diff options
Diffstat (limited to 'configs/samples/sip.conf.sample')
| -rw-r--r-- | configs/samples/sip.conf.sample | 44 |
1 files changed, 22 insertions, 22 deletions
diff --git a/configs/samples/sip.conf.sample b/configs/samples/sip.conf.sample index 27012614e..c5ffdcccd 100644 --- a/configs/samples/sip.conf.sample +++ b/configs/samples/sip.conf.sample @@ -15,7 +15,7 @@ ; - context - Which set of services you offer various users ; ; SIP dial strings -;----------------------------------------------------------- +; ---------------------------------------------------------- ; In the dialplan (extensions.conf) you can use several ; syntaxes for dialing SIP devices. ; SIP/devicename @@ -76,7 +76,7 @@ ; sip reload Reload configuration file ; sip show settings Show the current channel configuration ; -;------- Naming devices ------------------------------------------------------ +; ------ Naming devices ------------------------------------------------------ ; ; When naming devices, make sure you understand how Asterisk matches calls ; that come in. @@ -100,7 +100,7 @@ ; not needed at all. Check below. In later releases, it's renamed ; to "defaultuser" which is a better name, since it is used in ; combination with the "defaultip" setting. -;----------------------------------------------------------------------------- +; ---------------------------------------------------------------------------- ; ** Old configuration options ** ; The "call-limit" configuation option is considered old is replaced @@ -559,7 +559,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; are not purged during SIP reloads. ; -;------------------------ TLS settings ------------------------------------------------------------ +; ----------------------- TLS settings ------------------------------------------------------------ ;tlscertfile=</path/to/certificate.pem> ; Certificate chain (*.pem format only) to use for TLS connections ; The certificates must be sorted starting with the subject's certificate ; and followed by intermediate CA certificates if applicable. @@ -603,7 +603,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; Your distribution might have changed that list ; further. ; -;--------------------------- SIP timers ---------------------------------------------------- +; -------------------------- SIP timers ---------------------------------------------------- ; These timers are used primarily in INVITE transactions. ; The default for Timer T1 is 500 ms or the measured run-trip time between ; Asterisk and the device if you have qualify=yes for the device. @@ -617,7 +617,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; in this amount of time, the call will autocongest ; Defaults to 64*timert1 -;--------------------------- RTP timers ---------------------------------------------------- +; -------------------------- RTP timers ---------------------------------------------------- ; These timers are currently used for both audio and video streams. The RTP timeouts ; are only applied to the audio channel. ; The settings are settable in the global section as well as per device @@ -633,7 +633,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ;rtpkeepalive=<secs> ; Send keepalives in the RTP stream to keep NAT open ; (default is off - zero) -;--------------------------- SIP Session-Timers (RFC 4028)------------------------------------ +; -------------------------- SIP Session-Timers (RFC 4028)------------------------------------ ; SIP Session-Timers provide an end-to-end keep-alive mechanism for active SIP sessions. ; This mechanism can detect and reclaim SIP channels that do not terminate through normal ; signaling procedures. Session-Timers can be configured globally or at a user/peer level. @@ -662,7 +662,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ;session-minse=90 ;session-refresher=uac ; -;--------------------------- SIP DEBUGGING --------------------------------------------------- +; -------------------------- SIP DEBUGGING --------------------------------------------------- ;sipdebug = yes ; Turn on SIP debugging by default, from ; the moment the channel loads this configuration. ; NOTE: You cannot use the CLI to turn it off. You'll @@ -673,7 +673,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; SIP history is output to the DEBUG logging channel -;--------------------------- STATUS NOTIFICATIONS (SUBSCRIPTIONS) ---------------------------- +; -------------------------- STATUS NOTIFICATIONS (SUBSCRIPTIONS) ---------------------------- ; You can subscribe to the status of extensions with a "hint" priority ; (See extensions.conf.sample for examples) ; chan_sip support two major formats for notifications: dialog-info and SIMPLE @@ -718,7 +718,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ;callcounter = yes ; Enable call counters on devices. This can be set per ; device too. -;----------------------------------------- T.38 FAX SUPPORT ---------------------------------- +; ---------------------------------------- T.38 FAX SUPPORT ---------------------------------- ; ; This setting is available in the [general] section as well as in device configurations. ; Setting this to yes enables T.38 FAX (UDPTL) on SIP calls; it defaults to off. @@ -751,7 +751,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; faxdetect = cng ; Enables only CNG detection ; faxdetect = t38 ; Enables only T.38 detection ; -;----------------------------------------- OUTBOUND SIP REGISTRATIONS ------------------------ +; ---------------------------------------- OUTBOUND SIP REGISTRATIONS ------------------------ ; Asterisk can register as a SIP user agent to a SIP proxy (provider) ; Format for the register statement is: ; register => [peer?][transport://]user[@domain][:secret[:authuser]]@host[:port][/extension][~expiry] @@ -828,7 +828,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; 401 responses and continue retrying according to normal ; retry rules. -;----------------------------------------- OUTBOUND MWI SUBSCRIPTIONS ------------------------- +; ---------------------------------------- OUTBOUND MWI SUBSCRIPTIONS ------------------------- ; Asterisk can subscribe to receive the MWI from another SIP server and store it locally for retrieval ; by other phones. At this time, you can only subscribe using UDP as the transport. ; Format for the mwi register statement is: @@ -843,7 +843,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; MWI received will be stored in the 1234 mailbox of the SIP_Remote context. ; It can be used by other phones by following the below: ; mailbox=1234@SIP_Remote -;----------------------------------------- NAT SUPPORT ------------------------ +; ---------------------------------------- NAT SUPPORT ------------------------ ; ; WARNING: SIP operation behind a NAT is tricky and you really need ; to read and understand well the following section. @@ -981,7 +981,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; ; icesupport = yes -;----------------------------------- MEDIA HANDLING -------------------------------- +; ---------------------------------- MEDIA HANDLING -------------------------------- ; By default, Asterisk tries to re-invite media streams to an optimal path. If there's ; no reason for Asterisk to stay in the media path, the media will be redirected. ; This does not really work well in the case where Asterisk is outside and the @@ -1063,7 +1063,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; option may be specified at the global or peer scope. ;force_avp=yes ; Force 'RTP/AVP', 'RTP/AVPF', 'RTP/SAVP', and 'RTP/SAVPF' to be used for ; media streams when appropriate, even if a DTLS stream is present. -;----------------------------------------- REALTIME SUPPORT ------------------------ +; ---------------------------------------- REALTIME SUPPORT ------------------------ ; For additional information on ARA, the Asterisk Realtime Architecture, ; please read https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration ; @@ -1101,7 +1101,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; is still in memory (due to caching or other reasons), the ; information will not be removed from realtime storage -;----------------------------------------- SIP DOMAIN SUPPORT ------------------------ +; ---------------------------------------- SIP DOMAIN SUPPORT ------------------------ ; Incoming INVITE and REFER messages can be matched against a list of 'allowed' ; domains, each of which can direct the call to a specific context if desired. ; By default, all domains are accepted and sent to the default context or the @@ -1140,13 +1140,13 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; destinations which do not have a prior ; account relationship with your server. -;------------------------------ Advice of Charge CONFIGURATION -------------------------- +; ----------------------------- Advice of Charge CONFIGURATION -------------------------- ; snom_aoc_enabled = yes; ; This options turns on and off support for sending AOC-D and ; AOC-E to snom endpoints. This option can be used both in the ; peer and global scope. The default for this option is off. -;------------------------------ 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 @@ -1178,7 +1178,7 @@ 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 @@ -1197,7 +1197,7 @@ 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. @@ -1324,7 +1324,7 @@ 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. @@ -1379,7 +1379,7 @@ 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 |