Hosted SIP Phones with Full FXO Survivability and Full-time FXO Usage Sample Configuration

Version 5

    Hosted SIP Phones with Full FXO Survivability and Full-time FXO Usage Sample Configuration

     

     

    The sample configuration below contains the options needed for inbound FXO, outbound FXO, and local survivability of hosted SIP phones registered through the AOS SIP Proxy.  The configuration also allows for inbound and outbound calls on the FXO port under non failover conditions.  For the sample configuration, the following is assumed about the application:

     

     

    • IP interfaces, routing, and firewall features not referenced in this document have already been configured.
    • The unit is equipped with a PSTN grade FXO port.  Please reference the AOS Feature Matrix – Product Feature Matrix to confirm PSTN FXO capabilities.
    • The ERL tool, which determines the correct impedance value to use for the analog PSTN line, has already been run .  Details regarding usage of the ERL tool can be found in the Analog Configuration and Trunk Troubleshooting Guide.
    • There are no other voice trunks or voice grouped-trunks configured.
    • The SIP phones register to their SIP server using ten digit numbers (i.e. 256-555-1234), but the application requires local SIP phone to SIP phone calls be made using the last four digits of the phone number during survivability (i.e. 1234).
    • The number 256-555-1000 is associated to a ring all ring-group on the SIP server.  A ring-group with the same number has been configured on the AOS device to be used when in survivability.
    • 10 digit local dialing is in use on the FXO trunks.

     

     

    The following are key notes regarding the application:

     

     

    • The SIP phone facing IP address of the AOS device will need to be configured as the phones’ default gateway and DNS server.
    • The AOS device will be configured to act as a DNS proxy, which is required during survivability when a FQDN is used for the provisioning server, SIP server, outbound proxy, or registrar. The command voip name-service host <FQDN> sip udp ensures that the specified FQDN persists even when connectivity to the unit’s DNS server has been lost.  An entry will need to be made for every FQDN the SIP phones are configured for, whether that be for provisioning or to send SIP traffic to . If a phone fails to resolve a FQDN, the phone will potentially not boot up , register or make/receive phone calls .
    • The unit can be configured to allow the registration of SIP phones during failover with the sip proxy failover accept-registrations command. This configuration command is desirable if the end users reboot their phones during survivability. When choosing whether or not to use this command, be aware that the command opens the unit up to accept ALL registrations during survivability.  This means that any rogue SIP device on the LAN can register to the unit during survivability and make calls to other SIP proxy users or out the FXO trunk(s).
    • The SIP server will need to be provisioned to terminate a SIP trunk from the AOS device.  When not in survivability, all calls destined to the FXO or originating from the FXO will need to be routed through the SIP server.  It is not permissible for calls to bypass the SIP server under non-failover conditions.
    • If you would like to route emergency calls out the FXO port(s), you can optionally add the command sip proxy emergency-call-routing accept 911.

     

     

    Additional Resources:

     

    domain-proxy

    domain-proxy failover

    domain-lookup database local

    name-server 192.0.2.1

    !

    ip dhcp pool "PHONES"

      network 10.10.10.0 255.255.255.0

      dns-server 10.10.10.1

      default-router 10.10.10.1

      ntp-server 10.10.10.1

      timezone-offset -6:00

    !

    interface eth 0/2

      ip address 10.10.10.1  255.255.255.0

      ip access-policy PHONES

      media-gateway ip primary

      no shutdown

    !

    !

    voice forward-mode local

    !

    voice trunk-list FXO

      trunk T02

    !

    voice trunk T01 type sip

      sip-server primary 192.0.2.2

    !

    voice trunk T02 type analog supervision loop-start

      connect fxo 0/1

      trunk-number 2565551000

      prefer trunk-routing

    !

    voice grouped-trunk SIP

      trunk T01

      accept 2565551000

      deny proxy

      permit list FXO

    !

    voice grouped-trunk FXO

      trunk T02

      accept  411 cost 0

      accept 611 cost 0

      accept 911 cost 0

      accept NXX-NXX-XXXX cost 0

      accept 1-NXX-NXX-XXXX cost 0

      accept 011-X$ cost 0

    !

    voice ring-group 2565551000

      type all

      member 2565551200

      login-member 2565551200

      member 2565551201

      login-member 2565551201

      member 2565551202

      login-member 2565551202

    !

    sip

    sip proxy

    sip proxy transparent

    !

    sip proxy failover match-digits 4

    !

    sip proxy failover accept-registrations

    sip proxy failover direct-inbound

    !

    voip name-service host sip.example.com sip udp

    voip name-service host provisioning.example.com sip udp