################
-# Example of an ISDN interface on port 1
+# Example of an ISDN interface on port 0 used for external calls.
#[Ext]
-#port 1
+#extern
+#portnum 0
-# Example of a multilink ISDN interface (Anlagenanschluss) on port 3
-# and 4
+# Example of a multilink ISDN interface (Anlagenanschluss) on port 2
+# and 3
#[Ext]
-#port 3
-#port 4
+#extern
+#portnum 2
+#portnum 3
-# Example of an PTP ISDN interface on port 1
+
+# Example of an PTP ISDN interface on port 0
# Layer-2-Hold is required to keep link alive.
#[Ext]
-#port 1
+#extern
+#portnum 0
#ptp
#layer2hold yes
-# Example of an internal ISDN interface on port 2, which accepts all extensions
+
+# Example of an internal ISDN interface on port 1, which accepts all extensions
#[Int]
#extension
-#port 2
+#portnum 1
#nt
-# Example of an ISDN interface on port 2, which accepts only extensions
+# Example of an ISDN interface on port 1, which accepts only extensions
# 201, 202 and 203
#[Int]
#extension
#msn 201,202,203
-#port 2
+#portnum 1
#nt
-# Example of three ISDN interfaces on port 2, 3 and 4, which accept
+# Example of three ISDN interfaces on port 1, 2 and 3, which accept
# extension 201 on all interfaces, and extension 202 and 203 on the first port
# only.
# Hint: To make extension ring on all three interfaces parallel, use
#[Int1]
#extension
#msn 201,202,203
-#port 2
+#portnum 1
#nt
#[Int2]
#extension
#msn 201
-#port 3
+#portnum 2
#nt
#[Int3]
#extension
#msn 201
-#port 4
+#portnum 3
#nt
-# Example of an ISDN interface on port 1 that requires screening of caller IDs.
+# Example of an ISDN interface on port 0 that requires screening of caller IDs.
# 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
-#port 1
+#portnum 0
-# Example of an ISDN PRI interface on port 1 that has directed channels.
+# Example of an ISDN PRI interface on port 0 that has directed channels.
# 10 channels (channel 1-10) are incomming only.
# 10 channels (channel 11-15,17-21) are outgoing only.
# 10 channels (channel 22-31) are both way.
# 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]
-#port 1
+#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]
-#port 5
+#portnum 4
#nt
#layer2hold yes
#earlyb yes
#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 <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.
+
# Add your interfaces here:
[Ext]
-port 1
+extern
+portnum 0
[Int]
extension
msn 200,201,202,203
-port 2
+portnum 1
nt
projects / lcr.git / blobdiff