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| -rw-r--r-- | configs/sip.conf.sample | 867 |
1 files changed, 425 insertions, 442 deletions
diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample index 6b1d0a256..633cbe076 100644 --- a/configs/sip.conf.sample +++ b/configs/sip.conf.sample @@ -5,51 +5,51 @@ ;----------------------------------------------------------- ; In the dialplan (extensions.conf) you can use several ; syntaxes for dialing SIP devices. -; SIP/devicename -; SIP/username@domain (SIP uri) -; SIP/username[:password[:md5secret[:authname[:transport]]]]@host[:port] -; SIP/devicename/extension +; SIP/devicename +; SIP/username@domain (SIP uri) +; SIP/username[:password[:md5secret[:authname[:transport]]]]@host[:port] +; SIP/devicename/extension ; ; ; Devicename -; devicename is defined as a peer in a section below. +; devicename is defined as a peer in a section below. ; ; username@domain -; Call any SIP user on the Internet -; (Don't forget to enable DNS SRV records if you want to use this) +; Call any SIP user on the Internet +; (Don't forget to enable DNS SRV records if you want to use this) ; ; devicename/extension -; If you define a SIP proxy as a peer below, you may call -; SIP/proxyhostname/user or SIP/user@proxyhostname -; where the proxyhostname is defined in a section below -; This syntax also works with ATA's with FXO ports +; If you define a SIP proxy as a peer below, you may call +; SIP/proxyhostname/user or SIP/user@proxyhostname +; where the proxyhostname is defined in a section below +; This syntax also works with ATA's with FXO ports ; ; SIP/username[:password[:md5secret[:authname]]]@host[:port] -; This form allows you to specify password or md5secret and authname -; without altering any authentication data in config. -; Examples: +; This form allows you to specify password or md5secret and authname +; without altering any authentication data in config. +; Examples: ; -; SIP/*98@mysipproxy -; SIP/sales:topsecret::account02@domain.com:5062 -; SIP/12345678::bc53f0ba8ceb1ded2b70e05c3f91de4f:myname@192.168.0.1 +; SIP/*98@mysipproxy +; SIP/sales:topsecret::account02@domain.com:5062 +; SIP/12345678::bc53f0ba8ceb1ded2b70e05c3f91de4f:myname@192.168.0.1 ; ; All of these dial strings specify the SIP request URI. ; In addition, you can specify a specific To: header by adding an ; exclamation mark after the dial string, like ; -; SIP/sales@mysipproxy!sales@edvina.net +; SIP/sales@mysipproxy!sales@edvina.net ; ; CLI Commands ; ------------------------------------------------------------- ; 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 set debug Show all SIP messages +; sip set debug Show all SIP messages ; -; sip reload 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 ; ; ** Deprecated configuration options ** @@ -62,24 +62,24 @@ ; "setvar" to set variables that can be used in the dialplan for various limits. [general] -context=default ; Default context for incoming calls -;allowguest=no ; Allow or reject guest calls (default is yes) -;match_auth_username=yes ; if available, match user entry using the - ; 'username' field from the authentication line - ; instead of the From: field. -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 for unencrypted UDP - ; and TCP sessions is 5060) - ; bindport is the local UDP port that Asterisk will listen on -bindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all) - ; You can specify port here too, like 123.123.123.123:5080 +context=default ; Default context for incoming calls +;allowguest=no ; Allow or reject guest calls (default is yes) +;match_auth_username=yes ; if available, match user entry using the + ; 'username' field from the authentication line + ; instead of the From: field. +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 for unencrypted UDP + ; and TCP sessions is 5060) + ; bindport is the local UDP port that Asterisk will listen on +bindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all) + ; You can specify port here too, like 123.123.123.123:5080 ; ; Note that the TCP and TLS support for chan_sip is currently considered @@ -88,50 +88,50 @@ bindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds to al ; be reflected in this sample configuration file, as well as in the UPGRADE.txt file. ; tcpenable=no ; Enable server for incoming TCP connections (default is no) -tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces) +tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces) ; Optionally add a port number, 192.168.1.1:5062 (default is port 5060) ;tlsenable=no ; Enable server for incoming TLS (secure) connections (default is no) ;tlsbindaddr=0.0.0.0 ; IP address for TLS server to bind to (0.0.0.0) binds to all interfaces) ; Optionally add a port number, 192.168.1.1:5063 (default is port 5061) - ; Remember that the IP address must match the common name (hostname) in the - ; certificate, so you