SIP trunk call to CTI port media endpoint problem

Hi, I have the following scenario:
CUCM ---SIP-----CUBE----SIP----ITSP
CUCM cluster has two servers 10.1.9.5 and 10.1.9.6 (subscriber), we have a CTI application running on 10.1.9.8, we got a bunch of SIP DIDs from ITSP and those SIP DID numbers are mapped to CUCM internal DNs on CUBE through translation, inbound SIP calls to mapped SCCP phones work fine, media stream of internal call leg is established directly between SCCP phone and CUBE as expected. However, when inbound SIP calls to the number which maps to CTI extention, there is no audio, debug on CUBE shows that the media stream of internal call leg is between one of the CUCM servers. I want to understand why CUCM is telling CUBE to use the IP address which is NOT CTI port is registered from (in this case 10.1.9.8)? why would CUCM treat a CTI extention differently from a regular SCCP phone extension. For regular inbound call to SCCP phones codec between CUCM-CUBE and between CUBE-ITSP are both g11ulaw.
Thanks,

"debug on CUBE shows that the media stream of internal call leg is  between one of the CUCM servers" ==>
debug on CUBE shows that the media stream of internal call leg is  between 10.1.9.5 (happens to be MTP) and CUBE

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    State of The Call        : STATE_ACTIVE
    TCP Sockets Used         : NO
    Calling Number           : 27218091323
    Called Number            : 0862000000
    Source IP Address (Sig  ): 10.18.81.2
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    Destn SIP Resp Addr:Port : 10.111.111.254:5060
    Destination Name         : 10.111.111.254
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    Number of Media Streams: 1
    Media Stream             : 1
    Negotiated Codec         : g729br8
    Negotiated Codec Bytes   : 20
    Nego. Codec payload      : 18 (tx), 18 (rx)
    Negotiated Dtmf-relay    : 6
    Dtmf-relay Payload       : 101 (tx), 101 (rx)
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    Source IP Port    (Media): 22256
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    Destn  IP Port    (Media): 20074
    Orig Destn IP Address:Port (Media): [ - ]:0
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    Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e72075e3a02c1
    From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
    To: <sip:[email protected]>;tag=3C365010-1E42
    Date: Sun, 30 Nov 2014 11:44:29 GMT
    Call-ID: [email protected]
    Max-Forwards: 70
    CSeq: 101 ACK
    Allow-Events: presence, kpml
    Content-Type: application/sdp
    Content-Length: 236
    v=0
    o=CiscoSystemsCCM-SIP 9082578 1 IN IP4 10.18.81.11
    s=SIP Call
    c=IN IP4 10.18.80.40
    b=TIAS:8000
    b=AS:8
    t=0 0
    m=audio 21928 RTP/AVP 18 101
    a=rtpmap:18 G729/8000
    a=ptime:20
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    Nov 30 11:44:31.730: //64510/3CD5D2000001/SIP/Call/sipSPICallInfo: 
    The Call Setup Information is:
    Call Control Block (CCB) : 0x0x3D816D70
    State of The Call        : STATE_ACTIVE
    TCP Sockets Used         : NO
    Calling Number           : 0218091323
    Called Number            : 0862000000
    Source IP Address (Sig  ): 10.18.81.2
    Destn SIP Req Addr:Port  : 10.18.81.11:5060
    Destn SIP Resp Addr:Port : 10.18.81.11:5060
    Destination Name         : 10.18.81.11
    Nov 30 11:44:31.730: //64510/3CD5D2000001/SIP/Call/sipSPIMediaCallInfo: 
    Number of Media Streams: 1
    Media Stream             : 1
    Negotiated Codec         : g729br8
    Negotiated Codec Bytes   : 20
    Nego. Codec payload      : 18 (tx), 18 (rx)
    Negotiated Dtmf-relay    : 6
    Dtmf-relay Payload       : 101 (tx), 101 (rx)
    Source IP Address (Media): 10.18.81.2
    Source IP Port    (Media): 22350
    Destn  IP Address (Media): 10.18.80.40
    Destn  IP Port    (Media): 21928
    Orig Destn IP Address:Port (Media): [ - ]:0
    Nov 30 11:44:31.730: //64510/3CD5D2000001/SIP/Call/sipSPICallInfo: 
    The Call Setup Information is:
    Call Control Block (CCB) : 0x0x3D816D70
    State of The Call        : STATE_ACTIVE
    TCP Sockets Used         : NO
    Calling Number           : 0218091323
    Called Number            : 0862000000
    Source IP Address (Sig  ): 10.18.81.2
    Destn SIP Req Addr:Port  : 10.18.81.11:5060
    Destn SIP Resp Addr:Port : 10.18.81.11:5060
    Destination Name         : 10.18.81.11
    PM-HO-VG-01#
    Nov 30 11:44:31.730: //64510/3CD5D2000001/SIP/Call/sipSPIMediaCallInfo: 
    Number of Media Streams: 1
    Media Stream             : 1
    Negotiated Codec         : g729br8
    Negotiated Codec Bytes   : 20
    Nego. Codec payload      : 18 (tx), 18 (rx)
    Negotiated Dtmf-relay    : 6
    Dtmf-relay Payload       : 101 (tx), 101 (rx)
    Source IP Address (Media): 10.18.81.2
    Source IP Port    (Media): 22350
    Destn  IP Address (Media): 10.18.80.40
    Destn  IP Port    (Media): 21928
    Orig Destn IP Address:Port (Media): [ - ]:0
    PM-HO-VG-01#sh sip
    PM-HO-VG-01#sh sip-ua call
    PM-HO-VG-01#sh sip-ua calls 
    Total SIP call legs:2, User Agent Client:1, User Agent Server:1
    SIP UAC CALL INFO
    Call 1
    SIP Call ID                : [email protected]
       State of the call       : STATE_ACTIVE (7)
       Substate of the call    : SUBSTATE_NONE (0)
       Calling Number          : 27218091323
       Called Number           : 0862000000
       Bit Flags               : 0xC04018 0x10000100 0x0
       CC Call ID              : 64511
       Source IP Address (Sig ): 10.18.81.2
       Destn SIP Req Addr:Port : [10.111.111.254]:5060
       Destn SIP Resp Addr:Port: [10.111.111.254]:5060
       Destination Name        : 10.111.111.254
       Number of Media Streams : 1
       Number of Active Streams: 1
       RTP Fork Object         : 0x0
       Media Mode              : flow-through
       Media Stream 1
         State of the stream      : STREAM_ACTIVE
         Stream Call ID           : 64511
         Stream Type              : voice+dtmf (0)
         Stream Media Addr Type   : 1
         Negotiated Codec         : g729br8 (20 bytes)
         Codec Payload Type       : 18 
         Negotiated Dtmf-relay    : rtp-nte
         Dtmf-relay Payload Type  : 101
         QoS ID                   : -1
         Local QoS Strength       : BestEffort
         Negotiated QoS Strength  : BestEffort
         Negotiated QoS Direction : None
         Local QoS Status         : None
         Media Source IP Addr:Port: [10.18.81.2]:22256
         Media Dest IP Addr:Port  : [10.111.111.254]:20074
    Options-Ping    ENABLED:NO    ACTIVE:NO
       Number of SIP User Agent Client(UAC) calls: 1
    SIP UAS CALL INFO
    Call 1
    SIP Call ID                : [email protected]
       State of the call       : STATE_ACTIVE (7)
       Substate of the call    : SUBSTATE_NONE (0)
       Calling Number          : 0218091323
       Called Number           : 0862000000
       Bit Flags               : 0xC0401E 0x10000100 0x80004
       CC Call ID              : 64510
       Source IP Address (Sig ): 10.18.81.2
       Destn SIP Req Addr:Port : [10.18.81.11]:5060
       Destn SIP Resp Addr:Port: [10.18.81.11]:5060
       Destination Name        : 10.18.81.11
       Number of Media Streams : 1
       Number of Active Streams: 1
       RTP Fork Object         : 0x0
       Media Mode              : flow-through
       Media Stream 1
         State of the stream      : STREAM_ACTIVE
         Stream Call ID           : 64510
         Stream Type              : voice+dtmf (1)
         Stream Media Addr Type   : 1
         Negotiated Codec         : g729br8 (20 bytes)
         Codec Payload Type       : 18 
         Negotiated Dtmf-relay    : rtp-nte
         Dtmf-relay Payload Type  : 101
         QoS ID                   : -1
         Local QoS Strength       : BestEffort
         Negotiated QoS Strength  : BestEffort
         Negotiated QoS Direction : None
         Local QoS Status         : None
         Media Source IP Addr:Port: [10.18.81.2]:22350
         Media Dest IP Addr:Port  : [10.18.80.40]:21928
    Options-Ping    ENABLED:NO    ACTIVE:NO
       Number of SIP User Agent Server(UAS) calls: 1
    PM-HO-VG-01#
    PM-HO-VG-01#
    PM-HO-VG-01#
    As you can see, the call is connected and everything is working perfectly. When I press the hold button, here is what I get:
    NOTE: I have # debug ccsip messages and #debug ccsip calls (running)
    PM-HO-VG-01#
    PM-HO-VG-01#
    Nov 30 11:44:49.210: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received: 
    INVITE sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720852ab8b92
    From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
    To: <sip:[email protected]>;tag=3C365010-1E42
    Date: Sun, 30 Nov 2014 11:44:49 GMT
    Call-ID: [email protected]
    Supported: timer,resource-priority,replaces
    Min-SE:  1800
    User-Agent: Cisco-CUCM9.1
    Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
    CSeq: 102 INVITE
    Max-Forwards: 70
    Expires: 180
    Allow-Events: presence
    Supported: X-cisco-srtp-fallback
    Supported: Geolocation
    Contact: <sip:[email protected]:5060>
    Content-Type: application/sdp
    Content-Length: 244
    v=0
    o=CiscoSystemsCCM-SIP 9082578 2 IN IP4 10.18.81.11
    s=SIP Call
    c=IN IP4 0.0.0.0
    b=TIAS:8000
    b=AS:8
    t=0 0
    m=audio 21928 RTP/AVP 18 101
    a=rtpmap:18 G729/8000
    a=ptime:20
    a=inactive
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    Nov 30 11:44:49.218: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Sent: 
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720852ab8b92
    From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
    To: <sip:[email protected]>;tag=3C365010-1E42
    Date: Sun, 30 Nov 2014 11:44:49 GMT
    Call-ID: [email protected]
    CSeq: 102 INVITE
    Allow-Events: telephone-event
    Server: Cisco-SIPGateway/IOS-15.2.4.M5
    Content-Length: 0
    Nov 30 11:44:49.218: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Sent: 
    INVITE sip:[email protected]:5060;transport=udp SIP/2.0
    Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC9241
    From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
    To: <sip:[email protected]>;tag=71913148-1417348035284
    Date: Sun, 30 Nov 2014 11:44:49 GMT
    Call-ID: [email protected]
    Supported: 100rel,timer,resource-priority,replaces,sdp-anat
    Min-SE:  1800
    Cisco-Guid: 1020645888-0000065536-0000124117-0189862410
    User-Agent: Cisco-SIPGateway/IOS-15.2.4.M5
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    CSeq: 102 INVITE
    Max-Forwards: 70
    Timestamp: 1417347889
    Contact: <sip:[email protected]:5060>
    Call-Info: <sip:10.18.81.2:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
    Expires: 180
    Allow-Events: telephone-event
    Content-Type: application/sdp
    Content-Length: 271
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 7676 6959 IN IP4 10.18.81.2
    s=SIP Call
    c=IN IP4 0.0.0.0
    t=0 0
    m=audio 22256 RTP/AVP 18 101
    c=IN IP4 0.0.0.0
    a=inactive
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=yes
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=ptime:20
    Nov 30 11:44:49.278: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Received: 
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC9241
    From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
    To: <sip:[email protected]>;tag=71913148-1417348035284
    Call-ID: [email protected]
    CSeq: 102 INVITE
    Timestamp: 1417347889
    Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
    Supported: 
    Accept: application/media_control+xml,application/sdp,application/xml
    Contact: <sip:[email protected]:5060;transport=udp>
    X-BroadWorks-Correlation-Info: bbf94839-a234-4237-95e6-a7037322f0f4
    Content-Type: application/sdp
    Content-Length: 360
    v=0
    o=BroadWorks 316169737 2 IN IP4 10.111.111.254
    s=-
    c=IN IP4 0.0.0.0
    t=0 0
    m=audio 20074 RTP/AVP 18 101 100
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=rtpmap:100 X-NSE/8000
