X-Git-Url: http://git.eversberg.eu/gitweb.cgi?p=lcr.git;a=blobdiff_plain;f=default%2Finterface.conf;h=ab62ca85d41518cd8c0e9aadf8cc12745ff8bf38;hp=f6af8df735866d718a906f906d38c42cbe70ff10;hb=ef08f1213998f3bfd7bc3d95ab7c4917725fb3e2;hpb=f6aea744f84e702b3469393f007b9e1bf25f6737

diff --git a/default/interface.conf b/default/interface.conf
index f6af8df..ab62ca8 100644
--- a/default/interface.conf
+++ b/default/interface.conf
@@ -144,24 +144,26 @@
 #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 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
 
@@ -176,9 +178,16 @@
 ##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 interface requires mISDN_l1loop.ko to be loaded:
-#  modprobe mISDN_l1loop nchannel=8 # use up to 8 b-channels
 # 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. 
@@ -199,6 +208,48 @@
 #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.