clip no screening ( Link to this Config )
	Configuration Description This example shows which mapping tables are used to get the "clip no screening" working. The call comes in on isdn interface 01 and passes the complex function contatining 3 mapping tables. In the isdn setup message sent to the PSTN, the calling type of number will be set to national and calling numbering-plan to E164. In case the calling party number provided by the PBX is lead by a 0 (zero) it will be stripped away. Command Sequence to Copy, Edit, and Paste context cs switch routing-table called-e164 OUT route .T dest-interface IF_BRI_00 OUTBOUND mapping-table calling-e164 to calling-e164 SET_CNPN map 0(.%) to \1 mapping-table calling-type-of-number to calling-type-of-number TYPE map default to national mapping-table calling-numbering-plan to calling-numbering-plan PLAN map default to isdn-telephony complex-function OUTBOUND execute 1 PLAN execute 2 SET_CNPN execute 3 TYPE interface isdn IF_BRI_00 route call dest-interface IF_BRI_01 interface isdn IF_BRI_01 route call dest-table OUT
Call quality log ( Link to this Config )
	Configuration Description It is possible to store call logs with quality information in the local aaa data sink. To do so, a service aaa is needed where the calls are routed through. Create a profile aaa with the method local. Then create a service aaa to route the calls through. To see the collected logs use the command show accounting or use the Web-GUI to see the logs. In the Web-GUI, go to the 'Reports' page and select “Call Quality Log” in the tab bar on top of the page. On that page, you will also have the option to export the log as a comma separated text file. The output looks like this: 172.16.71.10(port)[bri]#show accounting AAA Log ======= Call --------- General Start time: 70-02-11 00:01:01 Duration: 21 State: T:16 (originator) Calling party Number : Name : IP Address : 0.0.0.0 CODEC : MOS : 450 Octets Rx/Tx : 0/0 Packets Rx/Tx : 0/0 Lost Packets Rx/Tx : 0/0 Jitter Rx/Tx : 0/0 RTT : 0 Called party Number : Name : IP Address : 172.16.90.3 CODEC : G.711 A-law MOS : 440 Octets Rx/Tx : 0/145280 Packets Rx/Tx : 0/908 Lost Packets Rx/Tx : 0/0 Jitter Rx/Tx : 0/0 RTT : 0 Command Sequence to Copy, Edit, and Paste profile aaa default method 1 local context cs switch routing-table called-e164 TAB_OUT route 200 dest-interface IF_SIP interface isdn IF_BRI_00 route call dest-service QoS_LOG.OUT.bri interface sip IF_SIP bind context sip-gateway sip route call dest-service QoS_LOG.OUT.ethernet service aaa QoS_LOG.OUT accounting use profile aaa LOCAL_AAA port bri route call dest-table TAB_OUT port ethernet route call dest-interface IF_BRI_00
Service Priority ( Link to this Config )
	Configuration Description The service ‘priority’ can automatically free resources if a high priority call needs to be established while no resources are available. The service ‘priority’ can have multiple ports. You can assign a priority level for each port. This priority level defines the priority level of each call, which is received through the port. If a call with higher priority fails to be established, the service tries dropping lower priority calls to free resources for the higher priority call. Subsequently it tries to establish the higher priority call again. The configuration below is a typical application for this service in which non-.emergency calls are dropped to free resources for emergency calls. The configuration below routes every call from the fxs-ports to a routing-table and if it is an emergency call (number 112, 117, 118, 119) to the higher port with the higher priority. Every other call can be dropped if an emergency call needs to be placed. Command Sequence to Copy, Edit, and Paste context cs routing-table called-e164 RT_CALLED_PARTY route 11[2789] dest-service SER_PRIO.priority route default dest-service SER_PRIO.standard interface fxs IF_FXS_00 route call dest-table RT_CALLED_PARTY interface fxs IF_FXS_01 route call dest-table RT_CALLED_PARTY interface fxo IF_FXO_00 service priority SER_PRIO port priority route call dest-interface IF_FXO_00 priority 10 port standard route call dest-interface IF_FXO_00
8 Phone Distibution configuration ( Link to this Config )
	Configuration Description Allows you to pick up Phone 0 and ring all other 7 ohones at the same time. Then once one of the other 7 phones is answered the call is connected and the other phones cease ringing. Command Sequence to Copy, Edit, and Paste context cs switch interface fxs PHONE0 route call dest-service DISTRIBUTION interface fxs PHONE1 interface fxs PHONE2 interface fxs PHONE3 interface fxs PHONE4 interface fxs PHONE5 interface fxs PHONE6 interface fxs PHONE7 service distribution-group DISTRIBUTION route call 1 dest-interface PHONE1 route call 2 dest-interface PHONE2 route call 3 dest-interface PHONE3 route call 4 dest-interface PHONE4 route call 5 dest-interface PHONE5 route call 6 dest-interface PHONE6 route call 7 dest-interface PHONE7 context cs switch no shutdown port ethernet 0 0 medium auto encapsulation ip bind interface eth0 router no shutdown port ethernet 0 1 medium auto encapsulation ip bind interface eth1 router no shutdown port fxs 0 0 caller-id format bell use profile fxs us encapsulation cc-fxs bind interface PHONE0 switch no shutdown port fxs 0 1 caller-id format bell use profile fxs us encapsulation cc-fxs bind interface PHONE1 switch no shutdown port fxs 0 2 caller-id format bell use profile fxs us encapsulation cc-fxs bind interface PHONE2 switch no shutdown port fxs 0 3 caller-id format bell use profile fxs us encapsulation cc-fxs bind interface PHONE3 switch no shutdown port fxs 0 4 caller-id format bell use profile fxs us encapsulation cc-fxs bind interface PHONE4 switch no shutdown port fxs 0 5 caller-id format bell use profile fxs us encapsulation cc-fxs bind interface PHONE5 switch no shutdown port fxs 0 6 caller-id format bell use profile fxs us encapsulation cc-fxs bind interface PHONE6 switch no shutdown port fxs 0 7 caller-id format bell use profile fxs us encapsulation cc-fxs bind interface PHONE7 switch no shutdown
Service limiter ( Link to this Config )
	Configuration Description The call-router ‘limiter’ service limit offers with a flexible technique to limit the maximum number of concurrent calls. It also limits the call-setup rate within a system. Calls exceeding the defined limits can either be simply dropped or they can be handled differently. A call is counted as an active call as soon as the call-setup message reaches the limiter service. The call remains active until the signaling has completed tearing down the call. Command Sequence to Copy, Edit, and Paste #-----------------------------------------------------------------# # # # SMARTWARE 5.X # # # # Service Limiter config example # # # # Similarly the call-setup-rate could be limited by using the # # ‘exceed max-call-rate’ instead of the ‘exceed maxcalls’ command # # in the limiter-port configuration mode. There is no limitation # # on the number of ports a limiter can have. You can create as # # many as you need for your application. # # # # DATE: 2011-01-05 # # AUTHOR: Thomas Tlapak # # PATTON-INALP-NETWORKS; CH-3172 NIEDERWANGEN # # WWW.PATTON.COM WWW.INALP.COM # # # # # #-----------------------------------------------------------------# context cs switch interface isdn VOICEMAIL interface isdn LOCALEXCHANGE route call dest-service mylimiter.inbound interface sip IF_SIP bind context sip-gateway GW_SIP route call dest-service mylimiter.outbound # Enter below the IP of the softswitch / sip server remote <0.0.0.0> service limiter mylimiter max-calls 20 port inbound route call dest-interface IF_SIP exceed max-calls route call dest-interface VOICEMAIL port outbound route call dest-interface LOCALEXCHANGE context cs switch no shutdown
CDR over Radius different ports ( Link to this Config )
	Configuration Description Command Sequence to Copy, Edit, and Paste #-----------------------------------------------------------------# # # # SMARTWARE 5.X # # # # ENHANCEMENT for CALL-DETAIL-RECORDS over RADIUS # # # # This config example is an enhancement to the CDR over Radius # # config. # # With it, the CDR traps will contain the port informaiton, where # # the call was coming from or was routed to. # # # # DATE: 2011-01-05 # # AUTHOR: Thomas Tlapak # # PATTON-INALP-NETWORKS; CH-3172 NIEDERWANGEN # # WWW.PATTON.COM WWW.INALP.COM # # # # # #-----------------------------------------------------------------# context cs switch routing-table called-e164 RT_OUT route .T dest-interface IF_SIP interface isdn IF_ISDN_0 route call dest-service AAA.OUT_ISDN_0 interface isdn IF_ISDN_1 route call dest-service AAA.OUT_ISDN_1 interface isdn IF_ISDN_2 route call dest-service AAA.OUT_ISDN_1 interface sip IF_SIP bind context sip-gateway GW_SIP route call dest-service HG # Enter below the SIP server / Softswitch IP address or FQDN remote <0.0.0.0> service hunt-group HG drop-cause normal-unspecified drop-cause no-circuit-channel-available drop-cause network-out-of-order drop-cause temporary-failure drop-cause switching-equipment-congestion drop-cause access-info-discarded drop-cause circuit-channel-not-available drop-cause resources-unavailable route call 1 dest-service AAA.IN_ISDN_0 route call 2 dest-service AAA.IN_ISDN_1 route call 3 dest-service AAA.IN_ISDN_2 service aaa AAA accounting use profile aaa default port IN_ISDN_0 route call dest-interface IF_ISDN_0 port IN_ISDN_1 route call dest-interface IF_ISDN_1 port IN_ISDN_2 route call dest-interface IF_ISDN_2 port OUT_ISDN_0 route call dest-table RT_OUT port OUT_ISDN_1 route call dest-table RT_OUT port OUT_ISDN_2 route call dest-table RT_OUT context cs switch no shutdown
CDR over Radius ( Link to this Config )
	Configuration Description The SmartNode sends a CDR message if a incoming or outgoing Call-Setup passes the SIP-Interface. You may define and configure a independent CDR-services-element for each call-routing-leg. Command Sequence to Copy, Edit, and Paste #-----------------------------------------------------------------# # # # SMARTWARE 5.X # # # # CALL-DETAIL-RECORDS over RADIUS # # # # Sample configuration of CDR service in a simple call-router # # configuration. You may enhance the call-router configuration # # according your requirements. # # # # After replacing all variables you may copy and paste this # # configuration part into your offline configuration file. # # # # The SmartNode sends a CDR message if a incoming # # or outgoing Call-Setup passes the SIP-Interface. # # You may define and configure a independent CDR-services-element # # for each call-routing-leg. # # # # DATE: 2011-01-05 # # AUTHOR: Thomas Tlapak # # PATTON-INALP-NETWORKS; CH-3172 NIEDERWANGEN # # WWW.PATTON.COM WWW.INALP.COM # # # # # #-----------------------------------------------------------------# radius-client accounting # Enter below the Radius Server IP and Port radius-server "ip-address" "port" # Enter below the Radius Password for authentication of SmartNode shared-secret authentication "password" profile aaa accounting method 1 radius accounting context cs switch routing-table called-e164 RT_to_SIP route .T dest-service SER_Radius_SIP.out interface sip IF_SIP_PROXY bind context sip-gateway GW_SIP route call dest-service SER_Radius_SIP.in # Enter below the SIP server / Softswitch IP address or FQDN remote <0.0.0.0> interface isdn IF_ISDN_PBX route call dest-table RT_to_SIP service aaa SER_Radius_SIP accounting start-trigger setup accounting use profile aaa accounting port in route call dest-interface IF_ISDN_PBX port out route call dest-dest-interface IF_SIP_PROXY context cs switch no shutdown
Distribution Group ( Link to this Config )
	Configuration Description Incoming SIP call rings four phones simultaneously. Command Sequence to Copy, Edit, and Paste #-----------------------------------------------------------------# # # # SMARTWARE 5.X # # # # Incoming SIP call rings 4 Analogue-phones simultaneously # # # # DATE: 2011-01-05 # # AUTHOR: Thomas Tlapak # # PATTON-INALP-NETWORKS; CH-3172 NIEDERWANGEN # # WWW.PATTON.COM WWW.INALP.COM # # # # # #-----------------------------------------------------------------# context cs switch interface sip IF_SIP bind context sip-gateway GW_SIP route call dest-service DISTRIBUTE interface fxs IF_FXS_00 route call dest-interface SIP_OUT subscriber-number 111 interface fxs IF_FXS_01 route call dest-interface SIP_OUT subscriber-number 222 interface fxs IF_FXS_02 route call dest-interface SIP_OUT subscriber-number 333 interface fxs IF_FXS_03 route call dest-interface SIP_OUT subscriber-number 444 service distribution-group DISTRIBUTE route call 1 dest-interface IF_FXS_00 route call 2 dest-interface IF_FXS_01 route call 3 dest-interface IF_FXS_02 route call 4 dest-interface IF_FXS_03 context cs switch no shutdown
Hunt Group ( Link to this Config )
	Configuration Description Inbound SIP call hunting across 4 FXS interfaces Command Sequence to Copy, Edit, and Paste #-----------------------------------------------------------------# # # # SMARTWARE 5.X # # # # Inbound SIP call hunting across 4 FXS interfaces # # # # DATE: 2011-01-05 # # AUTHOR: Thomas Tlapak # # PATTON-INALP-NETWORKS; CH-3172 NIEDERWANGEN # # WWW.PATTON.COM WWW.INALP.COM # # # # # #-----------------------------------------------------------------# context cs switch routing-table called-e164 RT_OUT route .T dest-interface IF_SIP interface sip IF_SIP route call dest-service SERV_HG # Enter below the SIP server / Softswitch IP address or FQDN remote <0.0.0.0> interface fxs IF_FXS_00 route call dest-table RT_OUT subscriber-number 111 interface fxs IF_FXS_01 route call dest-table RT_OUT subscriber-number 222 interface fxs IF_FXS_02 route call dest-table RT_OUT subscriber-number 333 interface fxs IF_FXS_03 route call dest-table RT_OUT subscriber-number 444 service hunt-group SERV_HG drop-cause normal-unspecified drop-cause no-circuit-channel-available drop-cause network-out-of-order drop-cause temporary-failure drop-cause switching-equipment-congestion drop-cause access-info-discarded drop-cause circuit-channel-not-available drop-cause resources-unavailable drop-cause user-busy route call 1 dest-interface IF_FXS_00 route call 2 dest-interface IF_FXS_01 route call 3 dest-interface IF_FXS_02 route call 4 dest-interface IF_FXS_03 context cs switch no shutdown
Second Dialtone Service ( Link to this Config )
	Configuration Description The command sequence below enables SmartNode to provide second-dial-tone service. Once a user connects to a SmartNode FXO port, SmartNode will generate and deliver a second dialtone to the user. The routing-table RT_CDPN decides where to deliver the call. Command Sequence to Copy, Edit, and Paste routing-table called-e164 RT_CDPN_TABLE route .T dest-interface <interface-name> interface fxo IF_FXO route call dest-service SER_SECOND_DIALTONE service second-dialtone SER_SECOND_DIALTONE route call dest-table RT_CDPN_TABLE

ISDN UD-64 Data connection over VOIP  ( Link to this Config )
	Configuration Description ENFORCE THE TRANSPARENT CODEC FOR ISDN UD-64 CALLS: Incoming ISDN calls will be split into two call-router paths to enforce transparent codec for ISDN unrestricted digital calls. Note that the VOIP-DATA interface has no route into the call-router, it operates as a one-way interface. However it has a special Voip-Profile linked, which only contains the transparent codec. The VOIP-DEFAULT interface will be used for incoming voip calls. It uses the default voip profile, which is linked in the voip gateway. The default profile contains the transparent codec with the lowest priority to ensure that voice calls never uses the transparent codec. Command Sequence to Copy, Edit, and Paste #-----------------------------------------------------------------# # # # SN4552 / 46XX / SN4960 / SN4940 / SN4950 / SN4660 / SN4670 # # R5.X / R6.X SIP /H323 # # # # ENFORCE THE TRANSPARENT CODEC FOR ISDN UD-64 CALLS # # # # Sample configuration of context CS only. Without gateway and IP # # configuration. # # # # After replacing all variables you may copy and paste this # # configuration part into your offline configuration file. # # # # Incoming ISDN calls will be split into two call-router paths to # # enforce transparent codec for ISDN unrestricted digital calls. # # Note that the VOIP-DATA interface has no route into the # # call-router, it operates as a one-way interface. But it has a # # special Voip-Profile linked, which only contains the # # transparent codec. # # The VOIP-DEFAULT interface will be used for incoming voip calls.# # It uses the default voip profile, which is linked in the voip # # gateway. The default profile contains the transparent codec # # with the lowest priority to ensure that voice calls never uses # # the transparent codec. # # # # # # DATE: 2006-10-04 / Dec. 2011 # # AUTHOR: Markus Zeier; Revision for 5.X, M. Aeberhard # # PATTON-INALP-NETWORKS; CH-3172 NIEDERWANGEN # # WWW.PATTON.COM WWW.INALP.COM # # # # # #-----------------------------------------------------------------# profile voip default codec 1 g726-32k-cisco rx-length 20 tx-length 20 codec 2 g711alaw64k rx-length 20 tx-length 20 codec 3 g711ulaw64k rx-length 20 tx-length 20 codec 4 transparent rx-length 20 tx-length 20 codec 5 transparent-cisco rx-length 20 tx-length 20 fax transmission 1 relay t38-udp fax redundancy low-speed 2 high-speed 1 modem transmission 1 bypass g711alaw64k modem bypass-method nse profile voip PF_VOIP_DATA codec 1 transparent rx-length 20 tx-length 20 codec 2 transparent-cisco rx-length 20 tx-length 20 context cs switch routing-table called-e164 RT_CDPN_TO_VOIP route .T dest-table RT_ITC_VOIP routing-table itc RT_ITC_VOIP route unrestricted-digital dest-interface IF_VOIP_DATA_OUT route default dest-interface IF_VOIP_DEFAULT interface isdn IF_ISDN_S01 route call dest-table RT_CDPN_TO_VOIP interface sip IF_VOIP_DEFAULT bind context sip-gateway <sip|h323> <gateway name> service default <service name> route call dest-interface IF_ISDN_S01 interface sip IF_VOIP_DATA_OUT bind context sip-gateway <sip|h323> <gateway name> service default <service name> use profile voip PF_VOIP_DATA context cs switch no shutdown
SN2X00 / 2*IC-E1Voip Gateway PBX to SIP with PSTN Fallback  ( Link to this Config )
	Configuration Description SmartNode 2X00 used as an ISDN E1 gateway between a PBX and a SIP Network. The second E1 port may be used for Fallback to the PSTN. Please read the comment before using the configuration. Command Sequence to Copy, Edit, and Paste #----------------------------------------------------------------# # # SN2X00 / 2*IC-E1Voip # R3.20 SIP # # Sample configuration of context CS only. Without gateway and IP # configuration. # # SmartNode connected as gateway between a SIP network # and PBX with 1 E1 port. The second Port is configured # as fallback and sync-port. It is configured in the user mode # and should be connected to a PSTN. # The first port is configured as net and should # be connected to a PBX. # # replace all variables including the brackets with a valid content # # # DATE: 2006-07-19 # AUTHOR: Markus Zeier; # PATTON-INALP-NETWORKS; CH-3172 NIEDERWANGEN # WWW.PATTON.COM WWW.INALP.COM # # #----------------------------------------------------------------# context cs switch digit-collection timeout 3 routing-table called-e164 RT_CDPN_OUT route .T dest-service HG_2_OUT interface sip IF_VOIP_GATEWAY bind gateway sip service default address-translation outgoing-call from-header user-part fix <account number> route call dest-interface IF_E1_10 interface isdn IF_E1_10 route call dest-table RT_CDPN_OUT interface isdn IF_E1_20 route call dest-interface IF_E1_10 service hunt-group HG_2_OUT drop-cause normal-unspecified drop-cause no-circuit-channel-available drop-cause network-out-of-order drop-cause temporary-failure drop-cause switching-equipment-congestion drop-cause access-info-discarded drop-cause circuit-channel-not-available drop-cause resources-unavailable route call 1 dest-interface IF_VOIP_GATEWAY route call 2 dest-interface IF_E1_20 context cs switch no shutdown port e1 <slot> 0 clock auto encapsulation q921 q921 uni-side auto encapsulation q931 q931 protocol dss1 uni-side net encapsulation cc-isdn bind interface IF_E1_10 port e1 <slot> 0 no shutdown port e1 <slot> 0 clock auto encapsulation q921 q921 protocol pp uni-side auto encapsulation q931 q931 protocol dss1 uni-side user encapsulation cc-isdn bind interface IF_E1_20 port e1 <slot> 0 no shutdown
SN1X00 / SN4552 / SN46XX; Phone Gateway to a SIP network with PSTN Fallback ( Link to this Config )
	Configuration Description SmartNode used as gateway between a SIP network and a phone, connected to port 0/1. The Port 0/0 is configured as user and can be used as fallback to a PSTN. Command Sequence to Copy, Edit, and Paste #----------------------------------------------------------------# # # SN1200 / SN4552 / 1400 / 46XX # R3.20 SIP # # Sample configuration of context CS only. Without gateway and IP # configuration. # # SmartNode connected as gateway between a SIP network # and a phone connected to port 0/1. The Port 0/0 is configured # as fallback. It is configured as user port # and should be connected to a PSTN. # # # DATE: 2006-05-08 # AUTHOR: Markus Zeier; # PATTON-INALP-NETWORKS; CH-3172 NIEDERWANGEN # WWW.PATTON.COM WWW.INALP.COM # # #----------------------------------------------------------------# context cs switch digit-collection timeout 3 routing-table called-e164 RT_CDPN_OUT route .T dest-service HG_2_OUT interface sip IF_VOIP_GATEWAY bind gateway sip service default address-translation outgoing-call from-header user-part fix <account number> route call dest-interface IF_S0_01 interface isdn IF_S0_00 route call dest-interface IF_S0_01 interface isdn IF_S0_01 route call dest-table RT_CDPN_OUT service hunt-group HG_2_OUT drop-cause normal-unspecified drop-cause no-circuit-channel-available drop-cause network-out-of-order drop-cause temporary-failure drop-cause switching-equipment-congestion drop-cause access-info-discarded drop-cause circuit-channel-not-available drop-cause resources-unavailable route call 1 dest-interface IF_VOIP_GATEWAY route call 2 dest-interface IF_S0_00 context cs switch no shutdown port bri 0 0 clock auto encapsulation q921 q921 protocol pmp uni-side auto encapsulation q931 q931 protocol dss1 uni-side user encapsulation cc-isdn bind interface IF_S0_00 port bri 0 0 no shutdown port bri 0 1 clock auto encapsulation q921 q921 protocol pmp uni-side auto encapsulation q931 q931 protocol dss1 uni-side net encapsulation cc-isdn bind interface IF_S0_01 port bri 0 1 no shutdown
SN1X00 / SN4552 / SN46XX; Phone Gateway to a SIP network with PSTN Fallback and Emercency call routing ( Link to this Config )
	Configuration Description SmartNode used as gateway between a SIP network and a phone, connected to port 0/1. The Port 0/0 is configured as user and can be used as fallback to a PSTN. Emercency calls will be routed to PSTN instead of VOIP. Command Sequence to Copy, Edit, and Paste #-----------------------------------------------------------------# # # # SMARTWARE 5.X # # # # Sample configuration of context CS and ISDN ports only. # # Without gateway and IP configuration. # # # # SmartNode connected as gateway between a SIP network # # and a phone connected to port 0/1. The Port 0/0 is configured # # as fallback. It is configured as user port # # and should be connected to a PSTN. # # # # DATE: 2011-01-05 # # AUTHOR: Thomas Tlapak # # PATTON-INALP-NETWORKS; CH-3172 NIEDERWANGEN # # WWW.PATTON.COM WWW.INALP.COM # # # # # #-----------------------------------------------------------------# system clock-source 1 bri 0 0 context cs switch digit-collection timeout 3 routing-table called-e164 RT_CDPN_OUT route .T dest-service HG_2_OUT interface sip IF_SIP # Enter below the context sip-gateway name bind context sip-gateway <gateway name> # Enter below the SIP server / Softswitch IP address or FQDN remote <0.0.0.0> # Enter below the SIP account username, to be used for authentication address-translation outgoing-call from-header user-part fix <account number> route call dest-interface IF_ISDN_01 interface isdn IF_ISDN_00 route call dest-interface IF_ISDN_01 interface isdn IF_ISDN_01 route call dest-table RT_CDPN_OUT service hunt-group HG_2_OUT drop-cause normal-unspecified drop-cause no-circuit-channel-available drop-cause network-out-of-order drop-cause temporary-failure drop-cause switching-equipment-congestion drop-cause access-info-discarded drop-cause circuit-channel-not-available drop-cause resources-unavailable route call 1 dest-interface IF_SIP route call 2 dest-interface IF_ISDN_00 context cs switch no shutdown port bri 0 0 clock auto encapsulation q921 q921 protocol pmp uni-side auto encapsulation q931 q931 protocol dss1 uni-side user encapsulation cc-isdn bind interface IF_ISDN_00 port bri 0 0 no shutdown port bri 0 1 clock auto encapsulation q921 q921 protocol pmp uni-side auto encapsulation q931 q931 protocol dss1 uni-side net encapsulation cc-isdn bind interface IF_ISDN_01 port bri 0 1 no shutdown
SN46x8 as Trunk Gateway SIP to PBX with Fallback ( Link to this Config )
	Configuration Description SN46X8 SMARTWARE 5.X Sample configuration SmartNode connected as trunk gateway between a SIP network and PBX with 4 S0 port. The Port S0 slot 0 port 0 is configured as fallback and sync-port. It is configured in the user mode and should be connected to a PSTN. The ports slot 0 port 1 to 4 are configured as net and should be connected to a PBX. Not all the ports have to be connected Command Sequence to Copy, Edit, and Paste #-----------------------------------------------------------------# # # # SN46X8 # # SMARTWARE 5.X # # # # Sample configuration of context CS and ISDN ports only. # # Without gateway and IP configuration. # # # # SmartNode connected as trunk gateway between a SIP network # # and ISDN PBX with 4 S0 port. The Port S0 slot 0 port 0 (BRI 0/0)# # is configured as fallback and sync-port. It is configured in # # user mode and should be connected to a PSTN. # # The ports slot 0 port 1 to 4 are configured as net and should # # be connected to a PBX. Not all the ports have to be connected # # # # DATE: 2011-01-05 # # AUTHOR: Thomas Tlapak # # PATTON-INALP-NETWORKS; CH-3172 NIEDERWANGEN # # WWW.PATTON.COM WWW.INALP.COM # # # # # #-----------------------------------------------------------------# system clock-source 1 bri 0 0 context cs switch routing-table called-e164 RT_CDPN_OUT route .T dest-service SERV_HG_2_OUT interface sip IF_SIP # Enter below the context sip-gateway name bind context sip-gateway <gateway name> # Enter below the SIP server / Softswitch IP address or FQDN remote <0.0.0.0> # Enter below the SIP account username, to be used for authentication address-translation outgoing-call from-header user-part fix <account number> route call dest-service SERV_HG_2_PBX interface isdn IF_ISDN_00 route call dest-service SERV_HG_2_PBX interface isdn IF_ISDN_01 route call dest-table RT_CDPN_OUT interface isdn IF_ISDN_02 route call dest-table RT_CDPN_OUT interface isdn IF_ISDN_03 route call dest-table RT_CDPN_OUT interface isdn IF_ISDN_04 route call dest-table RT_CDPN_OUT service hunt-group SERV_HG_2_OUT drop-cause normal-unspecified drop-cause no-circuit-channel-available drop-cause network-out-of-order drop-cause temporary-failure drop-cause switching-equipment-congestion drop-cause access-info-discarded drop-cause circuit-channel-not-available drop-cause resources-unavailable route call 1 dest-interface IF_SIP route call 2 dest-interface IF_ISDN_00 service hunt-group SERV_HG_2_PBX drop-cause normal-unspecified drop-cause no-circuit-channel-available drop-cause network-out-of-order drop-cause temporary-failure drop-cause switching-equipment-congestion drop-cause access-info-discarded drop-cause circuit-channel-not-available drop-cause resources-unavailable route call 1 dest-interface IF_ISDN_01 route call 2 dest-interface IF_ISDN_02 route call 2 dest-interface IF_ISDN_03 route call 2 dest-interface IF_ISDN_04 context cs switch no shutdown port bri 0 0 clock auto encapsulation q921 q921 protocol pp uni-side auto encapsulation q931 q931 protocol dss1 uni-side user encapsulation cc-isdn bind interface IF_ISDN_00 port bri 0 0 no shutdown port bri 0 1 clock auto encapsulation q921 q921 protocol pp uni-side auto encapsulation q931 q931 protocol dss1 uni-side net encapsulation cc-isdn bind interface IF_ISDN_01 port bri 0 1 no shutdown port bri 0 2 clock auto encapsulation q921 q921 protocol pp uni-side auto encapsulation q931 q931 protocol dss1 uni-side net encapsulation cc-isdn bind interface IF_ISDN_02 port bri 0 2 no shutdown port bri 0 3 clock auto encapsulation q921 q921 protocol pp uni-side auto encapsulation q931 q931 protocol dss1 uni-side net encapsulation cc-isdn bind interface IF_ISDN_03 port bri 0 3 no shutdown port bri 0 4 clock auto encapsulation q921 q921 protocol pp uni-side auto encapsulation q931 q931 protocol dss1 uni-side net encapsulation cc-isdn bind interface IF_ISDN_04 port bri 0 4 no shutdown

