X-Git-Url: http://git.eversberg.eu/gitweb.cgi?p=lcr.git;a=blobdiff_plain;f=default%2Finterface.conf;h=ab62ca85d41518cd8c0e9aadf8cc12745ff8bf38;hp=daa8c571f2a8cbdb260ed0599685e442357a40c4;hb=79bd731c0db3e3202cfeed2af3fb217ae744b70f;hpb=b2a665f8f1cdeb7d02c3f665d95e6a80297e21d1 diff --git a/default/interface.conf b/default/interface.conf index daa8c57..ab62ca8 100644 --- a/default/interface.conf +++ b/default/interface.conf @@ -2,26 +2,31 @@ ################ -# Example of an ISDN interface on port 0 +# Example of an ISDN interface on port 0 used for external calls. #[Ext] +#extern #portnum 0 # Example of a multilink ISDN interface (Anlagenanschluss) on port 2 # and 3 #[Ext] +#extern #portnum 2 #portnum 3 + # Example of an PTP ISDN interface on port 0 # Layer-2-Hold is required to keep link alive. #[Ext] +#extern #portnum 0 #ptp #layer2hold yes + # Example of an internal ISDN interface on port 1, which accepts all extensions #[Int] @@ -69,9 +74,11 @@ # This is required if the connected line doesn't screen caller IDs. # Also this interface will connect bchannel during call setup, so tones are # required. + #[Ext] +#extern #screen-out unknown 300 national 21250993300 -#screen-out unknown 2* national 212509932* +#screen-out unknown 2% national 212509932% #tones yes #portnum 0 @@ -83,17 +90,21 @@ # We prefer to use directed channels first, then we request any channel. # Outgoing calls on both-way-channels shall be indicated as "any channel". # (Siemens EWSD will select bot-way-channels when indicated that way.) + #[Ext] +#extern #portnum 0 #layer2hold #channel-in 1,2,3,4,5,6,7,8,9,10,22,23,24,25,26,27,28,29,30,31 #channel-out force,11,12,13,14,15,17,18,19,20,21,any + # Example of an ISDN interface that runs in NT-mode, but provides tones during # setup. Also we provide tones during setup also. # This is usefull to interconnect to another PBX. # Additinally the timeout values for the different call states are adjusted to 60 seconds. # They are: setup, dialing, proceeding, alerting, disconnect + #[PBX] #portnum 4 #nt @@ -102,17 +113,152 @@ #tones yes #timeouts 60 60 60 60 60 + +# Example of an interface on port 0 connected to another LCR +# This can be done by direct cross cable (terminated of course) or via L1oIP. +# Since the remote side (NT-mode) normally doesn't accept informations like +# redirected number or display facility, use 'te-special' to even transmit that +# against the ISDN specifications. A remote LCR can handle that. +# Now all information elements can be transmitted in both directions. + +#[Ext] +#extern +#portnum 0 +#te-special + + # Alternatively give port name. You will find the name with 'isdninfo' tool. + #[Int2] #portname hfc-s_usb.1 #nt + +# The remote switch may reject extreamly large numbers to be dialed during +# setup message. Define a limit of maximum numbers to dial. The rest of +# digits will be dialed after setup via overlap dialing. + +#[Ext] +#extern +#portnum 0 +#dialmax 20 + + +# Example of an ISDN interface on port 1, with alternate tones-dir to use. +# In this case, the tones are "german" tones generated by mISDN_dsp.ko. +# It is possible to give different sample sets, like "tones_german". + +#[Int] +#extension +#msn 201,202,203 +#tones-dir german +#portnum 1 +#nt + + +# A special case for GSM Network interface. +# optionally give "gsm-bs ". +# You may add 'extension' and 'msn' keywords to turn all your subscribers +# in you GSM network to internal 'extensions'. +# The MSN numbers will equal the subscriber number. +#[GSM] +#gsm-bs +#hr +#tones yes +#earlyb no + + +# A special case for GSM Mobile Station interface. +# give "gsm-ms ". +# You may add 'extern' to make this interface the external line by default. +#[GSM] +#gsm-ms 1 +#tones no +#earlyb yes +##extern + + +# Use chan_lcr (Asterisk PBX interface) as external interface. +#[Ext] +#remote asterisk +#exten from-lcr +#extern +#earlyb yes +#tones no + + +# Use chan_lcr (Asterisk PBX interface) as internal interface. +# The caller ID is used as extension, if "extension" parameter is given. +# Use "screen-in % xxx" to modify any caller id to xxx. +# An internal extension does not receive tones ("earlyb"), but sends them. +#[ast] +#remote asterisk +#exten from-lcr +##note: The following keyword means that this interface is an LCR internal extension +#extension +##screen-in % 209 +#earlyb no +#tones yes + + +# Use Sofia-SIP as SIP point-to-point interface +#[sip] +#sip [:] [:] +#sip 10.0.0.12 10.0.0.34 +#earlyb no +#tones no + +# Use Sofia-SIP as SIP client to register to a SIP gateway/proxy +#[sip] +## define source and destination IP to make a call +#sip 192.168.0.55 sipgate.de +## define [] to register to a SIP gateway +#register sipgate.de 300 +##define RTP port range or use default +#rtp-ports 30000 39999 +## use authentication credentials, use realm to authenticate remote +#authenticate [] +## define keepalive timer to keep INVITE/REGISTER alive +## this is also required to keep the NAT router's table alive +#options-interval 15 +## define asserted ID (real caller ID) to use no screening CLIP +##asserted-id +## define public IP (if behind NAT firewall) +#public 123.45.67.89 +## OR define stun server and resolving interval +#stun stun.sipgate.net 300 +## screen caller ID to SIP caller ID +#screen-out % +#tones yes +#earlyb yes + +# Use Sofia-SIP as SIP gateway/proxy to allow SIP clients to register +#[sip] +## define source +#sip 192.168.0.55 +##define RTP port range or use default +#rtp-ports 30000 39999 +## use authentication credentials and realm to authenticate remote +#authenticate +## define keepalive timer to keep INVITE/REGISTER alive +## this is also required to keep the NAT router's table alive +#options-interval 15 +## define public IP (if behind NAT firewall) +#public 123.45.67.89 +## OR define stun server and resolving interval +#stun stun.sipgate.net 300 +#tones yes +#earlyb yes + + # Hint: Enter "lcr interface" for quick help on interface options. + # Add your interfaces here: [Ext] +extern portnum 0