SN4554 configuration example R5.4

SmartNode 4554 patton configuration Example R5.4

The following script can be used to configure SN4554 R5.4 firmware to be used with asterisk.
It manages 2 BRI ISDN ports, through 1 SIP account: PATTON register as client to ASTERISK (192.168.1.5) with username 101 and password 12311.
The outgoing calls should be dialed to 1* when they should be addressed to BRI 1, 2* where they should be addressed to BRI 2.



;asterisk sip.conf : Patton register on asterisk with account 101, password 12311
[101]
  type=friend
  context=default
  host=dynamic
  secret=12311
  nat=no
  insecure=port,invite
  disallow=all
  allow=alaw
  permit=192.168.1.128/255.255.255.0



The following configuration should be applicable to SmartNode R5.X, and was tried on firmware SmartNode R5.4 2009-09-22 SIP.



#----------------------------------------------------------------#
#                                                                #
# SN4554/2BIS/EUI                                                #
# R5.2 2009-01-14 SIP                                            #
# 1970-01-02T03:26:53                                            #
# SN/00A0BA049189                                                #
# Generated configuration file                                   #
#                                                                #
#----------------------------------------------------------------#

cli version 3.20
# administration: user admin, password 12311
administrator admin password D2F2A1lO9/8= encrypted
gui type basic
dns-relay
webserver port 80 language en
sntp-client
# NTP server which provide the current date/time
sntp-client server primary 192.168.1.5 port 123 version 4

system

  ic voice 0

system
  clock-source 1 bri 0 0
  clock-source 2 bri 0 1

profile acl ACL_WAN_PERMIT_SEL_MGMT

profile service-policy SP_WAN_OUT
  no rate-limit

profile service-policy SP_WAN_IN
  no rate-limit

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

profile voip VOIP
  codec 1 g711alaw64k rx-length 20 tx-length 20
  codec 2 g711ulaw64k rx-length 20 tx-length 20
  dejitter-mode static
  dejitter-max-delay 120

profile pstn default

profile sip default

profile dhcp-server DHCPS_LAN

profile aaa default
  method 1 local
  method 2 none

context ip router

  interface IF_IP_WAN
                #IP address defined statically
    ipaddress 192.168.1.128 255.255.255.0
                #ipaddress dhcp
    use profile napt NAPT_WAN
    tcp adjust-mss rx mtu
    tcp adjust-mss tx mtu

  interface IF_IP_LAN
    ipaddress unnumbered

subscriber ppp SUB_PPPOE
  dial in
  no multilink

context cs switch
        #Avoid trouble with early dial
        digit-collection timeout 8
        no digit-collection terminating-char
        #used to get the complete dialed number for incoming calls
        national-prefix 0
        international-prefix 00

  routing-table called-e164 RT_TO_SIP
                # Route all incoming calls to the SIP account 101
    route .%T dest-interface IF_SIP_SERVICE

        routing-table called-e164 DIAL_OUT
                # routing table for outgoing calls:
                # SIP/101/1456789 -> 456789 on ISDN IF_S0_00
                # SIP/101/2456789 -> 456789 on ISDN IF_S0_01
                route 1.% dest-interface IF_S0_00 STRIP_MAP
                route 2.% dest-interface IF_S0_01 STRIP_MAP
                route default dest-interface IF_S0_00

        #strip the first digit from outgoing calls
        mapping-table called-e164 to called-e164 STRIP_MAP
          map .(.%) to \1

  interface isdn IF_S0_00
    route call dest-table RT_TO_SIP

  interface isdn IF_S0_01
    route call dest-table RT_TO_SIP

  interface sip IF_SIP_SERVICE
    bind context sip-gateway GW_SIP
    route call dest-table DIAL_OUT
    remote 192.168.1.5
    early-connect
    early-disconnect

        #maybe this section is useless
  service hunt-group BOUND_ISDN
    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_S0_00
    route call 2 dest-interface IF_S0_01
                route default dest-interface IF_S0_00


context cs switch
  no shutdown

# define authentication to asterisk SIP account 101 with password 12311
authentication-service AUTH_SVC
  username 101 password 12311

# asterisk server with IP address 192.168.1.5
# patton registers to asterisk
location-service LOCATION_SVC
  domain 1 192.168.1.5
  identity 101
    authentication outbound
      authenticate 1 authentication-service AUTH_SVC username 101
    registration outbound
    registrar 192.168.1.5
    lifetime 3600
    register auto

context sip-gateway GW_SIP
  interface IF_SIP
    bind interface IF_IP_WAN context router port 5060

context sip-gateway GW_SIP
  bind location-service LOCATION_SVC
  no shutdown

port ethernet 0 0
  bind interface IF_IP_WAN router
  pppoe
    session SES_PPPOE
      shutdown

port ethernet 0 0
  no shutdown

port bri 0 0
  clock auto
  encapsulation q921

  q921
    permanent-layer2
    protocol pmp
                #protocol pp
    uni-side auto
    encapsulation q931

    q931
      protocol dss1
      uni-side user
      bchan-number-order ascending
      encapsulation cc-isdn
      bind interface IF_S0_00 switch

port bri 0 0
  no shutdown

port bri 0 1
  clock auto
  encapsulation q921

  q921
    permanent-layer2
    protocol pmp
                #protocol pp
    uni-side auto
    encapsulation q931

    q931
      protocol dss1
      uni-side user
      bchan-number-order ascending
      encapsulation cc-isdn
      bind interface IF_S0_01 switch

port bri 0 1
  no shutdown




Note: improvements and comments to this configuration script are welcome!