Multilink PPP, MLPPP, over one subscriber ( Link to this Config )
	Configuration Description The example shows you, how to bundle multiple PPP lines into one, using one subscriber. You may add additional lines in the same way. Command Sequence to Copy, Edit, and Paste context ip router interface IF_MLPPP ipaddress unnumbered point-to-point use profile napt NAPT_WAN context ip router route 0.0.0.0 0.0.0.0 IF_MLPPP 0 subscriber ppp SUB_MLPPP dial out multilink max-links 2 authentication chap pap identification outbound your_username password your_password bind interface IF_MLPPP router port ethernet "slot" "port" medium auto encapsulation ppp pppoe session SES_PPP_ETH bind subscriber SUB_MLPPP no shutdown port ethernet 0 1 no shutdown port dsl 0 0 pvc vpi "vpi" vci "vci" pppoe session SES_PPP_ADSL bind subscriber SUB_MLPPP no shutdown
Multilink PPP, MLPPP, over 2 subscribers ( Link to this Config )
	Configuration Description The example shows you how to bundle multiple PPP lines into one, using two subscribers. You may add additional lines and subscribers in the same way. IMPORTANT: Both subscribers needs the same mulitlink settings Command Sequence to Copy, Edit, and Paste context ip router interface IF_MLPPP ipaddress unnumbered point-to-point use profile napt NAPT_WAN context ip router route 0.0.0.0 0.0.0.0 IF_MLPPP 0 subscriber ppp SUB_PPP_ETH dial out multilink max-links 2 authentication chap pap identification outbound your_username_1 password your_password_1 bind interface IF_MLPPP router subscriber ppp SUB_PPP_ADSL dial out multilink max-links 2 authentication chap pap identification outbound your_username_2 password your_password_2 bind interface IF_MLPPP router port ethernet "slot" "port" medium auto encapsulation ppp pppoe session SES_PPP_ETH bind subscriber SUB_PPP_ETH no shutdown port ethernet 0 1 no shutdown port dsl 0 0 pvc vpi "vpi" vci "vci" pppoe session SES_PPP_ADSL bind subscriber SUB_PPP_ADSL no shutdown
Connect SmartNode to the LAN as a DHCP Client ( Link to this Config )
	Configuration Description The command sequence below enables SmartNode to operate as a DHCP client by pointing a SmartNode IP-interface to a DHCP server on the directly connected LAN. As a DHCP client, upon power-up the SmartNode can automatically retreive an IP address, default gateway address, and DNS server address from the specified DHCP server. NOTE: Before you point a SmartNode IP-Interface [IF-NAME] to a DHCP server, you must ensure the interface is bound to an Ethernet port. Command Sequence to Copy, Edit, and Paste context ip interface [IF-NAME] ipaddress dhcp
Connect SmartNode to the LAN via static IP addressing with a default gateway address ( Link to this Config )
	Configuration Description The command sequence below enables static-IP-addressing by assigning a fixed IP address, netmask, and default gateway address to a SmartNode IP interface. To complete this configuration you will need the following information: The IP-Address and network mask for the SmartNode: [IF-IP], [IF-MASK] The Gateway address of the default gateway:[GW-IP] Use the command "show ip route" to display the routing table. 10.10.22.53(ctx-ip)[router]#show ip route Route Table of IP context 'router': Validity state (V): * valid, (*) duplicate Flags: U up, H host, G Gateway, L local, D default -------------------------------------------------------------- V Destination Nexthop Protocol Metric Flags -------------------------------------------------------------- * 10.10.22.53/32 loopback local 0 ULH * 192.168.2.1/32 loopback local 0 ULH * 127.0.0.1/32 loopback local 0 ULH * 192.168.2.0/24 eth1 local 1 UL * 10.10.0.0/16 eth0 local 1 UL * 127.0.0.0/8 loopback local 1 UL * 0.0.0.0/0 10.10.1.10 static 0 UDG (*) 0.0.0.0/0 10.10.1.1 dhcp 12 DG -------------------------------------------------------------- Command Sequence to Copy, Edit, and Paste context ip interface [IF-NAME] ipaddress [IF-IP] [IF-MASK] context ip route 0.0.0.0 0.0.0.0 [GW-IP]
Policy based routing ( Link to this Config )
	Configuration Description Sometimes you need to specify default routes for traffic in specific traffic-classes. SmartNodes have three built-in traffic-classes (expendable in configuration) which are used for Quality-of-Service purposes. For each traffic-class you can configure a default route. If your configure a default-route without mentioning of a traffic-class t is the default route for every untagged traffic. Command Sequence to Copy, Edit, and Paste context ip router route <destination> <netmask> <gateway> <metric> traffic-class <traffic-class>

Enable SmartNode as a DHCP Server ( Link to this Config )

Configuration Description The command sequence below enables SmartNode to operate as a DHCP server. In networks where the SmartNode is the Internet Gateway Router, this configuration is the best way to provide user devices with Internet access. Once enabled as a DHCP server, The SmartNode can provide a dynamically assigned IP address, DNS Server address, and Default Gateway address to users that are configured as DHCP clients (e.g. PCs, Laptops). NOTE: You must ensure the network address and the IP addresses in the DHCP server profile match one of the IP interfaces defined in the IP Context. When you start a DHCP server for the first time you will see the following warning: 2004-12-02T15:56:17 : LOGERROR : Error while loading lease database This is only a warning does not indicate a problem. Command Sequence to Copy, Edit, and Paste #create dhcp profile profile dhcp-server PF-DHCP1 network 192.168.1.0 255.255.255.0 include 192.168.1.50 192.168.1.100 lease 3 days default-router 192.168.1.1 #apply DHCP profile in the ip context context ip dhcp-server use PF-DHCP1

Dynamic Service ( Link to this Config )
	Configuration Description The command sequence below enables SmartNode to support dynamic service via DynDNS (i.e. http://www.dyndns.org). If your IP address changes more often than once every 28 days you should use this Service Dynamic configuration. If your IP address changes less often you should use the Service-Static configuration (shown below on this web page). When in doubt, use static configuration. NOTE: You must replace the following variables with your site-specific values: MyAccount.dyndns.org MyAccount MyPassword Command Sequence to Copy, Edit, and Paste context ip dyndns hostname MyAccount.dyndns.org wildcard authentication MyAccount MyPassword service dynamic observe eth0
Static Service ( Link to this Config )
	Configuration Description The command sequence below enables SmartNode to support static service via DynDNS (i.e. http://www.dyndns.org). If your IP address changes less often than every 28 days, you should use this Service-Static configuration. If your IP address changes more often, you should use ths Service Dynamic configuration. (shown above, on this web page). When in doubt, use static configuration. NOTE: You must replace the following variables with your site-specific values: MyAccount.dyndns.org MyAccount MyPassword Command Sequence to Copy, Edit, and Paste context ip dyndns hostname MyAccount.dyndns.org wildcard authentication MyAccount MyPassword service static observe eth0

