Hello Tony, thank you for your clarification.
I made a dial plan, but can't dial out. After dialing I'm getting "not found". Internal dialing works. Sorry if it sounds rude but debugging a gateway is horror. How can I see sip packets sent to the gateway. I used this command to make a file for sip viewer: sipx-trace --all-components --output test.xml sip After opening the file with sip viewer I can see both telefons with there ip's, the sipx server with ip, mediaserver, sipxproxy, sipregistrar, sipxbridge and the ip of a voip provider connected via trunk (can't call the provider also) The gateway has ip 192.168.3.250. I can ping it. From the gateway I can ping sipx. What must I do that sipx will send packets to the gateway an how can I see this. Thanks in advance Stephan -------- Original-Nachricht -------- > Datum: Tue, 26 May 2009 06:46:59 -0400 > Von: "Tony Graziano" <[email protected]> > An: [email protected], [email protected] > Betreff: Re: [sipx-users] Unable to register Patton 4552 > This is an ISDN gateway, it does not have any FXS port. Gateways do not > register (unless it has FXS ports, in which case it would). > > You should assign a dialing plan to it and test it. > > >>> "Stephan Bauer" <[email protected]> 05/26/09 6:34 AM >>> > Hello, > > I am unable to register a Patton 4552 to 4.0.0-015321. > > First I added a unmanaged gateway with name "Patton 4552" and the IP > 192.168.3.250. Other options default. > > Patton 4552 firmware is 5.3 > > Could anyone help me with this beast > > My config: > > #----------------------------------------------------------------# > # # > # SN4552/2BIS/EUI # > # R5.3 2009-03-18 SIP # > # 2009-05-26T12:25:34 # > # SN/00a0ba014a8d # > # Generated configuration file # > # # > #----------------------------------------------------------------# > > cli version 3.20 > gui type basic > clock local offset +02:00 > webserver port 80 language en > sntp-client > sntp-client server primary 192.168.3.254 port 123 version 4 > sntp-client local-clock-offset > timer DaylightSavingsOn 02:00 mar 25th next sunday every year "sntp-client > gmt-offset + 02:00:00" > timer DaylightSavingsOff 03:00 oct 25th next sunday every year > "sntp-client gmt-offset + 01:00:00" > system hostname mediagateway1.novanutria.zz > > system > > ic voice 0 > > profile acl ACL_WAN_PERMIT_ALL_MGMT > permit 1 ip any any "" > > profile acl ACL_WAN_PERMIT_SEL_MGMT > deny 1 tcp any any eq 23 "" > deny 2 tcp any any eq 80 "" > deny 3 udp any any eq 161 "" > permit 4 ip any any "" > > profile acl ACL_WAN_BLOCK_ALL_MGMT > deny 1 tcp any any eq 23 "" > deny 2 tcp any any eq 80 "" > deny 3 udp any any eq 161 "" > permit 4 ip any any "" > > profile service-policy SP_WAN_OUT > rate-limit 100000 header-length 18 voice-margin 0 > > source traffic-class local-voice > priority > > source traffic-class default > priority > > profile service-policy SP_WAN_IN > rate-limit 100000 header-length 18 voice-margin 200 > > source traffic-class local-voice > priority > > source traffic-class default > queue-limit 4 > > profile napt NAPT_WAN > > profile ppp default > > profile call-progress-tone US_DIAL_TONE > play 1 10 350 -13 440 -13 > > profile call-progress-tone US_RB_TONE > play 1 2000 440 -19 480 -19 > pause 2 4000 > > profile call-progress-tone US_BUSY_TONE > play 1 500 480 -24 620 -24 > pause 2 500 > > profile call-progress-tone US_CONGESTION_TONE > play 1 250 480 -24 620 -24 > pause 2 250 > > profile tone-set default > profile tone-set Europe > profile tone-set UnitedStates > map call-progress-tone dial-tone US_DIAL_TONE > map call-progress-tone ringback-tone US_RB_TONE > map