Third Party SIP Server connected to Avaya IP Office

phillip.bcs · Jun 21, 2018

Greetings!
I have an Avaya IP Office 500 V2 running 9.0700....
I'm also running a Brekeke SIP server with a SIP soft phone registered.
I have no problems calling the soft phone, but I cannot call from the soft phone any of the Avaya extensions.
I can make it work, however, if I setup an ICR on the SIP trunk and direct it to a specific extension. The issue is that I want to be able to dial any of my Avaya extensions on a softphone. Can this be done without setting up an ICR for every extension?

I have a single SIP URI configured on the SIP Trunk. It is set so that my trunk (Group 20) is both incoming and outgoing.
Local URI = *
Contact = *
Display Name = *

I've played with all these as "Use Internal Data", but it did not make a difference.
I've verified the SIP information is in place for my test users on the Avaya side.
SIP Line config has Call Routing Method set to "To Header" but I've tried "Request URI" with no success.
The trace is always returned 404 Not found.

IP Office : 10.20.245.1
Brekeke SIP : 10.20.240.8

I would really appreciate any assistance anyone could give.
(This trace was updated after it was pointed out to me that I had the wrong information)

Code:

  10207584mS SIP Rx: UDP 10.20.240.8:5060 -> 10.20.245.1:5060
                    INVITE sip:20211@10.20.245.1 SIP/2.0
                    Via: SIP/2.0/UDP 10.20.240.8:5060;rport;branch=z9hG4bK802d8fddbc32-30-1a2f6b
                    Via: SIP/2.0/UDP 10.20.252.112:5060;rport=5060;branch=z9hG4bKPj51ed07898c6c477bb66e7f661a09ba8c
                    Max-Forwards: 69
                    From: "Phillip" <sip:20899@10.20.240.8>;tag=5de879626607477783ce8d67f1ab4362
                    To: "20211" <sip:20211@10.20.245.1>
                    Contact: "Phillip" <sip:20899@10.20.240.8:5060;ob>
                    Call-ID: f325bfcd312c46818dbde46c648fbec7
                    CSeq: 22955 INVITE
                    Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
                    Supported: replaces, 100rel, timer, norefersub
                    Session-Expires: 1800
                    Min-SE: 90
                    User-Agent: MicroSIP/3.18.5
                    Record-Route: <sip:10.20.240.8:5060;ftag=5de879626607477783ce8d67f1ab4362;lr>
                    Content-Type: application/sdp
                    Content-Length: 305
                    
                    v=0
                    o=- 3738604773 3738604773 IN IP4 10.20.240.8
                    s=pjmedia
                    b=AS:84
                    t=0 0
                    a=X-nat:0
                    m=audio 10172 RTP/AVP 0 8 101
                    c=IN IP4 10.20.240.8
                    b=TIAS:64000
                    a=sendrecv
                    a=rtpmap:0 PCMU/8000
                    a=rtpmap:8 PCMA/8000
                    a=rtpmap:101 telephone-event/8000
                    a=fmtp:101 0-16
                    a=ssrc:873553672 cname:2089695b71411fea
  10207588mS Sip: Association found trunk: SIP Line (20)
  10207588mS SIP Call Rx: 20
                    INVITE sip:20211@10.20.245.1 SIP/2.0
                    Via: SIP/2.0/UDP 10.20.240.8:5060;rport;branch=z9hG4bK802d8fddbc32-30-1a2f6b
                    Via: SIP/2.0/UDP 10.20.252.112:5060;rport=5060;branch=z9hG4bKPj51ed07898c6c477bb66e7f661a09ba8c
                    Max-Forwards: 69
                    From: "Phillip" <sip:20899@10.20.240.8>;tag=5de879626607477783ce8d67f1ab4362
                    To: "20211" <sip:20211@10.20.245.1>
                    Contact: "Phillip" <sip:20899@10.20.240.8:5060;ob>
                    Call-ID: f325bfcd312c46818dbde46c648fbec7
                    CSeq: 22955 INVITE
                    Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
                    Supported: replaces, 100rel, timer, norefersub
                    Session-Expires: 1800
                    Min-SE: 90
                    User-Agent: MicroSIP/3.18.5
                    Record-Route: <sip:10.20.240.8:5060;ftag=5de879626607477783ce8d67f1ab4362;lr>
                    Content-Type: application/sdp
                    Content-Length: 305
                    