Configuration for the SmartNode 4638 as a PSTN gateway in combination with a GFI Faxmaker ( Link to this Config )
	Configuration Description This example includes a basic configuration for the SmartNode 4638 as a PSTN gateway in combination with a GFI Faxmaker. This example can be used for other SmartNodes too. The only thing which needs to be adjusted, are the ports and interfaces as far as required. Command Sequence to Copy, Edit, and Paste #----------------------------------------------------------------# # GFI Faxmaker # # # # This example includes a basic configuration for the SmartNode # # 4638 a PSTN gateway in combination with a GFI Faxmaker. # # This example can be used for other SmartNodes too. The only # # thing which needs to be adjusted, are the ports and interfaces # # as far as required. # # # # Author: Thomas Tlapak # # Patton-Inalp Neworks # # CH-3172 Niederwangen # # www.patton.com; support@patton.com # # # # Date: 11/18/2010 # #----------------------------------------------------------------# cli version 3.20 dns-client server <Primary_DNS_Server_If_you_use_hostnames> dns-client server <Secondary_DNS_Server_If_you_use_hostnames> webserver port 80 language en sntp-client sntp-client server primary 129.132.2.21 port 123 version 4 system ic voice 0 low-bitrate-codec g729 system clock-source 1 bri 0 0 clock-source 2 bri 0 1 clock-source 3 bri 0 2 clock-source 4 bri 0 3 clock-source 5 bri 0 4 profile napt NAPT_WAN profile ppp default profile tone-set default profile voip default codec 1 g711alaw64k rx-length 20 tx-length 20 codec 2 g711ulaw64k rx-length 20 tx-length 20 fax transmission 1 relay t38-udp fax transmission 2 bypass g711alaw64k fax redundancy low-speed 2 high-speed 1 fax bypass-method nse profile pstn default profile sip default no autonomous-transitioning profile aaa default method 1 local method 2 none context ip router interface WAN ipaddress <YOUR_IP_WAN_SIDE> <YOUR_NETWORK_MASK> use profile napt NAPT_WAN tcp adjust-mss rx mtu tcp adjust-mss tx mtu context ip router route 0.0.0.0 0.0.0.0 <YOUR_DEFAULT_GATEWAY> 0 context cs switch routing-table called-e164 RT_TO_SIP route .T2 dest-interface IF_SIP interface isdn IF_ISDN_01 route call dest-table RT_TO_SIP interface isdn IF_ISDN_00 route call dest-table RT_TO_SIP interface isdn IF_ISDN_02 route call dest-table RT_TO_SIP interface isdn IF_ISDN_03 route call dest-table RT_TO_SIP interface isdn IF_ISDN_04 route call dest-table RT_TO_SIP interface sip IF_SIP bind context sip-gateway GW_SIP route call dest-service SRV_HUNT remote <IP or domain of the xcapi> 5060 local <IP or domain of the xcapi> 5060 early-disconnect address-complete-indication accept set address-translation outgoing-call request-uri user-part call host-part fix <IP or domain of the xcapi> 5060 target-param none address-translation incoming-call calling-redir request-uri service hunt-group SRV_HUNT drop-cause normal-unspecified drop-cause no-circuit-channel-available drop-cause network-out-of-order drop-cause temporary-failure drop-cause switching-equipment-congestion drop-cause access-info-discarded drop-cause circuit-channel-not-available drop-cause resources-unavailable route call 1 dest-interface IF_ISDN_00 route call 2 dest-interface IF_ISDN_01 route call 3 dest-interface IF_ISDN_02 route call 4 dest-interface IF_ISDN_03 route call 5 dest-interface IF_ISDN_04 context cs switch no shutdown context sip-gateway GW_SIP interface IF_GW_SIP bind interface WAN context router port 5060 context sip-gateway GW_SIP no shutdown port ethernet 0 0 medium auto encapsulation ip bind interface WAN router no shutdown port ethernet 0 1 medium auto encapsulation ip no shutdown port bri 0 0 clock auto encapsulation q921 q921 permanent-layer2 protocol pp uni-side auto encapsulation q931 q931 protocol dss1 uni-side user bchan-number-order ascending encapsulation cc-isdn bind interface IF_ISDN_00 switch port bri 0 0 shutdown port bri 0 1 clock auto encapsulation q921 q921 permanent-layer2 protocol pp uni-side auto encapsulation q931 q931 protocol dss1 uni-side user bchan-number-order ascending encapsulation cc-isdn bind interface IF_ISDN_01 switch port bri 0 1 shutdown port bri 0 2 clock auto encapsulation q921 q921 permanent-layer2 protocol pp uni-side auto encapsulation q931 q931 protocol dss1 uni-side user bchan-number-order ascending encapsulation cc-isdn bind interface IF_ISDN_02 switch port bri 0 2 no shutdown port bri 0 3 clock auto encapsulation q921 q921 permanent-layer2 protocol pp uni-side auto encapsulation q931 q931 protocol dss1 uni-side user bchan-number-order ascending encapsulation cc-isdn bind interface IF_ISDN_03 switch port bri 0 3 no shutdown port bri 0 4 clock auto encapsulation q921 q921 permanent-layer2 protocol pp uni-side auto encapsulation q931 q931 protocol dss1 uni-side user bchan-number-order ascending encapsulation cc-isdn bind interface IF_ISDN_04 switch port bri 0 4 no shutdown
Configuration for the SmartNode 4112 as a PSTN gateway in combination with a GFI Faxmaker ( Link to this Config )
	Configuration Description This example includes a basic configuration for the SmartNode 4112 as a PSTN gateway in combination with a GFI Faxmaker. This example can be used for other SmartNodes too. The only thing which needs to be adjusted, are the ports and interfaces as far as required. Command Sequence to Copy, Edit, and Paste #----------------------------------------------------------------# # GFI Faxmaker # # # # This example includes a basic configuration for the SmartNode # # 4112 as a PSTN gateway in combination with a GFI Faxmaker. # # This example can be used for other SmartNodes too. The only # # thing which needs to be adjusted, are the ports and interfaces # # as far as required. # # # # Author: Thomas Tlapak # # Patton-Inalp Neworks # # CH-3172 Niederwangen # # www.patton.com; support@patton.com # # # # Date: 11/18/2010 # #----------------------------------------------------------------# cli version 3.20 dns-client server <Primary_DNS_Server_If_you_use_hostnames> dns-client server <Secondary_DNS_Server_If_you_use_hostnames> webserver port 80 language en sntp-client sntp-client server primary 129.132.2.21 port 123 version 4 system ic voice 0 low-bitrate-codec g729 profile napt NAPT_WAN profile ppp default profile tone-set default profile voip default codec 1 g711alaw64k rx-length 20 tx-length 20 codec 2 g711ulaw64k rx-length 20 tx-length 20 fax transmission 1 relay t38-udp fax transmission 2 bypass g711alaw64k fax redundancy low-speed 2 high-speed 1 fax bypass-method nse profile pstn default profile sip default no autonomous-transitioning profile aaa default method 1 local method 2 none context ip router interface WAN ipaddress <YOUR_IP_WAN_SIDE> <YOUR_NETWORK_MASK> use profile napt NAPT_WAN tcp adjust-mss rx mtu tcp adjust-mss tx mtu context ip router route 0.0.0.0 0.0.0.0 <YOUR_DEFAULT_GATEWAY> 0 context cs switch routing-table called-e164 RT_TO_SIP route .T2 dest-interface IF_SIP interface fxo IF_FXO_00 route call dest-table RT_TO_SIP interface fxo IF_FXO_01 route call dest-table RT_TO_SIP interface fxo IF_FXO_02 route call dest-table RT_TO_SIP interface fxo IF_FXO_03 route call dest-table RT_TO_SIP interface sip IF_SIP bind context sip-gateway GW_SIP route call dest-service SRV_HUNT remote <IP or domain of the xcapi> 5060 local <IP or domain of the xcapi> 5060 early-disconnect address-complete-indication accept set address-translation outgoing-call request-uri user-part call host-part fix <IP or domain of the xcapi> 5060 target-param none address-translation incoming-call calling-redir request-uri service hunt-group SRV_HUNT drop-cause normal-unspecified drop-cause no-circuit-channel-available drop-cause network-out-of-order drop-cause temporary-failure drop-cause switching-equipment-congestion drop-cause access-info-discarded drop-cause circuit-channel-not-available drop-cause resources-unavailable drop-cause user-busy route call 1 dest-interface IF_FXO_00 route call 2 dest-interface IF_FXO_01 route call 3 dest-interface IF_FXO_02 route call 4 dest-interface IF_FXO_03 context cs switch no shutdown context sip-gateway GW_SIP interface IF_GW_SIP bind interface WAN context router port 5060 context sip-gateway GW_SIP no shutdown port ethernet 0 0 medium auto encapsulation ip bind interface WAN router no shutdown port ethernet 0 1 medium auto encapsulation ip no shutdown port fxs 0 0 encapsulation cc-fxs bind interface IF_FXS_00 switch no shutdown port fxs 0 1 encapsulation cc-fxs bind interface IF_FXS_01 switch no shutdown port fxs 0 2 encapsulation cc-fxs bind interface IF_FXS_02 switch no shutdown port fxs 0 3 encapsulation cc-fxs bind interface IF_FXS_03 switch no shutdown
Template SN49XX PRI to SIP Trunking Cisco Call Manager with authentication ( Link to this Config )
	Configuration Description This example includes a basic configuration for the SmartNode SN49XX/ISDN for use as a PRI gateway in combination with a Cisco Call Manager. The configuration contains variables in brackets <>. Please Replace the variables including the brackets. Unused lines or variables have to be commented out or deleted Command Sequence to Copy, Edit, and Paste #----------------------------------------------------------------# # Cisco Call Manager # # # # This example includes a basic configuration for the SmartNode # # 49XX/ISDN in combination with a Cisco Call Manager with # # authentication and registration # # This example can be used for other SmartNodes too. The only # # thing which needs to be adjusted, are the ports and interfaces # # as far as required. # # # # Author: Olivier Gaudet # # Patton-Inalp Neworks # # CH-3172 Niederwangen # # www.patton.com; support@patton.com # # # # Date: 17/11/2011 # # # #----------------------------------------------------------------# cli version 3.20 clock local default-offset +00:00 dns-client server <Primary_DNS_Server_If_you_use_hostnames> dns-client server <Secondary_DNS_Server_If_you_use_hostnames> webserver port 80 language en sntp-client sntp-client server primary 129.132.2.21 port 123 version 4 system ic voice 0 system clock-source 1 e1t1 0 0 clock-source 1 e1t1 0 1 clock-source 1 e1t1 0 2 clock-source 1 e1t1 0 3 profile ppp default profile tone-set default profile voip default codec 1 g711alaw64k rx-length 20 tx-length 20 codec 2 g711ulaw64k rx-length 20 tx-length 20 fax transmission 1 relay t38-udp fax transmission 2 bypass g711alaw64k fax redundancy low-speed 2 high-speed 1 fax bypass-method nse modem transmission 1 bypass g711alaw64k modem bypass-method nse profile pstn default profile sip default no autonomous-transitioning profile aaa default method 1 local method 2 none context ip router interface LAN ipaddress <YOUR_IP_LAN_SIDE> <YOUR_NETWORK_MASK> tcp adjust-mss rx mtu tcp adjust-mss tx mtu context ip router route 0.0.0.0 0.0.0.0 <YOUR_DEFAULT_GATEWAY> 0 context cs switch digit-collection timeout 3 routing-table called-e164 RT_FROM_PSTN route default dest-interface IF_CISCO_CM routing-table called-e164 RT_FROM_CISCO route default dest-service HUNT_PSTN interface isdn IF_ISDN_00 route call dest-table RT_FROM_PSTN interface isdn IF_ISDN_01 route call dest-table RT_FROM_PSTN interface isdn IF_ISDN_02 route call dest-table RT_FROM_PSTN interface isdn IF_ISDN_03 route call dest-table RT_FROM_PSTN interface sip IF_CISCO_CM bind context sip-gateway CISCO_GW route call dest-table RT_FROM_CISCO remote <YOUR_CALL_MANAGER_IP> 5060 service hunt-group HUNT_PSTN timeout 3 drop-cause normal-unspecified drop-cause no-circuit-channel-available drop-cause network-out-of-order drop-cause temporary-failure drop-cause switching-equipment-congestion drop-cause access-info-discarded drop-cause circuit-channel-not-available drop-cause resources-unavailable route call 1 dest-interface IF_ISDN_00 route call 2 dest-interface IF_ISDN_01 route call 3 dest-interface IF_ISDN_02 route call 3 dest-interface IF_ISDN_03 context cs switch no shutdown authentication-service AUTH_SVC username <Username_SIP> password <Password_SIP> location-service LOCATION_SVC domain 1 <YOUR_CALL_MANAGER_IP> identity-group default authentication outbound authenticate 1 authentication-service AUTH_SVC username <Username_SIP> registration outbound register auto context sip-gateway CISCO_GW interface IF_sip_interface_GW bind interface LAN context router port 5060 context sip-gateway CISCO_GW bind location-service LOCATION_SVC no shutdown port ethernet 0 0 medium auto encapsulation ip bind interface LAN router no shutdown port e1t1 0 0 port-type e1 clock auto encapsulation q921 q921 permanent-layer2 uni-side auto encapsulation q931 q931 protocol dss1 uni-side user bchan-number-order ascending encapsulation cc-isdn bind interface IF_ISDN_00 switch port e1t1 0 0 no shutdown port e1t1 0 1 port-type e1 clock auto encapsulation q921 q921 permanent-layer2 uni-side auto encapsulation q931 q931 protocol dss1 uni-side user bchan-number-order ascending encapsulation cc-isdn bind interface IF_ISDN_01 switch port e1t1 0 1 no shutdown port e1t1 0 2 port-type e1 clock auto encapsulation q921 q921 permanent-layer2 uni-side auto encapsulation q931 q931 protocol dss1 uni-side user bchan-number-order ascending encapsulation cc-isdn bind interface IF_ISDN_02 switch port e1t1 0 2 no shutdown port e1t1 0 3 port-type e1 clock auto encapsulation q921 q921 permanent-layer2 uni-side auto encapsulation q931 q931 protocol dss1 uni-side user bchan-number-order ascending encapsulation cc-isdn bind interface IF_ISDN_03 switch port e1t1 0 3 no shutdown
Template SN49XX PRI to SIP Trunking with Cisco Call Manage without authentication ( Link to this Config )
	Configuration Description This example includes a basic configuration for the SmartNode SN49XX/ISDN to be used use as a PRI gateway in combination with a Cisco Call Manager. The configuration contains variables in brackets <>. Please Replace the variables including the brackets. Unused lines or variables have to be commented out or deleted Command Sequence to Copy, Edit, and Paste #----------------------------------------------------------------# # Cisco Call Manager # # # # This example includes a basic configuration for the SmartNode # # 49XX/ISDN in combination with a Cisco Call Manager without # # SIP authentication and registration # # This example can be used for other SmartNodes too. The only # # thing which needs to be adjusted, are the ports and interfaces # # as far as required. # # # # Author: Olivier Gaudet # # Patton-Inalp Neworks # # CH-3172 Niederwangen # # www.patton.com; support@patton.com # # # # Date: 17/11/2011 # # # #----------------------------------------------------------------# cli version 3.20 clock local default-offset +00:00 dns-client server <Primary_DNS_Server_If_you_use_hostnames> dns-client server <Secondary_DNS_Server_If_you_use_hostnames> webserver port 80 language en sntp-client sntp-client server primary 129.132.2.21 port 123 version 4 system ic voice 0 system clock-source 1 e1t1 0 0 clock-source 1 e1t1 0 1 clock-source 1 e1t1 0 2 clock-source 1 e1t1 0 3 profile ppp default profile tone-set default profile voip default codec 1 g711alaw64k rx-length 20 tx-length 20 codec 2 g711ulaw64k rx-length 20 tx-length 20 fax transmission 1 relay t38-udp fax transmission 2 bypass g711alaw64k fax redundancy low-speed 2 high-speed 1 fax bypass-method nse modem transmission 1 bypass g711alaw64k modem bypass-method nse profile pstn default profile sip default no autonomous-transitioning profile aaa default method 1 local method 2 none context ip router interface LAN ipaddress <YOUR_IP_LAN_SIDE> <YOUR_NETWORK_MASK> tcp adjust-mss rx mtu tcp adjust-mss tx mtu context ip router route 0.0.0.0 0.0.0.0 <YOUR_DEFAULT_GATEWAY> 0 context cs switch digit-collection timeout 3 routing-table called-e164 RT_FROM_PSTN route default dest-interface IF_CISCO_CM routing-table called-e164 RT_FROM_CISCO route default dest-service HUNT_PSTN interface isdn IF_ISDN_00 route call dest-table RT_FROM_PSTN interface isdn IF_ISDN_01 route call dest-table RT_FROM_PSTN interface isdn IF_ISDN_02 route call dest-table RT_FROM_PSTN interface isdn IF_ISDN_03 route call dest-table RT_FROM_PSTN interface sip IF_CISCO_CM bind context sip-gateway CISCO_GW route call dest-table RT_FROM_CISCO remote <YOUR_PRIMARY_CALL_MANAGER_IP> 5060 service hunt-group HUNT_PSTN timeout 3 drop-cause normal-unspecified drop-cause no-circuit-channel-available drop-cause network-out-of-order drop-cause temporary-failure drop-cause switching-equipment-congestion drop-cause access-info-discarded drop-cause circuit-channel-not-available drop-cause resources-unavailable route call 1 dest-interface IF_ISDN_00 route call 2 dest-interface IF_ISDN_01 route call 3 dest-interface IF_ISDN_02 route call 3 dest-interface IF_ISDN_03 context cs switch no shutdown context sip-gateway CISCO_GW interface IF_sip_interface_GW bind interface LAN context router port 5060 context sip-gateway CISCO_GW no shutdown port ethernet 0 0 medium auto encapsulation ip bind interface LAN router no shutdown port e1t1 0 0 port-type e1 clock auto encapsulation q921 q921 permanent-layer2 uni-side auto encapsulation q931 q931 protocol dss1 uni-side user bchan-number-order ascending encapsulation cc-isdn bind interface IF_ISDN_00 switch port e1t1 0 0 no shutdown port e1t1 0 1 port-type e1 clock auto encapsulation q921 q921 permanent-layer2 uni-side auto encapsulation q931 q931 protocol dss1 uni-side user bchan-number-order ascending encapsulation cc-isdn bind interface IF_ISDN_01 switch port e1t1 0 1 no shutdown port e1t1 0 2 port-type e1 clock auto encapsulation q921 q921 permanent-layer2 uni-side auto encapsulation q931 q931 protocol dss1 uni-side user bchan-number-order ascending encapsulation cc-isdn bind interface IF_ISDN_02 switch port e1t1 0 2 no shutdown port e1t1 0 3 port-type e1 clock auto encapsulation q921 q921 permanent-layer2 uni-side auto encapsulation q931 q931 protocol dss1 uni-side user bchan-number-order ascending encapsulation cc-isdn bind interface IF_ISDN_03 switch port e1t1 0 3 no shutdown
Template for SN49XX PRI to SIP Trunking with 2 Cisco Call Manager-primary and secondary- without authentication ( Link to this Config )
	Configuration Description This example includes a basic configuration for the SmartNode SN49XX/ISDN to be used as a PRI gateway in combination with two Cisco Call Manager (Primary and secondary). The configuration contains variables in brackets <>. Please Replace the variables including the brackets. Unused lines or variables have to be commented out or deleted Command Sequence to Copy, Edit, and Paste #----------------------------------------------------------------# # Cisco Call Manager # # # # This example includes a basic configuration for the SmartNode # # 49XX/ISDN in combination with two Cisco Call Manager without # # SIP authentication and registration. # # This example can be used for other SmartNodes too. The only # # thing which needs to be adjusted, are the ports and interfaces # # as far as required. # # # # Author: Olivier Gaudet # # Patton-Inalp Neworks # # CH-3172 Niederwangen # # www.patton.com; support@patton.com # # # # Date: 17/11/2011 # # # #----------------------------------------------------------------# cli version 3.20 clock local default-offset +00:00 dns-client server <Primary_DNS_Server_If_you_use_hostnames> dns-client server <Secondary_DNS_Server_If_you_use_hostnames> webserver port 80 language en sntp-client sntp-client server primary 129.132.2.21 port 123 version 4 system ic voice 0 system clock-source 1 e1t1 0 0 clock-source 1 e1t1 0 1 clock-source 1 e1t1 0 2 clock-source 1 e1t1 0 3 profile ppp default profile tone-set default profile voip default codec 1 g711alaw64k rx-length 20 tx-length 20 codec 2 g711ulaw64k rx-length 20 tx-length 20 fax transmission 1 relay t38-udp fax transmission 2 bypass g711alaw64k fax redundancy low-speed 2 high-speed 1 fax bypass-method nse modem transmission 1 bypass g711alaw64k modem bypass-method nse profile pstn default profile sip default no autonomous-transitioning profile aaa default method 1 local method 2 none context ip router interface LAN ipaddress <YOUR_IP_LAN_SIDE> <YOUR_NETWORK_MASK> tcp adjust-mss rx mtu tcp adjust-mss tx mtu context ip router route 0.0.0.0 0.0.0.0 <YOUR_DEFAULT_GATEWAY> 0 context cs switch digit-collection timeout 3 routing-table called-e164 RT_FROM_PSTN route default dest-service HUNT_CiscoCM routing-table called-e164 RT_FROM_CISCO route default dest-service HUNT_PSTN interface isdn IF_ISDN_00 route call dest-table RT_FROM_PSTN interface isdn IF_ISDN_01 route call dest-table RT_FROM_PSTN interface isdn IF_ISDN_02 route call dest-table RT_FROM_PSTN interface isdn IF_ISDN_03 route call dest-table RT_FROM_PSTN interface sip IF_CISCO_CM bind context sip-gateway CISCO_GW route call dest-table RT_FROM_CISCO remote <YOUR_PRIMARY_CALL_MANAGER_IP> 5060 interface sip IF_CISCO_CM2 bind context sip-gateway CISCO_GW route call dest-table RT_FROM_CISCO remote <YOUR_SECONDARY_CALL_MANAGER_IP> 5060 service hunt-group HUNT_CiscoCM timeout 3 drop-cause normal-unspecified drop-cause no-circuit-channel-available drop-cause network-out-of-order drop-cause temporary-failure drop-cause switching-equipment-congestion drop-cause access-info-discarded drop-cause circuit-channel-not-available drop-cause resources-unavailable route call 1 dest-interface IF_CISCO_CM route call 2 dest-interface IF_CISCO_CM2 service hunt-group HUNT_PSTN timeout 3 drop-cause normal-unspecified drop-cause no-circuit-channel-available drop-cause network-out-of-order drop-cause temporary-failure drop-cause switching-equipment-congestion drop-cause access-info-discarded drop-cause circuit-channel-not-available drop-cause resources-unavailable route call 1 dest-interface IF_ISDN_00 route call 2 dest-interface IF_ISDN_01 route call 3 dest-interface IF_ISDN_02 route call 3 dest-interface IF_ISDN_03 context cs switch no shutdown context sip-gateway CISCO_GW interface IF_sip_interface_GW bind interface LAN context router port 5060 context sip-gateway CISCO_GW no shutdown port ethernet 0 0 medium auto encapsulation ip bind interface LAN router no shutdown port e1t1 0 0 port-type e1 clock auto encapsulation q921 q921 permanent-layer2 uni-side auto encapsulation q931 q931 protocol dss1 uni-side user bchan-number-order ascending encapsulation cc-isdn bind interface IF_ISDN_00 switch port e1t1 0 0 no shutdown port e1t1 0 1 port-type e1 clock auto encapsulation q921 q921 permanent-layer2 uni-side auto encapsulation q931 q931 protocol dss1 uni-side user bchan-number-order ascending encapsulation cc-isdn bind interface IF_ISDN_01 switch port e1t1 0 1 no shutdown port e1t1 0 2 port-type e1 clock auto encapsulation q921 q921 permanent-layer2 uni-side auto encapsulation q931 q931 protocol dss1 uni-side user bchan-number-order ascending encapsulation cc-isdn bind interface IF_ISDN_02 switch port e1t1 0 2 no shutdown port e1t1 0 3 port-type e1 clock auto encapsulation q921 q921 permanent-layer2 uni-side auto encapsulation q931 q931 protocol dss1 uni-side user bchan-number-order ascending encapsulation cc-isdn bind interface IF_ISDN_03 switch port e1t1 0 3 no shutdown
Template for SN5XXX SBC between VOIP provider and 2 Cisco Call Manager -Primary and Secondary- ( Link to this Config )
