#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
##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.
#[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
# Use Sofia-SIP as SIP point-to-point interface
#[sip]
-#sip <local ip> <remote ip>[:<remote port>]
+#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.
projects / lcr.git / blobdiff