                    v=0
                    o=- 3738604773 3738604773 IN IP4 10.20.240.8
                    s=pjmedia
                    b=AS:84
                    t=0 0
                    a=X-nat:0
                    m=audio 10172 RTP/AVP 0 8 101
                    c=IN IP4 10.20.240.8
                    b=TIAS:64000
                    a=sendrecv
                    a=rtpmap:0 PCMU/8000
                    a=rtpmap:8 PCMA/8000
                    a=rtpmap:101 telephone-event/8000
                    a=fmtp:101 0-16
                    a=ssrc:873553672 cname:2089695b71411fea
  10207589mS Sip: SIPDialog f483a6a0 created, dialogs 1
  10207590mS Sip: License, Valid 1, Available 4, Consumed 0
  10207590mS Sip: CheckLineMonitors on SIP Endpoint - KEY & LAMP for SIP Trunk!
  10207591mS Sip: SIPTrunkEndpointDialogOwner::SetRemoteAddressForRequest from 10.20.240.8:5060 to 10.20.240.8:5060
  10207591mS Sip: SIPTrunkEndpointDialogOwner::SetRemoteAddressForResponse from 10.20.240.8:5060 to 10.20.240.8:5060
  10207591mS Sip: 20.1070.1 -1 SIPTrunk Endpoint(f483a6a0) PreProcessMsg calling CheckMinSEField, MinSE value is set to 90 in the header
  10207591mS Sip: 20.1070.1 -1 SIPTrunk Endpoint(f483a6a0) PreProcessMsg calling CheckSessionExpiresField, session expires is 1800 refresher_is_ipo 0
  10207591mS Sip: 20.1070.1 -1 SIPTrunk Endpoint(f483a6a0) PreProcessMsg calling CheckMinSEField, MinSE value is set to 1800 in the header
  10207592mS Sip: 20.1070.1 -1 SIPTrunk Endpoint(f483a6a0) Cloned 
  10207593mS Sip: 20.1070.1 -1 SIPTrunk Endpoint(f483a6a0) SendSIPResponse: INVITE code 100 SENT TO 10.20.240.8 5060
  10207593mS SIP Call Tx: 20
                    SIP/2.0 100 Trying
                    Via: SIP/2.0/UDP 10.20.240.8:5060;rport;branch=z9hG4bK802d8fddbc32-30-1a2f6b
                    Via: SIP/2.0/UDP 10.20.252.112:5060;rport=5060;branch=z9hG4bKPj51ed07898c6c477bb66e7f661a09ba8c
                    Record-Route: <sip:10.20.240.8:5060;ftag=5de879626607477783ce8d67f1ab4362;lr>
                    From: "Phillip" <sip:20899@10.20.240.8>;tag=5de879626607477783ce8d67f1ab4362
                    Call-ID: f325bfcd312c46818dbde46c648fbec7
                    CSeq: 22955 INVITE
                    Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,NOTIFY,UPDATE
                    Supported: timer
                    Server: IP Office 9.0.7.0 build 983
                    To: "20211" <sip:20211@10.20.245.1>;tag=d30a68f195527827
                    Content-Length: 0
                    
  10207594mS SIP Tx: UDP 10.20.245.1:5060 -> 10.20.240.8:5060
                    SIP/2.0 100 Trying
                    Via: SIP/2.0/UDP 10.20.240.8:5060;rport;branch=z9hG4bK802d8fddbc32-30-1a2f6b
                    Via: SIP/2.0/UDP 10.20.252.112:5060;rport=5060;branch=z9hG4bKPj51ed07898c6c477bb66e7f661a09ba8c
                    Record-Route: <sip:10.20.240.8:5060;ftag=5de879626607477783ce8d67f1ab4362;lr>
                    From: "Phillip" <sip:20899@10.20.240.8>;tag=5de879626607477783ce8d67f1ab4362
                    Call-ID: f325bfcd312c46818dbde46c648fbec7
                    CSeq: 22955 INVITE
                    Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,NOTIFY,UPDATE
                    Supported: timer
                    Server: IP Office 9.0.7.0 build 983
                    To: "20211" <sip:20211@10.20.245.1>;tag=d30a68f195527827
                    Content-Length: 0
                    
