Sample configs: Eliminate false multiline comment block starts.
Change-Id: Ie627def9604ae30abd80754f9e6f09874825aec6
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@ -46,7 +46,7 @@ extension=s
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; systems where there will be no return audio path, such as overhead pagers.
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;noaudiocapture=true
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;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
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; ----------------------------- JITTER BUFFER CONFIGURATION --------------------------
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; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of an
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; ALSA channel. Defaults to "no". An enabled jitterbuffer will
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; be used only if the sending side can create and the receiving
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@ -74,5 +74,5 @@ extension=s
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; network normally has low jitter, but occasionally has spikes.
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; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
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;-----------------------------------------------------------------------------------
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; ----------------------------------------------------------------------------------
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@ -64,9 +64,9 @@
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; PLEASE READ THIS!!!
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;===========================================
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;
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;---------------------------------------------------------------------
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; --------------------------------------------------------------------
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; Timers
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;---------------------------------------------------------------------
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; --------------------------------------------------------------------
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;There are three configurable timers for all types of CC: the
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;cc_offer_timer, the ccbs_available_timer, and the ccnr_available_timer.
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;In addition, when using a generic agent, there is a fourth timer,
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@ -98,9 +98,9 @@
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; only affects operation when using a generic agent.
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;
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;cc_recall_timer = 20
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;---------------------------------------------------------------------
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; --------------------------------------------------------------------
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; Policies
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;---------------------------------------------------------------------
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; --------------------------------------------------------------------
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; Policy settings tell Asterisk how to behave and what sort of
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; resources to allocate in order to facilitate CC. There are two
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; settings to control the actions Asterisk will take.
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@ -153,9 +153,9 @@
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;cc_monitor_policy=never
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;
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;
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;---------------------------------------------------------------------
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; --------------------------------------------------------------------
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; Limits
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;---------------------------------------------------------------------
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; --------------------------------------------------------------------
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;
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; The use of CC requires Asterisk to potentially use more memory than
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; some administrators would like. As such, it is a good idea to limit
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@ -175,9 +175,9 @@
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;
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;cc_max_monitors = 5
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;
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;---------------------------------------------------------------------
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; --------------------------------------------------------------------
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; Other
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;---------------------------------------------------------------------
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; --------------------------------------------------------------------
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;
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; When using a generic CC agent, the caller who requested CC will be
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; called back when a called party becomes available. When the caller
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@ -1220,7 +1220,7 @@ pickupgroup=1
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;
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;jitterbuffers=4
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;
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;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
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; ----------------------------- JITTER BUFFER CONFIGURATION --------------------------
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; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
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; DAHDI channel. Defaults to "no". An enabled jitterbuffer will
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; be used only if the sending side can create and the receiving
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@ -1248,7 +1248,7 @@ pickupgroup=1
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; network normally has low jitter, but occasionally has spikes.
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; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
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;-----------------------------------------------------------------------------------
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; ----------------------------------------------------------------------------------
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;
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; You can define your own custom ring cadences here. You can define up to 8
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; pairs. If the silence is negative, it indicates where the caller ID spill is
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@ -44,7 +44,7 @@
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;
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;mohinterpret=default
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;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
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; ----------------------------- JITTER BUFFER CONFIGURATION --------------------------
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; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of an
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; Console channel. Defaults to "no". An enabled jitterbuffer will
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; be used only if the sending side can create and the receiving
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@ -72,7 +72,7 @@
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; network normally has low jitter, but occasionally has spikes.
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; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
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;-----------------------------------------------------------------------------------
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; ----------------------------------------------------------------------------------
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;
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@ -11,12 +11,12 @@
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;cos=3 ; Sets 802.1p priority for signaling packets.
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;cos_audio=5 ; Sets 802.1p priority for RTP audio packets.
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;---------------------- DIGIT TIMEOUTS ----------------------------
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; --------------------- DIGIT TIMEOUTS ----------------------------
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firstdigittimeout = 30000 ; default 16000 = 16s
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gendigittimeout = 10000 ; default 8000 = 8s
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matchdigittimeout = 5000 ; defaults 3000 = 3s
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;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
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; ----------------------------- JITTER BUFFER CONFIGURATION --------------------------
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; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
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; MGCP channel. Defaults to "no". An enabled jitterbuffer will
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; be used only if the sending side can create and the receiving
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@ -48,7 +48,7 @@ matchdigittimeout = 5000 ; defaults 3000 = 3s
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; network normally has low jitter, but occasionally has spikes.