	Configuration Description This example includes a basic configuration for the SmartNode SN5XXX to be used as a Session Border Controller in combination with two Cisco Call Manager (Primary and secondary). The configuration contains variables in brackets <>. Please Replace the variables including the brackets. Unused lines or variables have to be commented out or deleted Command Sequence to Copy, Edit, and Paste #----------------------------------------------------------------# # Cisco Call Manager # # # # This example includes a basic configuration for the SmartNode # # SN5XXX in combination with two Cisco Call Manager on the LAN # # side and a SIP trunk to a VoIP provider on the WAN side. # # This example can be used for other SmartNodes too. The only # # thing which needs to be adjusted, are the ports and interfaces # # as far as required. # # # # Author: Olivier Gaudet # # Patton-Inalp Neworks # # CH-3172 Niederwangen # # www.patton.com; support@patton.com # # # # Date: 17/11/2011 # # # #----------------------------------------------------------------# cli version 3.20 clock local default-offset +00:00 dns-client server <Primary_DNS_Server_If_you_use_hostnames> dns-client server <Secondary_DNS_Server_If_you_use_hostnames> webserver port 80 language en sntp-client sntp-client server primary 129.132.2.21 port 123 version 4 system ic voice 0 profile napt NAPT_WAN profile ppp default profile tone-set default profile voip default codec 1 g711alaw64k rx-length 20 tx-length 20 codec 2 g711ulaw64k rx-length 20 tx-length 20 fax transmission 1 relay t38-udp fax transmission 2 bypass g711alaw64k fax redundancy low-speed 2 high-speed 1 fax bypass-method nse modem transmission 1 bypass g711alaw64k modem bypass-method nse profile pstn default profile sip default no autonomous-transitioning profile aaa default method 1 local method 2 none context ip router interface WAN ipaddress <YOUR_IP_WAN_SIDE> <YOUR_NETWORK_MASK> use profile napt NAPT_WAN tcp adjust-mss rx mtu tcp adjust-mss tx mtu interface LAN ipaddress <YOUR_IP_LAN_SIDE> <YOUR_NETWORK_MASK> tcp adjust-mss rx mtu tcp adjust-mss tx mtu context ip router route 0.0.0.0 0.0.0.0 <YOUR_DEFAULT_GATEWAY> 0 context cs switch routing-table called-e164 RT_FROM_CISCO route default dest-interface INTERFACE_TO_VOIP_PROVIDER routing-table called-e164 ROUTE_FROM_PROVIDER route default dest-service HUNT_In interface sip INTERFACE_TO_VOIP_PROVIDER bind context sip-gateway VOIP_PROVIDER_GW route call dest-table ROUTE_FROM_PROVIDER remote <YOUR_VOIP_PROVIDER_IP> 5060 interface sip IF_CISCO_CM bind context sip-gateway CISCO_GW route call dest-table RT_FROM_CISCO remote <YOUR_PRIMARY_CALL_MANAGER_IP> 5060 interface sip IF_CISCO_CM2 bind context sip-gateway CISCO_GW route call dest-table RT_FROM_CISCO remote <YOUR_SECONDARY_CALL_MANAGER_IP> 5060 service hunt-group HUNT_In timeout 3 allows-push-back drop-cause normal-unspecified drop-cause no-circuit-channel-available drop-cause network-out-of-order drop-cause temporary-failure drop-cause switching-equipment-congestion drop-cause access-info-discarded drop-cause circuit-channel-not-available drop-cause resources-unavailable drop-cause recovery-on-timer-expiry drop-cause no-route-to-destination drop-cause no-route-to-network drop-cause call-rejected drop-cause protocol-error route call 1 dest-interface IF_CISCO_CM route call 2 dest-interface IF_CISCO_CM2 context cs switch no shutdown authentication-service AUTH_VOIP_PROVIDER username <Username_SIP> password <Password_SIP> location-service LOCATION_PROVIDER domain 1 <YOUR_VOIP_PROVIDER_IP> identity-group default authentication outbound authenticate 1 authentication-service AUTH_VOIP_PROVIDER username <Username_SIP> registration outbound register auto context sip-gateway VOIP_PROVIDER_GW interface IF_sip_interface_voip_provider bind interface WAN context router port 5060 context sip-gateway VOIP_PROVIDER_GW bind location-service LOCATION_PROVIDER no shutdown context sip-gateway CISCO_GW interface IF_sip_interface_Cicso bind interface LAN context router port 5060 context sip-gateway CISCO_GW no shutdown port ethernet 0 0 medium auto encapsulation ip bind interface WAN router no shutdown port ethernet 0 1 medium auto encapsulation ip bind interface LAN router no shutdown
Avaya Open Office ( Link to this Config )
	Configuration Description Configuration for SN4112 for use with Avaya Open Office. ============================================ This example includes a basic configuration for the SmartNode SN4112/JS for use as an analog Terminal gateway (ATA for Fax and Telephones) in combination with the Avaya Open Office. The configuration contains variables in brackets <>. Please Replace the variables including the brackets. Unused lines or variables have to be commented out or deleted Command Sequence to Copy, Edit, and Paste #----------------------------------------------------------------# # Avaya Open Office # # # # This example includes a basic configuration for the SmartNode # # 4112 in combination with the Avaya Open Office. This example # # can be used for other SmartNodes too. The only thing which # # needs to be adjusted, are the ports and interfaces as far as # # required. # # # # Author: Thomas Tlapak # # Patton-Inalp Neworks # # CH-3172 Niederwangen # # www.patton.com; support@patton.com # # # # Date: 8/29/2011 # # # #----------------------------------------------------------------# cli version 3.20 webserver port 80 language en dns-client server 8.8.8.8 sntp-client sntp-client server primary 129.132.2.21 port 123 version 4 system ic voice 0 low-bitrate-codec g729 profile ppp default profile tone-set default profile voip default codec 1 g711alaw64k rx-length 20 tx-length 20 profile pstn default profile ringing-cadence default play 1 1000 pause 2 4000 profile sip default no autonomous-transitioning profile aaa default method 1 local method 2 none context ip router interface eth0 ipaddress dhcp tcp adjust-mss rx mtu tcp adjust-mss tx mtu context ip router context cs switch routing-table called-e164 RT_TO_SIP route .T dest-interface IF_SIP routing-table called-e164 RT_TO_FXS route <Phonenumber Port 0 0> dest-interface IF_FXS_00 route <Phonenumber Port 0 1> dest-interface IF_FXS_01 interface sip IF_SIP bind context sip-gateway GW_SIP route call dest-table RT_TO_FXS remote <IP or Domain> interface fxs IF_FXS_00 route call dest-table RT_TO_SIP caller-id-presentation mid-ring subscriber-number <Phonenumber Port 0 0> interface fxs IF_FXS_01 route call dest-table RT_TO_SIP caller-id-presentation mid-ring subscriber-number <Phonenumber Port 0 1> context cs switch no shutdown authentication-service AUTH_SVC username <Username 1> password <Password> username <Username 2> password <Password> location-service LOCATION_SVC domain 1 <IP or Domain> identity <Username 1> display-name <Username 1> authentication outbound authenticate 1 authentication-service AUTH_SVC username <Username 1> registration outbound register auto call outbound proxy 1 <IP or Domain> identity <Username 2> display-name <Username 2> authentication outbound authenticate 1 authentication-service AUTH_SVC username <Username 2> registration outbound registrar <IP or Domain> register auto call outbound proxy 1 <IP or Domain> context sip-gateway GW_SIP interface IF_GW_SIP bind interface eth0 context router port 5060 context sip-gateway GW_SIP bind location-service LOCATION_SVC no shutdown port ethernet 0 0 medium auto encapsulation ip bind interface eth0 router no shutdown port fxs 0 0 encapsulation cc-fxs bind interface IF_FXS_00 switch no shutdown port fxs 0 1 encapsulation cc-fxs bind interface IF_FXS_01 switch no shutdown
Stacking SmartNode`s ( Link to this Config )
	Configuration Description In some cases, more FXS, FXO or BRI ports are needed as available on one SmartNode for certain applications. This conifg example shows, how to stack two or more SmartNodes, achieving more channels / ports. The idea is to let the "Master" SmartNode handle all calls. It communicates with the SIP server and distinguishes calls to the slave SN over the second ethernet port (LAN) in case all TDM channels are occupied. The service hunt-group is in charge of this. ISDN ports, 0/0 on both SmartNodes are used for clock sync, while a PSTN line is plugged. IP network which connects the two SmartNodes is 192.168.20.0/24. There is no SIP call authentication between the two SmartNodes. Please make sure you set and replace all values which are commented as below with #. Command Sequence to Copy, Edit, and Paste MASTER SN46XX ------------- #----------------------------------------------------------------# # # # SN4638 MASTER # # R5.T # # 2010-05-12 # # Author M. Aeberhard # # # #----------------------------------------------------------------# cli version 3.20 banner # enter below the DNS server IP dns-client server [dns-serverIP] dns-relay webserver port 80 language en sntp-client sntp-client server primary swisstime.ethz.ch port 123 version 4 system hostname MASTER_SN system ic voice 0 low-bitrate-codec g729 system clock-source 1 bri 0 0 profile napt NAPT_WAN profile ppp default profile tone-set default profile voip default codec 1 g711alaw64k rx-length 20 tx-length 20 codec 2 g711ulaw64k rx-length 20 tx-length 20 fax transmission 1 relay t38-udp modem transmission 1 bypass g711alaw64k modem bypass-method nse profile pstn default profile sip default profile dhcp-server DHCP_LAN profile aaa default method 1 local method 2 none context ip router interface IF_WAN ipaddress dhcp use profile napt NAPT_WAN interface IF_LAN_VOIP ipaddress 192.168.20.1 255.255.255.0 context cs switch routing-table called-e164 RT_DIGIT_COLL route .T dest-interface IF_SIP_DEFAULT routing-table called-e164 RT_FROM_NET route default dest-service HG_TO_USR interface isdn IF_ISDN_S00 route call dest-table RT_FROM_NET isdn-date-time interface isdn IF_ISDN_S01 route call dest-table RT_DIGIT_COLL call-reroute accept no call-waiting network-type user isdn-date-time interface isdn IF_ISDN_S02 route call dest-table RT_DIGIT_COLL call-reroute accept no call-waiting network-type user isdn-date-time interface isdn IF_ISDN_S03 route call dest-table RT_DIGIT_COLL call-reroute accept no call-waiting network-type user isdn-date-time interface isdn IF_ISDN_S04 route call dest-table RT_DIGIT_COLL call-reroute accept no call-waiting network-type user isdn-date-time interface sip IF_SIP_DEFAULT bind context sip-gateway SIP_WAN route call dest-service HG_TO_USR # remote [registrar IP address or domain name] interface sip IF_SIP_LAN bind context sip-gateway SIP_LAN route call dest-interface IF_SIP_DEFAULT remote 192.168.20.2 local 192.168.20.2 service hunt-group HG_TO_USR allows-push-back drop-cause normal-unspecified drop-cause no-circuit-channel-available drop-cause network-out-of-order drop-cause temporary-failure drop-cause switching-equipment-congestion drop-cause access-info-discarded drop-cause circuit-channel-not-available drop-cause resources-unavailable drop-cause destination-out-of-order unavailable drop user-busy route call 1 dest-interface IF_ISDN_S01 route call 2 dest-interface IF_ISDN_S02 route call 2 dest-interface IF_ISDN_S03 route call 2 dest-interface IF_ISDN_S04 route call 3 dest-interface IF_SIP_LAN context cs switch no shutdown authentication-service AUTH_DEF # Enter below the username and passwort you got by VOIP provider username [XXX] password [YYY] location-service LOC_DEF # The line below requires the VOIP Providers domain name domain 1 [registrar IP or domain] identity-group default authentication outbound # Replace the XXX with the username of Authentication service authenticate 1 authentication-service AUTH_DEF username [XXX] registration outbound # Set the Registrar IP or domainname of VOIP provider registrar [registrar address] 5060 register auto # Set the right username while replacing the XXX identity [XXX] inherits default context sip-gateway SIP_WAN interface GW_SIP_WAN bind interface IF_WAN context router port 5060 context sip-gateway SIP_WAN bind location-service LOC_DEF no shutdown context sip-gateway SIP_LAN interface GW_SIP_LAN bind interface IF_LAN_VOIP context router port 5060 context sip-gateway SIP_LAN no shutdown port ethernet 0 0 medium auto encapsulation ip bind interface IF_WAN router no shutdown port ethernet 0 1 medium auto encapsulation ip bind interface IF_LAN_VOIP router no shutdown port bri 0 0 clock auto encapsulation q921 q921 permanent-layer2 protocol pp uni-side auto encapsulation q931 q931 protocol dss1 uni-side user bchan-number-order ascending encapsulation cc-isdn bind interface IF_ISDN_S00 switch port bri 0 0 no shutdown port bri 0 1 clock auto encapsulation q921 q921 permanent-layer2 protocol pp uni-side auto encapsulation q931 q931 protocol dss1 uni-side net bchan-number-order ascending encapsulation cc-isdn bind interface IF_ISDN_S01 switch port bri 0 1 no shutdown port bri 0 2 clock auto encapsulation q921 q921 permanent-layer2 protocol pp uni-side auto encapsulation q931 q931 protocol dss1 uni-side net bchan-number-order ascending encapsulation cc-isdn bind interface IF_ISDN_S02 switch port bri 0 2 no shutdown port bri 0 3 clock auto encapsulation q921 q921 permanent-layer2 protocol pp uni-side auto encapsulation q931 q931 protocol dss1 uni-side net bchan-number-order ascending encapsulation cc-isdn bind interface IF_ISDN_S03 switch port bri 0 3 no shutdown port bri 0 4 clock auto encapsulation q921 q921 permanent-layer2 protocol pp uni-side auto encapsulation q931 q931 protocol dss1 uni-side net bchan-number-order ascending encapsulation cc-isdn bind interface IF_ISDN_S03 switch port bri 0 4 no shutdown ------------------------------------------------- SLAVE SN46XX #----------------------------------------------------------------# # # # SN4634 SLAVE # # R5.T # # 2010-05-12 # # Author M. Aeberhard # # # #----------------------------------------------------------------# cli version 3.20 banner # enter below the DNS server IP dns-client server [dns-serverIP] dns-relay webserver port 80 language en sntp-client sntp-client server primary swisstime.ethz.ch port 123 version 4 system hostname SLAVE_SN system ic voice 0 low-bitrate-codec g729 system clock-source 1 bri 0 0 profile napt NAPT_WAN profile ppp default profile tone-set default profile voip default codec 1 g711alaw64k rx-length 20 tx-length 20 codec 2 g711ulaw64k rx-length 20 tx-length 20 fax transmission 1 relay t38-udp modem transmission 1 bypass g711alaw64k modem bypass-method nse profile pstn default profile sip default profile dhcp-server DHCP_LAN network 192.168.1.0 255.255.255.0 include 1 192.168.1.33 192.168.1.64 lease 1 days default-router 1 192.168.1.1 domain-name-server 1 192.168.1.1 profile aaa default method 1 local method 2 none context ip router interface IF_WAN ipaddress 192.168.20.2 255.255.255.0 use profile napt NAPT_WAN interface IF_LAN ipaddress 192.168.1.1 255.255.255.0 context ip router dhcp-server use profile DHCP_LAN route 0.0.0.0 0.0.0.0 192.168.20.1 0 context cs switch routing-table called-e164 RT_DIGIT_COLL route .T dest-interface IF_SIP_DEFAULT routing-table called-e164 RT_FROM_NET route default dest-service HG_TO_USR interface isdn IF_ISDN_S00 route call dest-table RT_FROM_NET network-type user isdn-date-time interface isdn IF_ISDN_S01 route call dest-table RT_DIGIT_COLL call-reroute accept no call-waiting network-type user isdn-date-time interface isdn IF_ISDN_S02 route call dest-table RT_DIGIT_COLL call-reroute accept no call-waiting network-type user isdn-date-time interface sip IF_SIP_DEFAULT bind context sip-gateway sip route call dest-table RT_FROM_NET remote 192.168.20.1 local 192.168.20.1 service hunt-group HG_TO_USR drop-cause normal-unspecified drop-cause no-circuit-channel-available drop-cause network-out-of-order drop-cause temporary-failure drop-cause switching-equipment-congestion drop-cause access-info-discarded drop-cause circuit-channel-not-available drop-cause resources-unavailable drop-cause destination-out-of-order unavailable drop user-busy route call 1 dest-interface IF_ISDN_S01 route call 2 dest-interface IF_ISDN_S02 context cs switch no shutdown context sip-gateway sip interface GW_SIP_WAN bind interface IF_WAN context router port 5060 context sip-gateway sip bind location-service LOC_DEF no shutdown port ethernet 0 0 medium auto encapsulation ip bind interface IF_WAN router no shutdown port ethernet 0 1 medium auto encapsulation ip bind interface IF_LAN router no shutdown port bri 0 0 clock slave encapsulation q921 q921 permanent-layer2 protocol pp uni-side auto encapsulation q931 q931 protocol dss1 uni-side user bchan-number-order ascending encapsulation cc-isdn bind interface IF_ISDN_S00 switch port bri 0 0 no shutdown port bri 0 1 clock auto encapsulation q921 q921 permanent-layer2 protocol pp uni-side auto encapsulation q931 q931 protocol dss1 uni-side net bchan-number-order ascending encapsulation cc-isdn bind interface IF_ISDN_S01 switch port bri 0 1 no shutdown port bri 0 2 clock auto encapsulation q921 q921 permanent-layer2 protocol pp uni-side auto encapsulation q931 q931 protocol dss1 uni-side net bchan-number-order ascending encapsulation cc-isdn bind interface IF_ISDN_S02 switch port bri 0 2 no shutdown
SmartNode configuration for use with OCS ( Link to this Config )
	Configuration Description Configuration for SN4638 for use with OCS. A more detailed explanation you will find on the following link. Command Sequence to Copy, Edit, and Paste #----------------------------------------------------------------# # # # SN4638/5BIS/UI # # R5.X # # # # OCS sampe configuration # # Author: Thomas Tlapak # # Patton-Inalp Neworks # # CH-3172 Niederwangen # # www.patton.com; support@patton.com # # # # Date: 5/11/2010 # # # #----------------------------------------------------------------# cli version 3.20 administrator administrator password abersogeheim! encrypted clock local offset +01:00 dns-relay webserver port 80 language en sntp-client sntp-client server primary 193.84.178.254 port 123 version 4 system hostname patton-tb system ic voice 0 low-bitrate-codec g729 system clock-source 1 bri 0 0 clock-source 2 bri 0 1 clock-source 3 bri 0 2 clock-source 4 bri 0 3 clock-source 5 bri 0 4 profile napt NAPT_WAN profile ppp default profile tone-set default profile voip default codec 1 g711alaw64k rx-length 20 tx-length 20 no-silence-suppression codec 2 g711ulaw64k rx-length 20 tx-length 20 no-silence-suppression rtp traffic-class local-default profile pstn default profile sip default profile dhcp-server DHCPS_LAN network 192.168.1.0 255.255.255.0 include 1 192.168.1.10 192.168.1.99 lease 2 hours default-router 1 192.168.1.1 domain-name-server 1 192.168.1.1 profile aaa default method 1 local method 2 none context ip router interface WAN ipaddress dhcp use profile napt NAPT_WAN tcp adjust-mss rx mtu tcp adjust-mss tx mtu interface IF_IP_LAN ipaddress 192.168.1.1 255.255.255.0 context ip router # Set below the default gateway IP address route 0.0.0.0 0.0.0.0 <default gateway IP> 0 context cs switch routing-table called-e164 RT_FROM_PSTN route .T dest-interface IF_SIP_OCS routing-table calling-e164 RT_TO_PSTN_OCS route .T dest-service SER_HUNT_PSTN interface isdn IF_TE_00 route call dest-table RT_FROM_PSTN interface isdn IF_TE_01 route call dest-table RT_FROM_PSTN interface isdn IF_TE_02 route call dest-table RT_FROM_PSTN interface isdn IF_TE_03 route call dest-table RT_FROM_PSTN interface isdn IF_TE_04 route call dest-table RT_FROM_PSTN interface sip IF_SIP_OCS bind context sip-gateway GW_OCS route call dest-table RT_TO_PSTN_OCS # Enter below the OCS server IP remote <Domain or IP> early-connect early-disconnect service hunt-group SER_HUNT_PSTN cyclic timeout 6 drop-cause normal-unspecified drop-cause no-circuit-channel-available drop-cause network-out-of-order drop-cause temporary-failure drop-cause switching-equipment-congestion drop-cause access-info-discarded drop-cause circuit-channel-not-available drop-cause resources-unavailable route call 1 dest-interface IF_TE_00 route call 2 dest-interface IF_TE_01 route call 3 dest-interface IF_TE_02 route call 4 dest-interface IF_TE_03 route call 5 dest-interface IF_TE_04 context cs switch no shutdown location-service SER_OCS domain 1 <Domain or IP> <Port> identity-group default call outbound preferred-transport-protocol tcp context sip-gateway GW_OCS interface IF_IP_LAN bind interface IF_IP_LAN context router port 5060 context sip-gateway GW_OCS bind location-service SER_OCS no shutdown port ethernet 0 0 medium auto encapsulation ip bind interface WAN router no shutdown port ethernet 0 1 medium auto encapsulation ip bind interface IF_IP_LAN router no shutdown port bri 0 0 clock auto encapsulation q921 q921 permanent-layer2 protocol pp uni-side user encapsulation q931 q931 protocol dss1 uni-side user bchan-number-order ascending encapsulation cc-isdn bind interface IF_TE_00 switch port bri 0 0 no shutdown port bri 0 1 clock auto encapsulation q921 q921 permanent-layer2 protocol pp uni-side user encapsulation q931 q931 protocol dss1 uni-side user bchan-number-order ascending encapsulation cc-isdn bind interface IF_TE_01 switch port bri 0 1 no shutdown port bri 0 2 clock auto encapsulation q921 q921 permanent-layer2 protocol pp uni-side auto encapsulation q931 q931 protocol dss1 uni-side user bchan-number-order ascending encapsulation cc-isdn bind interface IF_TE_02 switch port bri 0 2 no shutdown port bri 0 3 clock auto encapsulation q921 q921 permanent-layer2 protocol pp uni-side auto encapsulation q931 q931 protocol dss1 uni-side user bchan-number-order ascending encapsulation cc-isdn bind interface IF_TE_03 switch port bri 0 3 no shutdown port bri 0 4 clock auto encapsulation q921 q921 permanent-layer2 protocol pp uni-side auto encapsulation q931 q931 protocol dss1 uni-side user bchan-number-order ascending encapsulation cc-isdn bind interface IF_TE_04 switch port bri 0 4 no shutdown
Dial up connection  ( Link to this Config )
	Configuration Description !Additional license required for Dial-up! This example configuration can be used as a backup solution, in case a specific host in a specific network is not available anymore. In this case an ISDN Dial-up connection will be set up. The default route will be removed and the default route with the next higher metric will be used. This is done by the feature "link supervision" in addition with "dial-up". As soon as the specified host is available again, the Dial-up connection will be terminated. The traffic will be routed over the regular IP interface as it was initially. The default route with the lower metric is used again. Relpace: Z IP-Address WAN-port SmartNode Y IP-Address Specific host (which needs to be checked) X IP-Address default Gateway for Network Z W Username for Dial-up connection V Password for Dial-up connetion U Local e164 Number T Remote e164 Number to dial (ISP) Command Sequence to Copy, Edit, and Paste #----------------------------------------------------------------# # # # SN XXXX # # R5.X H323 SIP BRI # # # # Example configuration file # # # #----------------------------------------------------------------# cli version 3.20 clock local offset +01:00 dns-relay webserver port 80 language en sntp-client server primary 129.132.2.21 port 123 version 4 system hostname Dialup__Unit system ic voice 0 low-bitrate-codec g729 system clock-source 1 bri 0 0 profile napt NAPT_WAN profile ppp default profile tone-set default profile voip default codec 1 g711alaw64k rx-length 20 tx-length 20 codec 2 g711ulaw64k rx-length 20 tx-length 20 fax transmission 1 relay t38-udp modem transmission 1 bypass g711alaw64k modem bypass-method nse profile pstn default profile sip default profile dhcp-server DHCPS_LAN network 192.168.1.0 255.255.255.0 include 1 192.168.1.10 192.168.1.99 lease 2 hours default-router 1 192.168.1.1 domain-name-server 1 192.168.1.1 profile aaa default method 1 local method 2 none context ip router interface WAN ipaddress unnumbered point-to-point use profile napt NAPT_WAN tcp adjust-mss rx mtu tcp adjust-mss tx mtu dial-up monitor interface IF_WAN dial delay 5 drop delay 10 queue-limit 4 interface LAN ipaddress 192.168.1.1 255.255.255.0 tcp adjust-mss rx mtu tcp adjust-mss tx mtu interface IF_WAN ipaddress <ZZZZZZZ> <mask> use profile napt NAPT_WAN check-connectivity ping <YYYYYYY> tolerance-down 3 tolerance-up 1 timeout 3 interval 3 context ip router dhcp-server use profile DHCPS_LAN route 0.0.0.0 0.0.0.0 <XXXXXXX> 0 route 0.0.0.0 0.0.0.0 WAN 1 subscriber ppp Dialup dial out no multilink authentication chap authentication pap identification outbound <WWWWWWW> password <VVVVVVV> timeout idle 100 bind interface WAN router context cs switch interface isdn IF_BRI_00 route call dest-interface IF_SIP2 interface dialer IF_DIALUP outbound 1 DialupNet local-e164 <UUUUUUU> remote-e164 <TTTTTTT> route call dest-interface IF_BRI_00 use profile ppp default encapsulation ppp bind subscriber Dialup context cs switch no shutdown port ethernet 0 0 medium auto encapsulation ip bind interface IF_WAN router no shutdown port ethernet 0 1 medium auto encapsulation ip bind interface LAN router no shutdown port bri 0 0 clock auto encapsulation q921 q921 uni-side user encapsulation q931 q931 protocol dss1 uni-side user bchan-number-order ascending encapsulation cc-isdn bind interface IF_BRI_00 switch port bri 0 0 no shutdown
FXO To SIP TRUNK (with HUNT) ( Link to this Config )