    a=fmtp:100 200-202
    a=X-sqn:0
    a=X-cap: 1 audio RTP/AVP 100
    a=X-cpar: a=rtpmap:100 X-NSE/8000
    a=X-cpar: a=fmtp:100 200-202
    a=X-cap: 2 image udptl t38
    a=inactive
    Nov 30 11:44:49.278: //64511/3CD5D2000001/SIP/Call/sipSPICallInfo: 
    The Call Setup Information is:
    Call Control Block (CCB) : 0x0x3D7B1458
    State of The Call        : STATE_ACTIVE
    TCP Sockets Used         : NO
    Calling Number           : 27218091323
    Called Number            : 0862000000
    Source IP Address (Sig  ): 10.18.81.2
    Destn SIP Req Addr:Port  : 10.111.111.254:5060
    Destn SIP Resp Addr:Port : 10.111.111.254:5060
    Destination Name         : 10.111.111.254
    Nov 30 11:44:49.278: //64511/3CD5D2000001/SIP/Call/sipSPIMediaCallInfo: 
    Number of Media Streams: 1
    Media Stream             : 1
    Negotiated Codec         : g729br8
    Negotiated Codec Bytes   : 20
    Nego. Codec payload      : 18 (tx), 18 (rx)
    Negotiated Dtmf-relay    : 6
    Dtmf-relay Payload       : 101 (tx), 101 (rx)
    Source IP Address (Media): 10.18.81.2
    Source IP Port    (Media): 22256
    Destn  IP Address (Media): 0.0.0.0
    Destn  IP Port    (Media): 20074
    Orig Destn IP Address:Port (Media): [ - ]:0
    Nov 30 11:44:49.282: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Sent: 
    ACK sip:[email protected]:5060;transport=udp SIP/2.0
    Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECA2633
    From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
    To: <sip:[email protected]>;tag=71913148-1417348035284
    Date: Sun, 30 Nov 2014 11:44:49 GMT
    Call-ID: [email protected]
    Max-Forwards: 70
    CSeq: 102 ACK
    Allow-Events: telephone-event
    Content-Length: 0
    Nov 30 11:44:49.282: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Sent: 
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720852ab8b92
    From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
    To: <sip:[email protected]>;tag=3C365010-1E42
    Date: Sun, 30 Nov 2014 11:44:49 GMT
    Call-ID: [email protected]
    CSeq: 102 INVITE
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    Allow-Events: telephone-event
    Contact: <sip:[email protected]:5060>
    Supported: replaces
    Supported: sdp-anat
    Server: Cisco-SIPGateway/IOS-15.2.4.M5
    Supported: timer
    Content-Type: application/sdp
    Content-Length: 271
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 7965 2748 IN IP4 10.18.81.2
    s=SIP Call
    c=IN IP4 0.0.0.0
    t=0 0
    m=audio 22350 RTP/AVP 18 101
    c=IN IP4 0.0.0.0
    a=inactive
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=yes
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=ptime:20
    Nov 30 11:44:49.282: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received: 
    ACK sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e72094953dfea
    From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
    To: <sip:[email protected]>;tag=3C365010-1E42
    Date: Sun, 30 Nov 2014 11:44:49 GMT
    Call-ID: [email protected]
    Max-Forwards: 70
    CSeq: 102 ACK
    Allow-Events: presence
    Content-Length: 0
    Nov 30 11:44:49.290: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received: 
    INVITE sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720a6918040f
    From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
    To: <sip:[email protected]>;tag=3C365010-1E42
    Date: Sun, 30 Nov 2014 11:44:49 GMT
    Call-ID: [email protected]
    Supported: timer,resource-priority,replaces
    Min-SE:  1800
    User-Agent: Cisco-CUCM9.1
    Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
    CSeq: 103 INVITE
    Max-Forwards: 70
    Expires: 180
    Allow-Events: presence
    Supported: X-cisco-srtp-fallback
    Supported: Geolocation
    Contact: <sip:[email protected]:5060>
    Content-Length: 0
    Nov 30 11:44:49.294: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Sent: 
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720a6918040f
    From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
    To: <sip:[email protected]>;tag=3C365010-1E42
    Date: Sun, 30 Nov 2014 11:44:49 GMT
    Call-ID: [email protected]
    CSeq: 103 INVITE
    Allow-Events: telephone-event
    Server: Cisco-SIPGateway/IOS-15.2.4.M5
    Content-Length: 0
    Nov 30 11:44:49.294: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Sent: 
    INVITE sip:[email protected]:5060;transport=udp SIP/2.0
    Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECB16F3
    From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
    To: <sip:[email protected]>;tag=71913148-1417348035284
    Date: Sun, 30 Nov 2014 11:44:49 GMT
    Call-ID: [email protected]
    Supported: timer,resource-priority,replaces,sdp-anat
    Min-SE:  1800
    Cisco-Guid: 1020645888-0000065536-0000124117-0189862410
    User-Agent: Cisco-SIPGateway/IOS-15.2.4.M5
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    CSeq: 103 INVITE
    Max-Forwards: 70
    Timestamp: 1417347889
    Contact: <sip:[email protected]:5060>
    Expires: 180
    Allow-Events: telephone-event
    Content-Length: 0
    Nov 30 11:44:49.338: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Received: 
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECB16F3
    From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
    To: <sip:[email protected]>;tag=71913148-1417348035284
    Call-ID: [email protected]
    CSeq: 103 INVITE
    Timestamp: 1417347889
    Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
    Supported: 
    Accept: application/media_control+xml,application/sdp,application/xml
    Contact: <sip:[email protected]:5060;transport=udp>
    Content-Type: application/sdp
    Content-Length: 306
    v=0
    o=BroadWorks 316169737 3 IN IP4 10.111.111.254
    s=-
    c=IN IP4 10.111.111.254
    t=0 0
    m=audio 20074 RTP/AVP 18 101
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=X-sqn:0
    a=X-cap: 1 audio RTP/AVP 100
    a=X-cpar: a=rtpmap:100 X-NSE/8000
    a=X-cpar: a=fmtp:100 200-202
    a=X-cap: 2 image udptl t38
    Nov 30 11:44:49.342: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Sent: 
    SIP/2.0 2
    PM-HO-VG-01#00 OK
    Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720a6918040f
    From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
    To: <sip:[email protected]>;tag=3C365010-1E42
    Date: Sun, 30 Nov 2014 11:44:49 GMT
    Call-ID: [email protected]
    CSeq: 103 INVITE
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    Allow-Events: telephone-event
    Contact: <sip:[email protected]:5060>
    Supported: replaces
    Supported: sdp-anat
    Server: Cisco-SIPGateway/IOS-15.2.4.M5
    Supported: timer
    Content-Type: application/sdp
    Content-Length: 289
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 7965 2749 IN IP4 10.18.81.2
    s=SIP Call
    c=IN IP4 10.18.81.2
    t=0 0
    m=audio 22350 RTP/AVP 18 101 19
    c=IN IP4 10.18.81.2
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=yes
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=rtpmap:19 CN/8000
    a=ptime:20
    Nov 30 11:44:49.350: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received: 
    ACK sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720b594cd517
    From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
    To: <sip:[email protected]>;tag=3C365010-1E42
    Date: Sun, 30 Nov 2014 11:44:49 GMT
    Call-ID: [email protected]
    Max-Forwards: 70
    CSeq: 103 ACK
    Allow-Events: presence
    Content-Type: application/sdp
    Content-Length: 213
    v=0
    o=CiscoSystemsCCM-SIP 9082578 3 IN IP4 10.18.81.11
    s=SIP Call
    c=IN IP4 10.18.81.10
    t=0 0
    m=audio 4000 RTP/AVP 18
    a=X-cisco-media:umoh
    a=rtpmap:18 G729/8000
    a=ptime:20
    a=fmtp:18 annexb=no
    a=sendonly
    Nov 30 11:44:49.354: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Sent: 
    BYE sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECC55
    From: <sip:[email protected]>;tag=3C365010-1E42
    To: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
    Date: Sun, 30 Nov 2014 11:44:49 GMT
    Call-ID: [email protected]
    User-Agent: Cisco-SIPGateway/IOS-15.2.4.M5
    Max-Forwards: 70
    Timestamp: 1417347889
    CSeq: 101 BYE
    Reason: Q.850;cause=86
    P-RTP-Stat: PS=874,OS=17480,PR=872,OR=17440,PL=0,JI=0,LA=0,DU=17
    Content-Length: 0
    Nov 30 11:44:49.354: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Sent: 
    BYE sip:[email protected]:5060;transport=udp SIP/2.0
    Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECD1ECD
    From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
    To: <sip:[email protected]>;tag=71913148-1417348035284
    Date: Sun, 30 Nov 2014 11:44:49 GMT
    Call-ID: [email protected]
    User-Agent: Cisco-SIPGateway/IOS-15.2.4.M5
    Max-Forwards: 70
    Timestamp: 1417347889
    CSeq: 104 BYE
    Reason: Q.850;cause=65
    P-RTP-Stat: PS=872,OS=17440,PR=952,OR=19040,PL=0,JI=0,LA=0,DU=17
    Content-Length: 0
    Nov 30 11:44:49.374: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Received: 
    SIP/2.0 200 Race Condition
    Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECD1ECD
    From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
    To: <sip:[email protected]>;tag=71913148-1417348035284
    Call-ID: [email protected]
    Timestamp: 1417347889
    CSeq: 104 BYE
    Content-Length: 0
    Nov 30 11:44:49.374: //64511/3CD5D2000001/SIP/Call/sipSPICallInfo: 
    The Call Setup Information is:
    Call Control Block (CCB) : 0x0x3D7B1458
    State of The Call        : STATE_DEAD
    TCP Sockets Used         : NO
    Calling Number           : 27218091323
    Called Number            : 0862000000
    Source IP Address (Sig  ): 10.18.81.2
    Destn SIP Req Addr:Port  : 10.111.111.254:5060
    Destn SIP Resp Addr:Port : 10.111.111.254:5060
    Destination Name         : 10.111.111.254
    Nov 30 11:44:49.374: //64511/3CD5D2000001/SIP/Call/sipSPIMediaCallInfo: 
    Number of Media Streams: 1
    Media Stream             : 1
    Negotiated Codec         : g729br8
    Negotiated Codec Bytes   : 20
    Nego. Codec payload      : 18 (tx), 18 (rx)
    Negotiated Dtmf-relay    : 6
    Dtmf-relay Payload       : 101 (tx), 101 (rx)
    Source IP Address (Media): 10.18.81.2
    Source IP Port    (Media): 22256
    Destn  IP Address (Media): 10.111.111.254
    Destn  IP Port    (Media): 20074
    Orig Destn IP Address:Port (Media): [ - ]:0
    Nov 30 11:44:49.374: //64511/3CD5D2000001/SIP/Call/sipSPICallInfo: 
    Disconnect Cause (CC)    : 65
    Disconnect Cause (SIP)   : 200
    Nov 30 11:44:49.406: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Received: 
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECC55
    From: <sip:[email protected]>;tag=3C365010-1E42
    To: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
    Date: Sun, 30 Nov 2014 11:44:49 GMT
    Call-ID: [email protected]
    CSeq: 101 BYE
    Content-Length: 0
    Nov 30 11:44:49.406: //64510/3CD5D2000001/SIP/Call/sipSPICallInfo: 
    The Call Setup Information is:
    Call Control Block (CCB) : 0x0x3D816D70
    State of The Call        : STATE_DEAD
    TCP Sockets Used         : NO
    Calling Number           : 0218091323
    Called Number            : 0862000000
    Source IP Address (Sig  ): 10.18.81.2
    Destn SIP Req Addr:Port  : 10.18.81.11:5060
    Destn SIP Resp Addr:Port : 10.18.81.11:5060
    Destination Name         : 10.18.81.11
    Nov 30 11:44:49.406: //64510/3CD5D2000001/SIP/Call/sipSPIMediaCallInfo: 
    Number of Media Streams: 1
    Media Stream             : 1
    Negotiated Codec         : g729br8
    Negotiated Codec Bytes   : 20
    Nego. Codec payload      : 18 (tx), 18 (rx)
    Negotiated Dtmf-relay    : 6
    Dtmf-relay Payload       : 101 (tx), 101 (rx)
    Source IP Address (Media): 10.18.81.2
    Source IP Port    (Media): 22350
    Destn  IP Address (Media): 0.0.0.0
    Destn  IP Port    (Media): 21928
    Orig Destn IP Address:Port (Media): [ - ]:0
    Nov 30 11:44:49.406: //64510/3CD5D2000001/SIP/Call/sipSPICallInfo: 
    Disconnect Cause (CC)    : 86
    Disconnect Cause (SIP)   : 200
    PM-HO-VG-01#

