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Summary:ASTERISK-30312: Not able to Call
Reporter:Ajaykrishna P (ajaykrishna.p@pragtech.co.in)Labels:webrtc
Date Opened:2022-11-14 04:50:09.000-0600Date Closed:2022-11-14 04:50:14.000-0600
Priority:MajorRegression?
Status:Closed/CompleteComponents:. I did not set the category correctly.
Versions:16.29.0 Frequency of
Occurrence
Related
Issues:
Environment:UbuntuAttachments:( 0) Screenshot_from_2022-11-14_15-29-59.png
( 1) Screenshot_from_2022-11-14_15-31-53.png
( 2) Screenshot_from_2022-11-14_15-32-06.png
( 3) Screenshot_from_2022-11-14_15-33-29.png
( 4) Screenshot_from_2022-11-14_15-33-56.png
Description: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@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@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
Comments:By: Asterisk Team (asteriskteam) 2022-11-14 04:50:12.931-0600

We appreciate the difficulties you are facing, however information request type issues would be better served in a different forum.

The Asterisk community provides support over IRC, mailing lists, and forums as described at http://asterisk.org/community. The Asterisk issue tracker is used specifically to track issues concerning bugs and documentation errors.

If this issue is actually a bug please use the Bug issue type instead.

Please see the Asterisk Issue Guidelines [1] for instruction on the intended use of the Asterisk issue tracker.

Thanks!

[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines
By: Asterisk Team (asteriskteam) 2022-11-14 04:50:18.456-0600

Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution. Please note that log messages and other files should not be sent to the Sangoma Asterisk Team unless explicitly asked for. All files should be placed on this issue in a sanitized fashion as needed.

A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report.

Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process].

Please note that once your issue enters an open state it has been accepted. As Asterisk is an open source project there is no guarantee or timeframe on when your issue will be looked into. If you need expedient resolution you will need to find and pay a suitable developer. Asking for an update on your issue will not yield any progress on it and will not result in a response. All updates are posted to the issue when they occur.

Please note that by submitting data, code, or documentation to Sangoma through JIRA, you accept the Terms of Use present at [https://www.asterisk.org/terms-of-use/|https://www.asterisk.org/terms-of-use/].
By: Ajaykrishna P (ajaykrishna.p@pragtech.co.in) 2022-11-14 05:01:51.012-0600

All configuration file