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; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
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;-----------------------------------------------------------------------------------
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; ----------------------------------------------------------------------------------
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;[dlinkgw]
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;host = 192.168.0.64
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@ -12,7 +12,7 @@
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; this configuration file or realtime. The idea is to build voicemail as building blocks so that
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; a complete and adaptive voicemail system can be built in the dialplan
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;
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;------------------------------ Variables to use in subject, from and message body ------------------
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; ----------------------------- Variables to use in subject, from and message body ------------------
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; Change the from, body and/or subject, variables:
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; MVM_NAME, MVM_DUR, MVM_MSGNUM, VM_MAILBOX, MVM_CALLERID, MVM_CIDNUM,
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; MVM_CIDNAME, MVM_DATE
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@ -24,7 +24,7 @@
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; Note: The emailbody config row can only be up to 512 characters due to a
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; limitation in the Asterisk configuration subsystem.
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; To create longer mails, use the templatefile option when creating the template
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;----------------------------------------------------------------------------------------------------
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; ---------------------------------------------------------------------------------------------------
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[general]
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; Default format for storing and sending voicemail
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@ -64,7 +64,7 @@ silencethreshold=128
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; This is used both for e-mail and pager messages
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;mailcmd=/usr/sbin/sendmail -t
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;
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;--------------Default e-mail message template (used if no templates are used) ------
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; -------------Default e-mail message template (used if no templates are used) ------
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;fromstring=The Asterisk PBX
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;
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@ -82,7 +82,7 @@ emaildateformat=%A, %B %d, %Y at %r
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; 24h date format
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;emaildateformat=%A, %d %B %Y at %H:%M:%S
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;
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;--------------Default pager message template (used if no templates are used) ------
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; -------------Default pager message template (used if no templates are used) ------
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; You can also change the Pager From: string, the pager body and/or subject.
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; The above defined variables also can be used here
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;pagerfromstring=The Asterisk PBX
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@ -90,7 +90,7 @@ emaildateformat=%A, %B %d, %Y at %r
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;pagerbody=New ${MVM_DUR} long msg in box ${MVM_MAILBOX}\nfrom ${MVM_CALLERID}, on ${MVM_DATE}
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;
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;
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;--------------Timezone definitions (used in voicemail accounts) -------------------
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; -------------Timezone definitions (used in voicemail accounts) -------------------
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;
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; Users may be located in different timezones, or may have different
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; message announcements for their introductory message when they enter
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@ -133,7 +133,7 @@ central=America/Chicago|'vm-received' Q 'digits/at' IMp
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central24=America/Chicago|'vm-received' q 'digits/at' H N 'hours'
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military=Zulu|'vm-received' q 'digits/at' H N 'hours' 'phonetic/z_p'
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;----------------------- Message body templates---------------------
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; ---------------------- Message body templates---------------------
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; [template-name] ; "template-" is a verbatim marker
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; fromaddress = Your Friendly Asterisk Server
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; fromemail = asteriskvm@digium.com
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@ -187,7 +187,7 @@ dateformat=%A, %B %d, %Y at %r
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;subject = Dear old chap, you've got an electronic communique
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;charset=ascii
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;----------------------- Mailbox accounts --------------------------
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; ---------------------- Mailbox accounts --------------------------
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;Template for mailbox definition - all options
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;
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; [username@domain] ; Has to be unique within domain (MWM_USERNAME, MWM_DOMAIN)
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@ -109,7 +109,7 @@ crypt_prefix=**
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;
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crypt_keys=test,muh
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;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
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; ----------------------------- JITTER BUFFER CONFIGURATION --------------------------
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; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
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; SIP channel. Defaults to "no". An enabled jitterbuffer will
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; be used only if the sending side can create and the receiving
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@ -140,7 +140,7 @@ crypt_keys=test,muh
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; network normally has low jitter, but occasionally has spikes.