  10207594mS Sip: 20.1070.1 -1 SIPTrunk Endpoint(f483a6a0) INVITE Received ep 20.1070.1 -1 SIPTrunk Endpoint(f4778118), dialog f483a6a0 
  10207595mS Sip: 20.1070.1 -1 SIPTrunk Endpoint(f483a6a0) LR is On and route is Route: <sip:10.20.240.8:5060;ftag=5de879626607477783ce8d67f1ab4362;lr>
  10207595mS Sip: SIPTrunkEndpointDialogOwner::SetRemoteAddressForRequest from 10.20.240.8:5060 to 10.20.240.8:5060
  10207595mS Sip: 20.1070.1 -1 SIPTrunk Endpoint(f483a6a0) UpdateSIPCallState SIPDialog::INITIAL(0) -> SIPDialog::INVITE_RCVD(9)
  10207595mS Sip: 20.1070.1 -1 SIPTrunk Endpoint(f483a6a0) UpdateSDPState SIPDialog::IDLE(0) -> SIPDialog::OFFER_RCVD(2)
  10207596mS Sip: 20.1070.1 -1 SIPTrunk Endpoint(f483a6a0) SdpClone 
  10207596mS Sip: 20.1070.1 -1 SIPTrunk Endpoint(f483a6a0) SIPDialog::BuildFastStartFromSDPOffer sdpmsg f475a148
  10207596mS Sip: 20.1070.1 -1 SIPTrunk Endpoint(f483a6a0) FindConnectionParameters: found bandwidth info, bw_count=1, memorizing only first one
  10207596mS Sip: 20.1070.1 -1 SIPTrunk Endpoint(f483a6a0) SIPDialog::FindConnectionParameters: found media proto <RTP/AVP>
  10207596mS Sip: 20.1070.1 -1 SIPTrunk Endpoint(f483a6a0) SetRfc2833TxPayload: use RFC2833 for dtmf 
  10207596mS Sip: 20.1070.1 -1 SIPTrunk Endpoint(f483a6a0) SIPDialog::BuildFastStartFromAudioMediaSDPOffer reinvite 0 msg f53bd558 fs in msg 1 mRTP_PType 255
  10207597mS Sip: 20.1070.1 -1 SIPTrunk Endpoint(f483a6a0) SetRemoteRTPAddress to 10.20.240.8:10172 
  10207597mS Sip: 20.1070.1 -1 SIPTrunk Endpoint(f483a6a0) FindConnectionParameters: found bandwidth info, bw_count=1, memorizing only first one
  10207597mS Sip: 20.1070.1 -1 SIPTrunk Endpoint(f483a6a0) Process SIP request dialog f483a6a0, method INVITE in state SIPDialog::INVITE_RCVD(9)
  10207598mS Sip: 20.1070.1 -1 SIPTrunk Endpoint(f483a6a0) ProcessInboundSIPRequest calling CheckMinSEField, MinSE value is set to 90 in the header
  10207598mS Sip: 20.1070.1 -1 SIPTrunk Endpoint(f483a6a0) ProcessInboundSIPRequest calling CheckSessionExpiresField, session expires is 1800 refresher_is_ipo 0
  10207598mS Sip: 20.1070.1 -1 SIPTrunk Endpoint(f483a6a0) ProcessInboundSIPRequest calling CheckMinSEField, MinSE value is set to 1800 in the header
  10207599mS Sip: 20.1070.1 -1 SIPTrunk Endpoint(f483a6a0) UpdateClone mMesg f4831568 smsg f48317f0 
  10207599mS Sip: 20.1070.1 -1 SIPTrunk Endpoint(f483a6a0) CMSetup forwarded to call model owner_ep 20.1070.1 -1 SIPTrunk Endpoint(f4778118), dialog f483a6a0 has sdp 1
  10207605mS CMLOGGING:     CALL:2018/06/2121:19,00:00:00,000,20899@10.20.240.8,I,20211,20211,Phillip,,,0,,""n/a,0
  10207606mS Sip: 20.1070.1 -1 SIPTrunk Endpoint(f4778118) received CMReleaseComp
  10207606mS Sip: 20.1070.1 -1 SIPTrunk Endpoint(f483a6a0) Terminating dialog f483a6a0, state SIPDialog::INVITE_RCVD(9) for cause CMCauseUnallocatedNumber
  10207606mS Sip: 20.1070.1 -1 SIPTrunk Endpoint(f483a6a0) SendSIPResponse: INVITE code 404 SENT TO 10.20.240.8 5060
  10207607mS SIP Call Tx: 20
                    SIP/2.0 404 Not Found
                    Via: SIP/2.0/UDP 10.20.240.8:5060;rport;branch=z9hG4bK802d8fddbc32-30-1a2f6b
                    Via: SIP/2.0/UDP 10.20.252.112:5060;rport=5060;branch=z9hG4bKPj51ed07898c6c477bb66e7f661a09ba8c
                    Record-Route: <sip:10.20.240.8:5060;ftag=5de879626607477783ce8d67f1ab4362;lr>
                    From: "Phillip" <sip:20899@10.20.240.8>;tag=5de879626607477783ce8d67f1ab4362
                    Call-ID: f325bfcd312c46818dbde46c648fbec7
                    CSeq: 22955 INVITE
                    Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,NOTIFY,UPDATE
                    Supported: timer
                    Server: IP Office 9.0.7.0 build 983
                    Reason: Q.850;cause=1;text="Unallocated (unassigned) number"
                    To: "20211" <sip:20211@10.20.245.1>;tag=d30a68f195527827
                    Content-Length: 0
                    
