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viernes, 3 de septiembre de 2021

Asterisk webrtc configuration

 https://wiki.asterisk.org/wiki/display/AST/Configuring+Asterisk+for+WebRTC+Clients





asterisk -vvvvvvvvvgc”  | grep http

https://community.asterisk.org/t/asterisk-15-https-server/72517/6


https://feeding.cloud.geek.nz/posts/using-letsencrypt-cert-with-asterisk/

--------------------channel driver= pjsip


Transport  = 0.0.0.0-wss


Enable AVPF = yes


Enable ICE Support =Yes


Enable rtcp Mux =Yes


RTP Symmetric = Yes

 

Rewrite Contact = Yes 


Force rport=  Yes 


Enable WebRTC defaults =Yes



Media Encryption  = SRTP


Allow Non-Encrypted Media (Opportunistic SRTP) = Yes



Enable DTLS= No 

Auto Generate Certificate =No


Use Certificate = default


DTLS Verify  = fingerprint


DTLS Setup = Act/Pass


DTLS Rekey Interval= 0

SIP explained

 https://datatracker.ietf.org/doc/html/rfc7329

martes, 31 de agosto de 2021

OPTION REQUEST explained




11.1 Construction of OPTIONS Request

   An OPTIONS request is constructed using the standard rules for a SIP
   request as discussed in Section 8.1.1.

   A Contact header field MAY be present in an OPTIONS.

   An Accept header field SHOULD be included to indicate the type of
   message body the UAC wishes to receive in the response.  Typically,
   this is set to a format that is used to describe the media
   capabilities of a UA, such as SDP (application/sdp).

   The response to an OPTIONS request is assumed to be scoped to the
   Request-URI in the original request.  However, only when an OPTIONS
   is sent as part of an established dialog is it guaranteed that future
   requests will be received by the server that generated the OPTIONS
   response.

   Example OPTIONS request:

      OPTIONS sip:carol@chicago.com SIP/2.0
      Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKhjhs8ass877
      Max-Forwards: 70
      To: <sip:carol@chicago.com>
      From: Alice <sip:alice@atlanta.com>;tag=1928301774
      Call-ID: a84b4c76e66710
      CSeq: 63104 OPTIONS
      Contact: <sip:alice@pc33.atlanta.com>
      Accept: application/sdp
      Content-Length: 0




Rosenberg, et. al.          Standards Track                    [Page 67]


RFC 3261            SIP: Session Initiation Protocol           June 2002


11.2 Processing of OPTIONS Request

   The response to an OPTIONS is constructed using the standard rules
   for a SIP response as discussed in Section 8.2.6.  The response code
   chosen MUST be the same that would have been chosen had the request
   been an INVITE.  That is, a 200 (OK) would be returned if the UAS is
   ready to accept a call, a 486 (Busy Here) would be returned if the
   UAS is busy, etc.  This allows an OPTIONS request to be used to
   determine the basic state of a UAS, which can be an indication of
   whether the UAS will accept an INVITE request.

   An OPTIONS request received within a dialog generates a 200 (OK)
   response that is identical to one constructed outside a dialog and
   does not have any impact on the dialog.

   This use of OPTIONS has limitations due to the differences in proxy
   handling of OPTIONS and INVITE requests.  While a forked INVITE can
   result in multiple 200 (OK) responses being returned, a forked
   OPTIONS will only result in a single 200 (OK) response, since it is
   treated by proxies using the non-INVITE handling.  See Section 16.7
   for the normative details.

   If the response to an OPTIONS is generated by a proxy server, the
   proxy returns a 200 (OK), listing the capabilities of the server.
   The response does not contain a message body.

   Allow, Accept, Accept-Encoding, Accept-Language, and Supported header
   fields SHOULD be present in a 200 (OK) response to an OPTIONS
   request.  If the response is generated by a proxy, the Allow header
   field SHOULD be omitted as it is ambiguous since a proxy is method
   agnostic.  Contact header fields MAY be present in a 200 (OK)
   response and have the same semantics as in a 3xx response.  That is,
   they may list a set of alternative names and methods of reaching the
   user.  A Warning header field MAY be present.

   A message body MAY be sent, the type of which is determined by the
   Accept header field in the OPTIONS request (application/sdp is the
   default if the Accept header field is not present).  If the types
   include one that can describe media capabilities, the UAS SHOULD
   include a body in the response for that purpose.  Details on the
   construction of such a body in the case of application/sdp are
   described in [13].









Rosenberg, et. al.          Standards Track                    [Page 68]


RFC 3261            SIP: Session Initiation Protocol           June 2002


   Example OPTIONS response generated by a UAS (corresponding to the
   request in Section 11.1):

      SIP/2.0 200 OK
      Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKhjhs8ass877
       ;received=192.0.2.4
      To: <sip:carol@chicago.com>;tag=93810874
      From: Alice <sip:alice@atlanta.com>;tag=1928301774
      Call-ID: a84b4c76e66710
      CSeq: 63104 OPTIONS
      Contact: <sip:carol@chicago.com>
      Contact: <mailto:carol@chicago.com>
      Allow: INVITE, ACK, CANCEL, OPTIONS, BYE
      Accept: application/sdp
      Accept-Encoding: gzip
      Accept-Language: en
      Supported: foo
      Content-Type: application/sdp
      Content-Length: 274

      (SDP not shown)

https://datatracker.ietf.org/doc/html/rfc3261#page-67

martes, 24 de agosto de 2021

FreePBX not sending email from GUI but command line works

 


/usr/sbin/sendmail -t


https://community.freepbx.org/t/updated-to-freepbx-13-sendmail-working-extension-emails-are-not/56731/9




viernes, 30 de julio de 2021

FreePBX changing Asterisk version

 


Saltar al final de los metadatos
Ir al inicio de los metadatos

The FreePBX Distro has some built in features to allow you to change the Major Asterisk version you are using.

For example the 6.12.65 release track supports Asterisk 1.8, 11, 12 and 13

To change which Major Asterisk release you are using type;

asterisk-version-switch

 

You will then be prompted to pick which Asterisk Version you would like to switch to

Once the switch has been completed you will be informed which Version you are running and that Asterisk is restarting.

FreePBX will then perform a reload and re-write all the configuration files based on the version of Asterisk you switch to.

You can verify the running version of Asterisk at the bash prompt with:

#  asterisk -x "core show version"
Asterisk 13.9.1 built by mockbuild @ jenkins2.schmoozecom.net on a x86_64 running Linux on 2016-05-13 20:33:39 UTC

If the old Asterisk version is still shown, you can restart with:

fwconsole restart

The final step is to reload the system dial plan:

fwconsole reload

viernes, 23 de julio de 2021

block calls based on a blacklist file

 exten => _X.,1,NoOp(${CALLERID(num)}-${EXTEN})

same=>n,Set(BAN=${SHELL(cat /home/centos/spam.conf  | grep -w -F ${CALLERID(num)} | wc -l)})

same=>n,ExecIf($[${BAN}>0]?hangup())

same=>n,Noop(***${BAN}***)




*content of  /home/centos/spam.conf **

188222
22222
33333