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; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
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;-----------------------------------------------------------------------------------
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; ----------------------------------------------------------------------------------
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; users sections:
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;
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@ -46,7 +46,7 @@
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; queuesize = 10 ; frames in device driver
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; frags = 8 ; argument to SETFRAGMENT
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;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
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; ----------------------------- JITTER BUFFER CONFIGURATION --------------------------
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; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of an
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; OSS channel. Defaults to "no". An enabled jitterbuffer will
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; be used only if the sending side can create and the receiving
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@ -74,7 +74,7 @@
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; network normally has low jitter, but occasionally has spikes.
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; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
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;-----------------------------------------------------------------------------------
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; ----------------------------------------------------------------------------------
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; below is an entry for a second console channel
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; [card1]
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@ -129,7 +129,7 @@ monitor-type = MixMonitor
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;
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;penaltymemberslimit = 5
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;
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;----------------------QUEUE TIMING OPTIONS------------------------------------
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; ---------------------QUEUE TIMING OPTIONS------------------------------------
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; A Queue has two different "timeout" values associated with it. One is the
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; timeout parameter configured in queues.conf. This timeout specifies the
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; amount of time to try ringing a member's phone before considering the
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@ -181,7 +181,7 @@ monitor-type = MixMonitor
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;retry = 5
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;timeoutpriority = app|conf
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;
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;-----------------------END QUEUE TIMING OPTIONS---------------------------------
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; ----------------------END QUEUE TIMING OPTIONS---------------------------------
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; Weight of queue - when compared to other queues, higher weights get
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; first shot at available channels when the same channel is included in
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; more than one queue.
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@ -1,6 +1,6 @@
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;
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; Configuration file for res_snmp
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;---------------------------------
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; --------------------------------
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;
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; Res_snmp can run as a subagent or standalone SNMP agent. The standalone snmp
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; agent is based on net-snmp and will read a configuration file called
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@ -15,7 +15,7 @@
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; - context - Which set of services you offer various users
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;
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; SIP dial strings
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;-----------------------------------------------------------
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; ----------------------------------------------------------
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; In the dialplan (extensions.conf) you can use several
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; syntaxes for dialing SIP devices.
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; SIP/devicename
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@ -87,7 +87,7 @@
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; sip reload Reload configuration file
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; sip show settings Show the current channel configuration
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;
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;------- Naming devices ------------------------------------------------------
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; ------ Naming devices ------------------------------------------------------
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;
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; When naming devices, make sure you understand how Asterisk matches calls
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; that come in.
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@ -111,7 +111,7 @@
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; not needed at all. Check below. In later releases, it's renamed
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; to "defaultuser" which is a better name, since it is used in
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; combination with the "defaultip" setting.
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;-----------------------------------------------------------------------------
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; ----------------------------------------------------------------------------
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; ** Old configuration options **
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; The "call-limit" configuation option is considered old is replaced
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@ -573,7 +573,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
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; are not purged during SIP reloads.
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;
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;------------------------ TLS settings ------------------------------------------------------------
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; ----------------------- TLS settings ------------------------------------------------------------
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;tlscertfile=</path/to/certificate.pem> ; Certificate chain (*.pem format only) to use for TLS connections
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; The certificates must be sorted starting with the subject's certificate
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; and followed by intermediate CA certificates if applicable. If the
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@ -622,7 +622,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
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; Your distribution might have changed that list
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; further.
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;
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;--------------------------- SIP timers ----------------------------------------------------
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; -------------------------- SIP timers ----------------------------------------------------
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; These timers are used primarily in INVITE transactions.
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; The default for Timer T1 is 500 ms or the measured run-trip time between
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; Asterisk and the device if you have qualify=yes for the device.
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@ -636,7 +636,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
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; in this amount of time, the call will autocongest
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; Defaults to 64*timert1
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;--------------------------- RTP timers ----------------------------------------------------
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; -------------------------- RTP timers ----------------------------------------------------
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; These timers are currently used for both audio and video streams. The RTP timeouts
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; are only applied to the audio channel.
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; The settings are settable in the global section as well as per device
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@ -652,7 +652,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
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;rtpkeepalive=<secs> ; Send keepalives in the RTP stream to keep NAT open
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; (default is off - zero)
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;--------------------------- SIP Session-Timers (RFC 4028)------------------------------------
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; -------------------------- SIP Session-Timers (RFC 4028)------------------------------------
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; SIP Session-Timers provide an end-to-end keep-alive mechanism for active SIP sessions.