    Hi Manish,
    Again, excellent feedback. Much appreciated.
    I will try the commands suggested above and see if I can get DTMF to work correctly while interworking H.323 and SIP.
    But my ultimate goal is to have SIP all way from the CUCM to the CUBE and from the CUBE to the ITSP.
    If I enable SIP Early Offer with MTP on the CUCM going to the CUBE, all SIP Invite sent from the CUCM to the CUBE uses G.729r8 as the codec and once the call is established using G.729r8 should the ITSP reply with that codec, the call succeed and I am able to see an active MTP session using G.729 when I issue the command # show sccp connections.
    One thing that I saw is that my ITSP love so much sending G.729br8 most of the times, so even if using SIP EO with MTP on the SIP Trunk to the CUBE, when I sent my INVITE out from the CUCM to the CUBE using G.729r8, specially on call center numbers such as 0800 numbers, you will see that the call established but the codec being negotiated is G.729br8 which is voice only (missing DTMF).
    I will be doing some intensive test again later on this week and will send the logs. 
    Here is my question to both of you:
    Which is the best way of having a proper SIP to SIP setup all the way that will not pose any problem?
    Do I have to enable Early Offer on the SIP Profile used by the CUBE SIP Trunk or should I use the normal Standard SIP Profile? Do I need to enable MTP on my CUBE SIP Trunk or not?
    From the CUBE point of view, I have a voice class codec that support G.729r8 or G.729br8 and the DTMF Relay method supported by the ITSP is RFC 2833.
    I will send more logs for each scenario. I think that we are getting close to the resolution of this problem.
    Thanks again for your support fellows.

  • Calls via Gamme sip trunk

    Has anyone set up Lync server 2010 to use the Gamma SIP trunks, that dont require the use of a gateway?
    No requirement for an additional gateway device, with direct MS Lync connectivity
    The trouble is i cannot get Lync to connect to the trunks. We have purchased the SIP trunks from a gamma supplier(we didnt now they were a supplier, until recently when we asked for support and they went 'duhhhhhh me no know, we just
    sell things dunow how to set things up' what a PAIN IN THE A***), and they say that the SIP trunks are pointed at our EFM IP address. which also has the DDIs assigned to it.
    So, i setup a PSTN gateway on lync topology using IP of EFM, Listening port 5060 using TCP. Are these ports and protocol okay?
    The VoIP phones seem to want to call, they just lack any sound, no ringing tone, no dissconnected tone. Just says calling "+44157322****" So the dial plan is changing 22**** to the correct local code and whatever the +44 thing
    is.
    Any advice on how i can find the problem, or how to setup the trunks up would be hugely appreciated.
    P.S We initially tried to use an audiocodes mediant 1000, which was what we asked our trunk supplier about, and then they informed us about being a gamma supplier, and that the gamma trunks do not require a gateway. Followed setting
    up guide for mediant 1000 with gamma trunks through audiocodes blah, to no success. I think thats because it was changing the coders, which was not needed if the trunks are directly compatable.

    Hi,
    Please review the SIP trunk topology.
    http://technet.microsoft.com/en-us/library/gg398720.aspx
    To
     implement SIP trunking, you must route the connection through a Mediation Server, which proxies communications sessions between Lync Server 2010 clients and the service provider and transcodes media when necessary. Each Mediation Server has
    an internal and an external network interface. The internal interface connects to the Front End Servers. The external interface is commonly called the gateway interface because it has traditionally been used to connect the Mediation Server to a PSTN gateway
    or an IP-PBX. To implement a SIP trunk, you connect the external interface of the Mediation Server to the external edge component of the ITSP. The external edge component of the ITSP could be a Session Border Controller (SBC), a router, or a gateway.
    Generally the gateway is not required in your organization. You need to configure Mediation Server setting. For the details about
    the SIP trunk configuration of ITSP side, you need to contact Gamme Support for further assistance.
    Regards,
    Kent Huang
    TechNet Community Support ************************************************************************************************************************ Please remember to click “Mark as Answer” on the post that helps you, and to click “Unmark as Answer” if a
    marked post does not actually answer your question.