call-progress-tone busy-tone US_BUSY_TONE > map call-progress-tone release-tone US_BUSY_TONE > map call-progress-tone congestion-tone US_CONGESTION_TONE > > 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 g729 rx-length 20 tx-length 20 > codec 2 g711alaw64k rx-length 20 tx-length 20 > codec 3 g711ulaw64k rx-length 20 tx-length 20 > dejitter-mode static > dejitter-max-delay 120 > fax transmission 1 relay t38-udp > > profile pstn default > > profile sip default > > profile dhcp-server DHCPS_LAN > network 192.168.1.0 255.255.255.0 > include 1 192.168.1.10 192.168.1.19 > lease 2 hours > default-router 1 192.168.1.1 > domain-name patton.com > domain-name-server 1 192.168.1.1 > > profile aaa default > method 1 local > method 2 none > > context ip router > > interface IF_IP_WAN > ipaddress dhcp > use profile acl ACL_WAN_PERMIT_ALL_MGMT in > use profile service-policy SP_WAN_IN in > use profile service-policy SP_WAN_OUT out > use profile napt NAPT_WAN > tcp adjust-mss rx 582 > tcp adjust-mss tx 1440 > > interface IF_IP_LAN > ipaddress dhcp > icmp router-discovery > > subscriber ppp SUB_PPPOE > dial out > no multilink > authentication chap > authentication pap > bind interface IF_IP_WAN router > > context cs switch > > routing-table called-e164 RT_FROM_PSTN > route 4149080 dest-interface IF_S0_01 > route 41490.+ dest-interface IF_SIP_SIPX > > mapping-table called-e164 to called-e164 MT_SPEED_DIAL > mapping-table calling-e164 to calling-e164 MT_SET_CNPN > mapping-table called-e164 to called-e164 MT_NR_TRANSLATION > > interface isdn IF_S0_00 > route call dest-table RT_FROM_PSTN > > interface isdn IF_S0_01 > route call dest-interface IF_S0_00 > use profile tone-set Europe > isdn-date-time > > interface sip IF_SIP_SIPX > bind context sip-gateway GW_SIPX > route call dest-interface IF_S0_00 > remote 192.168.3.7 > use profile voip VOIP > use profile tone-set Europe > > service hunt-group SER_HG_PSTN_FALLBACK > 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 > drop-cause no-route-to-destination > route call 1 dest-interface IF_SIP_SERVICE > route call 2 dest-interface IF_S0_00 > > context cs switch > no shutdown > > authentication-service AUTH_SVC > > location-service LOCATION_SVC > > identity-group default > > authentication outbound > authenticate 1 authentication-service AUTH_SVC > > registration outbound > register auto > > call outbound > > context sip-gateway GW_SIPX > > interface IF_LAN > bind interface IF_IP_LAN context router port 5060 > > context sip-gateway GW_SIPX > no shutdown > > port ethernet 0 0 > bind interface IF_IP_WAN router > > pppoe > > session SES_PPPOE > bind subscriber SUB_PPPOE > shutdown > > port ethernet 0 0 > no shutdown > > port ethernet 0 1 > bind interface IF_IP_LAN router > no shutdown > > port bri 0 0 > clock auto > encapsulation q921 > > q921 > 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 > protocol pp > uni-side auto > encapsulation q931 > > q931 > protocol dss1 > uni-side net > bchan-number-order ascending > encapsulation cc-isdn > bind interface IF_S0_01 switch > > port bri 0 1 > no shutdown > > > > > > -- > Neu: GMX FreeDSL Komplettanschluss mit DSL 6.000 Flatrate + > Telefonanschluss für nur 17,95 Euro/mtl.!* http://portal.gmx.net/de/go/dsl02 > _______________________________________________ > sipx-users mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users > Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users -- Neu: GMX FreeDSL Komplettanschluss mit DSL 6.000 Flatrate + Telefonanschluss für nur 17,95 Euro/mtl.!* http://portal.gmx.net/de/go/dsl02 _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users