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; This mechanism can detect and reclaim SIP channels that do not terminate through normal
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; signaling procedures. Session-Timers can be configured globally or at a user/peer level.
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@ -681,7 +681,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
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;session-minse=90
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;session-refresher=uac
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;
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;--------------------------- SIP DEBUGGING ---------------------------------------------------
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; -------------------------- SIP DEBUGGING ---------------------------------------------------
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;sipdebug = yes ; Turn on SIP debugging by default, from
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; the moment the channel loads this configuration.
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; NOTE: You cannot use the CLI to turn it off. You'll
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@ -692,7 +692,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
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; SIP history is output to the DEBUG logging channel
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;--------------------------- STATUS NOTIFICATIONS (SUBSCRIPTIONS) ----------------------------
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; -------------------------- STATUS NOTIFICATIONS (SUBSCRIPTIONS) ----------------------------
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; You can subscribe to the status of extensions with a "hint" priority
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; (See extensions.conf.sample for examples)
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; chan_sip support two major formats for notifications: dialog-info and SIMPLE
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@ -741,7 +741,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
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;callcounter = yes ; Enable call counters on devices. This can be set per
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; device too.
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;----------------------------------------- T.38 FAX SUPPORT ----------------------------------
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; ---------------------------------------- T.38 FAX SUPPORT ----------------------------------
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;
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; This setting is available in the [general] section as well as in device configurations.
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; Setting this to yes enables T.38 FAX (UDPTL) on SIP calls; it defaults to off.
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@ -774,7 +774,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
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; faxdetect = cng ; Enables only CNG detection
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; faxdetect = t38 ; Enables only T.38 detection
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;
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;----------------------------------------- OUTBOUND SIP REGISTRATIONS ------------------------
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; ---------------------------------------- OUTBOUND SIP REGISTRATIONS ------------------------
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; Asterisk can register as a SIP user agent to a SIP proxy (provider)
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; Format for the register statement is:
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; register => [peer?][transport://]user[@domain][:secret[:authuser]]@host[:port][/extension][~expiry]
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@ -851,7 +851,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
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; 401 responses and continue retrying according to normal
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; retry rules.
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;----------------------------------------- OUTBOUND MWI SUBSCRIPTIONS -------------------------
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; ---------------------------------------- OUTBOUND MWI SUBSCRIPTIONS -------------------------
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; Asterisk can subscribe to receive the MWI from another SIP server and store it locally for retrieval
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; by other phones. At this time, you can only subscribe using UDP as the transport.
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; Format for the mwi register statement is:
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@ -866,7 +866,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
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; MWI received will be stored in the 1234 mailbox of the SIP_Remote context.
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; It can be used by other phones by following the below:
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; mailbox=1234@SIP_Remote
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;----------------------------------------- NAT SUPPORT ------------------------
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; ---------------------------------------- NAT SUPPORT ------------------------
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;
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; WARNING: SIP operation behind a NAT is tricky and you really need
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; to read and understand well the following section.
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@ -1008,7 +1008,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
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;
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; icesupport = yes
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;----------------------------------- MEDIA HANDLING --------------------------------
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; ---------------------------------- MEDIA HANDLING --------------------------------
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; By default, Asterisk tries to re-invite media streams to an optimal path. If there's
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; no reason for Asterisk to stay in the media path, the media will be redirected.
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; This does not really work well in the case where Asterisk is outside and the
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@ -1090,7 +1090,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
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; option may be specified at the global or peer scope.
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;force_avp=yes ; Force 'RTP/AVP', 'RTP/AVPF', 'RTP/SAVP', and 'RTP/SAVPF' to be used for
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; media streams when appropriate, even if a DTLS stream is present.
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;----------------------------------------- REALTIME SUPPORT ------------------------
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; ---------------------------------------- REALTIME SUPPORT ------------------------
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; For additional information on ARA, the Asterisk Realtime Architecture,
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; please read https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration
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;
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@ -1128,7 +1128,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
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; is still in memory (due to caching or other reasons), the
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; information will not be removed from realtime storage
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;----------------------------------------- SIP DOMAIN SUPPORT ------------------------
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; ---------------------------------------- SIP DOMAIN SUPPORT ------------------------
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; Incoming INVITE and REFER messages can be matched against a list of 'allowed'
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; domains, each of which can direct the call to a specific context if desired.