Other documentation


SmartNode 4554 patton configuration Example R5.4

The following script can be used to configure SN4554 R5.4 firmware to be used with asterisk.
It manages 2 BRI ISDN ports, through 1 SIP account: PATTON register as client to ASTERISK (192.168.1.5) with username 101 and password 12311.
The outgoing calls should be dialed to 1* when they should be addressed to BRI 1, 2* where they should be addressed to BRI 2.



;asterisk sip.conf : Patton register on asterisk with account 101, password 12311
[101]
  type=friend
  context=default
  host=dynamic
  secret=12311
  nat=no
  insecure=port,invite
  disallow=all
  allow=alaw
  permit=192.168.1.128/255.255.255.0



The following configuration should be applicable to SmartNode R5.X, and was tried on firmware SmartNode R5.4 2009-09-22 SIP.



#----------------------------------------------------------------#
#                                                                #
# SN4554/2BIS/EUI                                                #
# R5.2 2009-01-14 SIP                                            #
# 1970-01-02T03:26:53                                            #
# SN/00A0BA049189                                                #
# Generated configuration file                                   #
#                                                                #
#----------------------------------------------------------------#

cli version 3.20
# administration: user admin, password 12311
administrator admin password D2F2A1lO9/8= encrypted
gui type basic
dns-relay
webserver port 80 language en
sntp-client
# NTP server which provide the current date/time
sntp-client server primary 192.168.1.5 port 123 version 4

system

  ic voice 0

system
  clock-source 1 bri 0 0
  clock-source 2 bri 0 1

profile acl ACL_WAN_PERMIT_SEL_MGMT

profile service-policy SP_WAN_OUT
  no rate-limit

profile service-policy SP_WAN_IN
  no rate-limit

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

profile voip VOIP
  codec 1 g711alaw64k rx-length 20 tx-length 20
  codec 2 g711ulaw64k rx-length 20 tx-length 20
  dejitter-mode static
  dejitter-max-delay 120

profile pstn default

profile sip default

profile dhcp-server DHCPS_LAN

profile aaa default
  method 1 local
  method 2 none

context ip router

  interface IF_IP_WAN
                #IP address defined statically
    ipaddress 192.168.1.128 255.255.255.0
                #ipaddress dhcp
    use profile napt NAPT_WAN
    tcp adjust-mss rx mtu
    tcp adjust-mss tx mtu

  interface IF_IP_LAN
    ipaddress unnumbered

subscriber ppp SUB_PPPOE
  dial in
  no multilink

context cs switch
        #Avoid trouble with early dial
        digit-collection timeout 8
        no digit-collection terminating-char
        #used to get the complete dialed number for incoming calls
        national-prefix 0
        international-prefix 00

  routing-table called-e164 RT_TO_SIP
                # Route all incoming calls to the SIP account 101
    route .%T dest-interface IF_SIP_SERVICE

        routing-table called-e164 DIAL_OUT
                # routing table for outgoing calls:
                # SIP/101/1456789 -> 456789 on ISDN IF_S0_00
                # SIP/101/2456789 -> 456789 on ISDN IF_S0_01
                route 1.% dest-interface IF_S0_00 STRIP_MAP
                route 2.% dest-interface IF_S0_01 STRIP_MAP
                route default dest-interface IF_S0_00

        #strip the first digit from outgoing calls
        mapping-table called-e164 to called-e164 STRIP_MAP
          map .(.%) to \1

  interface isdn IF_S0_00
    route call dest-table RT_TO_SIP

  interface isdn IF_S0_01
    route call dest-table RT_TO_SIP

  interface sip IF_SIP_SERVICE
    bind context sip-gateway GW_SIP
    route call dest-table DIAL_OUT
    remote 192.168.1.5
    early-connect
    early-disconnect

        #maybe this section is useless
  service hunt-group BOUND_ISDN
    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_S0_00
    route call 2 dest-interface IF_S0_01
                route default dest-interface IF_S0_00


context cs switch
  no shutdown

# define authentication to asterisk SIP account 101 with password 12311
authentication-service AUTH_SVC
  username 101 password 12311

# asterisk server with IP address 192.168.1.5
# patton registers to asterisk
location-service LOCATION_SVC
  domain 1 192.168.1.5
  identity 101
    authentication outbound
      authenticate 1 authentication-service AUTH_SVC username 101
    registration outbound
    registrar 192.168.1.5
    lifetime 3600
    register auto

context sip-gateway GW_SIP
  interface IF_SIP
    bind interface IF_IP_WAN context router port 5060

context sip-gateway GW_SIP
  bind location-service LOCATION_SVC
  no shutdown

port ethernet 0 0
  bind interface IF_IP_WAN router
  pppoe
    session SES_PPPOE
      shutdown

port ethernet 0 0
  no shutdown

port bri 0 0
  clock auto
  encapsulation q921

  q921
    permanent-layer2
    protocol pmp
                #protocol pp
    uni-side auto
    encapsulation q931

    q931
      protocol dss1
      uni-side user
      bchan-number-order ascending
      encapsulation cc-isdn
      bind interface IF_S0_00 switch

port bri 0 0
  no shutdown

port bri 0 1
  clock auto
  encapsulation q921

  q921
    permanent-layer2
    protocol pmp
                #protocol pp
    uni-side auto
    encapsulation q931

    q931
      protocol dss1
      uni-side user
      bchan-number-order ascending
      encapsulation cc-isdn
      bind interface IF_S0_01 switch

port bri 0 1
  no shutdown




Note: improvements and comments to this configuration script are welcome!

Other documentation


Created by: mesfet, Last modification: Mon 21 of Dec, 2009 (17:15 UTC)
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