Configuring a Virtual User in AOS to respond to SIP OPTIONS over a SIP Trunk

Version 2

    Configuring a Virtual User in AOS to respond to SIP OPTIONS over a SIP Trunk


    SIP “OPTIONS” messages are occasionally used by ITSP providers to verify the availability of the trunk from the softswitch. A “200 OK” would be returned if the AOS device is ready to accept a call, a “486 (Busy Here)” would be returned if the AOS device is busy, etc. This allows an OPTIONS request to be used to determine the basic state of the SIP trunk, which can be an indication of whether the AOS device will accept an INVITE request.

     

    By default, the AOS device will respond with a “501 Not Implemented”:                                         

     

    Rx: UDP src=205.XXX.XXX.XXX:5060 dst=172.28.10.230:5060

    OPTIONS sip:metaswitch@172.28.10.230:5060;transport=udp SIP/2.0

    Via: SIP/2.0/UDP 205.XXX.XXX.XXX:5060;branch=z9hG4bK+ca17548f01433280d09cd2f163cc84791+sip+1+a6511224

    From:<sip:metaswitch@205.XXX.XXX.XXX:5060>;tag=205.XXX.XXX.XXX+1+1496460d+5a488fc9

    Content-Length: 0

    Supported: resource-priority, 100rel

    To: <sip:metaswitch@172.28.10.230>

    Contact: <sip:fa26977a002b9668986d84eda7e705d1@205.XXX.XXX.XXX:5060>

    Allow-Events: message-summary, refer, dialog, line-seize, presence, call-info, as-feature-event

    Max-Forwards: 69

    Call-ID: 0gQAAC8WAAACBAAALxYAAFOiyp/GRYq6B+rNtSlH+xHw7SXdCOf9lXGh8hoavbDd@205.XXX.XXX.XXX

    CSeq: 591356001 OPTIONS

    Organization: Metaswitch Networks

    Accept: application/sdp, application/dtmf-relay

     

     

    Tx: UDP src=172.28.10.230:5060 dst=205.XXX.XXX.XXX:5060

    SIP/2.0 501 Not Implemented

    From:<sip:metaswitch@205.XXX.XXX.XXX:5060>;tag=205.XXX.XXX.XXX+1+1496460d+5a488fc9

    To:<sip:metaswitch@172.28.10.230>;tag=4d8bf88-7f000001-13c4-1ab942-2ae9c2b7-1ab942

    Call-ID: 0gQAAC8WAAACBAAALxYAAFOiyp/GRYq6B+rNtSlH+xHw7SXdCOf9lXGh8hoavbDd@205.XXX.XXX.XXX

    CSeq: 591356001 OPTIONS

    Via: SIP/2.0/UDP 205.XXX.XXX.XXX:5060;branch=z9hG4bK+ca17548f01433280d09cd2f163cc84791+sip+1+a6511224

    Content-Length: 0

     

    In order for the AOS device to respond with a 200 OK, indicating that the trunk is available to receive calls, a “dummy” voice user needs to be configured:

     

    (config)#voice user 1000

    (config-1000)#sip-identity metaswitch T01

     

    The voice user can be any extension that is not in use currently on the system. The sip-identity value must match on the user portion of the SIP URI (bolded red above). The SIP trunk value must match on the trunk the response should be sent.


    The AOS device should respond with a “200 OK” if the user portion of the SIP URI is blank, or is already configured on the unit.