	Configuration Description The following will take calls in the FXO port, and send it out sip. It will also take calls in SIP and then hunt across all the FXO ports to find a open port This config also contains the USA tonesets, you can change that as needed for local tones. Finally, if you choose to use DNS names in the SIP interfaces or in the SIP gateway, you will need to specify a DNS server for any config to work. Command Sequence to Copy, Edit, and Paste #----------------------------------------------------------------# # # # SN4114/JO/EUI # # R5.X # # Config description above on Config library page # # # #----------------------------------------------------------------# cli version 3.20 webserver port 80 language en sntp-client sntp-client server primary 192.168.41.20 port 123 version 4 system ic voice 0 low-bitrate-codec g729 profile ppp default profile call-progress-tone US_Dialtone play 0 350 -13 440 -13 profile call-progress-tone US_Alertingtone play 2000 440 -19 480 -19 pause 4000 profile call-progress-tone US_Busytone play 500 480 -24 620 -24 pause 500 profile tone-set default profile tone-set US map call-progress-tone dial-tone US_Dialtone map call-progress-tone ringback-tone US_Alertingtone map call-progress-tone busy-tone US_Busytone map call-progress-tone release-tone US_Busytone map call-progress-tone congestion-tone US_Busytone profile voip default codec 1 g711alaw64k rx-length 20 tx-length 20 codec 2 g711ulaw64k rx-length 20 tx-length 20 profile pstn default profile sip default profile aaa default method 1 local method 2 none context ip router interface eth0 ipaddress dhcp tcp adjust-mss rx mtu tcp adjust-mss tx mtu context cs switch digit-collection timeout 2 interface sip IF_SIP bind context sip-gateway GW_SIP route call dest-service FXO-HUNT-OUT # Set below the server IP address remote <remote IP> interface fxo IF_FXO_00 route call dest-interface IF_SIP use profile tone-set US interface fxo IF_FXO_01 route call dest-interface IF_SIP use profile tone-set US interface fxo IF_FXO_02 route call dest-interface IF_SIP use profile tone-set US interface fxo IF_FXO_03 route call dest-interface IF_SIP use profile tone-set US service hunt-group FXO-HUNT-OUT drop-cause normal-unspecified drop-cause no-circuit-channel-available drop-cause network-out-of-order drop-cause temporary-failure drop-cause switching-equipment-congestion drop-cause access-info-discarded drop-cause circuit-channel-not-available drop-cause resources-unavailable drop-cause user-busy route call 1 dest-interface IF_FXO_00 route call 2 dest-interface IF_FXO_01 route call 3 dest-interface IF_FXO_02 route call 4 dest-interface IF_FXO_03 context sip-gateway GW_SIP interface IF_GWSIP bind interface eth0 context router port 5060 context sip-gateway GW_SIP no shutdown port ethernet 0 0 medium auto encapsulation ip bind interface eth0 router no shutdown port fxo 0 0 use profile fxo us caller-id format bell encapsulation cc-fxo bind interface IF_FXO_00 switch no shutdown port fxo 0 1 use profile fxo us caller-id format bell encapsulation cc-fxo bind interface IF_FXO_01 switch no shutdown port fxo 0 2 use profile fxo us caller-id format bell encapsulation cc-fxo bind interface IF_FXO_02 switch no shutdown port fxo 0 3 use profile fxo us caller-id format bell encapsulation cc-fxo bind interface IF_FXO_03 switch no shutdown
SmartNode 4552 with SIP-Proxy Sipgate.de and fallback to PSTN ( Link to this Config )
	Configuration Description Configuration for SN4552 with registration to www.sipgate.de Every national and international long distance call goes to www.sipgate.de. Every local call goes out to www.sipgate.de. All emergency calls and special Service Numbers like 110 = police germany, 112 = firebrigade, Health Car germany, 18XX = costable Service numbers are going out directly to local PSTN ISDN interface. In case of sipgate failure, all sipgate numbers goes as fallback to the local PSTN ISDN Interface. Command Sequence to Copy, Edit, and Paste #----------------------------------------------------------------# # # # SN4552/2BIS/EUI # # R5.X # # # # DATE: 2011-01-10 # # AUTHOR: Thomas Tlapak # # PATTON-INALP-NETWORKS; CH-3172 NIEDERWANGEN # # WWW.PATTON.COM WWW.INALP.COM # # # #----------------------------------------------------------------# cli version 3.20 # Set below the DNS server IP address dns-client server <ip-address> webserver port 80 language en system ic voice 0 profile ppp default profile tone-set default profile voip default codec 1 g711alaw64k rx-length 20 tx-length 20 codec 2 g711ulaw64k rx-length 20 tx-length 20 profile voip SIPGATE codec 1 g729 rx-length 20 tx-length 20 codec 2 g711ulaw64k rx-length 20 tx-length 20 codec 3 g711alaw64k rx-length 20 tx-length 20 profile pstn default profile sip default profile sip sipgate profile authentication default method 1 local method 2 none context ip router interface IF_IP_WAN # Set the Smartnode WAN IP ipaddress <ip-address> <subnet-mask> interface IF_IP_LAN # Set the Smartnode LAN IP ipaddress <ip-address> <subnet-mask> context ip router # Set the default gateway IP address route 0.0.0.0 0.0.0.0 <default-gateway> 0 context cs switch routing-table called-e164 RT_ISDN_TO_SIP route 1[1-9].T3 dest-interface IF_ISDN_00_PSTN route [2-9].T3 dest-service SERV_HG_OUT route default dest-service SERV_HG_OUT interface isdn IF_ISDN_00_PSTN route call dest-interface IF_ISDN_01_PHONE interface isdn IF_ISDN_01_PHONE route call dest-table RT_ISDN_TO_SIP interface sip IF_SIP_SIPGATE bind context sip-gateway GW_SIP remote sipgate.de route call dest-interface IF_ISDN_01_PHONE # Set the SIPgate user identity below address-translation outgoing-call from-header user-part fix <user-id sipgate> host-part call use profile voip SIPGATE service hunt-group SERV_HG_OUT timeout 6 drop-cause normal-unspecified drop-cause no-circuit-channel-available drop-cause network-out-of-order drop-cause temporary-failure drop-cause switching-equipment-congestion drop-cause access-info-discarded drop-cause circuit-channel-not-available drop-cause resources-unavailable route call 1 dest-interface IF_SIP_SIPGATE route call 2 dest-interface IF_ISDN_00_PSTN context cs switch no shutdown authentication-service AUTH_SVC # Set username and password accordingly username <user-id sipgate> password <authentification-password SIP security sipgate.de> location-service LOCATION_SVC domain 1 sipgate.de # Set the SIPgate user identity below identity <user-id sipgate> authentication outbound # Set the SIPgate user identity below authenticate 1 authentication-service AUTH_SVC username <user-id sipgate> registration outbound registrar 1 sipgate.de register auto context sip-gateway GW_SIP interface IF_GW_SIP bind interface WAN context router port 5060 context sip-gateway GW_SIP bind location-service LOCATION_SVC no shutdown port ethernet 0 0 encapsulation ip bind interface IF_IP_WAN router no shutdown port ethernet 0 1 encapsulation ip bind interface IF_IP_LAN router no shutdown port bri 0 0 clock auto encapsulation q921 q921 protocol pmp uni-side auto encapsulation q931 q931 protocol dss1 uni-side user encapsulation cc-isdn bind interface IF_ISDN_00_PSTN switch port bri 0 0 no shutdown port bri 0 1 clock auto encapsulation q921 q921 protocol pmp uni-side auto encapsulation q931 q931 protocol dss1 uni-side net encapsulation cc-isdn bind interface IF_ISDN_01_PHONE switch port bri 0 1 no shutdown
Altigen H.323 SOHO Router ( Link to this Config )
	Configuration Description Generic Altigen H.323 SOHO Router Config # Replace ‹Altigen IP› Command Sequence to Copy, Edit, and Paste cli version 3.00 banner "My Altigen Config" dns-client enabled dns-client cache 30 dns-relay webserver port 80 language en snmp community public ro sntp-client sntp-client server primary 192.43.244.18 port 123 version 4 sntp-client server secondary 63.149.208.50 port 123 version 4 sntp-client gmt-offset - 05:00:00 system ic voice 0 profile service-policy rateVoiceIn rate-limit 3000 header-length 18 voice-margin 200 queue-limit 8 debug queue statistics detail 3 source traffic-class default source traffic-class local-voice priority profile service-policy rateVoiceOut rate-limit 256 header-length 18 voice-margin 0 debug queue statistics detail 3 source traffic-class local-voice priority set ip tos 8 source traffic-class local-default priority source traffic-class default profile napt NAPT-ETH0 profile ppp default profile call-progress-tone US_Dialtone play 0 350 -13 440 -13 profile call-progress-tone US_Alertingtone play 2000 440 -19 480 -19 no play 4000 profile call-progress-tone US_Busytone play 500 480 -24 620 -24 no play 500 profile tone-set default profile tone-set US map call-progress-tone dial-tone US_Dialtone map call-progress-tone ringback-tone US_Alertingtone map call-progress-tone busy-tone US_Busytone map call-progress-tone release-tone US_Busytone map call-progress-tone congestion-tone US_Busytone profile voip default codec 1 g723-6k3 rx-length 30 tx-length 30 codec 2 g711alaw64k rx-length 20 tx-length 20 codec 3 g711ulaw64k rx-length 20 tx-length 20 dejitter-max-delay 300 profile voip VOIP-IF-Patton codec 1 g723-6k3 rx-length 30 tx-length 30 profile voip VOIP-IF-incom codec 1 g723-6k3 rx-length 30 tx-length 30 codec 2 g726-32k rx-length 20 tx-length 20 codec 3 g729 rx-length 20 tx-length 20 codec 4 g711ulaw64k rx-length 20 tx-length 20 codec 5 g711alaw64k rx-length 20 tx-length 20 profile pstn default profile dhcp-server dhcp_serv network 192.168.1.0 255.255.255.0 include 1 192.168.1.200 192.168.1.254 lease 60 days default-router 1 192.168.1.1 domain-name-server 1 192.168.1.1 profile authentication default method 1 local method 2 none context ip router interface eth0 ipaddress dhcp mtu 1480 icmp router-discovery icmp redirect accept use profile service-policy rateVoiceIn in use profile service-policy rateVoiceOut out use profile napt NAPT-ETH0 tcp adjust-mss rx mtu tcp adjust-mss tx mtu interface eth1 ipaddress 192.168.1.1 255.255.255.0 mtu 1480 icmp router-discovery icmp redirect accept tcp adjust-mss rx mtu tcp adjust-mss tx mtu # dyndns # hostname ‹myname›.dyndns.org wildcard # authentication ‹myname› ‹mypassword› # service dynamic # observe eth0 context ip router dhcp-server use dhcp_serv context cs switch routing-table called-e164 TAB-outgoing route default dest-table TAB-DIGCOL-IF-Patton routing-table called-e164 TAB-DIGCOL-IF-incom route default dest-interface IF-incom routing-table called-e164 TAB-DIGCOL-IF-terminal route default dest-interface IF-terminal routing-table called-e164 TAB-DIGCOL-IF-Patton route default dest-interface IF-Patton mapping-table called-e164 to called-e164 FN-RemPrefix map ..(.%)$ to \1 mapping-table called-e164 to calling-e164 FN-COPY-CNPN-TO-CDPN map (.%)$ to \1 interface h323 IF-Patton bind gateway h323 route call dest-interface IF-terminal remote ‹Altigen IP› 1720 use profile voip VOIP-IF-Patton use profile tone-set US interface h323 IF-incom bind gateway h323 route call dest-interface IF-terminal use profile voip VOIP-IF-incom use profile tone-set US interface fxs IF-terminal route call dest-table TAB-DIGCOL-IF-Patton caller-id-presentation mid-ring use profile tone-set US interface fxo IF-FXO no disconnect-signal loop-break ring-number 5 dial-after timeout 1 no use profile tone-set service distribution-group IF-Dist context cs switch no shutdown gateway h323 h323 bind interface eth0 router no shutdown port ethernet 0 0 medium auto encapsulation ip bind interface eth0 router no shutdown port ethernet 0 1 medium auto encapsulation ip bind interface eth1 router no shutdown port fxs 0 0 caller-id format bell use profile fxs us encapsulation cc-fxs bind interface IF-terminal switch no shutdown port fxs 0 1 encapsulation cc-fxs no shutdown
BroadVoice Long Distance FXS or FXO to PBX ( Link to this Config )
	Configuration Description SN4528 R5.X -should also load in any SN452x -ignore errors for missing ports -uncommment lines in SVC-DG-BROADVOICE for ports used) -Replace with telephone number -Replace ‹PASSWORD› with Password -Replace ‹SMARTNODE IP = x.x.x.x› with SmartNode IP -Replace ‹GATEWAY IP = x.x.x.x› with Gatway IP -Replace ‹DNS IP = x.x.x.x› with DNS IP -Uncommment lines in SVC-DG-BROADVOICE for ports used Command Sequence to Copy, Edit, and Paste #----------------------------------------------------------------# # # # SN4528 # # R5.X # # # # -should also load in any SN452x # # -ignore errors for missing ports # # -uncommment lines in SVC-DG-BROADVOICE for ports used) # # -Replace <user-id> with telephone number # # -Replace ‹PASSWORD› with Password # # -Replace ‹SMARTNODE IP = x.x.x.x› with SmartNode IP # # -Replace ‹GATEWAY IP = x.x.x.x› with Gatway IP # # -Replace ‹DNS IP = x.x.x.x› with DNS IP # # -Uncommment lines in SVC-DG-BROADVOICE for ports used # # # # DATE: 2010-06-01 # # AUTHOR: Thomas Tlapak # # PATTON-INALP-NETWORKS; CH-3172 NIEDERWANGEN # # WWW.PATTON.COM WWW.INALP.COM # # # #----------------------------------------------------------------# cli version 3.00 banner "BroadVoice Long Distance FXS or FXO to PBX" dns-client server ‹DNS IP = x.x.x.x› dns-client enabled dns-client cache 150 dns-relay webserver port 80 language en snmp community public ro sntp-client sntp-client server primary 192.43.244.18 port 123 version 4 sntp-client server secondary 63.149.208.50 port 123 version 4 system ic voice 0 profile ppp default profile call-progress-tone US_Dialtone play 1 0 350 -13 440 -13 profile call-progress-tone US_Alertingtone play 1 2000 440 -19 480 -19 pause 2 4000 profile call-progress-tone US_Busytone play 1 500 480 -24 620 -24 pause 2 500 profile tone-set default profile tone-set US map call-progress-tone dial-tone US_Dialtone map call-progress-tone ringback-tone US_Alertingtone map call-progress-tone busy-tone US_Busytone map call-progress-tone release-tone US_Busytone map call-progress-tone congestion-tone US_Busytone profile voip default codec 1 g711alaw64k rx-length 20 tx-length 20 codec 2 g711ulaw64k rx-length 20 tx-length 20 profile voip VOIP-BroadVoice codec 1 g711alaw64k rx-length 20 tx-length 20 codec 2 g711ulaw64k rx-length 20 tx-length 20 no dtmf-relay no dtmf-mute-encoder dejitter-max-delay 200 profile pstn default profile sip default profile authentication default method 1 local method 2 none context ip router interface eth0 ipaddress ‹SMARTNODE IP = x.x.x.x› interface eth1 ipaddress 192.168.1.1 255.255.255.0 context ip router route 0.0.0.0 0.0.0.0 ‹GATWEAY IP = x.x.x.x› 0 context cs switch routing-table called-e164 TAB-SIP-OUTGOING route 011.T dest-interface IF-BroadVoice route 1.......... dest-interface IF-BroadVoice route 240....... dest-interface IF-BroadVoice route 301....... dest-interface IF-BroadVoice route default dest-interface IF-BroadVoice routing-table called-e164 TAB-IN-BROADVOICE route default dest-service SVC-DG-BROADVOICE routing-table called-e164 TAB-IN-PBX route default dest-interface IF-BroadVoice interface sip IF_SIP bind context sip-gateway GW_SIP route call dest-table TAB-IN-BROADVOICE remote proxy.mia.broadvoice.com address-translation outgoing-call from-header user-part fix <user-id> host-part fix sip.broadvoice.com address-translation outgoing-call to-header user-part call host-part domain use profile voip VOIP-BroadVoice interface fxs IF_FXS_00 route call dest-table TAB-SIP-OUTGOING no call-hold no call-waiting caller-id-presentation mid-ring use profile tone-set US interface fxs IF_FXS_01 route call dest-table TAB-SIP-OUTGOING no call-hold no call-waiting caller-id-presentation mid-ring use profile tone-set US interface fxs IF_FXS_02 route call dest-table TAB-SIP-OUTGOING no call-hold no call-waiting caller-id-presentation mid-ring use profile tone-set US interface fxs IF_FXS_03 route call dest-table TAB-SIP-OUTGOING no call-hold no call-waiting caller-id-presentation mid-ring use profile tone-set US interface fxo IF_FXO_00 route call dest-service SVC-DIALTONE disconnect-signal battery-reversal no disconnect-signal loop-break disconnect-signal busy-tone connect-signal battery-reversal ring-number on-caller-id mute-dialing use profile tone-set US interface fxo IF_FXO_01 route call dest-service SVC-DIALTONE disconnect-signal battery-reversal no disconnect-signal loop-break disconnect-signal busy-tone connect-signal battery-reversal ring-number on-caller-id mute-dialing use profile tone-set US interface fxo IF_FXO_02 route call dest-service SVC-DIALTONE disconnect-signal battery-reversal no disconnect-signal loop-break disconnect-signal busy-tone connect-signal battery-reversal ring-number on-caller-id mute-dialing use profile tone-set US interface fxo IF_FXO_03 route call dest-service SVC-DIALTONE disconnect-signal battery-reversal no disconnect-signal loop-break disconnect-signal busy-tone connect-signal battery-reversal ring-number on-caller-id mute-dialing use profile tone-set US service distribution-group SVC-DG-BROADVOICE max-concurrent 1 route call 1 dest-interface IF_FXO_00 # route call 1 dest-interface IF_FXO_01 # route call 1 dest-interface IF_FXO_02 # route call 1 dest-interface IF_FXO_03 service second-dialtone SVC-DIALTONE route call dest-table TAB-SIP-OUTGOING use profile tone-set US context cs switch no shutdown authentication-service AUTH_SVC username <user-id> password <password> location-service LOCATION_SVC domain 1 <IP or Domain> identity <user-id> authentication outbound authenticate 1 authentication-service AUTH_SVC username <user-id> registration outbound registrar 1 <IP or Domain> register auto context sip-gateway GW_SIP interface IF_GW_SIP bind interface eth0 context router port 5062 context sip-gateway GW_SIP bind location-service LOCATION_SVC no shutdown port ethernet 0 0 medium auto encapsulation ip bind interface eth0 router no shutdown port ethernet 0 1 medium auto encapsulation ip bind interface eth1 router no shutdown port fxs 0 0 caller-id format bell use profile fxs us encapsulation cc-fxs bind interface IF_FXS_00 switch no shutdown port fxs 0 1 shutdown port fxs 0 2 shutdown port fxs 0 3 shutdown port fxo 0 0 use profile fxo us caller-id format bell encapsulation cc-fxo bind interface IF_FXO_00 switch no shutdown port fxo 0 1 use profile fxo us caller-id format bell encapsulation cc-fxo bind interface IF_FXO_01 switch no shutdown port fxo 0 2 use profile fxo us caller-id format bell encapsulation cc-fxo bind interface IF_FXO_02 switch no shutdown port fxo 0 3 use profile fxo us caller-id format bell encapsulation cc-fxo bind interface IF_FXO_03 switch no shutdown
Patton Avalanche (H.323 Gatekeeper) ( Link to this Config )
	Configuration Description Patton Avalanche Test Config Command Sequence to Copy, Edit, and Paste cli version 3.00 banner PattonTEST71122 dns-client enabled dns-client cache 30 dns-relay webserver port 80 language en snmp community public ro sntp-client sntp-client server primary 192.5.41.40 port 123 version 4 sntp-client server secondary 192.43.244.18 port 123 version 4 sntp-client local-clock-offset system hostname PattonTEST71122 system location "Gaithersburg, MD" system ic voice 0 profile service-policy rateVoice rate-limit 256 header-length 18 voice-margin 0 source traffic-class local-voice priority source traffic-class local-default priority source traffic-class default profile ppp default profile call-progress-tone US_Dialtone play 1 0 350 -13 440 -13 profile call-progress-tone US_Alertingtone play 1 2000 440 -19 480 -19 pause 2 4000 profile call-progress-tone US_Busytone play 1 500 480 -24 620 -24 pause 2 500 profile tone-set default profile tone-set US map call-progress-tone dial-tone US_Dialtone map call-progress-tone ringback-tone US_Alertingtone map call-progress-tone busy-tone US_Busytone map call-progress-tone release-tone US_Busytone map call-progress-tone congestion-tone US_Busytone profile voip VOIP-IF-VoIP codec 1 g723-6k3 rx-length 30 tx-length 30 codec 2 g711ulaw64k rx-length 20 tx-length 20 codec 3 g729 rx-length 20 tx-length 20 codec 4 g711alaw64k rx-length 20 tx-length 20 fax transmission 1 relay t38-udp fax max-bit-rate 9600 modem transmission 1 bypass g711alaw64k profile pstn default profile sip default profile dhcp-server DHCP-SERVER network 192.168.1.0 255.255.255.0 include 1 192.168.1.10 192.168.1.254 lease 60 days default-router 1 192.168.1.1 domain-name-server 1 192.168.1.1 profile authentication default method 1 local method 2 none context ip router interface eth0 ipaddress dhcp mtu 1480 icmp router-discovery icmp redirect accept tcp adjust-mss rx 760 tcp adjust-mss tx 1380 interface eth1 ipaddress 192.168.1.1 255.255.255.0 icmp router-discovery context ip router dhcp-server use DHCP-SERVER context cs switch routing-table calling-e164 TAB-TO-PHONES route default dest-interface IF-PHONE1 routing-table called-e164 TAB-DIGCOL-TO-VoIP route .T2 dest-interface IF-VoIP interface h323 IF-VoIP bind gateway h323 route call dest-table TAB-TO-PHONES use profile voip VOIP-IF-VoIP use profile tone-set US interface fxs IF-PHONE1 route call dest-table TAB-DIGCOL-TO-VoIP caller-id-presentation mid-ring subscriber-number 71005 use profile tone-set US context cs switch no shutdown gateway h323 h323 alias h323-id PattonTEST71122 faststart ras gatekeeper-discovery manual 209.49.110.207 1719 PattonAVIGK bind interface eth0 router no shutdown gateway sip sip shutdown port ethernet 0 0 medium auto encapsulation ip bind interface eth0 router no shutdown port ethernet 0 1 medium auto encapsulation ip bind interface eth1 router no shutdown port fxs 0 0 caller-id format bell use profile fxs us encapsulation cc-fxs bind interface IF-PHONE1 switch no shutdown
PSTN and Terminal Gateway for e-Phone applications ( Link to this Config )
	Configuration Description SmartNode used as a PSTN/ISDN and terminal gateway in a Media-Streams e-Phone enviroment. Command Sequence to Copy, Edit, and Paste #----------------------------------------------------------------# # # # SN1x00 # # R3.10 # # # # e-Phone PSTN and Terminal Gateway # # Connect the PSTN to port BRI0 and the Phones to Port BRI1. # # Connect the LAN to port ETH1 and if a WAN is available # # connect it to port ETHO # # # # The configuration contains variables in brackets. # # Please read all the comments and replace the variables # # including the brackets. # # Unused lines or variables have to be commented out or deleted # # # # Date: 2005-03-01 # # Author: MZ, PATTON-INALP NETWORKS, CH-3172 Niederwangen # # WWW.PATTON.COM SUPPORT@INALP.COM # # # #----------------------------------------------------------------# cli version 3.00 banner "WELCOME\nPLEASE LOG ON" # Set the Domain-Name-Server IP-Addresses dns-client server <IP-Address of primary DNS-server> dns-client server <IP-Address of secondary DNS-server> dns-client enabled dns-client cache 100 webserver port 80 language en # Set the time server IP-Addresses sntp-client server primary <primary time server IP-Address> port 123 version 4 sntp-client server secondary <secondary time server IP-Address> port 123 version 4 sntp-client operating-mode unicast sntp-client anycast-address 224.0.1.1 port 123 sntp-client local-port 123 # Set the System-Hostname system hostname <hostname> profile napt test profile ppp default profile tone-set default map call-progress-tone dial-tone defaultBusytone profile voip default codec 1 g711alaw64k rx-length 20 tx-length 20 profile pstn default profile authentication default method 1 local method 2 none context ip router # Set WAN-IP-Address and IP-Mask interface IF_WAN ipaddress <global IP-Address> <global IP-Mask> icmp router-discovery # Set LAN-IP-Address and IP-Mask interface IF_LAN ipaddress <local IP-Address> <local IP-Mask> icmp router-discovery # Set the default Gateway IP-Address context ip router route 0.0.0.0 0.0.0.0 <default Gateway IP-Address> 0 context cs switch routing-table called-e164 RT_CDPN_FROM_TERMINAL route T dest-interface IF_SIP_EPHONE MT_ADD_PREF routing-table called-e164 RT_CDPN_2_SIP route default dest-interface IF_SIP_EPHONE MT_CDPN_TRUNC routing-table called-e164 RT_CDPN_2_ISDN route default dest-interface IF_SO_TERMINAL route 5* dest-interface IF_S0_PSTN MT_REM_PREF mapping-table called-e164 to called-e164 MT_CDPN_ADD_PREF map 0(.%) to 5*\1 mapping-table called-e164 to called-e164 MT_CDPN_REM_PREF map 5*(.%) to \1 # Each dot within the brackets refers to one digit. # The number of dot within the brackets depends on your mumbering plan mapping-table called-e164 to called-e164 MT_CDPN_TRUNC map .%(...) to \1 interface isdn IF_SO_TERMINAL route call dest-table RT_CDPN_FROM_TERMINAL isdn-date-time interface isdn IF_S0_PSTN route call dest-table RT_CDPN_2_SIP use profile pstn default interface sip IF_SIP_EPHONE bind gateway sip route call dest-table RT_CDPN_2_ISDN early-disconnect address-translation incoming-call called-e164 to-header gateway sip sip # If the addresses are defined as an URL, # one or two domanin name servers have to be configured. # (See top of the configuration file.) # set SIP server addresses domain <URL or IP-ADDRESS of the domain> default-server <URL or IP-ADDRESS of the default server> 5060 loose-router registrar <URL or IP-ADDRESS of the default server> 5060 registration-lifetime 120 user 5* display-name PstnGateway phone-context 5* # Define the terminal user settings user <phone number of phone 1> display-name "<phone number and/or name of phone 1>" phone-context "" user <phone number of phone 2> display-name "<phone number and/or name of phone 2>" phone-context "" user <phone number of phone x> display-name "<phone number and/or name of phone x>" phone-context "" transport tcp bind interface IF_LAN router no shutdown port ethernet 0 0 medium 10 half encapsulation ip bind interface IF_WAN router no shutdown port ethernet 0 1 medium 10 half encapsulation ip bind interface IF_LAN router no shutdown port bri 0 0 clock auto encapsulation q921 q921 permanent-layer2 # set the ISDN Layer2 protocol protocol <pp|pmp> uni-side user encapsulation q931 q931 protocol dss1 uni-side user encapsulation cc-isdn bind interface IF_SO_PSTN switch port bri 0 0 no shutdown port bri 0 1 clock auto encapsulation q921 q921 protocol pmp uni-side auto encapsulation q931 q931 protocol dss1 uni-side net encapsulation cc-isdn bind interface IF_SO_TERMINAL switch port bri 0 1 no shutdown

