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integration problem between Call Manager 10.0 and third-party SIP

Marione77
Level 1
Level 1

Hi,

we are configuring a third-party SIP device (EXTRON DMP 128 Plus - the Extron guide for CUCM side configuration attached) but we encounter problems in internal calls coming into this device, while outgoing calls work regularly. We have 2 CUCM platforms one version is CUCM 10.0 and another 11.5 on both there is the same problem, the logs of this device in the incoming calls are the following

2018-06-22T16:27:25Z

    --- SIP-MESSAGE ( IN) ---

    INVITE sip:7770@172.17.80.91:5060 SIP/2.0

    Via: SIP/2.0/UDP 172.17.0.35:5060;branch=z9hG4bKc387be331f6da2

    Max-Forwards: 22

    Contact: <sip:03356936381@172.17.0.35:5060>

    To: <sip:7770@172.17.0.35:5060>

    From: <sip:03356936381@172.17.0.35>;tag=138770598~02b651c0-ca69-4ef4-b5d2-7f6506ffcd7d-50367657

    Call-ID: 4eea6700-b2d1072f-48c249-230011ac@172.17.0.35

   CSeq: 101 INVITE

    Expires: 180

    Min-SE: 3600

    Alert-Info: <file://Bellcore-dr2/>

    Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY

    Date: Fri, 22 Jun 2018 14:26:55 GMT

    Supported: timer, resource-priority, replaces

    User-Agent: Cisco-CUCM10.0

    Allow-Events: presence

    Remote-Party-ID: <sip:03356936381@172.17.0.35;x-cisco-callback-number=03356936381>;party=calling;screen=yes;privacy=off

    Send-Info: conference, x-cisco-conference

    Content-Length: 0

   

 2018-06-22T16:27:25Z

    --- SIP-MESSAGE (OUT) ---

    SIP/2.0 501 Not Implemented

    Via: SIP/2.0/UDP 172.17.0.35:5060;branch=z9hG4bKc387be331f6da2

    To: <sip:7770@172.17.0.35:5060>;tag=652c643d

    From: <sip:03356936381@172.17.0.35>;tag=138770598~02b651c0-ca69-4ef4-b5d2-7f6506ffcd7d-50367657

    Call-ID: 4eea6700-b2d1072f-48c249-230011ac@172.17.0.35

    CSeq: 101 INVITE

    User-Agent: EESB 1.03.0009

    Content-Length: 0

their guide, in the configuration of the Device SIP Profile, specifies the following:
When configuring the Trunk, Early Offer Support must be set to Mandatory. An MTP should be used if required.

we actually configure ourselves as required but their engineering tells us that incoming calls do not use the Early Offer.

 

below are the logs of a call generated by the DMP 128 Plus in which the early generated offer can be clearly seen

2018-06-22T16:26:29Z

    --- SIP-MESSAGE (OUT) ---

    INVITE sip:03356936381@172.17.0.35:5060 SIP/2.0

    Via: SIP/2.0/ 172.17.80.91:5060;branch=z9hG4bK-524287-1---8ce5fe704a6c8d64;rport

    Max-Forwards: 70

    Contact: <sip:7770;ob>;+sip.instance="<urn:uuid:229239d7-4aa4-4cb0-b5d8-2ab8a2b01ea9>"

    To: <sip:03356936381@172.17.0.35>;tag=138769876~02b651c0-ca69-4ef4-b5d2-7f6506ffcd7d-43592825

    From: "Sala CDA NPU"<sip:7770@172.17.0.35:5060>;tag=02d51c37

    Call-ID: s63q4WiyxlxP1OQyGd0s1g..

    CSeq: 2 INVITE

    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, INFO

    Content-Type: application/sdp

    Supported: outbound, path

    User-Agent: EESB 1.03.0009

    Content-Length: 218

   

    v=0

    o=user 1792 1793 IN IP4 172.17.80.91

    s=-

    c=IN IP4 172.17.80.91

    t=0 0

    m=audio 50050 RTP/AVP 18 110

    c=IN IP4 172.17.80.91

    a=ptime:20

    a=rtpmap:18 G729/8000

    a=rtpmap:110 telephone-event/8000

    a=fmtp:110 0-16

What should be done to get the callmanager to use the early offer ? 

 

Thanks for the help,

Francesco

 

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