  10207607mS SIP Tx: UDP 10.20.245.1:5060 -> 10.20.240.8:5060
                    SIP/2.0 404 Not Found
                    Via: SIP/2.0/UDP 10.20.240.8:5060;rport;branch=z9hG4bK802d8fddbc32-30-1a2f6b
                    Via: SIP/2.0/UDP 10.20.252.112:5060;rport=5060;branch=z9hG4bKPj51ed07898c6c477bb66e7f661a09ba8c
                    Record-Route: <sip:10.20.240.8:5060;ftag=5de879626607477783ce8d67f1ab4362;lr>
                    From: "Phillip" <sip:20899@10.20.240.8>;tag=5de879626607477783ce8d67f1ab4362
                    Call-ID: f325bfcd312c46818dbde46c648fbec7
                    CSeq: 22955 INVITE
                    Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,NOTIFY,UPDATE
                    Supported: timer
                    Server: IP Office 9.0.7.0 build 983
                    Reason: Q.850;cause=1;text="Unallocated (unassigned) number"
                    To: "20211" <sip:20211@10.20.245.1>;tag=d30a68f195527827
                    Content-Length: 0
                    
  10207608mS Sip: 20.1070.1 -1 SIPTrunk Endpoint(f483a6a0) UpdateSIPCallState SIPDialog::INVITE_RCVD(9) -> SIPDialog::FINAL(28)
  10207609mS Sip: 20.1070.1 -1 SIPTrunk Endpoint(f483a6a0) KeepDlgOnCmCallLost SIPDialog::FINAL
  10207609mS Sip: SIPDialog f483a6a0 deleted, dialogs 0
  10207610mS Sip: ~SipTrunkEndpoint 20.1070.1 -1 SIPTrunk Endpoint
  10207613mS SIP Rx: UDP 10.20.240.8:5060 -> 10.20.245.1:5060
                    ACK sip:20211@10.20.245.1 SIP/2.0
                    Via: SIP/2.0/UDP 10.20.240.8:5060;rport;branch=z9hG4bK802d8fddbc32-30-1a2f6b
                    Via: SIP/2.0/UDP 10.20.252.112:5060;rport=5060;branch=z9hG4bKPj51ed07898c6c477bb66e7f661a09ba8c
                    Max-Forwards: 69
                    From: "Phillip" <sip:20899@10.20.240.8>;tag=5de879626607477783ce8d67f1ab4362
                    To: "20211" <sip:20211@10.20.245.1>;tag=d30a68f195527827
                    Call-ID: f325bfcd312c46818dbde46c648fbec7
                    CSeq: 22955 ACK
                    Record-Route: <sip:10.20.240.8:5060;ftag=5de879626607477783ce8d67f1ab4362;lr>
                    Content-Length: 0
                    
  10207615mS Sip: Association found trunk: SIP Line (20)
  10207615mS SIP Call Rx: 20
                    ACK sip:20211@10.20.245.1 SIP/2.0
                    Via: SIP/2.0/UDP 10.20.240.8:5060;rport;branch=z9hG4bK802d8fddbc32-30-1a2f6b
                    Via: SIP/2.0/UDP 10.20.252.112:5060;rport=5060;branch=z9hG4bKPj51ed07898c6c477bb66e7f661a09ba8c
                    Max-Forwards: 69
                    From: "Phillip" <sip:20899@10.20.240.8>;tag=5de879626607477783ce8d67f1ab4362
                    To: "20211" <sip:20211@10.20.245.1>;tag=d30a68f195527827
                    Call-ID: f325bfcd312c46818dbde46c648fbec7
                    CSeq: 22955 ACK
                    Record-Route: <sip:10.20.240.8:5060;ftag=5de879626607477783ce8d67f1ab4362;lr>
                    Content-Length: 0
                    
  10212615mS Sip: sip_indicateTimeOut Timer 9
  10212615mS Sip: Timer 9 callback didn't find dialog, method INVITE, callid f325bfcd312c46818dbde46c648fbec7

janni78 · Jun 21, 2018

You trace just shows an OPTIONS message, it's not an actual call attempt.

"Trying is the first step to failure..." - Homer

phillip.bcs · Jun 21, 2018

You are correct. I posted the wrong trace. Sorry about that.
That is the keepalive being sent over the trunk.