  • Video only enabled when call is initiated from one direction across SIP Trunk

    wonder If anyone can shed some light on this.
    I have an issue between two cucm clusters, tied together with a SIP trunk. 
    If we dial from Australia to the US there is two way video and audio.  If the US calls Australia, there is only audio.   I have run a test call from the US through VLT and have found the following SDP's  (see below). When The US make a video enabled call to australia the message "Video is not available, Remote party has video off" on the US phone screen.
    Both clusters have the SIP trunk set up with the same codec settings and video bandwidth between reqions and locations.  the SIP trunk is configured pretty much stock standard and identical at both ends, yet the SDP seem to want to negotiate different Video Parameters  (again see SDP's below). CUCM in australia is 10.61.2.82.
    what other settings can I check to get video to work when calls get initiated from either direction,...................
    both phones are SIP 8941's, again audio is no problem in both directions.
    =======this is from the phone in Australia to the CUCM in australia phone IP 10.61.4.112======================================
    45870304.002 |09:02:07.941 |AppInfo  |SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 10.61.4.112 on port 34271 index 53563 with 2089 bytes:
    [344530309,NET]
    SIP/2.0 200 OK
    Via: SIP/2.0/TCP 10.61.2.82:5060;branch=z9hG4bKe0103892bbb75
    From: "Anonymous" <sip:[email protected]>;tag=109791678~1b5af941-cea2-4a00-a0bd-15a532224d7d-59374526
    To: <sip:[email protected]>;tag=5057a887bfdd550c0d321a20-7f843426
    Call-ID: [email protected]
    Date: Wed, 29 Apr 2015 23:02:07 GMT
    CSeq: 101 INVITE
    Server: Cisco-CP8941/9.4.2
    Contact: <sip:[email protected]:34271;transport=tcp>;video
    Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFO
    Remote-Party-ID: "Dennis Mink - 33935" <sip:[email protected]>;party=called;id-type=subscriber;privacy=off;screen=yes
    Supported: replaces,join,sdp-anat,norefersub,resource-priority,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-7.0.0,X-cisco-xsi-8.5.1
    Allow-Events: kpml,dialog
    Recv-Info: conference
    Recv-Info: x-cisco-conference
    Content-Length: 966
    Content-Type: application/sdp
    Content-Disposition: session;handling=optional
    v=0
    o=Cisco-SIPUA 28123 0 IN IP4 10.61.4.112
    s=SIP Call
    t=0 0
    m=audio 16736 RTP/AVP 0 8 18 102 9 116 101
    c=IN IP4 10.61.4.112
    a=trafficclass:conversational.audio.avconf.aq:admitted
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=no
    a=rtpmap:102 L16/16000
    a=rtpmap:9 G722/8000
    a=rtpmap:116 iLBC/8000
    a=fmtp:116 mode=20
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=sendrecv
    m=video 16738 RTP/AVP 126 97
    c=IN IP4 10.61.4.112
    b=TIAS:2000000 
    a=trafficclass:conversational.video.avconf.aq:admitted   <----this is missing from US SDP
    a=rtpmap:126 H264/90000
    a=fmtp:126 profile-level-id=428014;packetization-mode=1;level-asymmetry-allowed=1;max-mbps=36000;max-fs=1200;max-rcmd-nalu-size=1300
    a=imageattr:126 send * recv [x=640,y=480]
    a=rtpmap:97 H264/90000
    a=fmtp:97 profile-level-id=428014;packetization-mode=0;level-asymmetry-allowed=1;max-mbps=36000;max-fs=1200
    a=imageattr:97 send * recv [x=640,y=480]
    a=rtcp-fb:* ccm tmmbr
    a=sendrecv
    ============below is coming from the US (phone IP is 10.1.109.81)================
    04/30/2015 09:02:08.169 Send 10.61.4.112 SIP ACK bfa99a00-541162ed-71da57-52023d0a NotAvail
    SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 10.61.4.112 on port 34271 index 53563 
    [344530326,NET]
    ACK sip:[email protected]:34271;transport=tcp SIP/2.0
    Via: SIP/2.0/TCP 10.61.2.82:5060;branch=z9hG4bKe010481f320b08
    From: "Anonymous" <sip:[email protected]>;tag=109791678~1b5af941-cea2-4a00-a0bd-15a532224d7d-59374526
    To: <sip:[email protected]>;tag=5057a887bfdd550c0d321a20-7f843426
    Date: Wed, 29 Apr 2015 23:02:05 GMT
    Call-ID: [email protected]
    User-Agent: Cisco-CUCM10.0
    Max-Forwards: 70
    CSeq: 101 ACK
    Allow-Events: presence
    Content-Type: application/sdp
    Content-Length: 456
    SDP Message
    ====================================================
    v=0
    o=CiscoSystemsCCM-SIP 109791678 1 IN IP4 10.61.2.82
    s=SIP Call
    c=IN IP4 10.1.109.81
    b=TIAS:8000
    b=AS:8
    t=0 0
    m=audio 16412 RTP/AVP 18 101
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=no
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=trafficclass:conversational.audio.aq:admitted   <---what does this do here, and how?
    m=video 0 RTP/SAVP 31 34 96 97      <-----------port 0. why?
    a=rtpmap:31 H261/90000
    a=rtpmap:34 H263/90000
    a=rtpmap:96 H263-1998/90000
    a=rtpmap:97 H264/90000
    a=content:main
    a=inactive

    Hi Dennis,
    On US phone SDP media attribute is inactive.
    a=rtpmap:97 H264/90000
    a=content:main
    a=inactive
    Are you sure that audio works ? Can you please share all the SIP messages of both the scenarios.
    Thanks
    Manish

  • UC520 SIP trunk unable to make outgoing calls, incoming calls are ok

    I have an new SIP trunk set on an UC520 and the incoming calls are ok, but the outgoing calls are getting an busy tone(not working).
    The bellow trace is showing that the cause is "No route to destination (3) ". The question is this route has to do with the firewall(ip routing) or with the voice translation rules?
    001866: //3439/91242E51926F/SIP/Call/sipSPICallInfo:
    The Call Setup Information is:
    Call Control Block (CCB) : 0x843CE50C
    State of The Call        : STATE_DEAD
    TCP Sockets Used         : YES
    Calling Number           : 0777777777 <- main sip number
    Called Number            : 0888888888 <- called number
    Source IP Address (Sig  ): 0.0.0.0
    Destn SIP Req Addr:Port  : 0.0.0.0:0
    Destn SIP Resp Addr:Port : 0.0.0.0:0
    Destination Name         :
    001867: //3439/91242E51926F/SIP/Call/sipSPICallInfo:
    Disconnect Cause (CC)    : 3
    Disconnect Cause (SIP)   : 200

    Hi Emil,
    I've added bind control and media interface but outgoing calls are the same blocked, strange thing is that the cause is still no route to destination (3)
    but
    UC_520#show sip-ua status
    SIP User Agent Status
    SIP User Agent for UDP : ENABLED
    SIP User Agent for TCP : ENABLED
    SIP User Agent for TLS over TCP : ENABLED
    SIP User Agent bind status(signaling): ENABLED 10.10.10.5 <- fa0/0 IP address
    SIP User Agent bind status(media): ENABLED 10.10.10.5 <- fa0/0 IP address
    SIP early-media for 180 responses with SDP: ENABLED
    SIP max-forwards : 70
    SIP DNS SRV version: 2 (rfc 2782)
    NAT Settings for the SIP-UA
    Role in SDP: NONE
    Check media source packets: DISABLED
    Maximum duration for a telephone-event in NOTIFYs: 2000 ms
    SIP support for ISDN SUSPEND/RESUME: ENABLED
    Redirection (3xx) message handling: ENABLED
    Reason Header will override Response/Request Codes: DISABLED
    Out-of-dialog Refer: DISABLED
    Presence support is DISABLED
    SDP application configuration:
     Version line (v=) required
     Owner line (o=) required
     Timespec line (t=) required
     Media supported: audio image
     Network types supported: IN
     Address types supported: IP4
     Transport types supported: RTP/AVP udptl

  • ILBC calls via SIP Trunk with CUBE and CUCM7

    hi there,
    our SIP Provider offers the IBLC codec which promises to provide better quality compard to G.729.
    I'm using this scenario:
    IP-Phone(G711) --- CUCM7 --- (SIP-Trunk1) --- CUBE --- (SIP-Trunk2) --- Provider
    Everything workes unless I'm configuring IBLC at the provider and on trunk2.
    I have the CUBE router acting as a trancoding device and also specified IBLC as codec to be handled.
    SIP trunk 2 was placed in a region with IBLC as codec.
    On the trunk configuration in CUCM the media ressource group with XCODE capability is configured
    Transcoding workes between two IP Phones in different regions with different codecs within the intranet.
    Unfortunately the CUBE router doesn't seem to use the transcoder to change internal G711u calls into IBLC codec
    so calls are blocked by the CUBE device:
    deb ccsip calls
    for incoming call:
    .Mar 13 17:43:17.231: //145/BDB403DD8134/SIP/Call/sipSPICallInfo:
    The Call Setup Information is:
    Call Control Block (CCB) : 0x4AE7AC98
    State of The Call        : STATE_DEAD
    TCP Sockets Used         : NO
    Calling Number           : 0237892992
    Called Number            : 036677725231
    Source IP Address (Sig  ): 10.100.100.50
    Destn SIP Req Addr:Port  : <IP SIP Provicer>
    Destn SIP Resp Addr:Port : <IP SIP Provicer>:5060
    Destination Name         : <IP SIP Provicer>
    .Mar 13 17:43:17.231: //145/BDB403DD8134/SIP/Call/sipSPIMediaCallInfo:
    Number of Media Streams: 1
    Media Stream             : 1
    Negotiated Codec         : ilbc
    Negotiated Codec Bytes   : 0
    Nego. Codec payload      : 255 (tx), 255 (rx)
    Negotiated Dtmf-relay    : 6
    Dtmf-relay Payload       : 101 (tx), 101 (rx)
    Source IP Address (Media): <IP CUBE>
    Source IP Port    (Media): 0
    Destn  IP Address (Media): <IP SIP Provicer>
    Destn  IP Port    (Media): 22022
    Orig Destn IP Address:Port (Media): [ - ]:0
    .Mar 13 17:43:17.231: //145/BDB403DD8134/SIP/Call/sipSPICallInfo:
    Disconnect Cause (CC)    : 65
    Disconnect Cause (SIP)   : 488
    (Output lookes similar to outgoing calls)
    I set up ccm on cube and assigned dsp ressources without success:
    Here are the relevant configuration parts:
    voice class codec 1
    codec preference 1 iblc
    voice service voip
    address-hiding
    allow-connections sip to sip
    allow-connections h323 to sip
    allow-connections sip to h323
    fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback cisco
    h323
    sip
      header-passing error-passthru
      no update-callerid
      midcall-signaling passthru
      privacy-policy passthru
    voice-card 0
    dspfarm
    dsp services dspfarm
    dial-peer voice 40991 voip
    description *** Incoming from SIP-Provider
    destination-pattern 03667772523.%
    session protocol sipv2
    session target ipv4:<IP_of_CUCM>
    voice-class codec 1
    voice-class sip asserted-id pai
    voice-class sip privacy-policy passthru
    dtmf-relay rtp-nte
    fax-relay sg3-to-g3
    fax rate 14400
    fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback cisco
    ip qos dscp cs5 media
    ip qos dscp cs5 signaling
    sccp local GigabitEthernet0/0
    sccp ccm 10.100.100.50 identifier 11 version 4.1
    sccp
    sccp ccm group 11
    description *** lokaler CCM fuer Codec-Konvertierung von SIP/DUS.NET
    associate ccm 11 priority 1
    associate profile 21 register DE_WGT_MTP02
    dspfarm profile 21 transcode
    codec g711ulaw
    codec g711alaw
    codec g729ar8
    codec g729abr8
    codec g729r8
    codec ilbc
    maximum sessions 10
    associate application SCCP
    telephony-service
    sdspfarm units 1
    sdspfarm transcode sessions 10
    sdspfarm tag 1 DE_WGT_MTP02
    max-ephones 30
    max-dn 30
    ip source-address 10.100.100.50 port 2000
    max-conferences 8 gain -6
    transfer-system full-consult
    create cnf-files version-stamp 7960 Mar 14 2010 02:10:34
    sh sccp
    SCCP Admin State: UP
    Gateway Local Interface: GigabitEthernet0/0
            IPv4 Address: 10.100.100.50
            Port Number: 2000
    IP Precedence: 5
    User Masked Codec list: None
    Call Manager: 10.100.100.50, Port Number: 2000
                    Priority: N/A, Version: 4.1, Identifier: 11
                    Trustpoint: N/A
    Call Manager: 10.1.1.55, Port Number: 2000
                    Priority: N/A, Version: 7.0, Identifier: 10
                    Trustpoint: N/A
    Transcoding Oper State: ACTIVE - Cause Code: NONE
    Active Call Manager: 10.100.100.50, Port Number: 2000
    TCP Link Status: CONNECTED, Profile Identifier: 21
    Reported Max Streams: 20, Reported Max OOS Streams: 0
    Supported Codec: g711ulaw, Maximum Packetization Period: 30
    Supported Codec: g711alaw, Maximum Packetization Period: 30
    Supported Codec: g729ar8, Maximum Packetization Period: 60
    Supported Codec: g729abr8, Maximum Packetization Period: 60
    Supported Codec: g729r8, Maximum Packetization Period: 60
    Supported Codec: ilbc, Maximum Packetization Period: 120
    Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
    Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30
    Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization Period: 30
    sh dspfarm dsp all
    SLOT DSP VERSION  STATUS CHNL USE   TYPE    RSC_ID BRIDGE_ID PKTS_TXED PKTS_RXED
    0    1   26.3.4   UP     N/A  FREE  xcode  1      -         -         -
    0    1   26.3.4   UP     N/A  FREE  xcode  1      -         -         -
    0    2   26.3.4   UP     N/A  FREE  xcode  2      -         -         -
    0    2   26.3.4   UP     N/A  FREE  xcode  2      -         -         -
    0    2   26.3.4   UP     N/A  FREE  xcode  2      -         -         -
    0    2   26.3.4   UP     N/A  FREE  xcode  2      -         -         -
    0    2   26.3.4   UP     N/A  FREE  xcode  2      -         -         -
    0    2   26.3.4   UP     N/A  FREE  xcode  2      -         -         -
    1    1   26.3.4   UP     N/A  FREE  xcode  1      -         -         -
    1    1   26.3.4   UP     N/A  FREE  xcode  1      -         -         -
    1    1   26.3.4   UP     N/A  FREE  xcode  2      -         -         -
    1    1   26.3.4   UP     N/A  FREE  xcode  2      -         -         -
    1    1   26.3.4   UP     N/A  FREE  xcode  2      -         -         -
    1    1   26.3.4   UP     N/A  FREE  xcode  2      -         -         -
    Thanks in advance,
    David