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; By default, all domains are accepted and sent to the default context or the
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@ -1167,13 +1167,13 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
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; destinations which do not have a prior
|
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; account relationship with your server.
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||||
;------------------------------ Advice of Charge CONFIGURATION --------------------------
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||||
; ----------------------------- Advice of Charge CONFIGURATION --------------------------
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||||
; snom_aoc_enabled = yes; ; This options turns on and off support for sending AOC-D and
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; AOC-E to snom endpoints. This option can be used both in the
|
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; peer and global scope. The default for this option is off.
|
||||
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||||
|
||||
;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
|
||||
; ----------------------------- JITTER BUFFER CONFIGURATION --------------------------
|
||||
; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
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; SIP channel. Defaults to "no". An enabled jitterbuffer will
|
||||
; be used only if the sending side can create and the receiving
|
||||
|
@ -1205,7 +1205,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
|
|||
|
||||
; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
|
||||
|
||||
;-----------------------------------------------------------------------------------
|
||||
; ----------------------------------------------------------------------------------
|
||||
|
||||
[authentication]
|
||||
; Global credentials for outbound calls, i.e. when a proxy challenges your
|
||||
|
@ -1224,7 +1224,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
|
|||
; You may also add auth= statements to [peer] definitions
|
||||
; Peer auth= override all other authentication settings if we match on realm
|
||||
|
||||
;------------------------------------------------------------------------------
|
||||
; -----------------------------------------------------------------------------
|
||||
; DEVICE CONFIGURATION
|
||||
;
|
||||
; SIP entities have a 'type' which determines their roles within Asterisk.
|
||||
|
@ -1351,7 +1351,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
|
|||
; ; from the peer's configuration.
|
||||
;
|
||||
|
||||
;------------------------------------------------------------------------------
|
||||
; -----------------------------------------------------------------------------
|
||||
; DTLS-SRTP CONFIGURATION
|
||||
;
|
||||
; DTLS-SRTP support is available if the underlying RTP engine in use supports it.
|
||||
|
@ -1409,7 +1409,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
|
|||
;port=80 ; The port number we want to connect to on the remote side
|
||||
; Also used as "defaultport" in combination with "defaultip" settings
|
||||
|
||||
;--- sample definition for a provider
|
||||
; -- sample definition for a provider
|
||||
;[provider1]
|
||||
;type=peer
|
||||
;host=sip.provider1.com
|
||||
|
|
|
@ -54,7 +54,7 @@ keepalive=120
|
|||
;cos_audio=5 ; Sets 802.1p priority for RTP audio packets.
|
||||
;cos_video=4 ; Sets 802.1p priority for RTP video packets.
|
||||
|
||||
;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
|
||||
; ----------------------------- JITTER BUFFER CONFIGURATION --------------------------
|
||||
;jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
|
||||
; skinny channel. Defaults to "no". An enabled jitterbuffer will
|
||||
; be used only if the sending side can create and the receiving
|
||||
|
@ -79,10 +79,10 @@ keepalive=120
|
|||
; Defaults to fixed.
|
||||
|
||||
;jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
|
||||
;-----------------------------------------------------------------------------------
|
||||
; ----------------------------------------------------------------------------------
|
||||
|
||||
[lines]
|
||||
;----------------------------------- LINES SECTION --------------------------------
|
||||
; ---------------------------------- LINES SECTION --------------------------------
|
||||
; Options set under [lines] apply to all lines unless explicitly set for a particular
|
||||
; device. The options that can be set under lines are specified in GENERAL LINE OPTIONS.