Below is the correct trace:

Code:

  10207584mS SIP Rx: UDP 10.20.240.8:5060 -> 10.20.245.1:5060
                    INVITE sip:20211@10.20.245.1 SIP/2.0
                    Via: SIP/2.0/UDP 10.20.240.8:5060;rport;branch=z9hG4bK802d8fddbc32-30-1a2f6b
                    Via: SIP/2.0/UDP 10.20.252.112:5060;rport=5060;branch=z9hG4bKPj51ed07898c6c477bb66e7f661a09ba8c
                    Max-Forwards: 69
                    From: "Phillip" <sip:20899@10.20.240.8>;tag=5de879626607477783ce8d67f1ab4362
                    To: "20211" <sip:20211@10.20.245.1>
                    Contact: "Phillip" <sip:20899@10.20.240.8:5060;ob>
                    Call-ID: f325bfcd312c46818dbde46c648fbec7
                    CSeq: 22955 INVITE
                    Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
                    Supported: replaces, 100rel, timer, norefersub
                    Session-Expires: 1800
                    Min-SE: 90
                    User-Agent: MicroSIP/3.18.5
                    Record-Route: <sip:10.20.240.8:5060;ftag=5de879626607477783ce8d67f1ab4362;lr>
                    Content-Type: application/sdp
                    Content-Length: 305
                    
                    v=0
                    o=- 3738604773 3738604773 IN IP4 10.20.240.8
                    s=pjmedia
                    b=AS:84
                    t=0 0
                    a=X-nat:0
                    m=audio 10172 RTP/AVP 0 8 101
                    c=IN IP4 10.20.240.8
                    b=TIAS:64000
                    a=sendrecv
                    a=rtpmap:0 PCMU/8000
                    a=rtpmap:8 PCMA/8000
                    a=rtpmap:101 telephone-event/8000
                    a=fmtp:101 0-16
                    a=ssrc:873553672 cname:2089695b71411fea
  10207588mS Sip: Association found trunk: SIP Line (20)
  10207588mS SIP Call Rx: 20
                    INVITE sip:20211@10.20.245.1 SIP/2.0
                    Via: SIP/2.0/UDP 10.20.240.8:5060;rport;branch=z9hG4bK802d8fddbc32-30-1a2f6b
                    Via: SIP/2.0/UDP 10.20.252.112:5060;rport=5060;branch=z9hG4bKPj51ed07898c6c477bb66e7f661a09ba8c
                    Max-Forwards: 69
                    From: "Phillip" <sip:20899@10.20.240.8>;tag=5de879626607477783ce8d67f1ab4362
                    To: "20211" <sip:20211@10.20.245.1>
                    Contact: "Phillip" <sip:20899@10.20.240.8:5060;ob>
                    Call-ID: f325bfcd312c46818dbde46c648fbec7
                    CSeq: 22955 INVITE
                    Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
                    Supported: replaces, 100rel, timer, norefersub
                    Session-Expires: 1800
                    Min-SE: 90
                    User-Agent: MicroSIP/3.18.5
                    Record-Route: <sip:10.20.240.8:5060;ftag=5de879626607477783ce8d67f1ab4362;lr>
                    Content-Type: application/sdp
                    Content-Length: 305
                    
                    v=0
                    o=- 3738604773 3738604773 IN IP4 10.20.240.8
                    s=pjmedia
                    b=AS:84
                    t=0 0
                    a=X-nat:0
                    m=audio 10172 RTP/AVP 0 8 101
                    c=IN IP4 10.20.240.8
                    b=TIAS:64000
                    a=sendrecv
                    a=rtpmap:0 PCMU/8000
                    a=rtpmap:8 PCMA/8000
                    a=rtpmap:101 telephone-event/8000
                    a=fmtp:101 0-16
                    a=ssrc:873553672 cname:2089695b71411fea
  10207589mS Sip: SIPDialog f483a6a0 created, dialogs 1
  10207590mS Sip: License, Valid 1, Available 4, Consumed 0
  10207590mS Sip: CheckLineMonitors on SIP Endpoint - KEY & LAMP for SIP Trunk!
  10207591mS Sip: SIPTrunkEndpointDialogOwner::SetRemoteAddressForRequest from 10.20.240.8:5060 to 10.20.240.8:5060
  10207591mS Sip: SIPTrunkEndpointDialogOwner::SetRemoteAddressForResponse from 10.20.240.8:5060 to 10.20.240.8:5060
  10207591mS Sip: 20.1070.1 -1 SIPTrunk Endpoint(f483a6a0) PreProcessMsg calling CheckMinSEField, MinSE value is set to 90 in the header
  10207591mS Sip: 20.1070.1 -1 SIPTrunk Endpoint(f483a6a0) PreProcessMsg calling CheckSessionExpiresField, session expires is 1800 refresher_is_ipo 0
  10207591mS Sip: 20.1070.1 -1 SIPTrunk Endpoint(f483a6a0) PreProcessMsg calling CheckMinSEField, MinSE value is set to 1800 in the header
  10207592mS Sip: 20.1070.1 -1 SIPTrunk Endpoint(f483a6a0) Cloned 
  10207593mS Sip: 20.1070.1 -1 SIPTrunk Endpoint(f483a6a0) SendSIPResponse: INVITE code 100 SENT TO 10.20.240.8 5060
  10207593mS SIP Call Tx: 20
                    SIP/2.0 100 Trying
                    Via: SIP/2.0/UDP 10.20.240.8:5060;rport;branch=z9hG4bK802d8fddbc32-30-1a2f6b
                    Via: SIP/2.0/UDP 10.20.252.112:5060;rport=5060;branch=z9hG4bKPj51ed07898c6c477bb66e7f661a09ba8c
                    Record-Route: <sip:10.20.240.8:5060;ftag=5de879626607477783ce8d67f1ab4362;lr>
                    From: "Phillip" <sip:20899@10.20.240.8>;tag=5de879626607477783ce8d67f1ab4362
                    Call-ID: f325bfcd312c46818dbde46c648fbec7
                    CSeq: 22955 INVITE
                    Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,NOTIFY,UPDATE
                    Supported: timer
                    Server: IP Office 9.0.7.0 build 983
                    To: "20211" <sip:20211@10.20.245.1>;tag=d30a68f195527827
                    Content-Length: 0
                    
