diff options
Diffstat (limited to 'configs')
| -rw-r--r-- | configs/sip.conf.sample | 668 |
1 files changed, 334 insertions, 334 deletions
diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample index 57dff4d61..08eaf50b1 100644 --- a/configs/sip.conf.sample +++ b/configs/sip.conf.sample @@ -13,65 +13,65 @@ ; where the proxyhostname is defined in a section below ; ; Useful CLI commands to check peers/users: -; sip show peers Show all SIP peers (including friends) -; sip show users Show all SIP users (including friends) -; sip show registry Show status of hosts we register with +; sip show peers Show all SIP peers (including friends) +; sip show users Show all SIP users (including friends) +; sip show registry Show status of hosts we register with ; -; sip debug Show all SIP messages +; sip debug Show all SIP messages ; -; reload chan_sip.so Reload configuration file -; Active SIP peers will not be reconfigured +; module reload chan_sip.so Reload configuration file +; Active SIP peers will not be reconfigured ; [general] -context=default ; Default context for incoming calls -;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 -;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 - ; Realms MUST be globally unique according to RFC 3261 - ; Set this to your host name or domain name -bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) - ; bindport is the local UDP port that Asterisk will listen on -bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) -srvlookup=yes ; Enable DNS SRV lookups on outbound calls - ; Note: Asterisk only uses the first host - ; in SRV records - ; Disabling DNS SRV lookups disables the - ; ability to place SIP calls based on domain - ; names to some other SIP users on the Internet - -;pedantic=yes ; Enable checking of tags in headers, - ; international character conversions in URIs - ; and multiline formatted headers for strict - ; SIP compatibility (defaults to "no") +context=default ; Default context for incoming calls +;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 +;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 + ; Realms MUST be globally unique according to RFC 3261 + ; Set this to your host name or domain name +bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) + ; bindport is the local UDP port that Asterisk will listen on +bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) +srvlookup=yes ; Enable DNS SRV lookups on outbound calls + ; Note: Asterisk only uses the first host + ; in SRV records + ; Disabling DNS SRV lookups disables the + ; ability to place SIP calls based on domain + ; names to some other SIP users on the Internet + +;pedantic=yes ; Enable checking of tags in headers, + ; international character conversions in URIs + ; and multiline formatted headers for strict + ; SIP compatibility (defaults to "no") ; See doc/ip-tos.txt for a description of these parameters. ;tos_sip=cs3 ; Sets TOS for SIP packets. ;tos_audio=ef ; Sets TOS for RTP audio packets. ;tos_video=af41 ; Sets TOS for RTP video packets. -;maxexpiry=3600 ; Maximum allowed time of incoming registrations - ; and subscriptions (seconds) -;minexpiry=60 ; Minimum length of registrations/subscriptions (default 60) -;defaultexpiry=120 ; Default length of incoming/outgoing registration -;t1min=100 ; Minimum roundtrip time for messages to monitored hosts - ; Defaults to 100 ms -;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY -;checkmwi=10 ; Default time between mailbox checks for peers -;buggymwi=no ; Cisco SIP firmware doesn't support the MWI RFC - ; fully. Enable this option to not get error messages - ; when sending MWI to phones with this bug. -;vmexten=voicemail ; dialplan extension to reach mailbox sets the - ; Message-Account in the MWI notify message - ; defaults to "asterisk" -;disallow=all ; First disallow all codecs -;allow=ulaw ; Allow codecs in order of preference -;allow=ilbc ; see doc/rtp-packetization for framing options -; +;maxexpiry=3600 ; Maximum allowed time of incoming registrations + ; and subscriptions (seconds) +;minexpiry=60 ; Minimum length of registrations/subscriptions (default 60) +;defaultexpiry=120 ; Default length of incoming/outgoing registration +;t1min=100 ; Minimum roundtrip time for messages to monitored hosts + ; Defaults to 100 ms +;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY +;checkmwi=10 ; Default time between mailbox checks for peers +;buggymwi=no ; Cisco SIP firmware doesn't support the MWI RFC + ; fully. Enable this option to not get error messages + ; when sending MWI to phones with this bug. +;vmexten=voicemail ; dialplan extension to reach mailbox sets the + ; Message-Account in the MWI notify message + ; defaults to "asterisk" +;disallow=all ; First disallow all codecs +;allow=ulaw ; Allow codecs in order of preference +;allow=ilbc ; see doc/rtp-packetization for framing