diff options
author | oej <oej@f38db490-d61c-443f-a65b-d21fe96a405b> | 2006-06-29 07:04:43 +0000 |
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committer | oej <oej@f38db490-d61c-443f-a65b-d21fe96a405b> | 2006-06-29 07:04:43 +0000 |
commit | e73494f5eb7c2d0e9736fe8ab1d968494b8bf374 (patch) | |
tree | cb3c148f0611de0f3c4feed0a686824470da7870 | |
parent | f06ba80021e0c99081106ca7f58514db8bb43543 (diff) |
reformatting sip.conf.sample a bit, adding dumphistory that was not documented
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@36251 f38db490-d61c-443f-a65b-d21fe96a405b
-rw-r--r-- | configs/sip.conf.sample | 80 |
1 files changed, 42 insertions, 38 deletions
diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample index 602a18c80..a91f2308b 100644 --- a/configs/sip.conf.sample +++ b/configs/sip.conf.sample @@ -25,9 +25,7 @@ [general] context=default ; Default context for incoming calls -;allowguest=no ; Allow or reject guest calls (default is yes, - ; this can also be set to 'osp' - ; if asterisk was compiled with OSP support.) +;allowguest=no ; Allow or reject guest calls (default is yes) allowoverlap=no ; Disable overlap dialing support. (Default is yes) ;allowtransfer=no ; Disable all transfers (unless enabled in peers or users) ; Default is enabled @@ -49,15 +47,6 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; If configured, Asterisk will only allow ; INVITE and REFER to non-local domains ; Use "sip show domains" to list local domains -;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 -;allowexternalinvites=no ; Disable INVITE and REFER to non-local domains - ; Default is yes -;autodomain=yes ; Turn this on to have Asterisk add local host - ; name and local IP to domain list. ;pedantic=yes ; Enable checking of tags in headers, ; international character conversions in URIs ; and multiline formatted headers for strict @@ -79,9 +68,6 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ;vmexten=voicemail ; dialplan extension to reach mailbox sets the ; Message-Account in the MWI notify message ; defaults to "asterisk" -;recordhistory=yes ; Record SIP history by default - ; (see sip history / sip no history) - ;disallow=all ; First disallow all codecs ;allow=ulaw ; Allow codecs in order of preference ;allow=ilbc ; @@ -114,8 +100,6 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; auto : Use rfc2833 if offered, inband otherwise ;compactheaders = yes ; send compact sip headers. -;sipdebug = yes ; Turn on SIP debugging by default, from - ; the moment the channel loads this configuration ; ;videosupport=yes ; Turn on support for SIP video ;maxcallbitrate=384 ; Maximum bitrate for video calls (default 384 kb/s) @@ -123,19 +107,6 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; for peers and users as well ;callevents=no ; generate manager events when sip ua ; performs events (e.g. hold) - -;--------------------------- 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 -; 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 - ; Useful to limit subscriptions to local extensions - ; Settable per peer/user also -;notifyringing = yes ; Notify subscriptions on RINGING state ;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 @@ -143,7 +114,8 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; ; If regcontext is specified, Asterisk will dynamically create and destroy a ; NoOp priority 1 extension for a given peer who registers or unregisters with -; us. Multiple contexts may be specified by separating them with '&'. The +; us and have a "regexten=" configuration item. +; Multiple contexts may be specified by separating them with '&'. The ; actual extension is the 'regexten' parameter of the registering peer or its ; name if 'regexten' is not provided. If more than one context is provided, ; the context must be specified within regexten by appending the desired @@ -151,6 +123,28 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; separated by '&'. Patterns may be used in regexten. ; ;regcontext=sipregistrations +; +;--------------------------- SIP DEBUGGING --------------------------------------------------- +;sipdebug = yes ; Turn on SIP debugging by default, from + ; the moment the channel loads this configuration +;recordhistory=yes ; Record SIP history by default + ; (see sip history / sip no history) +;dumphistory=yes ; Dump SIP history at end of SIP dialogue + ; SIP history is output to the DEBUG logging channel + + +;--------------------------- 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 +; 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 + ; Useful to limit subscriptions to local extensions + ; Settable per peer/user also +;notifyringing = yes ; Notify subscriptions on RINGING state ;----------------------------------------- T.38 FAX PASSTHROUGH SUPPORT ----------------------- ; ; These settings are available in the [general] section as well as in device configurations @@ -301,13 +295,23 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; To disallow requests for domains not serviced by this server: ; allowexternaldomains=no -; fromdomain=mydomain.tld ; When making outbound SIP INVITEs to - ; non-peers, use your primary domain "identity" - ; for From: headers instead of just your IP - ; address. This is to be polite and - ; it may be a mandatory requirement for some - ; destinations which do not have a prior - ; account relationship with your server. +;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 +;allowexternalinvites=no ; Disable INVITE and REFER to non-local domains + ; Default is yes +;autodomain=yes ; 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 + ; non-peers, use your primary domain "identity" + ; for From: headers instead of just your IP + ; address. This is to be polite and + ; it may be a mandatory requirement for some + ; destinations which do not have a prior + ; account relationship with your server. ;------------------------------ JITTER BUFFER CONFIGURATION -------------------------- ; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a |