don't want to bind a TLS socket to multiple IP addresses. + ; Remember that the IP address must match the common name (hostname) in the + ; certificate, so you don't want to bind a TLS socket to multiple IP addresses. -;tlscertfile=asterisk.pem ; Certificate file (*.pem only) to use for TLS connections - ; default is to look for "asterisk.pem" in current directory +;tlscertfile=asterisk.pem ; Certificate file (*.pem only) to use for TLS connections + ; default is to look for "asterisk.pem" in current directory ;tlscafile=</path/to/certificate> -; If the server your connecting to uses a self signed certificate -; you should have their certificate installed here so the code can -; verify the authenticity of their certificate. +; If the server your connecting to uses a self signed certificate +; you should have their certificate installed here so the code can +; verify the authenticity of their certificate. ;tlscadir=</path/to/ca/dir> -; A directory full of CA certificates. The files must be named with -; the CA subject name hash value. -; (see man SSL_CTX_load_verify_locations for more info) +; A directory full of CA certificates. The files must be named with +; the CA subject name hash value. +; (see man SSL_CTX_load_verify_locations for more info) ;tlsdontverifyserver=[yes|no] -; If set to yes, don't verify the servers certificate when acting as -; a client. If you don't have the server's CA certificate you can -; set this and it will connect without requiring tlscafile to be set. -; Default is no. +; If set to yes, don't verify the servers certificate when acting as +; a client. If you don't have the server's CA certificate you can +; set this and it will connect without requiring tlscafile to be set. +; Default is no. ;tlscipher=<SSL cipher string> -; A string specifying which SSL ciphers to use or not use -; A list of valid SSL cipher strings can be found at: -; http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS - -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") +; A string specifying which SSL ciphers to use or not use +; A list of valid SSL cipher strings can be found at: +; http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS + +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 qos.tex or Quality of Service section of asterisk.pdf for a description of these parameters. ;tos_sip=cs3 ; Sets TOS for SIP packets. @@ -144,24 +144,24 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ;cos_video=4 ; Sets 802.1p priority for RTP video packets. ;cos_text=3 ; Sets 802.1p priority for RTP text 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 +;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 ;qualifyfreq=60 ; Qualification: How often to check for the ; host to be up in seconds ; Set to low value if you use low timeout for ; NAT of UDP sessions -;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY -;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 +;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY +;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 @@ -178,63 +178,60 @@ 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 - ; The default user agent string also contains the Asterisk - ; version. If you don't want to expose this, change the - ; useragent string. -;sdpsession=Asterisk PBX ; Allows you to change the SDP session name string, (s=) - ; Like the useragent parameter, the default user agent string - ; also contains the Asterisk version. -;sdpowner=root ; Allows you to change the username field in the SDP owner string, (o=) - ; This field MUST NOT contain spaces -;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 (application/dtmf-relay) - ; shortinfo : SIP INFO messages (application/dtmf) - ; 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 :-( -;outboundproxy=proxy.provider.domain ; send outbound signaling to this proxy, not directly to the devices +;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 + ; The default user agent string also contains the Asterisk + ; version. If you don't want to expose this, change the + ; useragent string. +;sdpsession=Asterisk PBX ; Allows you to change the SDP session name string, (s=) + ; Like the useragent parameter, the default user agent string + ; also contains the Asterisk version. +;sdpowner=root ; Allows you to change the username field in the SDP owner string, (o=) + ; This field MUST NOT contain spaces +;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 (application/dtmf-relay) + ; shortinfo : SIP INFO messages (application/dtmf) + ; inband : Inband audio (requires 64 kbit codec -alaw, ulaw) + ; auto : Use rfc2833 if offered, inband otherwise +;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 :-( +;outboundproxy=proxy.provider.domain ; send outbound signaling to this proxy, not directly to the devices ;outboundproxy=proxy.provider.domain:8080 ; send outbound signaling to this proxy, not