options + ; This option specifies a preference for which music on hold class this channel ; should listen to when put on hold if the music class has not been set on the ; channel with Set(CHANNEL(musicclass)=whatever) in the dialplan, and the peer @@ -87,50 +87,50 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; ;mohsuggest=default ; -;language=en ; Default language setting for all users/peers - ; This may also be set for individual users/peers -;relaxdtmf=yes ; Relax dtmf handling -;trustrpid = no ; If Remote-Party-ID should be trusted -;sendrpid = yes ; If Remote-Party-ID should be sent -;progressinband=never ; If we should generate in-band ringing always - ; use 'never' to never use in-band signalling, even in cases - ; where some buggy devices might not render it - ; Valid values: yes, no, never Default: never -;useragent=Asterisk PBX ; Allows you to change the user agent string -;promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP address - ; Note that promiscredir when redirects are made to the - ; local system will cause loops since Asterisk is incapable - ; of performing a "hairpin" call. -;usereqphone = no ; If yes, ";user=phone" is added to uri that contains - ; a valid phone number -;dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833 - ; Other options: - ; info : SIP INFO messages - ; inband : Inband audio (requires 64 kbit codec -alaw, ulaw) - ; auto : Use rfc2833 if offered, inband otherwise - -;compactheaders = yes ; send compact sip headers. -; -;videosupport=yes ; Turn on support for SIP video. You need to turn this on - ; in the this section to get any video support at all. - ; You can turn it off on a per peer basis if the general - ; video support is enabled, but you can't enable it for - ; one peer only without enabling in the general section. -;maxcallbitrate=384 ; Maximum bitrate for video calls (default 384 kb/s) - ; Videosupport and maxcallbitrate is settable - ; 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, - ; 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 - -;g726nonstandard = yes ; If the peer negotiates G726-32 audio, use AAL2 packing - ; order instead of RFC3551 packing order (this is required - ; for Sipura and Grandstream ATAs, among others). This is - ; contrary to the RFC3551 specification, the peer _should_ - ; be negotiating AAL2-G726-32 instead :-( +;language=en ; Default language setting for all users/peers + ; This may also be set for individual users/peers +;relaxdtmf=yes ; Relax dtmf handling +;trustrpid = no ; If Remote-Party-ID should be trusted +;sendrpid = yes ; If Remote-Party-ID should be sent +;progressinband=never ; If we should generate in-band ringing always + ; use 'never' to never use in-band signalling, even in cases + ; where some buggy devices might not render it + ; Valid values: yes, no, never Default: never +;useragent=Asterisk PBX ; Allows you to change the user agent string +;promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP address + ; Note that promiscredir when redirects are made to the + ; local system will cause loops since Asterisk is incapable + ; of performing a "hairpin" call. +;usereqphone = no ; If yes, ";user=phone" is added to uri that contains + ; a valid phone number +;dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833 + ; Other options: + ; info : SIP INFO messages + ; inband : Inband audio (requires 64 kbit codec -alaw, ulaw) + ; auto : Use rfc2833 if offered, inband otherwise + +;compactheaders = yes ; send compact sip headers. +; +;videosupport=yes ; Turn on support for SIP video. You need to turn this on + ; in the this section to get any video support at all. + ; You can turn it off on a per peer basis if the general + ; video support is enabled, but you can't enable it for + ; one peer only without enabling in the general section. +;maxcallbitrate=384 ; Maximum bitrate for video calls (default 384 kb/s) + ; Videosupport and maxcallbitrate is settable + ; 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, + ; 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 + +;g726nonstandard = yes ; If the peer negotiates G726-32 audio, use AAL2 packing + ; order instead of RFC3551 packing order (this is required + ; for Sipura and Grandstream ATAs, among others). This is + ; contrary to the RFC3551 specification, the peer _should_ + ; be negotiating AAL2-G726-32 instead :-( ;matchexterniplocally = yes ; Only substitute the externip or externhost setting if it matches ; your localnet setting. Unless you have some sort of strange network @@ -154,23 +154,23 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; 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 - ; 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 - ; on the audio channel - ; when we're on hold (must be > rtptimeout) -;rtpkeepalive=<secs> ; Send keepalives in the RTP stream