Enable SmartNode as H.323 Gateway with GateKeeper ( Link to this Config )

Configuration Description The command sequence below enables SmartNode to operate as an H.323 gateway for registration with a GateKeeper. NOTE: You must replace. . . xxx.xxx.xxx.xxx . . . with your GateKeeper's IP address. Command Sequence to Copy, Edit, and Paste configure context cs gateway h323 h323 alias h323-id <h323-id> faststart ras gatekeeper-discovery manual xxx.xxx.xxx.xxx 1720 <gatekeeper-name> bind interface eth0 router no shutdown

ISDN Trunk Gateway configuration  ( Link to this Config )
	Configuration Description SN46XX SIP to ISDN Trunk Gateway configuration without SIP registration. The ISDN PSTN may be connected to Port BRI 0/0 to 0/X. Copy the content below this introduction and save it to a text file for future upload to the SmartNode. Command Sequence to Copy, Edit, and Paste #----------------------------------------------------------------# # # # SN46XX/3BIS/UI # # R5.X # # 2011-06-15 # # IP-PBX Gateway, SIP Trunk to ISDN PSTN # # Default configuration # # Author: MZ; Rev;0 # # # #----------------------------------------------------------------# cli version 3.20 clock local default-offset +01:00 clock local dst-rule DST_SUMMER +02:00 from 02:00 mar last sunday until 03:00 oct last sunday dns-client server 8.8.8.8 dns-client cache 100 dns-relay webserver port 80 language en sntp-client sntp-client server primary swisstime.ee.ethz.ch port 123 version 4 sntp-client poll-interval 600 system hostname TRUNK_GATEWAY system ic voice 0 low-bitrate-codec g729 system clock-source 1 bri 0 0 clock-source 2 bri 0 1 clock-source 3 bri 0 2 profile napt NAPT_LAN profile voip default codec 1 g711alaw64k rx-length 20 tx-length 20 codec 2 g711ulaw64k rx-length 20 tx-length 20 codec 3 g726-32k rx-length 20 tx-length 20 fax transmission 1 relay t38-udp fax redundancy low-speed 2 high-speed 1 fax bypass-method nse modem transmission 1 bypass g711alaw64k modem bypass-method nse profile dhcp-server DHCP_MGT network 192.168.100.0 255.255.255.0 include 1 192.168.100.100 192.168.100.110 lease 1 days default-router 1 192.168.100.1 domain-name-server 1 192.168.100.1 context ip router interface IF_IP_LAN ipaddress dhcp interface IF_IP_MGT ipaddress 192.168.100.1 255.255.255.0 context ip router dhcp-server use profile DHCP_MGT context cs switch national-prefix 0 international-prefix 00 routing-table called-e164 RT_CDPN_SIP route default dest-interface IF_SIP_TRUNK MAP_NL mapping-table called-e164 to called-e164 MAP_NL map .%(.)$ to \1 interface isdn IF_ISDN_S00 route call dest-table RT_CDPN_SIP interface isdn IF_ISDN_S01 route call dest-table RT_CDPN_SIP interface isdn IF_ISDN_S02 route call dest-table RT_CDPN_SIP interface isdn IF_ISDN_S03 route call dest-table RT_CDPN_SIP interface isdn IF_ISDN_S04 route call dest-table RT_CDPN_SIP interface sip IF_SIP_TRUNK bind context sip-gateway SIP_GW route call dest-service HG_PSTN_TRUNK service hunt-group HG_PSTN_TRUNK drop-cause normal-unspecified drop-cause no-circuit-channel-available drop-cause network-out-of-order drop-cause temporary-failure drop-cause switching-equipment-congestion drop-cause access-info-discarded drop-cause circuit-channel-not-available drop-cause resources-unavailable route call 1 dest-interface IF_ISDN_S00 route call 2 dest-interface IF_ISDN_S01 route call 3 dest-interface IF_ISDN_S02 route call 3 dest-interface IF_ISDN_S03 route call 3 dest-interface IF_ISDN_S04 context cs switch no shutdown context sip-gateway SIP_GW interface IF_SIP_GW bind interface IF_IP_LAN context router port 5060 context sip-gateway SIP_GW no shutdown port ethernet 0 0 medium auto encapsulation ip bind interface IF_IP_LAN router no shutdown port ethernet 0 1 medium auto encapsulation ip bind interface IF_IP_MGT router no shutdown port bri 0 0 clock auto encapsulation q921 q921 uni-side auto encapsulation q931 q931 protocol dss1 uni-side user encapsulation cc-isdn bind interface IF_ISDN_S00 switch port bri 0 0 no shutdown port bri 0 1 clock auto encapsulation q921 q921 uni-side auto encapsulation q931 q931 protocol dss1 uni-side user encapsulation cc-isdn bind interface IF_ISDN_S01 switch port bri 0 1 no shutdown port bri 0 2 clock auto power-feed encapsulation q921 q921 uni-side auto encapsulation q931 q931 protocol dss1 uni-side user encapsulation cc-isdn bind interface IF_ISDN_S02 switch port bri 0 2 no shutdown port bri 0 3 clock auto power-feed encapsulation q921 q921 uni-side auto encapsulation q931 q931 protocol dss1 uni-side user encapsulation cc-isdn bind interface IF_ISDN_S03 switch port bri 0 3 no shutdown port bri 0 4 clock auto power-feed encapsulation q921 q921 uni-side auto encapsulation q931 q931 protocol dss1 uni-side user encapsulation cc-isdn bind interface IF_ISDN_S04 switch port bri 0 4 no shutdown
PSTN trunk gateway for Asterisk IP-PBX ( Link to this Config )
	Configuration Description PSTN trunk gateway for Aserisk IP-PBX application Command Sequence to Copy, Edit, and Paste #----------------------------------------------------------------# # # # Patton SmartNode SN46XX # # SWR5.X # # # # PSTN trunk gateway for Aserisk IP-PBX application # # # # DATE: 2010-06-01 # # AUTHOR: Thomas Tlapak # # PATTON-INALP-NETWORKS; CH-3172 NIEDERWANGEN # # WWW.PATTON.COM WWW.INALP.COM # # # #----------------------------------------------------------------# cli version 3.20 clock local default-offset +02:00 dns-relay webserver port 80 language en sntp-client sntp-client server primary 129.132.2.21 port 123 version 4 system ic voice 0 low-bitrate-codec g729 system clock-source 1 bri 0 0 profile ppp default profile tone-set default profile voip default codec 1 g711alaw64k rx-length 20 tx-length 20 codec 2 g711ulaw64k rx-length 20 tx-length 20 profile sip default no autonomous-transitioning profile dhcp-server DHCPS_LAN network 192.168.1.0 255.255.255.0 include 1 192.168.1.10 192.168.1.99 lease 2 hours default-router 1 192.168.1.1 domain-name-server 1 192.168.1.1 profile aaa default method 1 local method 2 none context ip router interface IF_WAN ipaddress dhcp tcp adjust-mss rx mtu tcp adjust-mss tx mtu interface IF_LAN ipaddress 192.168.1.1 255.255.255.0 tcp adjust-mss rx mtu tcp adjust-mss tx mtu context ip router dhcp-server use profile DHCPS_LAN context cs switch routing-table called-e164 RT_IN route .T2 dest-interface IF_SIP interface isdn IF_ISDN_00 route call dest-table RT_IN isdn-date-time interface isdn IF_ISDN_01 route call dest-table RT_IN isdn-date-time interface isdn IF_ISDN_02 route call dest-table RT_IN isdn-date-time interface isdn IF_ISDN_03 route call dest-table RT_IN isdn-date-time interface sip IF_SIP bind context sip-gateway GW_SIP route call dest-service SER_HUNT_PSTN # enter the remote IP of Asterisk PBX below remote <remote IP> privacy service hunt-group SER_HUNT_PSTN cyclic drop-cause normal-unspecified drop-cause no-circuit-channel-available drop-cause network-out-of-order drop-cause temporary-failure drop-cause switching-equipment-congestion drop-cause access-info-discarded drop-cause circuit-channel-not-available drop-cause resources-unavailable route call 1 dest-interface IF_ISDN_00 route call 2 dest-interface IF_ISDN_01 route call 3 dest-interface IF_ISDN_02 route call 4 dest-interface IF_ISDN_03 context cs switch no shutdown context sip-gateway GW_SIP interface IF_GWSIP bind interface IF_WAN context router port 5060 context sip-gateway GW_SIP no shutdown port ethernet 0 0 medium auto encapsulation ip bind interface IF_WAN router no shutdown port ethernet 0 1 medium auto encapsulation ip bind interface IF_LAN router no shutdown port bri 0 0 clock auto power-feed encapsulation q921 q921 uni-side auto encapsulation q931 q931 protocol dss1 uni-side user encapsulation cc-isdn bind interface IF_ISDN_00 switch port bri 0 0 no shutdown port bri 0 1 clock auto encapsulation q921 q921 uni-side auto encapsulation q931 q931 protocol dss1 uni-side user encapsulation cc-isdn bind interface IF_ISDN_01 port bri 0 1 no shutdown port bri 0 2 clock auto encapsulation q921 q921 uni-side auto encapsulation q931 q931 protocol dss1 uni-side user encapsulation cc-isdn bind interface IF_ISDN_02 port bri 0 2 no shutdown port bri 0 3 clock auto encapsulation q921 q921 uni-side auto encapsulation q931 q931 protocol dss1 uni-side user encapsulation cc-isdn bind interface IF_ISDN_03 port bri 0 3 no shutdown

SmartNode in front of a firewall ( Link to this Config )
	Configuration Description In case you need to put the SmartNode in front of a firewall you can forward every ip-packet which is not handled by SmartNode to the firewall. Command Sequence to Copy, Edit, and Paste profile napt <profile-name> dmz-host <ip-address of firewall>
Dynamic NAPT ( Link to this Config )
	Configuration Description The command sequence below will create a NAPT profile and apply it to the SmartNode's global IP Interface. A home office with the following characteristics will typically use Dynamic NAPT configuration: - a single global IP address (whether static or dynamic) - multiple host workstations (PCs, laptops) connected to the home LAN Command Sequence to Copy, Edit, and Paste #create NAPT profile profile napt PF-NAPT1 #apply profile to global IP interface context ip router interface eth0 use profile napt PF-NAPT1
Dynamic NAPT with static Port Mappings ( Link to this Config )
	Configuration Description The command sequence below will create a NAPT profile with port mappings and apply it to the SmartNode's global IP Interface. This configuration allows SmartNode to forward incoming IP connections on specified ports of the global IP Interface to a private host behind NAPT. NOTE: To determine which ports need to be mapped, please refer to the documentation for the applications you are running behind NAPT. Command Sequence to Copy, Edit, and Paste profile napt PF-NAPT1 static tcp 192.168.1.10 20 static udp 192.168.1.10 20 static tcp 192.168.1.10 21 static udp 192.168.1.10 21 static tcp 192.168.1.11 80 static udp 192.168.1.11 80 context ip router interface eth0 use profile napt PF-NAPT1
Static NAT Address Mapping ( Link to this Config )
	Configuration Description The command sequence below will create a NAT profile with static address mappings and apply it to an IP interface on the SmartNode. This configuration allows SmartNode to translate IP adresses one-to-one for incoming and outgoing packets. When SmartNode is configured for NAT, the defined port numbers remain unchanged. Note that he mapped IP addresses must be reachable on their respective interfaces. Command Sequence to Copy, Edit, and Paste profile napt PF-NAT2 static [LOCAL-IP-1] [GLOBAL-IP-1] static [LOCAL-IP-2] [GLOBAL-IP-2] static [LOCAL-IP-3] [GLOBAL-IP-3] context ip router interface eth0 ipaddress dhcp use profile napt PF-NAPT2

VLAN 802.1q on an Ethernet port ( Link to this Config )
	Configuration Description The configuration example below shows how to configure 802.1q Vlan's on a SmartNode Ethernet port. Command Sequence to Copy, Edit, and Paste context ip interface IF_IP_VLAN1 ipaddress 10.0.0.1 225.225.0.0 interface IF_IP_VLAN2 ipaddress 10.1.0.1 225.225.0.0 port ethernet 0 0 medium auto encapsulation ip no shutdown vlan 1 bind interface IF_IP_VLAN1 router no shutdown vlan 2 bind interface IF_IP_VLAN2 router no shutdown
SN465X G.SHDSL connection with PPPoE ( Link to this Config )
	Configuration Description Port and context ip configuration for a PPPoE connection over the SmartNodes G.SHDSL. Command Sequence to Copy, Edit, and Paste context ip router interface IF_WAN ipaddress unnumbered mtu 1492 point-to-point use profile service-policy SP_WAN_IN in use profile service-policy SP_WAN_OUT out tcp adjust-mss rx 1440 tcp adjust-mss tx 1440 context ip router dhcp-server use DHCP_LAN route 0.0.0.0 0.0.0.0 IF_WAN 0 source-address-map 0.0.0.0 0.0.0.0 IF_LAN subscriber ppp SUB_ADSL dial out authentication chap authentication pap identification outbound username password passwort bind interface IF_WAN router port dsl 0 0 pvc vpi 8 vci 35 pppoe session SES_ADSL use profile ppp default bind subscriber SUB_ADSL no shutdown
SN ADSL Port configuration ( Link to this Config )
	Configuration Description Port and context ip configuration for a ADSL link over a SmartNode VoIP-Gateway. Command Sequence to Copy, Edit, and Paste #----------------------------------------------------------------# # # # SN465X / SN485X ADSL-VOIP-GATEWAY # # # # Sample-Configuration, including the IP-Router # # # # Please replace all variables before you are using this # # configuration. # # # # Author: MZ, Patton-Inalp Netowrks, CH-3172 Niederwangen # # Date: 2006-02-23 # # # #----------------------------------------------------------------# profile napt NAPT_WAN profile ppp default context ip router interface IF_WAN ipaddress <set your ip-addres and ip-mask or use unnumbered> mtu 700 point-point use profile napt NAPT_WAN tcp adjust-mss rx 660 tcp adjust-mss tx 660 context ip router route 0.0.0.0 0.0.0.0 IF_WAN 0 subscriber ppp SUB_ADSL_1 dial out authentication chap identification outbound <ADSL-Username> password <ADSL-Password> bind interface IF_WAN router port dsl 0 0 pvc vpi <set your vpi number> vci <set your vci number> pppoe session SES_ADSL bind subscriber SUB_ADSL_1 no shutdown
FXS ( Link to this Config )
	Configuration Description FXS Port configuration. To be used in the US. For Europe usage, just replace "bell" with "etsi". Command Sequence to Copy, Edit, and Paste port fxs 0 0 caller-id format bell use profile fxs us encapsulation cc-fxs bind interface IF_FXS_00 switch no shutdown port fxs 0 1 caller-id format bell use profile fxs us encapsulation cc-fxs bind interface IF_FXS_01 switch no shutdown port fxs 0 2 caller-id format bell use profile fxs us encapsulation cc-fxs bind interface IF_FXS_02 switch no shutdown port fxs 0 3 caller-id format bell use profile fxs us encapsulation cc-fxs bind interface IF_FXS_03 switch no shutdown
FXO ( Link to this Config )
	Configuration Description FXO Port configuration. To be used in the US. For Europe usage, just replace "bell" with "etsi". Command Sequence to Copy, Edit, and Paste port fxo 0 0 use profile fxo us caller-id format bell encapsulation cc-fxo bind interface IF_FX0_00 switch no shutdown port fxo 0 1 use profile fxo us encapsulation cc-fxo bind interface IF_FX0_01 switch no shutdown port fxo 0 2 use profile fxo us encapsulation cc-fxo bind interface IF_FX0_02 switch no shutdown port fxo 0 3 use profile fxo us encapsulation cc-fxo bind interface IF_FX0_03 switch no shutdown
E1 Configuration ( Link to this Config )
	Configuration Description SN4960 E1 Configuration with DSS1 signaling protocol Sample Configuration; port 0 0 may be connected to a PSTN line port 0 1 may be connected to a ISDN E1 PBX Command Sequence to Copy, Edit, and Paste system clock-source 1 e1t1 0 0 port e1t1 0 0 port-type e1 clock auto framing crc4 encapsulation q921 q921 uni-side auto encapsulation q931 q931 protocol dss1 uni-side user bchan-number-order ascending encapsulation cc-isdn bind interface IF_ISDN_00 switch port e1t1 0 0 no shutdown port e1t1 0 1 port-type e1 clock auto framing crc4 encapsulation q921 q921 uni-side auto encapsulation q931 q931 protocol dss1 uni-side net bchan-number-order ascending encapsulation cc-isdn bind interface IF_ISDN_01 switch port e1t1 0 1 no shutdown
Frame-Relay over Serial X.21/V.35 ( Link to this Config )
	Configuration Description Port and context ip configuration for a Frame-Relay connection over a V.35 or X21 serial link. The transmit-data-on-edge setting depends on the remote modem device Command Sequence to Copy, Edit, and Paste context ip interface IF_WAN ipaddress <unnumbered | ip-address ip-mask | dhcp> point-point mtu 1500 context ip route 0.0.0.0 0.0.0.0 IF_WAN port serial 0 0 encapsulation framerelay hardware-port <x21 | v35> transmit-data-on-edge <positive | negativ> no shutdown framerelay lmi-type <ansi | itu | gov> keepalive 10 pvc 100 encapsulation rfc1490 bind interface IF_WAN router fragment 80 no shutdown
T1 Configuration NETWORK ( Link to this Config )
	Configuration Description 2300/2400 T1 Slot 3 0 Network-Side Command Sequence to Copy, Edit, and Paste port t1 1 0 no shutdown port t1 3 0 clock master encapsulation q921 q921 protocol pp uni-side net encapsulation q931 q931 protocol ni2 uni-side net encapsulation cc-isdn bind interface IF-PBX switch port t1 3 0 no shutdown
T1 Configuration USER ( Link to this Config )
	Configuration Description 2300/2400 T1 Card in Slot 3 0 User-Side DD Command Sequence to Copy, Edit, and Paste configure system clock-source internal exit context cs switch routing-table called-e164 TAB-FROM-SWITCH interface isdn IF-SWITCH route call dest-table TAB-FROM-SWITCH exit port t1 3 0 clock slave encapsulation q921 q921 protocol pp uni-side user encapsulation q931 q931 protocol ni2 uni-side user encapsulation cc-isdn bind interface IF-SWITCH switch port t1 3 0 no shutdown
PPP over Serial X.21/ V.35 ( Link to this Config )
	Configuration Description Port and context ip configuration for a PPP connection over a V.35 or X21 serial link. The transmit-data-on-edge setting depends on the remote modem device. Command Sequence to Copy, Edit, and Paste context ip interface IF_WAN ipaddress unnumbered | ip-address ip-mask | dhcp point-point mtu 1500 context ip route 0.0.0.0 0.0.0.0 IF_WAN port serial 0 0 encapsulation ppp hardware-port < x21 | v35 > transmit-data-on-edge <positive |negativ> bind interface IF_WAN router no shutdown
PPPoE ( Link to this Config )
	Configuration Description A standard PPPoE configuration including authentication. # Replace ‹account› and ‹password› Command Sequence to Copy, Edit, and Paste configure profile napt PF_NAPT_WAN port ethernet 0 0 encapulation pppoe context ip interface IF_WAN ipaddress unnumbered use profile napt PF_NAPT_WAN point-to-point mtu 1480 tcp adjust-mss rx mtu tcp adjust-mss tx mtu context ip router route 0.0.0.0 0.0.0.0 IF_WAN 0 subscriber ppp PPP-SUBSCRIBER dial out authentication chap pap identification outbound ‹account› password ‹password› bind interface IF_WAN router port ethernet 0 0 encapsulation pppoe | multi pppoe session PPPoE-Session bind subscriber PPP-SUBSCRIBER no shutdown port ethernet 0 0 no shutdown

Auto Provisioning with Timers and System MAC ( Link to this Config )

Configuration Description This configuration checks 2 minutes after boot up for new firmware and if available downloads and istalls it. Furthermore the server will be checked every minute for new firmware. Furthermore SmartNode checks 30 minutes after bootup and then every minute for a new configuration on the specified server. Should a newer configuration be available it will be download, installed and the SmartNode rebooted to activate the new configuration. Note: Reload takes only effect as long as there is no phone call ongoing. # Replace all <>-variables with appropriate IP-address or dns-name Command Sequence to Copy, Edit, and Paste timer FIRMWARE-UPDATE now + 60 minutes every minute "provisioning execute FIRMWARE" timer CONFIG-UPDATE now + 30 minutes every minute "provisioning execute CONFIG" profile provisioning CONFIG destination configuration location 1 tftp://<ip-address or dns-name>/<path if needed>/$(system.mac).cfg activation reload graceful profile provisioning FIRMWARE destination script location 1 tftp://<ip-address or dns-name>/<path if needed>/bw activation reload graceful

Voice Priority ( Link to this Config )

Configuration Description Quality-of-Service needs to be configured carefully. Especially the rate-limit values needs to be exact as this is the base for prioritizing the data/voice traffic. All the values in brackets <> needs to be replaced. Please have a look in the user-guide in order to see what exactly is possible to be configured in this profile... Note: The voice-margin MUST NOT be configured in the outgoing profile Command Sequence to Copy, Edit, and Paste profile service-policy PROF_QOS_DOWNSTREAM rate-limit <downstream-speed-in-bps> voice-margin <1/3 of the rate-limit> source traffic-class local-voice priority profile service-policy PROF_QOS_UPSTREAM rate-limit <upstream-speed-in-bps> source traffic-class local-voice priority context ip interface IF_WAN use profile service-policy PROF_QOS_DOWNSTREAM in use profile service-policy PROF_QOS_UPSTREAM out

