fixup sip

[lcr.git] / default / interface.conf

diff --git a/default/interface.conf b/default/interface.conf
index 51cc036..ab62ca8 100644 (file)
--- a/default/interface.conf
+++ b/default/interface.conf
@@ -2,15 +2,17 @@
 ################
 
 
-# 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
 
@@ -19,6 +21,7 @@
 # Layer-2-Hold is required to keep link alive.
 
 #[Ext]
+#extern
 #portnum 0
 #ptp
 #layer2hold yes
@@ -73,8 +76,9 @@
 # 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
 
@@ -88,6 +92,7 @@
 # (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
@@ -117,6 +122,7 @@
 # Now all information elements can be transmitted in both directions.
 
 #[Ext]
+#extern
 #portnum 0
 #te-special
 
@@ -133,9 +139,118 @@
 # 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 <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-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
+#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.
 
 
@@ -143,6 +258,7 @@
 
 
 [Ext]
+extern
 portnum 0