to keep NAT open - ; (default is off - zero) +;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 + ; on the audio channel + ; when we're on hold (must be > rtptimeout) +;rtpkeepalive=<secs> ; 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 - ; 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 +;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) ---------------------------- @@ -189,24 +189,24 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; 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 ; Control whether subscriptions already INUSE get sent - ; RINGING when another call is sent (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 - ; 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 peer limit instead of applying two call limits, - ; one for the peer and one for the user. - ; "sip show inuse" will only show active calls on - ; the peer side of a "type=friend" object if this - ; setting is turned on. +;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 ; Control whether subscriptions already INUSE get sent + ; RINGING when another call is sent (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 + ; 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 peer limit instead of applying two call limits, + ; one for the peer and one for the user. + ; "sip show inuse" will only show active calls on + ; the peer side of a "type=friend" object if this + ; setting is turned on. ;----------------------------------------- T.38 FAX PASSTHROUGH SUPPORT ----------------------- ; @@ -234,7 +234,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; ; Examples: ; -;register => 1234:password@mysipprovider.com +;register => 1234:password@mysipprovider.com ; ; This will pass incoming calls to the 's' extension ; @@ -249,34 +249,34 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; Tip 2: Use separate type=peer and type=user sections for SIP providers ; (instead of type=friend) if you have calls in both directions -;registertimeout=20 ; retry registration calls every 20 seconds (default) -;registerattempts=10 ; Number of registration attempts before we give up - ; 0 = continue forever, hammering the other server - ; until it accepts the registration - ; Default is 0 tries, continue forever +;registertimeout=20 ; retry registration calls every 20 seconds (default) +;registerattempts=10 ; Number of registration attempts before we give up + ; 0 = continue forever, hammering the other server + ; until it accepts the registration + ; Default is 0 tries, continue forever ;----------------------------------------- NAT SUPPORT ------------------------ ; The externip, externhost and localnet settings are used if you use Asterisk ; behind a NAT device to communicate with services on the outside. -;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 -;externhost=foo.dyndns.net ; Alternatively you can specify an - ; external host, and Asterisk will - ; perform DNS queries periodically. Not - ; recommended for production - ; environments! Use externip instead -;externrefresh=10 ; How often to refresh externhost if - ; used - ; You may add multiple local networks. A reasonable - ; set of defaults are: +;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 +;externhost=foo.dyndns.net ; Alternatively you can specify an + ; external host, and Asterisk will + ; perform DNS queries periodically. Not + ; recommended for production + ; environments! Use externip instead +;externrefresh=10 ; How often to refresh externhost if + ; 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=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 ; The nat= setting is used when Asterisk is on a public IP, communicating with @@ -285,12 +285,12 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; 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 - ; route = Assume NAT, don't send rport - ; (work around more UNIDEN bugs) + ; route = Assume NAT, don't send rport + ; (work around more UNIDEN bugs) ;----------------------------------- MEDIA HANDLING -------------------------------- ; By default, Asterisk tries to re-invite the audio to an optimal path. If there's @@ -298,72 +298,72 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; 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=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 - ; support this (especially if one of them is behind a NAT). - ; The default setting is YES. If you have all clients - ; behind a NAT, or for some other reason wants Asterisk to - ; stay in the audio path, you may want to turn this off. - - ; In Asterisk 1.4 this setting also affect direct RTP - ; at call setup (a new feature in 1.4 - setting up the - ; call directly between the endpoints instead of sending - ; a re-INVITE). - -;directrtpsetup=yes ; Enable the new experimental direct RTP setup. This sets up - ; the call directly with