Hi, I'm trying to configure a 4638 to pass inbound and outbound to and from ISDN and SIP interfaces. I'm using web interface at the moment. Setup is: ISDN -- <BRI> -- Patton 4638 -- <SIP> Asterisk -- <SIP> -- <IP Phone> I can call from IP phone but can't receive any incoming call : I can't see any SIP message coming in when a call comes in. Previously, with 4.2 firmware, you just have to edit routing table binding ISDN ports to SIP interface to get calls coming in but now with 5.3, configuration process changed. Here is an extract from my running config. Any idea ? Regards context cs switch routing-table called-e164 appels_provenance_ISDN route [0-9]+ dest-service ASTERISK_SRV route default dest-service ASTERISK_SRV routing-table called-uri appels_vers_ISDN route default dest-service isdnports mapping-table called-e164 to called-ip transfo map [0-9]+ to 192.168.100.254 mapping-table called-e164 to called-uri transfo2 interface isdn IF-PBX route call dest-table appels_provenance_ISDN interface isdn IF-PBX2 route call dest-table appels_provenance_ISDN interface isdn IF-PBX3 route call dest-table appels_provenance_ISDN interface isdn IF-PBX4 route call dest-table appels_provenance_ISDN interface sip IF-ASTERISK bind context sip-gateway ASTERISK route call dest-table appels_vers_ISDN service sip-location-service ASTERISK_SRV bind location-service ASTERISK_SRV mode hunt hunt-timeout 20 service hunt-group isdnports 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-PBX route call 2 dest-interface IF-PBX2 route call 3 dest-interface IF-PBX3 context cs switch no shutdown authentication-service patton realm 1 asterisk username patton password Otx2vJCEWP+8Bb6tqoGkwA== encrypted location-service ASTERISK_SRV domain 1 192.168.100.254 5060 domain 2 asterisk 5060 identity-group default identity patton alias name patton authentication outbound authenticate 1 authentication-service patton username patton registration outbound registrar 192.168.100.254 5060 proxy none lifetime 3600 register auto retry-timeout on-system-error 10 retry-timeout on-client-error 10 retry-timeout on-server-error 10 call outbound use profile tone-set default use profile voip default use profile sip default preferred-transport-protocol udp invite-transaction-timeout 32 non-invite-transaction-timeout 32 call inbound use profile tone-set default use profile voip default use profile sip default -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090225/1c862a6d/attachment.htm