    Hi there,
    Just wondering whether you ever got this resolved? I seem to have a very similiar problem.
    Regards
    Karen

  • SIP Trunk not accepting inbound calls

    I have a CME setup using Engin as a SIP provider
    I am able to dial out with no issue, however my inbound calls do not work, they divert to the Engin voicemail
    My SIP registration is OK and the number is configured as the primary DN on one of my phones
    Router#sh sip-ua register status
    Line                             peer       expires(sec) registered P-Associ-URI
    ================================ ========== ============ ========== ============
    038682XXXX                       -1         1124         yes       
    101                              20001      45           no        
    102                              20003      18           no        
    103                              20005      45           no        
    104                              20006      45           no        
    I do see the call come in if I debug the dial peer, but it only seems to match an outgoing dp
    I am seeing a couple of disconnect cause codes that I cant seem to find any relavent information on in the CCSIP debugs
    Router#
    Sep  8 18:15:15.009: //156/83D9548A803F/SIP/Event/Session-Timer/sipSTSLMain: Event: E_STSL_SESSION_REFRESH_REQ
    Sep  8 18:15:15.009: //156/83D9548A803F/SIP/Event/Session-Timer/sipSTSLMain: dir:2, method:102, resp_code:0, container:4F947560
    Sep  8 18:15:15.009: //156/83D9548A803F/SIP/Event/Session-Timer/sipSTSLPrintTDContainer: Peer-Event: E_STSL_PASS_ST_PARAMS, SE Value:1800, SE Refresher:uas, Min-SE Value:1800, flags:2001
    Sep  8 18:15:15.017: //156/83D9548A803F/SIP/Event/Session-Timer/sipSTSLMain: Event: E_STSL_SESSION_REFRESH_RESP
    Sep  8 18:15:15.017: //156/83D9548A803F/SIP/Event/Session-Timer/sipSTSLMain: dir:1, method:102, resp_code:100, container:4F947DF8
    Sep  8 18:15:15.025: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_DISCONNECT
    Sep  8 18:15:15.025: //156/83D9548A803F/SIP/Event/Session-Timer/sipSTSLMain: Event: E_STSL_SESSION_REFRESH_RESP
    Sep  8 18:15:15.025: //156/83D9548A803F/SIP/Event/Session-Timer/sipSTSLMain: dir:1, method:102, resp_code:403, container:4F947B38
    Sep  8 18:15:15.061: //156/83D9548A803F/SIP/Call/sipSPICallInfo:
    The Call Setup Information is:
    Call Control Block (CCB) : 0x4C39C570
    State of The Call        : STATE_DEAD
    TCP Sockets Used         : NO
    Calling Number           : 0417XXXXXX
    Called Number            : 038682XXXX
    Source IP Address (Sig  ): 211.30.48.136
    Destn SIP Req Addr:Port  : 203.161.164.69:5060
    Destn SIP Resp Addr:Port : 203.161.164.69:5060
    Destination Name         : 203.161.164.69
    Sep  8 18:15:15.061: //156/83D9548A803F/SIP/Call/sipSPIMediaCallInfo:
    Number of Media Streams: 1
    Media Stream             : 1
    Negotiated Codec         : g711ulaw
    Negotiated Codec Bytes   : 160
    Nego. Codec payload      : 0 (tx), 0 (rx)
    Negotiated Dtmf-relay    : 0
    Dtmf-relay Payload       : 0 (tx), 0 (rx)
    Source IP Address (Media): 211.30.48.136
    Source IP Port    (Media): 17768
    Destn  IP Address (Media): 203.161.164.69
    Destn  IP Port    (Media): 18314
    Orig Destn IP Address:Port (Media): [ - ]:0
    Sep  8 18:15:15.061: //156/83D9548A803F/SIP/Call/sipSPICallInfo:
    Disconnect Cause (CC)    : 21
    Disconnect Cause (SIP)   : 403
    Any Ideas
    Doug

