[general]
context=default ; Default context for incoming calls
-;allowguest=no ; Allow or reject guest calls (default is yes)
+allowguest=yes ; Allow or reject guest calls (default is yes)
allowoverlap=yes ; Disable overlap dialing support. (Default is yes)
allowtransfer=yes ; Disable all transfers (unless enabled in peers or users)
; Default is enabled
+
+; **** GESTION DES DOMAINES SIP : voir /etc/asterisk/auf/sip-general.local
+; **** pour y indiquer le nom du domaine SIP géré localement
+
;realm=mydomain.tld ; Realm for digest authentication
; defaults to "asterisk". If you set a system name in
; asterisk.conf, it defaults to that system name
; ability to place SIP calls based on domain
; names to some other SIP users on the Internet
+; **** GESTION DES DOMAINES SIP : voir /etc/asterisk/auf/sip-general.local
+; **** pour y indiquer le nom du domaine SIP géré localement
;domain=mydomain.tld ; Set default domain for this host
; If configured, Asterisk will only allow
; INVITE and REFER to non-local domains
; for peers and users as well
;callevents=no ; generate manager events when sip ua
; performs events (e.g. hold)
-;alwaysauthreject = yes ; When an incoming INVITE or REGISTER is to be rejected,
+alwaysauthreject = yes ; When an incoming INVITE or REGISTER is to be rejected,
; for any reason, always reject with '401 Unauthorized'
; instead of letting the requester know whether there was
; a matching user or peer for their request
; are only applied to the audio channel.
; The settings are settable in the global section as well as per device
;
-;rtptimeout=60 ; Terminate call if 60 seconds of no RTP or RTCP activity
+rtptimeout=60 ; Terminate call if 60 seconds of no RTP or RTCP activity
; on the audio channel
; when we're not on hold. This is to be able to hangup
; a call in the case of a phone disappearing from the net,
; like a powerloss or grandma tripping over a cable.
-;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP or RTCP activity
+rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP or RTCP activity
; on the audio channel
; when we're on hold (must be > rtptimeout)
-;rtpkeepalive=<secs> ; Send keepalives in the RTP stream to keep NAT open
+rtpkeepalive=60 ; Send keepalives in the RTP stream to keep NAT open
; (default is off - zero)
;--------------------------- SIP DEBUGGING ---------------------------------------------------
;sipdebug = yes ; Turn on SIP debugging by default, from
; Note: Subscriptions does not work if you have a realtime dialplan and use the
; realtime switch.
;
-;allowsubscribe=no ; Disable support for subscriptions. (Default is yes)
-;subscribecontext = default ; Set a specific context for SUBSCRIBE requests
+allowsubscribe = yes ; Disable support for subscriptions. (Default is yes)
+subscribecontext = AUF-local ; Set a specific context for SUBSCRIBE requests
; Useful to limit subscriptions to local extensions
; Settable per peer/user also
-;notifyringing = yes ; Notify subscriptions on RINGING state (default: no)
-;notifyhold = yes ; Notify subscriptions on HOLD state (default: no)
+notifyringing = yes ; Notify subscriptions on RINGING state (default: no)
+notifyhold = yes ; Notify subscriptions on HOLD state (default: no)
; Turning on notifyringing and notifyhold will add a lot
; more database transactions if you are using realtime.
-;limitonpeers = yes ; Apply call limits on peers only. This will improve
+limitonpeers = yes ; Apply call limits on peers only. This will improve
; status notification when you are using type=friend
; Inbound calls, that really apply to the user part
; of a friend will now be added to and compared with
; The externip, externhost and localnet settings are used if you use Asterisk
; behind a NAT device to communicate with services on the outside.
+; si votre serveur Asterisk est derrière un système DNAT, il faut indiquer
+; ici son adresse IP publique.
+
+; ********** A FAIRE DANS /etc/asterisk/auf/sip-general.local
+; ********** sinon la modification sera effacée à la prochaine mise à jour...
+
;externip = 200.201.202.203 ; Address that we're going to put in outbound SIP
; messages if we're behind a NAT
; The externip and localnet is used
; when registering and communicating with other proxies
; that we're registered with
+
+; Si cette IP est dynamique, vous pouvez essayer d'utiliser externhost
+; et un système de type DynDNS
+
;externhost=foo.dyndns.net ; Alternatively you can specify an
; external host, and Asterisk will
; perform DNS queries periodically. Not
; used
; You may add multiple local networks. A reasonable
; set of defaults are:
-;localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks
-;localnet=10.0.0.0/255.0.0.0 ; Also RFC1918
-;localnet=172.16.0.0/12 ; Another RFC1918 with CIDR notation
-;localnet=169.254.0.0/255.255.0.0 ;Zero conf local network
+
+; réseaux locaux avec lesquels il ne faut pas faire de NAT
+localnet=10.0.0.0/8
+localnet=172.16.0.0/12
+localnet=192.168.0.0/16
+localnet=169.254.0.0/16 ; ZeroConf
; The nat= setting is used when Asterisk is on a public IP, communicating with
; devices hidden behind a NAT device (broadband router). If you have one-way
; audio problems, you usually have problems with your NAT configuration or your
; firewall's support of SIP+RTP ports. You configure Asterisk choice of RTP
; ports for incoming audio in rtp.conf
-;
-;nat=no ; Global NAT settings (Affects all peers and users)
+
+nat=no ; Global NAT settings (Affects all peers and users)
; yes = Always ignore info and assume NAT
; no = Use NAT mode only according to RFC3581 (;rport)
; never = Never attempt NAT mode or RFC3581 support
; This does not really work with in the case where Asterisk is outside and have
; clients on the inside of a NAT. In that case, you want to set canreinvite=nonat
;
+
+canreinvite=no ; Asterisk reste sur le chemin du flux audio
+
;canreinvite=yes ; Asterisk by default tries to redirect the
; RTP media stream (audio) to go directly from
; the caller to the callee. Some devices do not
; To disallow requests for domains not serviced by this server:
; allowexternaldomains=no
+; **** GESTION DES DOMAINES SIP : voir /etc/asterisk/auf/sip-general.local
+; **** pour y indiquer le nom du domaine SIP géré localement
+
;domain=mydomain.tld,mydomain-incoming
; Add domain and configure incoming context
; for external calls to this domain
;domain=1.2.3.4 ; Add IP address as local domain
; You can have several "domain" settings
-;allowexternaldomains=no ; Disable INVITE and REFER to non-local domains
+allowexternaldomains=yes ; Disable INVITE and REFER to non-local domains
; Default is yes
-;autodomain=yes ; Turn this on to have Asterisk add local host
+autodomain=no ; Turn this on to have Asterisk add local host
; name and local IP to domain list.
; fromdomain=mydomain.tld ; When making outbound SIP INVITEs to
; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
;-----------------------------------------------------------------------------------
+#include "auf/sip-general.local"
+
[authentication]
; Global credentials for outbound calls, i.e. when a proxy challenges your
; Asterisk server for authentication. These credentials override
dtmfmode=info
;
-; AUF
+; Sortie vers SIPBroker http://www.sipbroker.com
+;
+
+[sipbroker-out]
+type=peer
+fromuser=voip
+fromdomain=auf.org
+host=sipbroker.com
+port=5060
+canreinvite=yes
+
+;
+;
;
; Comptes pour postes clients locaux SIP
projets / asterisk-config-auf.git / blobdiff