media peer-2-peer without re-invites. - ; Will not work for video and cases where the callee sends - ; RTP payloads and fmtp headers in the 200 OK that does not match the - ; callers INVITE. This will also fail if canreinvite is enabled when - ; the device is actually behind NAT. - -;canreinvite=nonat ; An additional option is to allow media path redirection - ; (reinvite) but only when the peer where the media is being - ; sent is known to not be behind a NAT (as the RTP core can - ; determine it based on the apparent IP address the media - ; arrives from). - -;canreinvite=update ; Yet a third option... use UPDATE for media path redirection, - ; instead of INVITE. This can be combined with 'nonat', as - ; 'canreinvite=update,nonat'. It implies 'yes'. +;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 + ; support this (especially if one of them is behind a NAT). + ; The default setting is YES. If you have all clients + ; behind a NAT, or for some other reason wants Asterisk to + ; stay in the audio path, you may want to turn this off. + + ; In Asterisk 1.4 this setting also affect direct RTP + ; at call setup (a new feature in 1.4 - setting up the + ; call directly between the endpoints instead of sending + ; a re-INVITE). + +;directrtpsetup=yes ; Enable the new experimental direct RTP setup. This sets up + ; the call directly with media peer-2-peer without re-invites. + ; Will not work for video and cases where the callee sends + ; RTP payloads and fmtp headers in the 200 OK that does not match the + ; callers INVITE. This will also fail if canreinvite is enabled when + ; the device is actually behind NAT. + +;canreinvite=nonat ; An additional option is to allow media path redirection + ; (reinvite) but only when the peer where the media is being + ; sent is known to not be behind a NAT (as the RTP core can + ; determine it based on the apparent IP address the media + ; arrives from). + +;canreinvite=update ; Yet a third option... use UPDATE for media path redirection, + ; instead of INVITE. This can be combined with 'nonat', as + ; 'canreinvite=update,nonat'. It implies 'yes'. ;----------------------------------------- REALTIME SUPPORT ------------------------ ; For additional information on ARA, the Asterisk Realtime Architecture, ; please read realtime.txt and extconfig.txt in the /doc directory of the ; source code. ; -;rtcachefriends=yes ; Cache realtime friends by adding them to the internal list - ; just like friends added from the config file only on a - ; as-needed basis? (yes|no) - -;rtsavesysname=yes ; Save systemname in realtime database at registration - ; Default= no - -;rtupdate=yes ; Send registry updates to database using realtime? (yes|no) - ; If set to yes, when a SIP UA registers successfully, the ip address, - ; the origination port, the registration period, and the username of - ; the UA will be set to database via realtime. - ; If not present, defaults to 'yes'. -;rtautoclear=yes ; Auto-Expire friends created on the fly on the same schedule - ; as if it had just registered? (yes|no|<seconds>) - ; If set to yes, when the registration expires, the friend will - ; vanish from the configuration until requested again. If set - ; to an integer, friends expire within this number of seconds - ; instead of the registration interval. - -;ignoreregexpire=yes ; Enabling this setting has two functions: - ; - ; For non-realtime peers, when their registration expires, the - ; information will _not_ be removed from memory or the Asterisk database - ; if you attempt to place a call to the peer, the existing information - ; will be used in spite of it having expired - ; - ; For realtime peers, when the peer is retrieved from realtime storage, - ; the registration information will be used regardless of whether - ; it has expired or not; if it expires while the realtime peer - ; is still in memory (due to caching or other reasons), the - ; information will not be removed from realtime storage +;rtcachefriends=yes ; Cache realtime friends by adding them to the internal list + ; just like friends added from the config file only on a + ; as-needed basis? (yes|no) + +;rtsavesysname=yes ; Save systemname in realtime database at registration + ; Default= no + +;rtupdate=yes ; Send registry updates to database using realtime? (yes|no) + ; If set to yes, when a SIP UA registers successfully, the ip address, + ; the origination port, the registration period, and the username of + ; the UA will be set to database via realtime. + ; If not present, defaults to 'yes'. +;rtautoclear=yes ; Auto-Expire friends created on the fly on the same schedule + ; as if it had just registered? (yes|no|<seconds>) + ; If set to yes, when the registration expires, the friend will + ; vanish from the configuration until requested again. If set + ; to an integer, friends expire within this number