[asterisk-users] Patton 5.3. How to get incoming calls ? [SOLVED]

[Olivier]

Hi,

Changing the line bellow helped to get incoming calls but I add to remove
secret= option in sip.conf (otherwise, Patton wouldn't respond to 407 Auth
required challenges).
If someone could enable secret and still get incoming calls (in any
SmartWare 5.X), please, do not hesitate to share here ...

  interface sip IF-ASTERISK
    bind context sip-gateway ASTERISK
    route call dest-table calls_from_SIP
    remote 192.168.100.254 5060


Cheers

2009/2/25 Olivier <oza-4h07 at myamail.com>

> 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
>
>
>
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