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how to configur DID Inbound Rules?

Discussion in '3CX Phone System - General' started by vsefcik, Feb 17, 2010.

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  1. vsefcik

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    I'm running 3CX version 8.0.10116 and can't figure out how to get the PSTN gateway Inbound Rules to work correctly. I'm using a DIalogic DMG2000 gateway, which processes calls in from and out to our PBX with no problems, as long as the call coming in is to a registered VoIP phone and the call going out is from a registered VoIP phone. I now want to configure some DID Inbound Rules to route calls to extension numbers that are not on registered VoIP phones to a phone that is registered. For example, I have extension 7070 configured in 3CX and it is on an Aastra phone, and the phone has registered. Calls to and from 7070 and the PSTN gateway work fine. I want a call to extension 7072 from the PSTN gateway to be routed to the phone with extension 7070. So I added a DID Inbound Rule that says DID/DDI number/mask is 7072 and Connect to Extension 7070. When I dial 7072 on the PBX, the SIP Invite gets to 3CX, which responds with "407 Proxy Authentication Required" and the Server Activity Log contains the following:

    09:46:48.603 [CM500002]: Unidentified incoming call. Review INVITE and adjust source identification:
    INVITE sip:7072@169.231.192.10;user=phone SIP/2.0
    Via: SIP/2.0/UDP 169.231.192.12:5060;branch=z9hG4bK801E438076C8097F2D084BF9A3D0261B
    Max-Forwards: 70
    Contact: <sip:7072@169.231.192.12:5060>
    To: <sip:7072@169.231.192.10;user=phone>
    From: <sip:4182@169.231.192.12:5060;user=phone>;tag=7C68324631353641000085EE;vnd.pimg.port=1
    Call-ID: 01B22719EE81400000000010@dialogic2.commserv.ucsb.edu
    CSeq: 1 INVITE
    Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REGISTER, INFO, COMET, PRACK, REFER, SUBSCRIBE, NOTIFY, MESSAGE
    Supported: replaces, early-session, 100rel
    User-Agent: PBX-IP Media Gateway
    Content-Length: 0
    Diversion: <tel:8937072>;reason=unconditional

    If I configure extension 7072 in 3CX, but do not add this extension to a real VoIP phone, a call to 7072 restuls in a SIP "480 Temporarily Unavailable" (because it hasn't registered) and the Server Activity Log shows:

    09:55:10.755 [CM503008]: Call(25): Call is terminated
    09:55:10.725 [CM503020]: Normal call termination. Reason: Not available
    09:55:10.725 [CM503016]: Call(25): Attempt to reach <sip:7072@169.231.192.10;user=phone> failed. Reason: Not Registered
    09:55:10.725 [CM503016]: Call(25): Attempt to reach <sip:7072@169.231.192.10;user=phone> failed. Reason: Not Registered
    09:55:10.725 [CM503017]: Call(25): Target is not registered: Ext:Ext.7072
    09:55:10.725 [CM503010]: Making route(s) to <sip:7072@169.231.192.10;user=phone>
    09:55:10.725 [MS210000] C:25.1:Offer received. RTP connection: 169.231.192.12:49038(49039)
    09:55:10.725 Remote SDP is set for legC:25.1
    09:55:10.715 [CM505001]: Ext.4182: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [PBX-IP Media Gateway] PBX contact: [sip:4182@169.231.192.10:5060]
    09:55:10.715 [CM503001]: Call(25): Incoming call from Sip.4182 to <sip:7072@169.231.192.10;user=phone>
    09:55:10.705 [CM500002]: Info on incoming INVITE:
    INVITE sip:7072@169.231.192.10;user=phone SIP/2.0
    Via: SIP/2.0/UDP 169.231.192.12:5060;branch=z9hG4bKD2D44D48D11F5E56207E3EC81AB4799D
    Max-Forwards: 70
    Contact: <sip:7072@169.231.192.12:5060>
    To: <sip:7072@169.231.192.10;user=phone>
    From: <sip:4182@169.231.192.12:5060;user=phone>;tag=61793246313536410000998D;vnd.pimg.port=1
    Call-ID: 01B2271BE481400000000012@dialogic2.commserv.ucsb.edu
    CSeq: 1 INVITE
    Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REGISTER, INFO, COMET, PRACK, REFER, SUBSCRIBE, NOTIFY, MESSAGE
    Supported: replaces, early-session, 100rel
    User-Agent: PBX-IP Media Gateway
    Content-Length: 0
    Diversion: <tel:8937072>;reason=unconditional

    So, my question is: can I have a call from a PSTN gateway be re-routed to a registered extension if the called number does not appear on a real VoIP phone that has Registered? Thanks for any guidance.
     
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