  10207594mS SIP Tx: UDP 10.20.245.1:5060 -> 10.20.240.8:5060
                    SIP/2.0 100 Trying
                    Via: SIP/2.0/UDP 10.20.240.8:5060;rport;branch=z9hG4bK802d8fddbc32-30-1a2f6b
                    Via: SIP/2.0/UDP 10.20.252.112:5060;rport=5060;branch=z9hG4bKPj51ed07898c6c477bb66e7f661a09ba8c
                    Record-Route: <sip:10.20.240.8:5060;ftag=5de879626607477783ce8d67f1ab4362;lr>
                    From: "Phillip" <sip:20899@10.20.240.8>;tag=5de879626607477783ce8d67f1ab4362
                    Call-ID: f325bfcd312c46818dbde46c648fbec7
                    CSeq: 22955 INVITE
                    Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,NOTIFY,UPDATE
                    Supported: timer
                    Server: IP Office 9.0.7.0 build 983
                    To: "20211" <sip:20211@10.20.245.1>;tag=d30a68f195527827
                    Content-Length: 0
                    
  10207594mS Sip: 20.1070.1 -1 SIPTrunk Endpoint(f483a6a0) INVITE Received ep 20.1070.1 -1 SIPTrunk Endpoint(f4778118), dialog f483a6a0 
  10207595mS Sip: 20.1070.1 -1 SIPTrunk Endpoint(f483a6a0) LR is On and route is Route: <sip:10.20.240.8:5060;ftag=5de879626607477783ce8d67f1ab4362;lr>
  10207595mS Sip: SIPTrunkEndpointDialogOwner::SetRemoteAddressForRequest from 10.20.240.8:5060 to 10.20.240.8:5060
  10207595mS Sip: 20.1070.1 -1 SIPTrunk Endpoint(f483a6a0) UpdateSIPCallState SIPDialog::INITIAL(0) -> SIPDialog::INVITE_RCVD(9)
  10207595mS Sip: 20.1070.1 -1 SIPTrunk Endpoint(f483a6a0) UpdateSDPState SIPDialog::IDLE(0) -> SIPDialog::OFFER_RCVD(2)
  10207596mS Sip: 20.1070.1 -1 SIPTrunk Endpoint(f483a6a0) SdpClone 
  10207596mS Sip: 20.1070.1 -1 SIPTrunk Endpoint(f483a6a0) SIPDialog::BuildFastStartFromSDPOffer sdpmsg f475a148
  10207596mS Sip: 20.1070.1 -1 SIPTrunk Endpoint(f483a6a0) FindConnectionParameters: found bandwidth info, bw_count=1, memorizing only first one
  10207596mS Sip: 20.1070.1 -1 SIPTrunk Endpoint(f483a6a0) SIPDialog::FindConnectionParameters: found media proto <RTP/AVP>
  10207596mS Sip: 20.1070.1 -1 SIPTrunk Endpoint(f483a6a0) SetRfc2833TxPayload: use RFC2833 for dtmf 
  10207596mS Sip: 20.1070.1 -1 SIPTrunk Endpoint(f483a6a0) SIPDialog::BuildFastStartFromAudioMediaSDPOffer reinvite 0 msg f53bd558 fs in msg 1 mRTP_PType 255
  10207597mS Sip: 20.1070.1 -1 SIPTrunk Endpoint(f483a6a0) SetRemoteRTPAddress to 10.20.240.8:10172 
  10207597mS Sip: 20.1070.1 -1 SIPTrunk Endpoint(f483a6a0) FindConnectionParameters: found bandwidth info, bw_count=1, memorizing only first one
  10207597mS Sip: 20.1070.1 -1 SIPTrunk Endpoint(f483a6a0) Process SIP request dialog f483a6a0, method INVITE in state SIPDialog::INVITE_RCVD(9)
  10207598mS Sip: 20.1070.1 -1 SIPTrunk Endpoint(f483a6a0) ProcessInboundSIPRequest calling CheckMinSEField, MinSE value is set to 90 in the header
  10207598mS Sip: 20.1070.1 -1 SIPTrunk Endpoint(f483a6a0) ProcessInboundSIPRequest calling CheckSessionExpiresField, session expires is 1800 refresher_is_ipo 0