General example configuration for INBOUND registrations in firmware 5.X ( Link to this Config )
	Configuration Description The firmware 5.X supports the registrar function what allows SmartNode to accept registration from any kind of SIP-clients (telephones, soft-clients). The example configuration shows how such an so called identity needs to be configured. Please refer to the configuration guide for a complete overview of what is possible. The example below does only show how to configure the SmartNode in order to accept inbound-registrations and how to route a call to such a registered client. Note: To register more than 3 SIP users, the SIP-registrar license needs to be installed. To get such a license please contact our sales-departement. Command Sequence to Copy, Edit, and Paste context cs routing-table called-e164 RT_2_SIP route default dest-service SER_SIP_LOCATION interface sip IF_SIP bind context sip-gateway GW_SIP route call dest-<interface | table | service> <call-routing destination> local <domain or IP of SmartNode> service sip-location-service SER_SIP_LOCATION bind location-service SER_LOCATION context cs switch no shutdown authentication-service SER_AUTH realm <enter any value> username <username> password <password> location-service SER_LOCATION domain 1 <domain-name> identity <identity> alias name <define an alias if needed> authentication inbound authenticate 1 authentication-service SER_AUTH username <username> registration inbound context sip-gateway GW_SIP interface INBOUND bind interface <ethernet-interface on where the registration will arrive> context router port 5060 context sip-gateway SIP bind location-service SER_LOCATION no shutdown
General example configuration for OUTBOUND registrations and authentication for a SIP trunk firmware 5.X ( Link to this Config )
	Configuration Description This configuration examples may be used in applications, where the gateway has to use the SIP-Username, e.g. root calling number, for registration and outbound authentication but the content of the SIP-From-Header may change; e.g. root calling number together with the extension. Command Sequence to Copy, Edit, and Paste #---------------------------------------------------------------- # # SmartWare release 5.X # All SmartNode Products # # This configuration examples may be used in applications, # where the gateway has to use the SIP-Username, # e.g. root calling number, for registration and # outbound authentication but the content of the # SIP-From-Header may change; # e.g. root calling number together with the extension. # # The configuration contains variables in brackets. # Please read all the comments and replace the variables # including the brackets. # Unused lines or variables have to be commented out or deleted # # Date: 2008-04-25 # Author: MZ, PATTON-INALP NETWORKS, CH-3172 Niederwangen # WWW.PATTON.COM SUPPORT@INALP.COM # #---------------------------------------------------------------- authentication-service AS_DEFAULT username <SIP-Username> password <password> location-service SER_LOC_DEFAULT domain 1 <domain-name> # Identity Group for outbound call authentication identity-group default authentication outbound authenticate 1 authentication-service AS_DEFAULT username <SIP-Username> # identity for outbound identity <SIP account name> authentication outbound authenticate 1 authentication-service AS_DEFAULT registration outbound registrar <URL or IP-address of registrar server> lifetime 300 register auto
General example configuration for OUTBOUND registrations in firmware 5.X ( Link to this Config )
	Configuration Description This configuration shows an example for how to route calls beetwen an ISDN and SIP network including registration on a SIP server using the new SIP gateway configuration architecture within the SWR5.T software version. Command Sequence to Copy, Edit, and Paste #----------------------------------------------------------------# # # SN411X / SN452X / SN453X / SN455X / SN463X / SN483X / SN4960 / # R 5 SIP / H323 # # NEW SIP GATEWAY CONFIGURATION INCLUDING CONTEXT CS # # After replacing all variables you may copy and paste this # configuration part into your offline configuration file. # # This configuration shows an example for how to route calls # beetwen an ISDN and SIP network including registration on a SIP # server using the new SIP gateway configuration architecture # within the SWR5.T software version. # # DATE: 2007-10-31 # AUTHOR: Gion Padrun; # PATTON-INALP-NETWORKS; CH-3172 NIEDERWANGEN # WWW.PATTON.COM WWW.INALP.COM # # #----------------------------------------------------------------# context cs switch routing-table called-e164 RT_TO_SIP route .T dest-interface IF_SIP interface isdn IF_ISDN route call dest-table RT_TO_SIP interface sip IF_SIP route call dest-interface IF_ISDN remote <remote domain-name or ip-address> local <remote domain-name or ip-address> bind context sip-gateway GW_SIP context cs switch no shutdown authentication-service SER_AUTH_OB username <authentication username> password <authentication password> location-service SER_LOC_CERT domain 1 <sip domain> identity <your sip identity> authentication outbound authenticate 1 authentication-service SER_AUTH_OB username <authentication username> registration outbound registrar <URL or IP-address of registrar server> lifetime 300 register auto context sip-gateway GW_SIP interface IF_GW_WAN bind interface <bind interface name> context router context sip-gateway GW_SIP bind location-service SER_LOC_CERT no shutdown
General example configuration for SIP-gateway configuration in firmware 3.2x / 4.x ( Link to this Config )
	Configuration Description Below you will find different configuration examples for a SIP-Proxy configuration. The type of the proxy does not matter as the configuration is the same. Please note the configuration is different depending of what firmware you use. In the examples below there are some variables what need to be adapted to fit into your requirements. The variables and their meaning are: # Replace ‹x.x.x.x› with IP address of registrar, e.g. 192.168.1.1 # Replace ‹user-account› with account, e.g., 1234 # Replace ‹user-password› with account password, e.g., mysecret # Replace ‹network-interface› with the corresponding network interface # Replace ‹interface-type› with fxs or isdn depending of the type of your SmartNode Command Sequence to Copy, Edit, and Paste Example for firmware 3.20 ************************* profile voip default codec g729 rx-length 20 tx-length 20 codec g711ulaw64k rx-length 20 tx-length 20 codec g711alaw64k rx-length 20 tx-length 20 context cs routing-table called-e164 RT_2_ASTERISK route .T dest-interface IF_SIP_ASTERISK interface ‹interface-type› IF_TELEPHONE route call dest-table RT_2_ASTERISK interface sip IF_SIP_ASTERISK bind gateway GW_SIP_ASTERISK service default route call dest-interface IF_TELEPHONE address-translation outgoing-call from-header user-part fix ‹user-account› host-part call gateway sip GW_SIP_ASTERISK bind interface ‹network-interface› router service default domain ‹domain-name-or-ip-of-Asterisk› defaultserver ‹domain-name-or-ip-of-Asterisk› registration ‹domain-name-or-ip-of-Asterisk› user ‹user-account› password ‹user-password› gateway sip GW_SIP_ASTERISK no shutdown ************************************************** Example for firmware 3.21, 4.1 and 4.2 ************************************************** profile voip default codec g729 rx-length 20 tx-length 20 codec g711ulaw64k rx-length 20 tx-length 20 codec g711alaw64k rx-length 20 tx-length 20 context cs routing-table called-e164 RT_2_ASTERISK route .T dest-interface IF_SIP_ASTERISK interface ‹interface-type› IF_TELEPHONE route call dest-table RT_2_ASTERISK interface sip IF_SIP_ASTERISK bind gateway GW_SIP_ASTERISK service default route call dest-interface IF_TELEPHONE address-translation outgoing-call from-header user-part fix ‹user-account› host-part call gateway sip GW_SIP_ASTERISK bind interface ‹network-interface› router service default domain ‹domain-name-or-ip-of-Asterisk› defaultserver manual ‹domain-name-or-ip-of-Asterisk› registration manual ‹domain-name-or-ip-of-Asterisk› user ‹user-account› authenticate password ‹user-password› register gateway sip GW_SIP_ASTERISK no shutdown
Example configuration of SIP-gateway with T1 ( Link to this Config )
	Configuration Description A Simple Asterisk Registration into PBX T1 on Port 0 0 # # Replace ‹0.0.0.0› with IP address or dns name of the registrar, e.g. 192.168.1.1 # Replace ‹XXX› with account, e.g., 1234 # Replace ‹YYY› with account password, e.g., mysecret # Replace < ethernet-interface > with the ethernet interface you use # # Example configuration is based on firmware 5.X, for SN4940, SN4950 and SN4960 Command Sequence to Copy, Edit, and Paste profile voip VOIP-ASTERISK codec 1 g729 rx-length 20 tx-length 20 codec 2 g711ulaw64k rx-length 20 tx-length 20 codec 3 g711alaw64k rx-length 20 tx-length 20 context cs switch routing-table called-e164 RT_PSTN_TO_SIP route .T dest-interface IF_SIP interface isdn IF_ISDN_00 route call dest-table RT_PSTN_TO_SIP interface sip IF_SIP bind context sip-gateway GW_SIP # Enter below the SIP server / Softswitch IP address or FQDN remote <0.0.0.0> route call dest-interface IF_ISDN_00 # Enter below the SIP username, to be used as user part in from header address-translation outgoing-call from-header user-part fix <username> host-part call context cs switch no shutdown authentication-service AUTH_SVC # Enter below the username and passwort you got by VOIP provider username [XXX] password [YYY] location-service LOCATION_SVC # The line below requires the VOIP Providers domain name domain 1 [registrar IP or domain] # Set the SIP username on the line below identity <username> authentication outbound # Replace the XXX with the username of Authentication service authenticate 1 authentication-service AUTH_SVC username <username> registration outbound # Set the Registrar IP or domainname of VOIP provider registrar 1 <IP or Domain> register auto context sip-gateway GW_SIP interface IF_GWSIP bind interface WAN context router port 5060 context sip-gateway GW_SIP bind location-service LOCATION_SVC no shutdown port t1 0 0 clock auto encapsulation q921 q921 protocol pp uni-side auto encapsulation q931 q931 protocol ni2 uni-side user encapsulation cc-isdn bind interface IF_ISDN_00 switch port t1 0 0 no shutdown
Example configuration of SIP-gateway with E1 ( Link to this Config )
	Configuration Description A Simple Asterisk Registration into PBX E1 on Port 0 0 # # Replace ‹0.0.0.0› with IP address or dns name of the registrar, e.g. 192.168.1.1 # Replace ‹XXX› with account, e.g., 1234 # Replace ‹YYY› with account password, e.g., mysecret # Replace < ethernet-interface > with the ethernet interface you use # # Example configuration is based on firmware 5.X for a SN4960, SN4940 and SN4950 Command Sequence to Copy, Edit, and Paste profile voip default codec 1 g729 rx-length 20 tx-length 20 codec 2 g711ulaw64k rx-length 20 tx-length 20 codec 3 g711alaw64k rx-length 20 tx-length 20 context cs switch routing-table called-e164 RT_PSTN_TO_SIP route .T dest-interface IF_SIP interface isdn IF_ISDN_00 route call dest-table RT_PSTN_TO_SIP interface sip IF_SIP bind context sip-gateway GW_SIP # Enter below the SIP server / Softswitch IP address or FQDN remote <0.0.0.0> route call dest-interface IF_ISDN_00 # Enter below the SIP username, to be used as user part in from header address-translation outgoing-call from-header user-part fix <username> host-part call context cs switch no shutdown authentication-service AUTH_SVC # Enter below the username and passwort you got by VOIP provider username [XXX] password [YYY] location-service LOCATION_SVC # The line below requires the VOIP Providers domain name domain 1 [registrar IP or domain] # Set the SIP username on the line below identity <username> authentication outbound # Replace the XXX with the username of Authentication service authenticate 1 authentication-service AUTH_SVC username <username> registration outbound # Set the Registrar IP or domainname of VOIP provider registrar 1 <IP or Domain> register auto context sip-gateway GW_SIP interface IF_GWSIP bind interface WAN context router port 5060 context sip-gateway GW_SIP bind location-service LOCATION_SVC no shutdown port e1t1 0 0 port-type e1 clock auto framing crc4 encapsulation q921 q921 uni-side auto encapsulation q931 q931 protocol dss1 uni-side user bchan-number-order ascending encapsulation cc-isdn bind interface IF_ISDN_00 port e1t1 0 0 no shutdown
Example of SIP-gateway with fxs ports ( Link to this Config )
	Configuration Description Configuration for a Pingtel / sipX PBX -US Dial Tones -SN45xx (firmware 5.X) based -Replace all <>-variables with appropriate values Command Sequence to Copy, Edit, and Paste profile call-progress-tone US_Dialtone play 1 0 350 -13 440 -13 profile call-progress-tone US_Alertingtone play 1 2000 440 -19 480 -19 pause 2 4000 profile call-progress-tone US_Busytone play 1 500 480 -24 620 -24 pause 2 500 profile tone-set default profile tone-set US map call-progress-tone dial-tone US_Dialtone map call-progress-tone ringback-tone US_Alertingtone map call-progress-tone busy-tone US_Busytone map call-progress-tone release-tone US_Busytone map call-progress-tone congestion-tone US_Busytone profile voip default codec 1 g723-6k3 rx-length 30 tx-length 30 codec 2 g711ulaw64k rx-length 20 tx-length 20 codec 3 g711alaw64k rx-length 20 tx-length 20 context cs switch routing-table called-e164 RT_PSTN_TO_SIP route .T dest-interface IF_SIP interface sip IF_SIP bind context sip-gateway GW_SIP # Enter below the SIP server / Softswitch IP address or FQDN remote <0.0.0.0> route call dest-interface IF_FXS_00 # Enter below the SIP username, to be used as user part in from header address-translation outgoing-call from-header user-part fix <username> host-part call interface fxs IF_FXS_00 route call dest-table RT_PSTN_TO_SIP context cs switch no shutdown authentication-service AUTH_SVC # Enter below the username and passwort you got by VOIP provider username [XXX] password [YYY] location-service LOCATION_SVC # The line below requires the VOIP Providers domain name domain 1 [registrar IP or domain] # Set the SIP username on the line below identity <username> authentication outbound # Replace the XXX with the username of Authentication service authenticate 1 authentication-service AUTH_SVC username <username> registration outbound # Set the Registrar IP or domainname of VOIP provider registrar 1 <IP or Domain> register auto context sip-gateway GW_SIP interface IF_GWSIP bind interface WAN context router port 5060 context sip-gateway GW_SIP bind location-service LOCATION_SVC no shutdown port ethernet 0 0 medium auto encapsulation ip bind interface WAN router no shutdown # Repeat the port fxs-part below if you have/need more fxs-interfaces # for each FXS port an interface in context cs is required port fxs 0 0 # (use etsi for europe instead of bell) caller-id format bell # (use etsi or other country if not US based) use profile fxs us encapsulation cc-fxs bind interface IF_FXS_00 no shutdown

Time offset rule for Central Europe ( Link to this Config )
	Configuration Description Smartware version R5.X not older than 2010-07-15 Use these two commands to achieve that the SmartNode automatically performs the daylight saving change. This example is for countries with default offset +01:00 and daylight saving of +02:00 between the last Sunday of March until the last Sunday of October. For all other time zones please change this value accordingly. Command Sequence to Copy, Edit, and Paste #----------------------------------------------------------------# # # # Smartware version # # R5.X not older than 2010-07-15 # # # # Use these two commands to achieve that the SmartNode # # automatically performs the daylight saving change. # # This example is for countries with default offset +01:00 and # # daylight saving of +02:00 between the last Sunday of March # # until the last Sunday of October. For all other time zones # # please change this value accordingly. # # # # DATE: 2011-01-05 # # AUTHOR: Thomas Tlapak, Marc Aeberhard # # PATTON-INALP-NETWORKS; CH-3172 NIEDERWANGEN # # WWW.PATTON.COM WWW.INALP.COM # # # #----------------------------------------------------------------# clock local default-offset +02:00 clock local dst-rule DST +01:00 from 03:00 oct 31st previous sunday until 02:00 mar 31st previous sunday
SNTP for Australia ( Link to this Config )
	Configuration Description SNTP configuration recomendation for Australia. This configuration uses a combination of one DNS name in Australia and one IP address in Australia. DNS server must work work for DNS to be resolved. SNTP should continue to work if DNS breaks or the IP address changes of either SNTP provider. DD ---- # DNS server must come for DHCP or the a specific DNS server configured in the line below (remove the line or change xxx.xxx.xxx.xxx to the IP address of the DNS server.) dns-client server sntp-client xxx.xxx.xxx.xxx sntp-client server primary cronos.cenam.mx port 123 version 4 sntp-client server secondary 192.43.244.18 port 123 version 4 sntp-client poll-interval 36000 sntp-client local-clock-offset sntp-client gmt-offset - 05:00:00 Command Sequence to Copy, Edit, and Paste # DNS server must come for DHCP or the a specific DNS server configured in the line below (remove the line or change xxx.xxx.xxx.xxx to the IP address of the DNS server.) dns-client server sntp-client xxx.xxx.xxx.xxx sntp-client server primary ntp.saard.net port 123 version 4 sntp-client server secondary 203.21.84.4 port 123 version 4 sntp-client poll-interval 36000 sntp-client local-clock-offset sntp-client gmt-offset + 11:00:00
SNTP for Europe & UK ( Link to this Config )
	Configuration Description SNTP configuration recomendation for Europe. This configuration uses a combination of one DNS name in Europe and one IP address in Europe. DNS server must work work for DNS to be resolved. SNTP should continue to work if DNS breaks or the IP address changes of either SNTP provider. DD Command Sequence to Copy, Edit, and Paste # DNS server must come for DHCP or the a specific DNS server configured in the line below (remove the line or change xxx.xxx.xxx.xxx to the IP address of the DNS server.) dns-client server sntp-client xxx.xxx.xxx.xxx sntp-client server primary swisstime.ethz.ch port 123 version 4 sntp-client server secondary 192.36.143.150 port 123 version 4 sntp-client poll-interval 36000 sntp-client local-clock-offset
SNTP for Japan ( Link to this Config )
	Configuration Description SNTP configuration recomendation for Japan. This configuration uses a combination of one DNS name in Japan and one IP address in Japan. DNS server must work work for DNS to be resolved. SNTP should continue to work if DNS breaks or the IP address changes of either SNTP provider. DD Command Sequence to Copy, Edit, and Paste # DNS server must come for DHCP or the a specific DNS server configured in the line below (remove the line or change xxx.xxx.xxx.xxx to the IP address of the DNS server.) dns-client server sntp-client xxx.xxx.xxx.xxx sntp-client server primary ntp.cybernet-fleet.net port 123 version 4 sntp-client server secondary 133.100.9.2 port 123 version 4 sntp-client poll-interval 36000 sntp-client local-clock-offset sntp-client gmt-offset + 09:00:00
SNTP for Mexico ( Link to this Config )
	Configuration Description SNTP configuration recomendation for Mexico. This configuration uses a combination of one DNS name in Mexico and one IP address in USA. DNS server must work work for DNS to be resolved. SNTP should continue to work if DNS breaks or the IP address changes of either SNTP provider. DD Command Sequence to Copy, Edit, and Paste # DNS server must come for DHCP or the a specific DNS server configured in the line below (remove the line or change xxx.xxx.xxx.xxx to the IP address of the DNS server.) dns-client server sntp-client xxx.xxx.xxx.xxx sntp-client server primary cronos.cenam.mx port 123 version 4 sntp-client server secondary 192.43.244.18 port 123 version 4 sntp-client poll-interval 36000 sntp-client local-clock-offset sntp-client gmt-offset - 05:00:00
SNTP for Russia ( Link to this Config )
	Configuration Description SNTP configuration recomendation for Russia. This configuration uses a combination of one DNS name in Russia and one IP address in Russia. DNS server must work work for DNS to be resolved. SNTP should continue to work if DNS breaks or the IP address changes of either SNTP provider. DD Command Sequence to Copy, Edit, and Paste # DNS server must come for DHCP or the a specific DNS server configured in the line below (remove the line or change xxx.xxx.xxx.xxx to the IP address of the DNS server.) dns-client server sntp-client xxx.xxx.xxx.xxx sntp-client server primary ntp.psn.ru port 123 version 4 sntp-client server secondary 193.233.46.10 port 123 version 4 sntp-client poll-interval 36000 sntp-client local-clock-offset sntp-client gmt-offset + 03:00:00
SNTP for South Africa ( Link to this Config )
	Configuration Description SNTP configuration recomendation for South Africa. This configuration uses a combination of one DNS name in Africa and one IP address in Africa. DNS server must work work for DNS to be resolved. SNTP should continue to work if DNS breaks or the IP address changes of either SNTP provider. DD Command Sequence to Copy, Edit, and Paste # DNS server must come for DHCP or the a specific sntp-client server primary ntp.is.co.za port 123 version 4 sntp-client server secondary 146.64.241.228 port 123 version 4 sntp-client anycast-address 224.0.1.1 port 123 sntp-client poll-interval 36000 sntp-client local-clock-offset sntp-client gmt-offset + 02:00:00
SNTP for South America ( Link to this Config )
	Configuration Description SNTP configuration recomendation for South America. This configuration uses a combination of one DNS name in South Americaand one IP address in South America. DNS server must work work for DNS to be resolved. SNTP should continue to work if DNS breaks or the IP address changes of either SNTP provider. DD Command Sequence to Copy, Edit, and Paste # DNS server must come for DHCP or the a specific DNS server configured in the line below (remove the line or change xxx.xxx.xxx.xxx to the IP address of the DNS server.) dns-client server sntp-client xxx.xxx.xxx.xxx sntp-client server primary ntp.linux.org.ve port 123 version 4 sntp-client server secondary 200.144.121.33 port 123 version 4 sntp-client poll-interval 36000 sntp-client local-clock-offset sntp-client gmt-offset - 02:00:00
SNTP for USA ( Link to this Config )
	Configuration Description SNTP configuration recomendation for USA. This configuration uses a combination of one DNS name in the USA and one IP address in USA. DNS server must work work for DNS to be resolved. SNTP should continue to work if DNS breaks or the IP address changes of either SNTP provider. DD Command Sequence to Copy, Edit, and Paste # DNS server must come for DHCP or the a specific DNS server configured in the line below (remove the line or change xxx.xxx.xxx.xxx to the IP address of the DNS server.) dns-client server sntp-client xxx.xxx.xxx.xxx sntp-client server primary pool.ntp.org port 123 version 4 sntp-client server secondary 88.198.124.92 port 123 version 4 sntp-client poll-interval 36000 sntp-client local-clock-offset sntp-client gmt-offset - 05:00:00