directly to the devices ;outboundproxy=proxy.provider.domain,force ; Send ALL outbound signalling to proxy, ignoring route: headers -;outboundproxy=tls://proxy.provider.domain ; same as '=proxy.provider.domain' except we try to connect with tls +;outboundproxy=tls://proxy.provider.domain ; same as '=proxy.provider.domain' except we try to connect with tls ; ; (could also be tcp,udp) - defining transports on the proxy line only ; ; applies for the global proxy, otherwise use the transport= option ;matchexterniplocally = yes ; Only substitute the externip or externhost setting if it matches @@ -253,40 +250,40 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; separated by '&'. Patterns may be used in regexten. ; ;regcontext=sipregistrations -;regextenonqualify=yes ; Default "no" - ; If you have qualify on and the peer becomes unreachable - ; this setting will enforce inactivation of the regexten - ; extension for the peer +;regextenonqualify=yes ; Default "no" + ; If you have qualify on and the peer becomes unreachable + ; this setting will enforce inactivation of the regexten + ; extension for the peer ; ;--------------------------- 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. ; -;t1min=100 ; Minimum roundtrip time for messages to monitored hosts - ; Defaults to 100 ms -;timert1=500 ; Default T1 timer - ; Defaults to 500 ms or the measured round-trip - ; time to a peer (qualify=yes). -;timerb=32000 ; Call setup timer. If a provisional response is not received - ; in this amount of time, the call will autocongest - ; Defaults to 64*timert1 +;t1min=100 ; Minimum roundtrip time for messages to monitored hosts + ; Defaults to 100 ms +;timert1=500 ; Default T1 timer + ; Defaults to 500 ms or the measured round-trip + ; time to a peer (qualify=yes). +;timerb=32000 ; Call setup timer. If a provisional response is not received + ; in this amount of time, the call will autocongest + ; Defaults to 64*timert1 ;--------------------------- 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 ; -;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 Session-Timers (RFC 4028)------------------------------------ ; SIP Session-Timers provide an end-to-end keep-alive mechanism for active SIP sessions. @@ -308,28 +305,14 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ;session-minse=90 ;session-refresher=uas ; -;--------------------------- HASH TABLE SIZES ------------------------------------------------ -; For maximum efficiency, adjust the following -; values to be slightly larger than the maximum number of in-memory objects (devices). -; Too large, and space is wasted. Too small, and things will run slower. -; 563 is probably way too big for small (home) applications, but it -; should cover most small/medium sites. -; It is recommended to make the sizes be a prime number! -; This was internally set to 17 for small-memory applications... -; All tables default to 563, except when compiled in LOW_MEMORY mode, -; in which case, they default to 17. You can override this by uncommenting -; the following, and changing the values. -;hash_users=563 -;hash_peers=563 -;hash_dialogs=563 ;--------------------------- 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) ---------------------------- @@ -350,26 +333,26 @@ 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. -;callcounter = yes ; Enable call counters on devices. This can be set per - ; device too. -;counteronpeer = yes ; Apply call counting 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 counter instead of applying two call counters, - ; 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. +;callcounter = yes ; Enable call counters on devices. This can be set per + ; device too. +;counteronpeer = yes ; Apply call counting 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 counter instead of applying two call counters, + ; 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 ----------------------- ; @@ -400,14 +383,14 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; A similar effect can be achieved by adding a "callbackextension" option in a peer section. ; this is equivalent to having the following line in the general section: ; -; register => username:secret@host/callbackextension +; register => username:secret@host/callbackextension ; ; and more readable because you don't have to write the parameters in two places ; (note that the "port" is ignored - this is a bug that should be fixed). ; ; Examples: ; -;register => 1234:password@mysipprovider.com +;register => 1234:password@mysipprovider.com ; ; This will pass incoming calls to the 's' extension ; @@ -422,11 +405,11 @@ 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 ------------------------ ; @@ -446,8 +429,8 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; Multiple entries are allowed, e.g. a reasonable set is the following: ; ; localnet=192.168.0.0/255.255.0.0 ; RFC 