|
||||
; These options can also be set for each individual device as well as those under SPECIFIC
|
||||
|
@ -95,15 +95,15 @@ keepalive=120
|
|||
; Where options are common to both lines and devices, the results typically take that of
|
||||
; the least permission. ie if a no is set for either line or device, the call will not be
|
||||
; able to use that permission
|
||||
;-------------------------------- GENERAL LINE OPTIONS -----------------------------
|
||||
; ------------------------------- GENERAL LINE OPTIONS -----------------------------
|
||||
;earlyrtp=1 ; whether audio signalling should be provided by asterisk
|
||||
; ; (earlyrtp=1) or device generated (earlyrtp=0). default=yes
|
||||
;transfer=1 ; whether the device is allowed to transfer. default=yes
|
||||
;context=default ; context to use for this line.
|
||||
;callfwdtimeout=20000 ; ms before cfwd_noans occurs (default 20 secs)
|
||||
;------------------------------- SPECIFIC LINE OPTIONS -----------------------------
|
||||
; ------------------------------ SPECIFIC LINE OPTIONS -----------------------------
|
||||
;setvar= ; allows for the setting of chanvars.
|
||||
;-----------------------------------------------------------------------------------
|
||||
; ----------------------------------------------------------------------------------
|
||||
|
||||
;[100]
|
||||
;nat=yes
|
||||
|
@ -149,7 +149,7 @@ keepalive=120
|
|||
|
||||
|
||||
[devices]
|
||||
;---------------------------------- DEVICES SECTION -------------------------------
|
||||
; --------------------------------- DEVICES SECTION -------------------------------
|
||||
; Options set under [devices] apply to all devices unless explicitly set for a particular
|
||||
; device. The options that can be set under devices are specified in GENERAL DEVICE OPTIONS.
|
||||
; These options can also be set for each individual device as well as those under SPECIFIC
|
||||
|
@ -162,16 +162,16 @@ keepalive=120
|
|||
; Where options are common to both lines and devices, the results typically take that of
|
||||
; the least permission. ie if a no is set for either line or device, the call will not be
|
||||
; able to use that permission
|
||||
;------------------------------- GENERAL DEVICE OPTIONS ----------------------------
|
||||
; ------------------------------ GENERAL DEVICE OPTIONS ----------------------------
|
||||
;earlyrtp=1 ; whether audio signalling should be provided by asterisk
|
||||
; ; (earlyrtp=1) or device generated (earlyrtp=0). default=yes
|
||||
;transfer=1 ; whether the device is allowed to transfer. default=yes
|
||||
;------------------------------ SPECIFIC DEVICE OPTIONS ----------------------------
|
||||
; ----------------------------- SPECIFIC DEVICE OPTIONS ----------------------------
|
||||
;device="SEPxxxxxxxxxxxx ; id of the device. Must be set.
|
||||
;version=P002G204 ; firmware version to be loaded. If this version is different
|
||||
; ; to the one on the device, the device will try to load this
|
||||
; ; version from the tftp server. Set to device firmware version.
|
||||
;-----------------------------------------------------------------------------------
|
||||
; ----------------------------------------------------------------------------------
|
||||
|
||||
; Typical config for 12SP+
|
||||
;[florian]
|
||||
|
|
|
@ -17,7 +17,7 @@ port=5000 ; UDP port
|
|||
;autoprovisioning=no ; Allow undeclared phones to register an extension. See README for important
|
||||
; informations. no (default), yes, tn.
|
||||
;mohsuggest=default
|
||||
;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
|
||||
; ----------------------------- JITTER BUFFER CONFIGURATION --------------------------
|
||||
; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
|
||||
; SIP channel. Defaults to "no". An enabled jitterbuffer will
|
||||
; be used only if the sending side can create and the receiving
|
||||
|
@ -41,7 +41,7 @@ port=5000 ; UDP port
|
|||
; variable size, actually the new jb of IAX2). Defaults to fixed.
|
||||
|
||||
; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
|
||||
;-----------------------------------------------------------------------------------
|
||||
; ----------------------------------------------------------------------------------
|
||||
|
||||
|
||||
;[black] ; name of the device
|
||||
|
|
|
@ -199,7 +199,7 @@ grunttimeout=3600
|
|||
;
|
||||
mode=immediate
|
||||
|
||||
;-------------------------------------------------------------------------
|
||||
; ------------------------------------------------------------------------
|
||||
; Channel definitions
|
||||
;
|
||||
; Each channel inherits the settings specified above, unless the are
|
||||
|
|
Loading…
Reference in New Issue