of seconds + ; instead of the registration interval. + +;ignoreregexpire=yes ; Enabling this setting has two functions: + ; + ; For non-realtime peers, when their registration expires, the + ; information will _not_ be removed from memory or the Asterisk database + ; if you attempt to place a call to the peer, the existing information + ; will be used in spite of it having expired + ; + ; For realtime peers, when the peer is retrieved from realtime storage, + ; the registration information will be used regardless of whether + ; it has expired or not; if it expires while the realtime peer + ; is still in memory (due to caching or other reasons), the + ; information will not be removed from realtime storage ;----------------------------------------- SIP DOMAIN SUPPORT ------------------------ ; Incoming INVITE and REFER messages can be matched against a list of 'allowed' @@ -387,22 +387,22 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; allowexternaldomains=no ;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 - ; 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. + ; 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 + ; 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 @@ -439,8 +439,8 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; realms. We match realm on the proxy challenge and pick an set of ; credentials from this list ; Syntax: -; auth = <user>:<secret>@<realm> -; auth = <user>#<md5secret>@<realm> +; auth = <user>:<secret>@<realm> +; auth = <user>#<md5secret>@<realm> ; Example: ;auth=mark:topsecret@digium.com ; @@ -454,7 +454,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; User config options: Peer configuration: ; -------------------- ------------------- ; context context -; callingpres callingpres +; callingpres callingpres ; permit permit ; deny deny ; secret secret @@ -474,15 +474,15 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; useclientcode useclientcode ; accountcode accountcode ; setvar setvar -; callerid callerid -; amaflags amaflags -; call-limit call-limit -; allowoverlap allowoverlap -; allowsubscribe allowsubscribe -; allowtransfer allowtransfer -; subscribecontext subscribecontext -; videosupport videosupport -; maxcallbitrate maxcallbitrate +; callerid callerid +; amaflags amaflags +; call-limit call-limit +; allowoverlap allowoverlap +; allowsubscribe allowsubscribe +; allowtransfer allowtransfer +; subscribecontext subscribecontext +; videosupport videosupport +; maxcallbitrate maxcallbitrate ; rfc2833compensate mailbox ; t38pt_usertpsource username ; template @@ -509,25 +509,25 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ;host=fwd.pulver.com ;[sip_proxy-out] -;type=peer ; we only want to call out, not be called +;type=peer ; we only want to call out, not be called ;secret=guessit -;username=yourusername ; Authentication user for outbound proxies -;fromuser=yourusername ; Many SIP providers require this! -;fromdomain=provider.sip.domain +;username=yourusername ; Authentication user for outbound proxies +;fromuser=yourusername ; Many SIP providers require this! +;fromdomain=provider.sip.domain ;host=box.provider.com -;usereqphone=yes ; This provider requires ";user=phone" on URI -;call-limit=5 ; permit only 5 simultaneous outgoing calls to this peer -;outboundproxy=proxy.provider.domain ; send outbound signaling to this proxy, not directly to the peer - ; Call-limits will not be enforced on real-time peers, - ; since they are not stored in-memory -;port=80 ; The port number we want to connect to on the remote side - ; Also used as "defaultport" in combination with "defaultip" settings +;usereqphone=yes ; This provider requires ";user=phone" on URI +;call-limit=5 ; permit only 5 simultaneous outgoing calls to this peer +;outboundproxy=proxy.provider.domain ; send outbound signaling to this proxy, not directly to the peer + ; Call-limits will not be enforced on real-time peers, + ; since they are not stored in-memory +;port=80 ; The port number we want to connect to on the remote side + ; Also used as "defaultport" in combination with "defaultip" settings ;------------------------------------------------------------------------------ ; Definitions of locally connected SIP devices ; -; type = user a device that authenticates to us by "from" field to place calls -; type = peer a device we place calls to or that calls us and we match by host +; type = user a device that authenticates to us by "from" field to place calls +; type = peer a device we place calls to or that calls us and we match by host ; type = friend two configurations (peer+user) in one ; ; For device names, we recommend using only a-z, numerics (0-9) and underscore @@ -540,129 +540,129 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; Also, turn on qualify=yes to keep the nat session open ;[grandstream1] -;type=friend -;context=from-sip ; Where to start in the dialplan when this phone calls -;callerid=John Doe <1234> ; Full caller ID, to override the phones config - ; on incoming calls to Asterisk -;host=192.168.0.23 ; we have a static but private IP address - ; No registration allowed -;nat=no ; there is not NAT between phone and Asterisk -;canreinvite=yes ; allow RTP voice traffic to bypass Asterisk -;dtmfmode=info ; either RFC2833 or INFO for the BudgeTone -;call-limit=1 ; permit only 1 outgoing call and 1 incoming call at a time - ; from the phone to asterisk - ; 1 for the explicit peer, 1 for the explicit user, - ; remember that a friend equals 1 peer and 1 user in - ; memory - ; This will affect your subscriptions as well. - ; There is no combined call counter for a "friend" - ; so there's currently no way in sip.conf to limit - ; to one inbound or outbound call per phone. Use - ; the group counters in the dial plan for that. - ; -;mailbox=1234@default ; mailbox 1234 in voicemail context "default" -;disallow=all ; need to disallow=all before we can use allow= -;allow=ulaw ; Note: In user sections the order of codecs - ; listed with allow= does NOT matter! +;type=friend +;context=from-sip ; Where to start in the dialplan when this phone calls +;callerid=John Doe <1234> ; Full caller ID, to override the phones config + ; on incoming calls to Asterisk +;host=192.168.0.23 ; we have a static but private IP address + ; No registration allowed +;nat=no ; there is not NAT between phone and Asterisk +;canreinvite=yes ; allow RTP voice traffic to bypass Asterisk +;dtmfmode=info ; either RFC2833 or INFO for the BudgeTone +;call-limit=1 ; permit only 1 outgoing call and 1 incoming call at a time + ; from the phone to asterisk + ; 1 for the explicit peer, 1 for the explicit user, + ; remember that a friend equals 1 peer and 1 user in + ; memory + ; This will affect your subscriptions as well. + ; There is no combined call counter for a "friend" + ; so there's currently no way in sip.conf to limit + ; to one inbound or outbound call per phone. Use + ; the group counters in the dial plan for that. + ; +;mailbox=1234@default ; mailbox 1234 in voicemail context "default" +;disallow=all ; need to disallow=all before we can use allow= +;allow=ulaw ; Note: In user sections the order of codecs + ; listed with allow= does NOT matter! ;allow=alaw -;allow=g723.1 ; Asterisk only supports g723.1 pass-thru! -;allow=g729 ; Pass-thru only unless g729 license obtained -;callingpres=allowed_passed_screen ; Set caller ID presentation - ; See doc/callingpres.txt for more information +;allow=g723.1 ; Asterisk only supports g723.1 pass-thru! +;allow=g729 ; Pass-thru only unless g729 license obtained +;callingpres=allowed_passed_screen ; Set caller ID presentation + ; See doc/callingpres.txt for more information ;[xlite1] ; Turn off silence suppression in X-Lite ("Transmit Silence"=YES)! ; Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed ;type=friend -;regexten=1234 ; When they register, create extension 1234 +;regexten=1234 ; When they register, create extension 1234 ;callerid="Jane Smith" <5678> -;host=dynamic ; This device needs to register -;nat=yes ; X-Lite is behind a NAT router -;canreinvite=no ; Typically set to NO if behind NAT +;host=dynamic ; This device needs to register +;nat=yes ; X-Lite is behind a NAT router +;canreinvite=no ; Typically set to NO if behind NAT ;disallow=all -;allow=gsm ; GSM consumes far less bandwidth than ulaw +;allow=gsm ; GSM consumes far less bandwidth than ulaw ;allow=ulaw ;allow=alaw -;mailbox=1234@default,1233@default ; Subscribe to status of multiple mailboxes +;mailbox=1234@default,1233@default ; Subscribe to status of multiple mailboxes ;[snom] -;type=friend ; Friends place calls and receive calls -;context=from-sip ; Context for incoming calls from this user +;type=friend ; Friends place calls and receive calls +;context=from-sip ; Context for incoming calls from this user ;secret=blah -;subscribecontext=localextensions ; Only allow SUBSCRIBE for local extensions -;language=de ; Use German prompts for this user -;host=dynamic ; This peer register with us -;dtmfmode=inband ; Choices are inband, rfc2833, or info -;defaultip=192.168.0.59 ; IP used until peer registers +;subscribecontext=localextensions ; Only allow SUBSCRIBE for local extensions +;language=de ; Use German prompts for this user +;host=dynamic ; This peer register with us +;dtmfmode=inband ; Choices are inband, rfc2833, or info +;defaultip=192.168.0.59 ; IP used until peer registers ;mailbox=1234@context,2345 ; Mailbox(-es) for message waiting indicator -;subscribemwi=yes ; Only send notifications if this phone - ; subscribes for mailbox notification -;vmexten=voicemail ; dialplan extension to reach mailbox - ; sets the Message-Account in the MWI notify message - ; defaults to global vmexten which defaults to "asterisk" +;subscribemwi=yes ; Only send notifications if this phone + ; subscribes for mailbox notification +;vmexten=voicemail ; dialplan extension to reach mailbox + ; sets the Message-Account in the MWI notify message + ; defaults to global vmexten which defaults to "asterisk" ;disallow=all -;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw! +;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw! ;[polycom] -;type=friend ; Friends place calls and receive calls -;context=from-sip ; Context for incoming calls from this user +;type=friend ; Friends place calls and receive calls +;context=from-sip ; Context for incoming calls from this user ;secret=blahpoly -;host=dynamic ; This peer register with us -;dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info -;username=polly ; Username to use in INVITE until peer registers - ; Normally you do NOT need to set this parameter +;host=dynamic ; This peer register with us +;dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info +;username=polly ; Username to use in INVITE until peer registers + ; Normally you do NOT need to set this parameter ;disallow=all ;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw! -;progressinband=no ; Polycom phones don't work properly with "never" +;progressinband=no ; Polycom phones don't work properly with "never" ;[pingtel] ;type=friend ;secret=blah ;host=dynamic -;insecure=port ; Allow matching of peer by IP address without - ; matching port number -;insecure=invite ; Do not require authentication of incoming INVITEs -;insecure=port,invite ; (both) -;qualify=1000 ; Consider it down if it's 1 second to reply - ; Helps with NAT session - ; qualify=yes uses default value +;insecure=port ; Allow matching of peer by IP address without + ; matching port number +;insecure=invite ; Do not require authentication of incoming INVITEs +;insecure=port,invite ; (both) +;qualify=1000 ; Consider it down if it's 1 second to reply + ; Helps with NAT session + ; qualify=yes uses default value ; ; Call group and Pickup group should be in the range from 0 to 63 ; -;callgroup=1,3-4 ; We are in caller groups 1,3,4 -;pickupgroup=1,3-5 ; We can do call pick-p for call group 1,3,4,5 -;defaultip=192.168.0.60 ; IP address to use if peer has not registered -;deny=0.0.0.0/0.0.0.0 ; ACL: Control access to this account based on IP address +;callgroup=1,3-4 ; We are in caller groups 1,3,4 +;pickupgroup=1,3-5 ; We can do call pick-p for call group 1,3,4,5 +;defaultip=192.168.0.60 ; IP address to use if peer has not registered +;deny=0.0.0.0/0.0.0.0 ; ACL: Control access to this account based on IP address ;permit=192.168.0.60/255.255.255.0 ;[cisco1] ;type=friend ;secret=blah -;qualify=200 ; Qualify peer is no more than 200ms away -;nat=yes ; This phone may be natted - ; Send SIP and RTP to the IP address that packet is - ; received from instead of trusting SIP headers -;host=dynamic ; This device registers with us -;canreinvite=no ; Asterisk by default tries to redirect the - ; RTP media stream (audio) to go directly from - ; the caller to the callee. Some devices do not - ; support this (especially if one of them is - ; behind a NAT). -;defaultip=192.168.0.4 ; IP address to use until registration -;username=goran ; Username to use when calling this device before registration - ; Normally you do NOT need to set this parameter -;setvar=CUSTID=5678 ; Channel variable to be set for all calls from this device +;qualify=200 ; Qualify peer is no more than 200ms away +;nat=yes ; This phone may be natted + ; Send SIP and RTP to the IP address that packet is + ; received from instead of trusting SIP headers +;host=dynamic ; This device registers with us +;canreinvite=no ; Asterisk by default tries to redirect the + ; RTP media stream (audio) to go directly from + ; the caller to the callee. Some devices do not + ; support this (especially if one of them is + ; behind a NAT). +;defaultip=192.168.0.4 ; IP address to use until registration +;username=goran ; Username to use when calling this device before registration + ; Normally you do NOT need to set this parameter +;setvar=CUSTID=5678 ; Channel variable to be set for all calls from this device ;[pre14-asterisk] ;type=friend ;secret=digium ;host=dynamic -;rfc2833compensate=yes ; Compensate for pre-1.4 DTMF transmission from another Asterisk machine. - ; You must have this turned on or DTMF reception will work improperly. +;rfc2833compensate=yes ; Compensate for pre-1.4 DTMF transmission from another Asterisk machine. + ; You must have this turned on or DTMF reception will work improperly. ;t38pt_usertpsource=yes ; Use the source IP address of RTP as the destination IP address for UDPTL packets ; if the nat option is enabled. If a single RTP packet is received Asterisk will know the ; external IP address of the remote device. If port forwarding is done at the client side |