  10207598mS Sip: 20.1070.1 -1 SIPTrunk Endpoint(f483a6a0) ProcessInboundSIPRequest calling CheckMinSEField, MinSE value is set to 1800 in the header
  10207599mS Sip: 20.1070.1 -1 SIPTrunk Endpoint(f483a6a0) UpdateClone mMesg f4831568 smsg f48317f0 
  10207599mS Sip: 20.1070.1 -1 SIPTrunk Endpoint(f483a6a0) CMSetup forwarded to call model owner_ep 20.1070.1 -1 SIPTrunk Endpoint(f4778118), dialog f483a6a0 has sdp 1
  10207605mS CMLOGGING:     CALL:2018/06/2121:19,00:00:00,000,20899@10.20.240.8,I,20211,20211,Phillip,,,0,,""n/a,0
  10207606mS Sip: 20.1070.1 -1 SIPTrunk Endpoint(f4778118) received CMReleaseComp
  10207606mS Sip: 20.1070.1 -1 SIPTrunk Endpoint(f483a6a0) Terminating dialog f483a6a0, state SIPDialog::INVITE_RCVD(9) for cause CMCauseUnallocatedNumber
  10207606mS Sip: 20.1070.1 -1 SIPTrunk Endpoint(f483a6a0) SendSIPResponse: INVITE code 404 SENT TO 10.20.240.8 5060
  10207607mS SIP Call Tx: 20
                    SIP/2.0 404 Not Found
                    Via: SIP/2.0/UDP 10.20.240.8:5060;rport;branch=z9hG4bK802d8fddbc32-30-1a2f6b
                    Via: SIP/2.0/UDP 10.20.252.112:5060;rport=5060;branch=z9hG4bKPj51ed07898c6c477bb66e7f661a09ba8c
                    Record-Route: <sip:10.20.240.8:5060;ftag=5de879626607477783ce8d67f1ab4362;lr>
                    From: "Phillip" <sip:20899@10.20.240.8>;tag=5de879626607477783ce8d67f1ab4362
                    Call-ID: f325bfcd312c46818dbde46c648fbec7
                    CSeq: 22955 INVITE
                    Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,NOTIFY,UPDATE
                    Supported: timer
                    Server: IP Office 9.0.7.0 build 983
                    Reason: Q.850;cause=1;text="Unallocated (unassigned) number"
                    To: "20211" <sip:20211@10.20.245.1>;tag=d30a68f195527827
                    Content-Length: 0
                    
  10207607mS SIP Tx: UDP 10.20.245.1:5060 -> 10.20.240.8:5060
                    SIP/2.0 404 Not Found
                    Via: SIP/2.0/UDP 10.20.240.8:5060;rport;branch=z9hG4bK802d8fddbc32-30-1a2f6b
                    Via: SIP/2.0/UDP 10.20.252.112:5060;rport=5060;branch=z9hG4bKPj51ed07898c6c477bb66e7f661a09ba8c
                    Record-Route: <sip:10.20.240.8:5060;ftag=5de879626607477783ce8d67f1ab4362;lr>
                    From: "Phillip" <sip:20899@10.20.240.8>;tag=5de879626607477783ce8d67f1ab4362
                    Call-ID: f325bfcd312c46818dbde46c648fbec7
                    CSeq: 22955 INVITE
                    Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,NOTIFY,UPDATE
                    Supported: timer
                    Server: IP Office 9.0.7.0 build 983
                    Reason: Q.850;cause=1;text="Unallocated (unassigned) number"
                    To: "20211" <sip:20211@10.20.245.1>;tag=d30a68f195527827
                    Content-Length: 0
                    