1918 addresses -; 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 "externally visible" address and port number to be used when talking @@ -463,9 +446,9 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; This approach can be useful if you have a NAT device where you can ; configure the mapping statically. Examples: ; -; externip = 12.34.56.78 ; use this address. -; externip = 12.34.56.78:9900 ; use this address and port. -; externip = mynat.my.org:12600 ; Public address of my nat box. +; externip = 12.34.56.78 ; use this address. +; externip = 12.34.56.78:9900 ; use this address and port. +; externip = mynat.my.org:12600 ; Public address of my nat box. ; ; b. "externhost = hostname[:port]" is similar to "externip" except ; that the hostname is looked up every "externrefresh" seconds @@ -474,8 +457,8 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; Beware, you might suffer from service disruption when the name server ; resolution fails. Examples: ; -; externhost=foo.dyndns.net ; refreshed periodically -; externrefresh=180 ; change the refresh interval +; externhost=foo.dyndns.net ; refreshed periodically +; externrefresh=180 ; change the refresh interval ; ; c. "stunaddr = stun.server[:port]" queries the STUN server specified ; as an argument to obtain the external address/port. @@ -483,8 +466,8 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; (as a side effect, sending the query also acts as a keepalive for ; the state entry on the nat box): ; -; stunaddr = foo.stun.com:3478 -; externrefresh = 15 +; stunaddr = foo.stun.com:3478 +; externrefresh = 15 ; ; Note that at the moment all these mechanism work only for the SIP socket. ; The IP address discovered with externip/externhost/STUN is reused for @@ -510,11 +493,11 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; However, this is only useful if the external traffic can reach us. ; The following settings are allowed (both globally and in individual sections): ; -; nat = no ; default. Use NAT mode only according to RFC3581 (;rport) -; nat = yes ; Always ignore info and assume NAT -; nat = never ; Never attempt NAT mode or RFC3581 support -; nat = route ; route = Assume NAT, don't send rport -; ; (work around more UNIDEN bugs) +; nat = no ; default. Use NAT mode only according to RFC3581 (;rport) +; nat = yes ; Always ignore info and assume NAT +; nat = never ; Never attempt NAT mode or RFC3581 support +; nat = route ; 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 @@ -522,72 +505,72 @@ 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. - - ; 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. + + ; 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' @@ -611,22 +594,22 @@ 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 @@ -663,8 +646,8 @@ 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 ; @@ -699,16 +682,16 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; useclientcode useclientcode ; accountcode accountcode ; setvar setvar -; callerid callerid -; amaflags amaflags -; call-limit call-limit (deprecated) +; callerid callerid +; amaflags amaflags +; call-limit call-limit (deprecated) ; callcounter callcounter -; allowoverlap allowoverlap -; allowsubscribe allowsubscribe -; allowtransfer allowtransfer -; subscribecontext subscribecontext -; videosupport videosupport -; maxcallbitrate maxcallbitrate +; allowoverlap allowoverlap +; allowsubscribe allowsubscribe +; allowtransfer allowtransfer +; subscribecontext subscribecontext +; videosupport videosupport +; maxcallbitrate maxcallbitrate ; rfc2833compensate mailbox ; session-timers busylevel ; session-expires @@ -746,38 +729,38 @@ 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 -;defaultuser=yourusername ; Authentication user for outbound proxies -;fromuser=yourusername ; Many SIP providers require this! -;fromdomain=provider.sip.domain +;defaultuser=yourusername ; Authentication user for outbound proxies +;fromuser=yourusername ; Many SIP providers require this! +;fromdomain=provider.sip.domain ;host=box.provider.com -;transport=udp,tcp ; This sets the transport type to udp for outgoing, and will -; ; accept both tcp and udp. Default is udp. The first transport -; ; listed will always be used for outgoing connections. -;usereqphone=yes ; This provider requires ";user=phone" on URI -;callcounter=yes ; Enable call counter -;busylevel=2 ; Signal busy at 2 or more calls -;outboundproxy=proxy.provider.domain ; send outbound signaling to this proxy, not directly to the peer -;port=80 ; The port number we want to connect to on the remote side - ; Also used as "defaultport" in combination with "defaultip" settings +;transport=udp,tcp ; This sets the transport type to udp for outgoing, and will +; ; accept both tcp and udp. Default is udp. The first transport +; ; listed will always be used for outgoing connections. +;usereqphone=yes ; This provider requires ";user=phone" on URI +;callcounter=yes ; Enable call counter +;busylevel=2 ; Signal busy at 2 or more calls +;outboundproxy=proxy.provider.domain ; send outbound signaling to this proxy, not directly to the peer +;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 ;[provider1] ;type=peer ;host=sip.provider1.com -;fromuser=4015552299 ; how your provider knows you +;fromuser=4015552299 ; how your provider knows you ;secret=youwillneverguessit -;callbackextension=123 ; Register with this server and require calls coming back to this extension -;transport=udp,tcp ; This sets the transport type to udp for outgoing, and will -; ; accept both tcp and udp. Default is udp. The first transport -; ; listed will always be used for outgoing connections. +;callbackextension=123 ; Register with this server and require calls coming back to this extension +;transport=udp,tcp ; This sets the transport type to udp for outgoing, and will +; ; accept both tcp and udp. Default is udp. The first transport +; ; listed will always be used for outgoing connections. ;------------------------------------------------------------------------------ ; 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 @@ -794,172 +777,172 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; the the various sections. Examples are below, and we can even leave ; the templates uncommented as they will not harm: -[basic-options](!) ; a template - dtmfmode=rfc2833 - context=from-office - type=friend +[basic-options](!) ; a template + dtmfmode=rfc2833 + context=from-office + type=friend -[natted-phone](!,basic-options) ; another template inheriting basic-options - nat=yes - canreinvite=no - host=dynamic +[natted-phone](!,basic-options) ; another template inheriting basic-options + nat=yes + canreinvite=no + host=dynamic -[public-phone](!,basic-options) ; another template inheriting basic-options - nat=no - canreinvite=yes +[public-phone](!,basic-options) ; another template inheriting basic-options + nat=no + canreinvite=yes -[my-codecs](!) ; a template for my preferred codecs - disallow=all - allow=ilbc - allow=g729 - allow=gsm - allow=g723 - allow=ulaw +[my-codecs](!) ; a template for my preferred codecs + disallow=all + allow=ilbc + allow=g729 + allow=gsm + allow=g723 + allow=ulaw -[ulaw-phone](!) ; and another one for ulaw-only - disallow=all - allow=ulaw +[ulaw-phone](!) ; and another one for ulaw-only + disallow=all + allow=ulaw ; and finally instantiate a few phones ; ; [2133](natted-phone,my-codecs) -; secret = peekaboo +; secret = peekaboo ; [2134](natted-phone,ulaw-phone) -; secret = not_very_secret +; secret = not_very_secret ; [2136](public-phone,ulaw-phone) -; secret = not_very_secret_either +; secret = not_very_secret_either ; ... ; ; Standard configurations not using templates look like this: ; ;[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 (deprecated) - ; 1 for the explicit peer, 1 for the explicit user, - ; remember that a friend equals 1 peer and 1 user in - ; memory - ; 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 (deprecated) + ; 1 for the explicit peer, 1 for the explicit user, + ; remember that a friend equals 1 peer and 1 user in + ; memory + ; 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 README.callingpres 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 README.callingpres 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 -;registertrying=yes ; Send a 100 Trying when the device registers. +;mailbox=1234@default,1233@default ; Subscribe to status of multiple mailboxes +;registertrying=yes ; Send a 100 Trying when the device registers. ;[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 -;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" +;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" ;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 -;defaultuser=polly ; Username to use in INVITE until peer registers +;host=dynamic ; This peer register with us +;dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info +;defaultuser=polly ; Username to use in INVITE until peer registers ;defaultip=192.168.40.123 - ; Normally you do NOT need to set this parameter + ; 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" +;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw! +;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 -;qualifyfreq=60 ; Qualification: How often to check for the - ; host to be up in seconds - ; Set to low value if you use low timeout for - ; NAT of UDP sessions +;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 +;qualifyfreq=60 ; Qualification: How often to check for the + ; host to be up in seconds + ; Set to low value if you use low timeout for + ; NAT of UDP sessions ; ; 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 -;defaultuser=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 +;defaultuser=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 |