DTLS cert file is
> dtlsprivatekey=/etc/asterisk/keys/asterisk.key ; Tell Asterisk where your DTLS private key is
> dtlssetup=actpass ; Tell Asterisk to use actpass SDP parameter when setting up DTLS
> rtp_symmetric=yes
> insecure=very
> [1061] ; This will be WebRTC client
> type=friend
> username=1061 ; The Auth user for SIP.js
> host=dynamic ; Allows any host to register
> secret=password ; The SIP Password for SIP.js
> encryption=yes ; Tell Asterisk to use encryption for this peer
> servername=Asterisk
> enabled=yes
> tlsenable=yes
> transport= tls,ws,wss ;
> rtp_symmetric=yes
> insecure=very
> directmedia=no
> http.conf --->
> bindport=8088;
> bindaddr=0.0.0.0
> tlscertfile=/etc/asterisk/keys/asterisk.crt
> tlsprivatekey=/etc/asterisk/keys/asterisk.key
> sessionlimit=300
> session_inactivity=30000
> session_keep_alive=15000
> extension.conf -->
> [default]
> exten => 1060,1,Dial(SIP/1060) ; Dialing 1060 will call the SIP client registered to 1060
> exten => 1060,2,Answer()
> exten => 1060,3,Playback(vm-nobodyavail)
> exten => 1060,4,VoiceMail(1060 at main)
> exten => 1060,5,Hangup()
> exten => 1061,1,Dial(SIP/1061) ; Dialing 1061 will call the SIP client registered to 1061
> exten => 1061,2,Answer()
> exten => 1061,3,Playback(vm-nobodyavail)
> exten => 1061,4,VoiceMail(1061 at main)
> exten => 1061,5,Hangup()
> manage.conf --->
> [general]
> enabled = no
> port = 5038
> bindaddr = 0.0.0.0
> voicemail.conf --->
> [general]
> format = wav49|gsm|wav
> [default]
> 1060 => 1060,Admin
> 1061 => 1061,Abc
> 1062 => 1062,Bcd
> [example]
> ; Voicemail context for all internal users in the example.com domain.
> 1060 = 1060,Admin
> 1061 = 1061,Abc
> 1062 = 1062,Bcd
> 1101 = 0717,Maria Berny
> 1102 = 7085,Tommie Briar
> 1103 = 1809,Penelope Bronte
> 1104 = 0039,Richard Casey
> 1105 = 6618,Garnet Claude
> 1106 = 9805,Aaron Courtney
> 1107 = 7484,Lindsey Freddie
> 1108 = 7788,Colby Hildred
> 1109 = 5750,Terry Jules
> 1110 = 3702,Hollis Justy
> 1111 = 1878,Temple Morgan
> 1112 = 5497,Franny Ocean
> 1113 = 1637,Laverne Roberts
> 1114 = 3717,Sal Smith
> 1115 = 3088,Dusty Williams



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