  10207608mS Sip: 20.1070.1 -1 SIPTrunk Endpoint(f483a6a0) UpdateSIPCallState SIPDialog::INVITE_RCVD(9) -> SIPDialog::FINAL(28)
  10207609mS Sip: 20.1070.1 -1 SIPTrunk Endpoint(f483a6a0) KeepDlgOnCmCallLost SIPDialog::FINAL
  10207609mS Sip: SIPDialog f483a6a0 deleted, dialogs 0
  10207610mS Sip: ~SipTrunkEndpoint 20.1070.1 -1 SIPTrunk Endpoint
  10207613mS SIP Rx: UDP 10.20.240.8:5060 -> 10.20.245.1:5060
                    ACK sip:20211@10.20.245.1 SIP/2.0
                    Via: SIP/2.0/UDP 10.20.240.8:5060;rport;branch=z9hG4bK802d8fddbc32-30-1a2f6b
                    Via: SIP/2.0/UDP 10.20.252.112:5060;rport=5060;branch=z9hG4bKPj51ed07898c6c477bb66e7f661a09ba8c
                    Max-Forwards: 69
                    From: "Phillip" <sip:20899@10.20.240.8>;tag=5de879626607477783ce8d67f1ab4362
                    To: "20211" <sip:20211@10.20.245.1>;tag=d30a68f195527827
                    Call-ID: f325bfcd312c46818dbde46c648fbec7
                    CSeq: 22955 ACK
                    Record-Route: <sip:10.20.240.8:5060;ftag=5de879626607477783ce8d67f1ab4362;lr>
                    Content-Length: 0
                    
  10207615mS Sip: Association found trunk: SIP Line (20)
  10207615mS SIP Call Rx: 20
                    ACK sip:20211@10.20.245.1 SIP/2.0
                    Via: SIP/2.0/UDP 10.20.240.8:5060;rport;branch=z9hG4bK802d8fddbc32-30-1a2f6b
                    Via: SIP/2.0/UDP 10.20.252.112:5060;rport=5060;branch=z9hG4bKPj51ed07898c6c477bb66e7f661a09ba8c
                    Max-Forwards: 69
                    From: "Phillip" <sip:20899@10.20.240.8>;tag=5de879626607477783ce8d67f1ab4362
                    To: "20211" <sip:20211@10.20.245.1>;tag=d30a68f195527827
                    Call-ID: f325bfcd312c46818dbde46c648fbec7
                    CSeq: 22955 ACK
                    Record-Route: <sip:10.20.240.8:5060;ftag=5de879626607477783ce8d67f1ab4362;lr>
                    Content-Length: 0
                    
  10212615mS Sip: sip_indicateTimeOut Timer 9
  10212615mS Sip: Timer 9 callback didn't find dialog, method INVITE, callid f325bfcd312c46818dbde46c648fbec7

holdmusic34 · Jun 22, 2018

Incoming call route will be something like this:
Received number 20XXX
Destination 20N

Also advise running monitor with a default filter as well as enabling the SIP filters. You will pick up CM and targeting events as well then.

A madman with a taste for speed.

phillip.bcs · Jun 22, 2018

holdmusic34,
Thank you! I was told that putting wildcards in ICRs was bad practice. I definitely agree when working with incoming calls, but when it's between two internal systems, I figured there had to be a better way.

Your suggestion put me on the right track. You cannot use 20N as a destination because "N" is not a valid destination in ICR. You can, however use "#". In this case, 20XXX for an incoming number is looking for "20" and three digits "XXX". The destination as 20# then translates to "20" and whatever was dialed as the three trailing digits. Thus, a single "#" represents whatever and however many "X" are in the dialed number.

I've got things working! I very much appreciate everyone's help.

sizbut · Jun 22, 2018

putting wildcards in ICRs was bad practice" - tell us who said that and we'll happily educate them back into the real world.

Nothing wrong with it/a lot to recommend it.

Stuck in a never ending cycle of file copying.

phillip.bcs · Jun 22, 2018

Specifically, it relates to incoming DIDs. I was taught (and it makes sense to me) that it is better to put each number in fully so you can easily make changes, detect issues, put in temporary redirects, etc. It was my mistake to take that to mean you should not use them at all.

holdmusic34 · Jun 22, 2018

The more accurate rule can take priority.
Genitaly, I would make a full number iCR for the main number, the fax line and any other important services.
I don't call it a wildcard. I call it a catch all as it cleans up the rest of the unassigned pool and any numbers that haven't been correctly configured.

A madman with a taste for speed.

IPGuru · Jun 26, 2018

I prefer to see the full number in the ICR for each DDI as it efectivy self documents the system when working on it remotely

That said a wild card entry can also be appropriate in certain instances, this is probably one of them.

Do things on the cheap & it will cost you dear

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