Bulgaria 1 Style Tone Set ( Link to this Config )
	Configuration Description There are Three distinct tones sets used in Bulgaria. Bulgaria1, Bulgaria2 & Bulgaria3. The command sequence below enables SmartNode to provide the standard Norway1 style tone set for dial-tones and call-progress tones. Command Sequence to Copy, Edit, and Paste profile call-progress-tone BG1_Dialtone play 250 425 -12 pause 300 play 700 425 -12 pause 800 profile call-progress-tone BG1_Alertingtone play 1000 425 -12 pause 9000 profile call-progress-tone BG1_Busytone play 250 425 -12 pause 250 profile tone-set default profile tone-set BG1 map call-progress-tone dial-tone BG1_Dialtone map call-progress-tone ringback-tone BG1_Alertingtone map call-progress-tone busy-tone BG1_Busytone map call-progress-tone release-tone BG1_Busytone map call-progress-tone congestion-tone BG1_Busytone
Bulgaria 2 Style Tone Set ( Link to this Config )
	Configuration Description There are Three distinct tones sets used in Bulgaria. Bulgaria1, Bulgaria2 & Bulgaria3. The command sequence below enables SmartNode to provide the standard Norway1 style tone set for dial-tones and call-progress tones. Command Sequence to Copy, Edit, and Paste profile call-progress-tone BG2_Dialtone play 250 425 -12 pause 750 play 750 425 -12 pause 1000 profile call-progress-tone BG2_Alertingtone play 1000 425 -12 pause 4000 profile call-progress-tone BG2_Busytone play 500 425 -12 pause 500 profile tone-set default profile tone-set BG2 map call-progress-tone dial-tone BG2_Dialtone map call-progress-tone ringback-tone BG2_Alertingtone map call-progress-tone busy-tone BG2_Busytone map call-progress-tone release-tone BG2_Busytone map call-progress-tone congestion-tone BG2_Busytone
Bulgaria 3 Style Tone Set ( Link to this Config )
	Configuration Description There are Three distinct tones sets used in Bulgaria. Bulgaria1, Bulgaria2 & Bulgaria3. The command sequence below enables SmartNode to provide the standard Norway1 style tone set for dial-tones and call-progress tones. Command Sequence to Copy, Edit, and Paste profile call-progress-tone BG3_Dialtone play 5000 425 -12 profile call-progress-tone BG3_Alertingtone play 1830 425 -12 pause 3000 profile call-progress-tone BG3_Busytone play 200 425 -12 pause 500 profile tone-set default profile tone-set BG3 map call-progress-tone dial-tone BG3_Dialtone map call-progress-tone ringback-tone BG3_Alertingtone map call-progress-tone busy-tone BG3_Busytone map call-progress-tone release-tone BG3_Busytone map call-progress-tone congestion-tone BG3_Busytone
Czech Republic Style Tone Set ( Link to this Config )
	Configuration Description The command sequence below enables SmartNode to provide the standard Czech Republic style tone set for dial-tones and call-progress tones. Command Sequence to Copy, Edit, and Paste profile call-progress-tone CZ_Dialtone play 330 425 -12 pause 330 play 660 425 -12 pause 660 profile call-progress-tone CZ_Alertingtone play 1000 425 -12 pause 4000 profile call-progress-tone CZ_Busytone play 330 425 0 pause 330 profile tone-set default profile tone-set CZ map call-progress-tone dial-tone CZ_Dialtone map call-progress-tone ringback-tone CZ_Alertingtone map call-progress-tone busy-tone CZ_Busytone map call-progress-tone release-tone CZ_Busytone map call-progress-tone congestion-tone CZ_Busytone
France Style Tone Set ( Link to this Config )
	Configuration Description The command sequence below enables SmartNode to provide the standard France style tone set for dial-tones and call-progress tones. Command Sequence to Copy, Edit, and Paste profile call-progress-tone FR_Dialtone play 5000 440 -10 profile call-progress-tone FR_Alertingtone play 1500 440 -10 pause 3500 profile call-progress-tone FR_Busytone play 500 440 -10 pause 500 profile tone-set default profile tone-set TS_FRANCE map call-progress-tone dial-tone FR_Dialtone map call-progress-tone ringback-tone FR_Alertingtone map call-progress-tone busy-tone FR_Busytone map call-progress-tone release-tone FR_Busytone map call-progress-tone congestion-tone FR_Busytone
Hungary Stle Tone Sets ( Link to this Config )
	Configuration Description The command sequence below enables SmartNode to provide the standard Hungary-style tone set for dial-tones and call-progress tones. Command Sequence to Copy, Edit, and Paste profile call-progress-tone HUN_Dialtone play 5000 425 0 profile call-progress-tone HUN_Alertingtone play 1200 425 -12 pause 3700 profile call-progress-tone HUN_Busytone play 300 425 -12 pause 300 profile tone-set default profile tone-set HUN map call-progress-tone dial-tone HUN_Dialtone map call-progress-tone ringback-tone HUN_Alertingtone map call-progress-tone busy-tone HUN_Busytone map call-progress-tone release-tone HUN_Busytone map call-progress-tone congestion-tone HUN_Busytone
Italy Style Tone Set ( Link to this Config )
	Configuration Description The command sequence below enables SmartNode to provide the standard Italy-style tone set for dial-tones and call-progress tones. Command Sequence to Copy, Edit, and Paste profile call-progress-tone IT_Dialtone play 200 425 -12 pause 200 play 600 425 -12 pause 1000 profile call-progress-tone IT_Alertingtone play 1000 425 -12 pause 4000 profile call-progress-tone IT_Busytone play 500 425 -12 pause 500 profile tone-set default profile tone-set IT map call-progress-tone dial-tone IT_Dialtone map call-progress-tone ringback-tone IT_Alertingtone map call-progress-tone busy-tone IT_Busytone map call-progress-tone release-tone IT_Busytone map call-progress-tone congestion-tone IT_Busytone
Mexico Style Tone Set  ( Link to this Config )
	Configuration Description The command sequence below enables SmartNode to provide the standard Mexico style tone set for dial-tones and call-progress tones. Command Sequence to Copy, Edit, and Paste profile call-progress-tone MX_Dialtone play 1 200 425 -12 profile call-progress-tone MX_Alertingtone play 1 1000 425 -12 pause 2 4000 profile call-progress-tone MX_Busytone play 1 250 425 -12 pause 2 250 profile call-progress-tone MX_Congestiontone play 1 250 425 -12 pause 2 250 profile tone-set default profile tone-set MX map call-progress-tone dial-tone MX_Dialtone map call-progress-tone ringback-tone MX_Alertingtone map call-progress-tone busy-tone MX_Busytone map call-progress-tone release-tone MX_Busytone map call-progress-tone congestion-tone MX_Congestiontone
Netherlands Style Tone Set ( Link to this Config )
	Configuration Description The command sequence below enables SmartNode to provide the standard Netherlands style tone set for dial-tones and call-progress tones. Command Sequence to Copy, Edit, and Paste profile call-progress-tone NL_Dialtone play 1 200 425 -12 450 -12 profile call-progress-tone NL_Alertingtone play 1 1000 425 -12 450 -12 pause 2 4000 profile call-progress-tone NL_Busytone play 1 500 425 -12 450 -12 pause 2 500 profile call-progress-tone NL_Congestiontone play 1 250 425 -12 450 -12 pause 2 250 profile call-progress-tone NL_Waitingtone play 1 500 425 -12 450 -12 pause 2 250 profile tone-set default profile tone-set NL map call-progress-tone dial-tone NL_Dialtone map call-progress-tone ringback-tone NL_Alertingtone map call-progress-tone busy-tone NL_Busytone map call-progress-tone release-tone NL_Busytone map call-progress-tone congestion-tone NL_Congestiontone map call-progress-tone waiting-tone NL_Waitingtone
New Zealand-Style Tone Set ( Link to this Config )
	Configuration Description The command sequence below enables SmartNode to provide the standard New Zealand-style tone set for dial-tones and call-progress tones Command Sequence to Copy, Edit, and Paste profile call-progress-tone NZ_Dialtone play 1 5000 400 -15 profile call-progress-tone NZ_Alertingtone play 1 400 400 -15 450 -15 pause 2 200 play 3 400 400 -15 450 -15 pause 4 2000 profile call-progress-tone NZ_Busytone play 1 500 400 -15 pause 2 500 profile call-progress-tone NZ_SDTone play 1 10000 425 -19 620 -19 profile call-progress-tone NZ_Callwaiting play 1 200 400 -15 pause 2 200 profile call-progress-tone NZ_Releasetone play 1 250 400 -15 pause 2 250 profile tone-set NZ map call-progress-tone dial-tone NZ_Dialtone map call-progress-tone ringback-tone NZ_Alertingtone map call-progress-tone busy-tone NZ_Busytone map call-progress-tone waiting-tone NZ_Callwaiting map call-progress-tone release-tone NZ_Releasetone map call-progress-tone congestion-tone NZ_Releasetone map call-progress-tone special-dial-tone NZ_SDTone
Norway 1 Style Tone Set ( Link to this Config )
	Configuration Description There are Two distinct tones sets used in Norway. Norway1 and Norway2. The command sequence below enables SmartNode to provide the standard Norway1 style tone set for dial-tones and call-progress tones. Command Sequence to Copy, Edit, and Paste profile call-progress-tone NOR1_Dialtone play 5000 425 -12 425 -12 profile call-progress-tone NOR1_Alertingtone play 1000 425 -12 pause 3000 profile call-progress-tone NOR1_Busytone play 500 425 0 pause 500 profile tone-set default profile tone-set NOR1 map call-progress-tone dial-tone NOR1_Dialtone map call-progress-tone ringback-tone NOR1_Alertingtone map call-progress-tone busy-tone NOR1_Busytone map call-progress-tone release-tone NOR1_Busytone map call-progress-tone congestion-tone NOR1_Busytone
Norway 2 Style Tone Set ( Link to this Config )
	Configuration Description There are Two distinct tones sets used in Norway. Norway1 and Norway2 The command sequence below enables SmartNode to provide the standard Norway2 style tone set for dial-tones and call-progress tones. Command Sequence to Copy, Edit, and Paste profile call-progress-tone NOR2_Dialtone play 5000 425 -12 425 -12 profile call-progress-tone NOR2_Alertingtone play 1000 450 -12 pause 3000 profile call-progress-tone NOR2_Busytone play 200 450 0 pause 400 profile tone-set default profile tone-set NOR2 map call-progress-tone dial-tone NOR2_Dialtone map call-progress-tone ringback-tone NOR2_Alertingtone map call-progress-tone busy-tone NOR2_Busytone map call-progress-tone release-tone NOR2_Busytone map call-progress-tone congestion-tone NOR2_Busytone
Poland Style Tone Sets ( Link to this Config )
	Configuration Description The command sequence below enables SmartNode to provide the standard Poland-style tone set for dial-tones and call-progress tones. Command Sequence to Copy, Edit, and Paste profile call-progress-tone POL_Dialtone play 5000 425 0 profile call-progress-tone POL_Alertingtone play 1000 425 -12 pause 4000 profile call-progress-tone POL_Busytone play 500 425 -12 pause 500 profile tone-set default profile tone-set POL map call-progress-tone dial-tone POL_Dialtone map call-progress-tone ringback-tone POL_Alertingtone map call-progress-tone busy-tone POL_Busytone map call-progress-tone release-tone POL_Busytone map call-progress-tone congestion-tone POL_Busytone
Russia Style Tone Sets ( Link to this Config )
	Configuration Description The command sequence below enables SmartNode to provide the standard Russia-style tone set for dial-tones and call-progress tones. Command Sequence to Copy, Edit, and Paste profile call-progress-tone RUS_Dialtone play 5000 425 0 profile call-progress-tone RUS_Alertingtone play 800 425 -12 pause 3200 profile call-progress-tone RUS_Busytone play 400 425 -12 pause 400 profile tone-set default profile tone-set RUS map call-progress-tone dial-tone RUS_Dialtone map call-progress-tone ringback-tone RUS_Alertingtone map call-progress-tone busy-tone RUS_Busytone map call-progress-tone release-tone RUS_Busytone map call-progress-tone congestion-tone RUS_Busytone
Slovenia Type Tone Sets ( Link to this Config )
	Configuration Description The command sequence below enables SmartNode to provide the standard Slovenia-style tone set for dial-tones and call-progress tones. Command Sequence to Copy, Edit, and Paste profile call-progress-tone SLVN_Dialtone play 700 425 -12 pause 800 play 200 425 -12 pause 300 profile call-progress-tone SLVN_Alertingtone play 1000 425 -12 pause 4000 profile call-progress-tone SLVN_Busytone play 500 425 -12 pause 500 profile tone-set default profile tone-set SLVN map call-progress-tone dial-tone SLVN_Dialtone map call-progress-tone ringback-tone SLVN_Alertingtone map call-progress-tone busy-tone SLVN_Busytone map call-progress-tone release-tone SLVN_Busytone map call-progress-tone congestion-tone SLVN_Busytone
South Africa Style Tone Sets ( Link to this Config )
	Configuration Description The command sequence below enables SmartNode to provide the standard South Africa-style tone set for dial-tones and call-progress tones. Command Sequence to Copy, Edit, and Paste profile call-progress-tone ZA_Dialtone play 1 0 400 -13 profile call-progress-tone ZA_Alertingtone play 1 400 400 -13 pause 2 200 play 3 400 400 -13 pause 4 2000 profile call-progress-tone ZA_Busytone play 1 500 400 -13 pause 2 500 profile call-progress-tone ZA_Releasetone play 1 250 400 -13 pause 2 250 profile call-progress-tone ZA_Holdtone play 1 30 400 -13 pause 2 60 play 3 30 400 -13 pause 4 4880 profile tone-set default profile tone-set ZA map call-progress-tone dial-tone ZA_Dialtone map call-progress-tone ringback-tone ZA_Alertingtone map call-progress-tone busy-tone ZA_Busytone map call-progress-tone release-tone ZA_Releasetone map call-progress-tone hold-tone ZA_Holdtone
Spain Style Tone Set  ( Link to this Config )
	Configuration Description The command sequence below enables SmartNode to provide the standard Spain style tone set for dial-tones and call-progress tones. Command Sequence to Copy, Edit, and Paste profile call-progress-tone ES_Dialtone play 1 200 425 -12 profile call-progress-tone ES_Alertingtone play 1 1500 425 -12 pause 2 3000 profile call-progress-tone ES_Busytone play 1 200 425 -12 pause 2 200 profile call-progress-tone ES_Congestiontone play 1 200 425 -12 pause 2 200 play 3 200 425 -12 pause 4 200 play 5 200 425 -12 pause 4 600 profile call-progress-tone ES_Waitingtone play 1 200 425 -12 pause 2 600 profile tone-set default profile tone-set ES map call-progress-tone dial-tone ES_Dialtone map call-progress-tone ringback-tone ES_Alertingtone map call-progress-tone busy-tone ES_Busytone map call-progress-tone release-tone ES_Busytone map call-progress-tone congestion-tone ES_Congestiontone map call-progress-tone waiting-tone ES_Waitingtone
Sri Lanka Type Tone Sets ( Link to this Config )
	Configuration Description The command sequence below enables SmartNode to provide the standard Sri Lanka-style tone set for dial-tones and call-progress tones. Command Sequence to Copy, Edit, and Paste profile call-progress-tone LK_Dialtone play 5000 425 profile call-progress-tone LK_Alertingtone play 400 425 pause 200 profile call-progress-tone LK_Busytone play 750 425 pause 750 profile call-progress-tone LK_Congestiontone play 250 425 pause 250 profile tone-set LK map call-progress-tone dial-tone LK_Dialtone map call-progress-tone ringback-tone LK_Alertingtone map call-progress-tone busy-tone LK_Busytone map call-progress-tone release-tone LK_Busytone map call-progress-tone congestion-tone LK_Congestiontone
Sweden Style Tone Set ( Link to this Config )
	Configuration Description The command sequence below enables SmartNode to provide the standard Sweden-style tone set for dial-tones and call-progress tones. Command Sequence to Copy, Edit, and Paste profile call-progress-tone SE_DIAL play 1 5000 425 -12 profile call-progress-tone SE_BUSY play 1 250 425 0 pause 2 250 profile call-progress-tone SE_ALERTING play 1 1000 425 -12 pause 2 5000 profile tone-set SVERIGE map call-progress-tone dial-tone SE_DIAL map call-progress-tone ringback-tone SE_ALERTING map call-progress-tone busy-tone SE_BUSY
UK-Style Tone Set ( Link to this Config )
	Configuration Description The command sequence below enables SmartNode to provide the standard UK-style tone set for dial-tones and call-progress tones. Command Sequence to Copy, Edit, and Paste profile call-progress-tone UK_Dialtone play 1 5000 350 -13 440 -13 profile call-progress-tone UK_Alertingtone play 1 400 400 0 450 0 pause 2 200 play 3 400 400 0 450 0 pause 4 2000 profile call-progress-tone UK_Busytone play 1 375 400 0 pause 2 375 profile call-progress-tone UK_CongestionTone play 1 400 400 0 pause 2 350 play 3 225 400 0 pause 4 525 profile call-progress-tone UK_WaitingTone play 1 100 400 0 pause 2 5000 profile call-progress-tone UK_ReleaseTone play 1 1500 400 0 profile tone-set UK map call-progress-tone dial-tone UK_Dialtone map call-progress-tone ringback-tone UK_Alertingtone map call-progress-tone busy-tone UK_Busytone map call-progress-tone special-information-tone UK_SpecialInfoTone map call-progress-tone waiting-tone UK_WaitingTone map call-progress-tone release-tone UK_ReleaseTone map call-progress-tone congestion-tone UK_CongestionTone map call-progress-tone special-dial-tone UK_SpecialDialTone
USA-Style Tone Set ( Link to this Config )
	Configuration Description The command sequence below enables SmartNode to provide the standard USA-style tone set for dial-tones and call-progress tones. Command Sequence to Copy, Edit, and Paste configure system profile call-progress-tone US_Dialtone play 1 1000 350 -13 440 -13 profile call-progress-tone US_Alertingtone play 1 1000 440 -19 480 -19 pause 2 3000 profile call-progress-tone US_Busytone play 1 500 480 -24 620 -24 pause 2 500 profile call-progress-tone US_Releasetone play 1 250 480 -24 620 -24 pause 2 250 profile tone-set default profile tone-set US map call-progress-tone dial-tone US_Dialtone map call-progress-tone ringback-tone US_Alertingtone map call-progress-tone busy-tone US_Busytone map call-progress-tone release-tone US_Releasetone map call-progress-tone congestion-tone US_Busytone

IKE with AES-128 (Side #1)  ( Link to this Config )
	Configuration Description The command sequence below defines a VPN Tunnel that uses AES-128 to route traffic from Side #1 (192.168.1.0) to Side #2 (192.168.2.0). NOTE: -You must replace ‹X.X.X.X› in the template with the IP address for Side #2. -You must replace ‹MySharedKey› in the template with text string as the initial shared key -You must replace ‹nnnn› in the template with unique number from 1-99999 like 12345 -You must purchase the appropriate software license for your SmartNode to support VPN service. To see the current licenses active for your SmartNode, execute the "show license" command. Command Sequence to Copy, Edit, and Paste profile ipsec-transform AES-128 esp-encryption aes-cbc 128 key-lifetime-seconds 3600 profile isakmp-transform AES-128 encryption aes-cbc 128 authentication-algorithm sha1 key-lifetime-seconds 86400 profile ipsec-policy-isakmp VPN-IKE authentication-method pre-shared-key ‹MySharedKey› # SET IP ADDRESS of SIDE #2 peer ‹X.X.X.X› protection-group ‹nnnn› mode tunnel diffie-hellman-group group2 use profile ipsec-transform 1 AES-128 use profile isakmp-transform 1 AES-128 profile acl VPN-Out permit ip 192.168.1.0 0.0.0.255 192.168.2.0 0.0.0.255 ipsec-policy VPN-IKE permit ip any any profile acl VPN-In permit esp any any permit ah any any permit ip 192.168.2.0 0.0.255.255 192.168.1.0 0.0.0.255 permit ip any any context ip interface eth0 use profile acl VPN-In in use profile acl VPN-Out out
IKE with AES-128 (Side #2)  ( Link to this Config )
	Configuration Description The command sequence below defines a VPN Tunnel that uses AES-128 to route traffic from Side #2 (192.168.4.0) to Side #1 (192.168.1.0). NOTE: -You must replace ‹X.X.X.X› in the template with the IP address for Side #2. -You must replace ‹MySharedKey› in the template with text string as the initial shared key -You must replace ‹nnnn› in the template with unique number from 1-99999 like 12345 -You must purchase the appropriate software license for your SmartNode to support VPN service. To see the current licenses active for your SmartNode, execute the "show license" command. Command Sequence to Copy, Edit, and Paste profile ipsec-transform AES-128 esp-encryption aes-cbc 128 key-lifetime-seconds 3600 profile isakmp-transform AES-128 encryption aes-cbc 128 authentication-algorithm sha1 key-lifetime-seconds 86400 profile ipsec-policy-isakmp VPN-IKE authentication-method pre-shared-key ‹MySharedKey› # SET IP ADDRESS of SIDE #1 peer ‹X.X.X.X› protection-group ‹nnnn› mode tunnel diffie-hellman-group group2 use profile ipsec-transform 1 AES-128 use profile isakmp-transform 1 AES-128 profile acl VPN-Out permit ip 192.168.4.0 0.0.0.255 192.168.1.0 0.0.0.255 ipsec-policy VPN-IKE permit ip any any profile acl VPN-In permit esp any any permit ah any any permit ip 192.168.1.0 0.0.255.255 192.168.4.0 0.0.0.255 permit ip any any context ip interface eth0 use profile acl VPN-In in use profile acl VPN-Out out context ip router route 192.168.1.0 255.255.255.0 eth0
Manual Key Tunnel Mode with DES (Side # 1) ( Link to this Config )
	Configuration Description The command sequence below defines a VPN Tunnel that uses DES to route traffic from Side #1 (192.168.1.0) to Side #2 (192.168.2.0). NOTE: You must replace X.X.X.X in the template with the IP address for Side #2. NOTE: You must purchase the appropriate software license for your SmartNode to support VPN service. To see the current licenses active for your SmartNode, execute the "show license" command. Command Sequence to Copy, Edit, and Paste profile ipsec-transform DES esp-encryption des-cbc 64 profile ipsec-policy-manual VPN_DES use profile ipsec-transform DES session-key inbound esp-encryption A242353012531297 session-key outbound esp-encryption B242353012531297 spi inbound esp 1111 spi outbound esp 2222 # SET IP ADDRESS of SIDE #2 peer X.X.X.X mode tunnel profile acl VPN_Out permit ip 192.168.1.0 0.0.0.255 192.168.2.0 0.0.0.255 ipsec-policy VPN_DES permit ip any any profile acl VPN_In permit esp any any permit ah any any permit ip 192.168.2.0 0.0.0.255 192.168.1.0 0.0.0.255 permit ip any any context ip interface eth0 use profile acl VPN_In in use profile acl VPN_Out out context ip router route 192.168.2.0 255.255.255.0 eth0
Manual Key Tunnel Mode with DES (Side # 2) ( Link to this Config )
	Configuration Description The command sequence below defines a VPN Tunnel that uses DES to route traffic from Side #2 (192.168.2.0) to Side #1 (192.168.1.0). NOTE: You must replace X.X.X.X in the template with the IP address for Side #1. NOTE: You must purchase the appropriate software license for your SmartNode to support VPN service. To see the current licenses active for your SmartNode, execute the "show license" command. Command Sequence to Copy, Edit, and Paste profile ipsec-transform DES esp-encryption des-cbc 64 profile ipsec-policy-manual VPN_DES use profile ipsec-transform DES session-key inbound esp-encryption B242353012531297 session-key outbound esp-encryption A242353012531297 spi inbound esp 2222 spi outbound esp 1111 # SET IP ADDRESS of SIDE #1 peer X.X.X.X mode tunnel profile acl VPN_Out permit ip 192.168.2.0 0.0.0.255 192.168.1.0 0.0.0.255 ipsec-policy VPN_DES permit ip any any profile acl VPN_In permit esp any any permit ah any any permit ip 192.168.1.0 0.0.0.255 192.168.2.0 0.0.0.255 permit ip any any context ip interface eth0 use profile acl VPN_In in use profile acl VPN_Out out context ip router route 192.168.1.0 255.255.255.0 eth0