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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 --------------------------
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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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@ -1205,7 +1205,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
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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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[authentication]
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; Global credentials for outbound calls, i.e. when a proxy challenges your
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@ -1224,7 +1224,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
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; You may also add auth= statements to [peer] definitions
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; Peer auth= override all other authentication settings if we match on realm
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;------------------------------------------------------------------------------
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; -----------------------------------------------------------------------------
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; DEVICE CONFIGURATION
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;
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; SIP entities have a 'type' which determines their roles within Asterisk.
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@ -1351,7 +1351,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
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; ; from the peer's configuration.
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;
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;------------------------------------------------------------------------------
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; -----------------------------------------------------------------------------
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; DTLS-SRTP CONFIGURATION
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;
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; DTLS-SRTP support is available if the underlying RTP engine in use supports it.
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@ -1409,7 +1409,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
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;port=80 ; The port number we want to connect to on the remote side
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; Also used as "defaultport" in combination with "defaultip" settings
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;--- sample definition for a provider
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; -- sample definition for a provider
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;[provider1]
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;type=peer
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;host=sip.provider1.com
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Richard Mudgett
9 years ago