    Hi Tapan,
    Firstly the topology is as follows
    ISP/VOIP provider - Internet - Cable modem - 2800 CME router - IP Phone
    The VM is provided by the ISP
    debug ccsip messages
    Sep  9 10:21:41.488: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    INVITE sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 203.161.164.69:5060;branch=z9hG4bKv0hlbo2030i1c85sm7b0.1
    From: "0417XXXXXX"[email protected];user=phone>;tag=SD1uttd01-353510938-1315527701453-
    To: "Doug Goding"[email protected]>
    Call-ID: SD1uttd01-c27fdbf3f1de4ad1f0f2e1342b210494-au418e3
    CSeq: 633854439 INVITE
    Contact:
    Supported: 100rel,timer
    Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
    Accept: multipart/mixed,application/media_control+xml,application/sdp
    Min-SE: 60
    Session-Expires: 1800;refresher=uas
    Max-Forwards: 9
    Content-Type: application/sdp
    Content-Disposition: session;handling=required
    Content-Length: 317
    v=0
    o=BroadWorks 18275729 1 IN IP4 203.161.164.69
    s=-
    c=IN IP4 203.161.164.69
    t=0 0
    m=audio 18128 RTP/AVP 18 8 0 101
    c=IN IP4 203.161.164.69
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=yes
    a=rtpmap:8 PCMA/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=bsoft: 1 image udptl t38
    Sep  9 10:21:41.508: //4375/867C64EB8067/SIP/Msg/ccsipDisplayMsg:
    Sent:
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 203.161.164.69:5060;branch=z9hG4bKv0hlbo2030i1c85sm7b0.1
    From: "0417XXXXXX"[email protected];user=phone>;tag=SD1uttd01-353510938-1315527701453-
    To: "Doug Goding"[email protected]>
    Date: Fri, 09 Sep 2011 00:21:41 GMT
    Call-ID: SD1uttd01-c27fdbf3f1de4ad1f0f2e1342b210494-au418e3
    CSeq: 633854439 INVITE
    Allow-Events: telephone-event
    Server: Cisco-SIPGateway/IOS-12.x
    Content-Length: 0
    Sep  9 10:21:41.516: //4375/867C64EB8067/SIP/Msg/ccsipDisplayMsg:
    Sent:
    SIP/2.0 403 Forbidden
    Via: SIP/2.0/UDP 203.161.164.69:5060;branch=z9hG4bKv0hlbo2030i1c85sm7b0.1
    From: "0417XXXXXX"[email protected];user=phone>;tag=SD1uttd01-353510938-1315527701453-
    To: "Doug Goding"[email protected]>;tag=39373A4-586
    Date: Fri, 09 Sep 2011 00:21:41 GMT
    Call-ID: SD1uttd01-c27fdbf3f1de4ad1f0f2e1342b210494-au418e3
    CSeq: 633854439 INVITE
    Allow-Events: telephone-event
    Server: Cisco-SIPGateway/IOS-12.x
    Reason: Q.850;cause=21
    Content-Length: 0
    Sep  9 10:21:41.544: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    ACK sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 203.161.164.69:5060;branch=z9hG4bKv0hlbo2030i1c85sm7b0.1
    CSeq: 633854439 ACK
    From: "0417XXXXXX"[email protected];user=phone>;tag=SD1uttd01-353510938-1315527701453-
    To: "Doug Goding"[email protected]>;tag=39373A4-586
    Call-ID: SD1uttd01-c27fdbf3f1de4ad1f0f2e1342b210494-au418e3
    Max-Forwards: 9
    Content-Length: 0
    Voice Config
    Router#
    voice service voip
    allow-connections h323 to sip
    allow-connections sip to h323
    allow-connections sip to sip
    sip
      registrar server expires max 3600 min 3600
      localhost dns:mel.byo.engin.com.au
      no call service stop
    voice class codec 1
    codec preference 1 g711ulaw
    voice translation-rule 10
    rule 1 /^0/ //
    voice translation-rule 11
    rule 1 /^.*/ /0386821234/
    voice translation-profile PSTN_Outgoing
    translate calling 11
    voice-card 0
    dsp services dspfarm
    mgcp profile default
    sccp local Vlan100
    sccp ccm 10.1.100.1 identifier 1 version 7.0
    sccp
    sccp ccm group 1
    bind interface Vlan100
    associate ccm 1 priority 1
    associate profile 1 register confdsp
    dspfarm profile 1 conference 
    codec g711ulaw
    codec g711alaw
    codec g729ar8
    codec g729abr8
    codec g729r8
    codec g729br8
    maximum sessions 4
    associate application SCCP
    dial-peer voice 99 voip
    translation-profile outgoing PSTN_Outgoing
    destination-pattern .T
    session protocol sipv2
    session target sip-server
    dtmf-relay rtp-nte
    codec g711ulaw
    no vad
    dial-peer voice 100 voip
    session protocol sipv2
    session target dns:mel.byo.engin.com.au
    incoming called-number 0386821234
    dtmf-relay sip-notify
    codec g711ulaw
    no vad
    dial-peer voice 110 voip
    description ** Incoming call from SIP trunk (Generic SIP Trunk Provider) **
    session protocol sipv2
    session target dns:mel.byo.engin.com.au
    incoming called-number .%
    dtmf-relay rtp-nte
    no vad
    dial-peer voice 90 voip
    description Melbourne 03 Numbers
    translation-profile outgoing PSTN_Outgoing
    destination-pattern [89].......
    session protocol sipv2
    session target sip-server
    dtmf-relay rtp-nte
    codec g711ulaw
    no vad
    dial-peer voice 91 voip
    description National Numbers
    translation-profile outgoing PSTN_Outgoing
    destination-pattern 0[278]........
    session protocol sipv2
    session target sip-server
    dtmf-relay rtp-nte
    codec g711ulaw
    no vad
    dial-peer voice 92 voip
    description Vic/Tas 03 numbers
    translation-profile outgoing PSTN_Outgoing
    destination-pattern [56].......
    session protocol sipv2
    session target sip-server
    dtmf-relay rtp-nte
    codec g711ulaw
    no vad
    dial-peer voice 93 voip
    description Mobile numbers
    translation-profile outgoing PSTN_Outgoing
    destination-pattern 04........
    session protocol sipv2
    session target sip-server
    dtmf-relay rtp-nte
    codec g711ulaw
    no vad
    dial-peer voice 94 voip
    description 13XXXX numbers
    translation-profile outgoing PSTN_Outgoing
    destination-pattern 13[1-9]...
    session protocol sipv2
    session target sip-server
    dtmf-relay rtp-nte
    codec g711ulaw
    no vad
    dial-peer voice 96 voip
    description 1300/1800 numbers
    translation-profile outgoing PSTN_Outgoing
    destination-pattern 1[38]00......
    session protocol sipv2
    session target sip-server
    dtmf-relay rtp-nte
    codec g711ulaw
    no vad
    dial-peer voice 98 voip
    description Emergency 000
    translation-profile outgoing PSTN_Outgoing
    destination-pattern 000
    session protocol sipv2
    session target sip-server
    dtmf-relay rtp-nte
    codec g711ulaw
    no vad
    sip-ua
    credentials username 0386821234 password 7 XXXX realm voice.mibroadband.com.au
    authentication username 0386821234 password 7 XXXX
    nat symmetric role active
    nat symmetric check-media-src
    no remote-party-id
    retry invite 2
    retry register 10
    timers connect 100
    mwi-server dns:mel.byo.engin.com.au expires 3600 port 5060 transport udp unsolicited
    registrar dns:mel.byo.engin.com.au expires 3600
    sip-server dns:mel.byo.engin.com.au
    connection-reuse
    telephony-service
    sdspfarm conference mute-on #1 mute-off #2
    sdspfarm units 2
    sdspfarm tag 1 confdsp
    conference hardware
    max-ephones 42
    max-dn 144
    ip source-address 10.1.100.1 port 2000
    calling-number initiator
    service phone videoCapability 1
    service phone displayOnDuration 00:01
    service phone displayOnTime 08:30
    service phone displayOffTime 17:30
    service phone displayIdleTimeout 00:01
    service phone displayOnWhenIncomingCall 1
    system message Cisco CME
    load 7941 SCCP41.8-4-2S
    load 7942 SCCP42.8-4-2S
    load 7945 SCCP45.8-4-2S
    load 7961 SCCP41.8-4-2S
    load 7962 SCCP42.8-4-2S
    load 7965 SCCP45.8-4-2S
    load ata ATA030204SCCP090202A
    time-zone 48
    date-format dd-mm-yy
    voicemail 90125200
    mwi relay
    max-conferences 8 gain -6
    call-forward pattern .T
    call-forward system redirecting-expanded
    moh music-on-hold.au
    web admin system name cisco secret 5 $1$d8/H$glhLiCCWXmFSUp6BtwGho0
    dn-webedit
    time-webedit
    transfer-system full-consult
    transfer-pattern 0.T
    create cnf-files version-stamp 7960 Jul 06 2011 10:32:45
    ephone-dn  1  dual-line
    number 038682XXXX
    label 101
    name 7965
    mwi sip
    ephone-dn  2  dual-line
    number 102
    label 102
    name 7941
    ephone-dn  3  dual-line
    number 103
    label 103
    name 7920
    ephone  1
    device-security-mode none
    video
    mac-address 0023.5EB8.6E4E
    type 7965
    button  1:2 2:1
    ephone  3
    device-security-mode none
    mac-address 0019.0633.A933
    max-calls-per-button 2
    type 7920
    button  1:3
    ephone  10
    device-security-mode none
    mac-address 0019.E7B7.BAB3
    max-calls-per-button 2
    type ata
    button  1:1

  • Cucm , Cube via Sip and Sip Trunk to ISP , Outgoing calls not working

    Hi
    We have issue with the outgoing calls to sip trunk
    Below is the config and the debugs
    It will be great if you give your thoughts since we have stuck here
    My thoughts are:
    i see that for unknown reason the called number is going with 4 digits instead of 8 digits
    i dont see any sip message comming from ISP
    Maybe the call not going there ? to isp trunk? From the trace the call hit the correct dialpeer 888 but i see 4 digits as a called number , but i dodnt understant the reason to translated in 4 digits the called number.Not apply a translation rule for that
    confused!!!
    Calling Numbner:22324086
    Called Number: 23823690
    CUCM:192.168.1.241 and 242
    CUBE:192.168.1.10
    voice service voip
    ip address trusted list
      ipv4 0.0.0.0 0.0.0.0
    dtmf-interworking rtp-nte
    allow-connections h323 to h323
    allow-connections h323 to sip
    allow-connections sip to h323
    allow-connections sip to sip
    supplementary-service h450.12
    no supplementary-service h225-notify cid-update
    no supplementary-service sip moved-temporarily
    no supplementary-service sip refer
    fax protocol none
    no fax-relay sg3-to-g3
    h323
    sip
      registrar server
      localhost dns:bbtb.cyta.com.cy
      outbound-proxy dns:sbg.bbtb.cyta.com.cy
      no update-callerid
      early-offer forced
    voice class codec 2
    codec preference 1 g711alaw
    codec preference 2 g711ulaw
    codec preference 3 g729br8
    codec preference 4 g729r8
    voice translation-rule 1
    rule 1 /.*\(....\)/ /\1/
    voice translation-rule 3
    rule 1 /^9/ //
    voice translation-rule 4
    rule 1 /\+/ /900/
    rule 2 /^\(9\)\(.......$\)/ /99\2/
    rule 3 /^\(2\)\(.......$\)/ /92\2/
    rule 4 /^0/ /90/
    rule 5 /^1/ /9001/
    rule 6 /^3/ /9003/
    rule 7 /^4/ /9004/
    rule 8 /^5/ /9005/
    rule 9 /^6/ /9006/
    rule 10 /^7/ /9007/
    rule 11 /^8/ /9008/
    rule 12 /^9/ /9009/
    rule 13 /^2/ /9002/
    voice translation-rule 5
    rule 1 // /2232/
    rule 2 /^9/ //
    voice translation-profile SIP_Incoming
    translate calling 4
    translate called 1
    voice translation-profile SIP_Outgoing
    translate calling 5
    translate called 3
    interface FastEthernet0/0
    ip address 192.168.1.10 255.255.255.0
    duplex auto
    speed auto
    interface FastEthernet0/1
    description **SIP TRUNK WITH CYTA**
    ip address 10.249.13.130 255.255.255.252
    duplex auto
    speed auto
    interface FastEthernet0/0
    ip address 192.168.1.10 255.255.255.0
    duplex auto
    speed auto
    interface FastEthernet0/1
    description **SIP TRUNK WITH CYTA**
    ip address 10.249.13.130 255.255.255.252
    duplex auto
    speed auto
    dial-peer voice 889 voip
    description **SIP Trunk to CUCM**
    destination-pattern 4086
    session protocol sipv2
    session target ipv4:192.168.1.242:5060
    voice-class codec 2 
    voice-class sip dtmf-relay force rtp-nte
    no voice-class sip outbound-proxy  
    voice-class sip bind control source-interface FastEthernet0/0
    voice-class sip bind media source-interface FastEthernet0/0
    dtmf-relay sip-notify
    no vad
    dial-peer voice 890 voip
    description **SIP Trunk to CUCM2**
    destination-pattern 4086
    session protocol sipv2
    session target ipv4:192.168.1.241:5060
    voice-class codec 2 
    voice-class sip dtmf-relay force rtp-nte
    no voice-class sip outbound-proxy  
    voice-class sip bind control source-interface FastEthernet0/0
    voice-class sip bind media source-interface FastEthernet0/0
    dtmf-relay sip-notify
    no vad
    dial-peer voice 888 voip
    description **SIP Trunk to CYTA OUTGOING**
    translation-profile incoming SIP_Incoming
    translation-profile outgoing SIP_Outgoing
    destination-pattern 9T
    session protocol sipv2
    session target sip-server
    incoming called-number .
    voice-class codec 2 
    voice-class sip dtmf-relay force rtp-nte
    dtmf-relay rtp-nte
    no vad
    voice service voip
    ip address trusted list
      ipv4 0.0.0.0 0.0.0.0
    dtmf-interworking rtp-nte
    allow-connections h323 to h323
    allow-connections h323 to sip
    allow-connections sip to h323
    allow-connections sip to sip
    supplementary-service h450.12
    no supplementary-service h225-notify cid-update
    no supplementary-service sip moved-temporarily
    no supplementary-service sip refer
    fax protocol none
    no fax-relay sg3-to-g3
    h323
    sip
      registrar server
      localhost dns:bbtb.cyta.com.cy
      outbound-proxy dns:sbg.bbtb.cyta.com.cy
      no update-callerid
      early-offer forced
    voice class codec 2
    codec preference 1 g711alaw
    codec preference 2 g711ulaw
    codec preference 3 g729br8
    codec preference 4 g729r8
    voice translation-rule 1
    rule 1 /.*\(....\)/ /\1/
    voice translation-rule 3
    rule 1 /^9/ //
    voice translation-rule 4
    rule 1 /\+/ /900/
    rule 2 /^\(9\)\(.......$\)/ /99\2/
    rule 3 /^\(2\)\(.......$\)/ /92\2/
    rule 4 /^0/ /90/
    rule 5 /^1/ /9001/
    rule 6 /^3/ /9003/
    rule 7 /^4/ /9004/
    rule 8 /^5/ /9005/
    rule 9 /^6/ /9006/
    rule 10 /^7/ /9007/
    rule 11 /^8/ /9008/
    rule 12 /^9/ /9009/
    rule 13 /^2/ /9002/
    voice translation-rule 5
    rule 1 // /2232/
    rule 2 /^9/ //
    voice translation-profile SIP_Incoming
    translate calling 4
    translate called 1
    voice translation-profile SIP_Outgoing
    translate calling 5
    translate called 3
    dial-peer voice 889 voip
    description **SIP Trunk to CUCM**
    destination-pattern 4086
    session protocol sipv2
    session target ipv4:192.168.1.242:5060
    voice-class codec 2 
    voice-class sip dtmf-relay force rtp-nte
    no voice-class sip outbound-proxy  
    voice-class sip bind control source-interface FastEthernet0/0
    voice-class sip bind media source-interface FastEthernet0/0
    dtmf-relay sip-notify
    no vad
    dial-peer voice 890 voip
    description **SIP Trunk to CUCM2**
    destination-pattern 4086
    session protocol sipv2
    session target ipv4:192.168.1.241:5060
    voice-class codec 2 
    voice-class sip dtmf-relay force rtp-nte
    no voice-class sip outbound-proxy  
    voice-class sip bind control source-interface FastEthernet0/0
    voice-class sip bind media source-interface FastEthernet0/0
    dtmf-relay sip-notify
    no vad
    dial-peer voice 888 voip
    description **SIP Trunk to CYTA OUTGOING**
    translation-profile incoming SIP_Incoming
    translation-profile outgoing SIP_Outgoing
    destination-pattern 9T
    session protocol sipv2
    session target sip-server
    incoming called-number .
    voice-class codec 2 
    voice-class sip dtmf-relay force rtp-nte
    dtmf-relay rtp-nte
    no vad

