X-Git-Url: http://git.eversberg.eu/gitweb.cgi?p=lcr.git;a=blobdiff_plain;f=default%2Finterface.conf;h=ab62ca85d41518cd8c0e9aadf8cc12745ff8bf38;hp=85e53f1ec15ca62c56bfe639d044c9e1dad2f04d;hb=ef08f1213998f3bfd7bc3d95ab7c4917725fb3e2;hpb=feea42c7f7f2e171c1490bd0d3af09beba629f21

diff --git a/default/interface.conf b/default/interface.conf
index 85e53f1..ab62ca8 100644
--- a/default/interface.conf
+++ b/default/interface.conf
@@ -2,17 +2,17 @@
 ################
 
 
-# Example of an ISDN interface on port 0
+# Example of an ISDN interface on port 0 used for external calls.
 
 #[Ext]
-#external
+#extern
 #portnum 0
 
 # Example of a multilink ISDN interface (Anlagenanschluss) on port 2
 # and 3
 
 #[Ext]
-#external
+#extern
 #portnum 2
 #portnum 3
 
@@ -21,7 +21,7 @@
 # Layer-2-Hold is required to keep link alive.
 
 #[Ext]
-#external
+#extern
 #portnum 0
 #ptp
 #layer2hold yes
@@ -76,9 +76,9 @@
 # required.
 
 #[Ext]
-#external
+#extern
 #screen-out unknown 300  national 21250993300
-#screen-out unknown 2*  national 212509932*
+#screen-out unknown 2%  national 212509932%
 #tones yes
 #portnum 0
 
@@ -92,7 +92,7 @@
 # (Siemens EWSD will select bot-way-channels when indicated that way.)
 
 #[Ext]
-#external
+#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
@@ -122,7 +122,7 @@
 # Now all information elements can be transmitted in both directions.
 
 #[Ext]
-#external
+#extern
 #portnum 0
 #te-special
 
@@ -139,39 +139,116 @@
 # digits will be dialed after setup via overlap dialing.
 
 #[Ext]
-#external
+#extern
 #portnum 0
 #dialmax 20
 
 
-# Example of an ISDN interface on port 1, with alternate tones_dir to use.
+# 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
+#tones-dir german
 #portnum 1
 #nt
 
 
-# A special case for GSM interface.
-# Don't remove/change the settings, they will cause undefined behaviour
-# of LCR. The actual interface is defined in gsm.conf.
-# You may add 'extension' and 'mns' keywords to turn all your subscribers
+# A special case for GSM Network interface.
+# optionally give "gsm-bs <name of network instance>".
+# 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
-#nt
-#layer1hold no
-#layer2hold no
+#gsm-bs
+#hr
+#tones yes
+#earlyb no
+
+
+# A special case for GSM Mobile Station interface.
+# give "gsm-ms <name of mobile instance>".
+# 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 <local ip>[:<local port>] <remote ip>[:<remote port>]
+#sip 10.0.0.12 10.0.0.34
 #earlyb no
-#channel-in free
-#channel-out any
-#nodtmf
+#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 <user> <host> [<options-interval>] to register to a SIP gateway
+#register <user> sipgate.de 300
+##define RTP port range or use default
+#rtp-ports 30000 39999
+## use authentication credentials, use realm to authenticate remote
+#authenticate <user> <password> [<realm>]
+## 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 <my real phone number>
+## 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 % <callerID>
+#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 <user> <password> <realm>
+## 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.
@@ -181,7 +258,7 @@
 
 
 [Ext]
-external
+extern
 portnum 0