    Hi Aok
    I change the default value for IPVMS from g711ulaw to g711alaw but the results remained the same
    Also i have  restarted the IPVMS
    SIP-GW#
    SIP-GW#
    *Mar  5 14:19:57.854: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    INVITE sip:[email protected]:5060;transport=tcp SIP/2.0
    Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK79816bbd4196
    From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
    To: ;tag=125E62C-1354
    Date: Tue, 05 Mar 2013 13:52:31 GMT
    Call-ID: [email protected]
    Supported: timer,resource-priority,replaces
    Min-SE:  1800
    User-Agent: Cisco-CUCM9.0
    Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
    CSeq: 102 INVITE
    Max-Forwards: 70
    Expires: 180
    Allow-Events: presence
    Supported: X-cisco-srtp-fallback
    Supported: Geolocation
    Session-Expires:  1800;refresher=uac
    P-Asserted-Identity:
    Remote-Party-ID: ;party=calling;screen=yes;privacy=off
    Contact:
    Content-Type: application/sdp
    Content-Length: 244
    v=0
    o=CiscoSystemsCCM-SIP 38874 2 IN IP4 192.168.1.241
    s=SIP Call
    c=IN IP4 0.0.0.0
    b=TIAS:64000
    b=AS:64
    t=0 0
    m=audio 24784 RTP/AVP 8 101
    a=rtpmap:8 PCMA/8000
    a=ptime:20
    a=inactive
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    *Mar  5 14:19:57.878: //717/DC740C800000/SIP/Msg/ccsipDisplayMsg:
    Sent:
    INVITE sip:[email protected]:5060;transport=udp SIP/2.0
    Via: SIP/2.0/UDP 10.249.13.130:5060;branch=z9hG4bK355253C
    From: [email protected]>;tag=125E594-5C7
    To: [email protected]>;tag=h7g4Esbg_945723725-1362491526714
    Date: Tue, 05 Mar 2013 14:19:57 GMT
    Call-ID: [email protected]
    Route:
    Supported: 100rel,timer,resource-priority,replaces,sdp-anat
    Min-SE:  1800
    Cisco-Guid: 3698592896-0000065536-0000000107-4043417792
    User-Agent: Cisco-SIPGateway/IOS-12.x
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    CSeq: 102 INVITE
    Max-Forwards: 70
    Timestamp: 1362493197
    Contact:
    Expires: 60
    Allow-Events: telephone-event
    Content-Type: application/sdp
    Content-Length: 262
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 6506 3807 IN IP4 10.249.13.130
    s=SIP Call
    c=IN IP4 10.249.13.130
    t=0 0
    m=audio 19234 RTP/AVP 8 101
    c=IN IP4 10.249.13.130
    a=inactive
    a=rtpmap:8 PCMA/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=ptime:20
    *Mar  5 14:19:57.878: //716/DC740C800000/SIP/Msg/ccsipDisplayMsg:
    Sent:
    SIP/2.0 100 Trying
    Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK79816bbd4196
    From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
    To: ;tag=125E62C-1354
    Date: Tue, 05 Mar 2013 14:19:57 GMT
    Call-ID: [email protected]
    CSeq: 102 INVITE
    Allow-Events: telephone-event
    Server: Cisco-SIPGateway/IOS-12.x
    Content-Length: 0
    *Mar  5 14:19:57.926: //717/DC740C800000/SIP/Msg/ccsipDisplayMsg:
    Received:
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 10.249.13.130:5060;branch=z9hG4bK355253C
    To: [email protected]>;tag=h7g4Esbg_945723725-1362491526714
    From: [email protected]>;tag=125E594-5C7
    Call-ID: [email protected]
    CSeq: 102 INVITE
    Contact:
    Require: timer
    Session-Expires: 1800;refresher=uac
    Content-Type: application/sdp
    Content-Length: 213
    Allow: ACK, BYE, CANCEL, INVITE, OPTIONS, PRACK, REFER, NOTIFY, UPDATE, INFO
    Accept: application/media_control+xml
    Accept: application/sdp
    Accept: application/x-broadworks-call-center+xml
    v=0
    o=BroadWorks 96335268 2 IN IP4 10.224.42.164
    s=-
    c=IN IP4 10.224.42.72
    t=0 0
    m=audio 54932 RTP/AVP 8 101
    a=rtpmap:8 PCMA/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=ptime:20
    a=inactive
    *Mar  5 14:19:57.942: //716/DC740C800000/SIP/Msg/ccsipDisplayMsg:
    Sent:
    SIP/2.0 200 OK
    Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK79816bbd4196
    From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
    To: ;tag=125E62C-1354
    Date: Tue, 05 Mar 2013 14:19:57 GMT
    Call-ID: [email protected]
    CSeq: 102 INVITE
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    Allow-Events: telephone-event
    Contact:
    Supported: replaces
    Supported: sdp-anat
    Server: Cisco-SIPGateway/IOS-12.x
    Session-Expires:  1800;refresher=uac
    Require: timer
    Supported: timer
    Content-Type: application/sdp
    Content-Length: 259
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 9410 5774 IN IP4 192.168.1.10
    s=SIP Call
    c=IN IP4 192.168.1.10
    t=0 0
    m=audio 19314 RTP/AVP 8 101
    c=IN IP4 192.168.1.10
    a=inactive
    a=rtpmap:8 PCMA/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=ptime:20
    *Mar  5 14:19:57.946: //717/DC740C800000/SIP/Msg/ccsipDisplayMsg:
    Sent:
    ACK sip:[email protected]:5060;transport=udp SIP/2.0
    Via: SIP/2.0/UDP 10.249.13.130:5060;branch=z9hG4bK3562A4
    From: [email protected]>;tag=125E594-5C7
    To: [email protected]>;tag=h7g4Esbg_945723725-1362491526714
    Date: Tue, 05 Mar 2013 14:19:57 GMT
    Call-ID: [email protected]
    Route:
    Max-Forwards: 70
    CSeq: 102 ACK
    Allow-Events: telephone-event
    Content-Length: 0
    *Mar  5 14:19:57.946: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    ACK sip:[email protected]:5060;transport=tcp SIP/2.0
    Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK798246ab3597
    From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
    To: ;tag=125E62C-1354
    Date: Tue, 05 Mar 2013 13:52:31 GMT
    Call-ID: [email protected]
    Max-Forwards: 70
    CSeq: 102 ACK
    Allow-Events: presence
    Content-Length: 0
    *Mar  5 14:19:58.146: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    INVITE sip:[email protected]:5060;transport=tcp SIP/2.0
    Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK7983739137ab
    From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
    To: ;tag=125E62C-1354
    Date: Tue, 05 Mar 2013 13:52:31 GMT
    Call-ID: [email protected]
    Supported: timer,resource-priority,replaces
    Min-SE:  1800
    User-Agent: Cisco-CUCM9.0
    Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
    CSeq: 103 INVITE
    Max-Forwards: 70
    Expires: 180
    Allow-Events: presence
    Supported: X-cisco-srtp-fallback
    Supported: Geolocation
    Session-Expires:  1800;refresher=uac
    P-Asserted-Identity:
    Remote-Party-ID: ;party=calling;screen=yes;privacy=off
    Contact:
    Content-Length: 0
    *Mar  5 14:19:58.158: //717/DC740C800000/SIP/Msg/ccsipDisplayMsg:
    Sent:
    INVITE sip:[email protected]:5060;transport=udp SIP/2.0
    Via: SIP/2.0/UDP 10.249.13.130:5060;branch=z9hG4bK3571933
    From: [email protected]>;tag=125E594-5C7
    To: [email protected]>;tag=h7g4Esbg_945723725-1362491526714
    Date: Tue, 05 Mar 2013 14:19:58 GMT
    Call-ID: [email protected]
    Route:
    Supported: timer,resource-priority,replaces,sdp-anat
    Min-SE:  1800
    Cisco-Guid: 3698592896-0000065536-0000000107-4043417792
    User-Agent: Cisco-SIPGateway/IOS-12.x
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    CSeq: 103 INVITE
    Max-Forwards: 70
    Timestamp: 1362493198
    Contact:
    Expires: 60
    Allow-Events: telephone-event
    Content-Length: 0
    *Mar  5 14:19:58.158: //716/DC740C800000/SIP/Msg/ccsipDisplayMsg:
    Sent:
    SIP/2.0 100 Trying
    Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK7983739137ab
    From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
    To: ;tag=125E62C-1354
    Date: Tue, 05 Mar 2013 14:19:58 GMT
    Call-ID: [email protected]
    CSeq: 103 INVITE
    Allow-Events: telephone-event
    Server: Cisco-SIPGateway/IOS-12.x
    Content-Length: 0
    *Mar  5 14:19:58.218: //717/DC740C800000/SIP/Msg/ccsipDisplayMsg:
    Received:
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 10.249.13.130:5060;branch=z9hG4bK3571933
    To: [email protected]>;tag=h7g4Esbg_945723725-1362491526714
    From: [email protected]>;tag=125E594-5C7
    Call-ID: [email protected]
    CSeq: 103 INVITE
    Contact:
    Require: timer
    Session-Expires: 1800;refresher=uac
    Content-Type: application/sdp
    Content-Length: 216
    Allow: ACK, BYE, CANCEL, INVITE, OPTIONS, PRACK, REFER, NOTIFY, UPDATE, INFO
    Accept: application/media_control+xml
    Accept: application/sdp
    Accept: application/x-broadworks-call-center+xml
    v=0
    o=BroadWorks 96335268 3 IN IP4 10.224.42.164
    s=-
    c=IN IP4 10.224.42.72
    t=0 0
    m=audio 54932 RTP/AVP 8 18 96 99
    a=rtpmap:96 AMR/8000
    a=rtpmap:99 telephone-event/8000
    a=fmtp:99 0-15
    a=ptime:20
    a=sendrecv
    *Mar  5 14:19:58.234: //716/DC740C800000/SIP/Msg/ccsipDisplayMsg:
    Sent:
    SIP/2.0 200 OK
    Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK7983739137ab
    From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
    To: ;tag=125E62C-1354
    Date: Tue, 05 Mar 2013 14:19:58 GMT
    Call-ID: [email protected]
    CSeq: 103 INVITE
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    Allow-Events: telephone-event
    Contact:
    Supported: replaces
    Supported: sdp-anat
    Server: Cisco-SIPGateway/IOS-12.x
    Session-Expires:  1800;refresher=uac
    Require: timer
    Supported: timer
    Content-Type: application/sdp
    Content-Length: 283
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 9410 5775 IN IP4 192.168.1.10
    s=SIP Call
    c=IN IP4 192.168.1.10
    t=0 0
    m=audio 19314 RTP/AVP 8 18 101
    c=IN IP4 192.168.1.10
    a=rtpmap:8 PCMA/8000
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=yes
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    *Mar  5 14:19:58.242: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    ACK sip:[email protected]:5060;transport=tcp SIP/2.0
    Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK7985648033f2
    From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
    To: ;tag=125E62C-1354
    Date: Tue, 05 Mar 2013 13:52:31 GMT
    Call-ID: [email protected]
    Max-Forwards: 70
    CSeq: 103 ACK
    Allow-Events: presence
    Content-Type: application/sdp
    Content-Length: 192
    v=0
    o=CiscoSystemsCCM-SIP 38874 3 IN IP4 192.168.1.241
    s=SIP Call
    c=IN IP4 192.168.1.241
    t=0 0
    m=audio 4000 RTP/AVP 8
    a=X-cisco-media:umoh
    a=rtpmap:8 PCMA/8000
    a=ptime:20
    a=sendonly
    *Mar  5 14:19:58.262: //717/DC740C800000/SIP/Msg/ccsipDisplayMsg:
    Sent:
    ACK sip:[email protected]:5060;transport=udp SIP/2.0
    Via: SIP/2.0/UDP 10.249.13.130:5060;branch=z9hG4bK358582
    From: [email protected]>;tag=125E594-5C7
    To: [email protected]>;tag=h7g4Esbg_945723725-1362491526714
    Date: Tue, 05 Mar 2013 14:19:58 GMT
    Call-ID: [email protected]
    Route:
    Max-Forwards: 70
    CSeq: 103 ACK
    Allow-Events: telephone-event
    Content-Type: application/sdp
    Content-Length: 259
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 6506 3808 IN IP4 10.249.13.130
    s=SIP Call
    c=IN IP4 10.249.13.130
    t=0 0
    m=audio 19234 RTP/AVP 8 99
    c=IN IP4 10.249.13.130
    a=sendonly
    a=rtpmap:8 PCMA/8000
    a=rtpmap:99 telephone-event/8000
    a=fmtp:99 0-15
    a=ptime:20
    SIP-GW#
    SIP-GW#sh voip rtp connections
    VoIP RTP active connections :
    No. CallId     dstCallId  LocalRTP RmtRTP     LocalIP                                RemoteIP
    1     716        717        19314    4000     192.168.1.10                           192.168.1.241
    2     717        716        19234    54932    10.249.13.130                          10.224.42.72
    Found 2 active RTP connections

  • Calls from Sip Trunk to UC540 and then to CUE returned ** Service Unavailable**

    Hi to all
    i have something strange here and i need your assistance
    Call Flow:
    Sip trunk-->UC540--> CUE
    When calls coming to UC540 from outside and then going to cue then we send back service unavailable.I made a translation and i sent directly the incoming calls to CUE
    The same behavior is also if i send the calls to dummy number and then from there set forward all to voice mail.
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    Uc540: 8.6
    CUE: 8.6.5
    A number: 99999999
    B number: 22777777
    Voice Mail Number:111
    Attached is the trace
    i see that we hit the correct dial peers .
    I have enable only trancoder since MTP is not register ( don't know why , but i don't think also that is necessary..
    voice service voip
     ip address trusted list
      ipv4 172.16.80.0 255.255.255.0
      ipv4 172.16.81.0 255.255.255.0
     allow-connections sip to sip
     supplementary-service h450.12
     no supplementary-service sip moved-temporarily
     no supplementary-service sip refer
     supplementary-service media-renegotiate
     sip
      no update-callerid
    dial-peer voice 1000 voip
     description **SIP TRUNK**
     translation-profile incoming SIP-INCOMING
     translation-profile outgoing SIP-OUTGOING
     destination-pattern 9T
     modem passthrough nse codec g711alaw
     session protocol sipv2
     session target sip-server
     incoming called-number .T
     voice-class codec 2  
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     fax-relay ecm disable
     no fax-relay sg3-to-g3
     fax rate 9600
     fax protocol pass-through g711alaw
     no vad
    dial-peer voice 2001 voip
     description ** cue voicemail pilot number **
     destination-pattern 111
     b2bua
     session protocol sipv2
     session target ipv4:10.1.10.1
     incoming called-number 111
     no voice-class sip outbound-proxy   
     dtmf-relay sip-notify
     codec g711ulaw
     no vad
    Regards
    chrysostomos

    Hi
    Interface                  IP-Address      OK? Method Status                Protocol
    FastEthernet0/0            unassigned      YES NVRAM  up                    up
    FastEthernet0/0.10         192.168.0.10    YES DHCP   up                    up   ----> For internet
    FastEthernet0/0.20         10.151.5.130    YES NVRAM  up                    up  ------> For sip trunk
    In0/0                      10.1.10.2       YES unset  up                    up    --------> default gw for cue
    Vlan1                      unassigned      YES unset  up                    up
    Vlan100                    unassigned      YES unset  up                    up
    Vlan200                    unassigned      YES unset  up                    down
    Vlan300                    unassigned      YES unset  up                    down
    NVI0                       10.1.10.2       YES unset  up                    up
    BVI1                       192.168.20.1    YES NVRAM  up                    up
    BVI100                     10.1.1.1        YES NVRAM  up                    up   ---------> ip for cme
    Loopback0                  10.1.10.2       YES NVRAM  up                    up   ------> default gw for cue
    dial-peer voice 2001 voip
     description ** cue voicemail pilot number **
     destination-pattern 111
     b2bua
     session protocol sipv2
     session target ipv4:10.1.10.1
     incoming called-number 111
     no voice-class sip outbound-proxy
     voice-class sip bind control source-interface BVI100
     voice-class sip bind media source-interface BVI100
     dtmf-relay sip-notify
     codec g711ulaw
     no vad
    interface FastEthernet0/0.10
     description **FOR INTERNET**
     encapsulation dot1Q 10
     ip address dhcp
     ip access-group 105 in
     ip nat outside
     ip inspect SDM_LOW out
     ip virtual-reassembly in
    interface FastEthernet0/0.20
     description **FOR SIP TRUNK WITH ISP**
     encapsulation dot1Q 20
     ip address 10.151.5.130 255.255.255.240
    ip route 10.1.10.1 255.255.255.255 Integrated-Service-Engine0/0
    ping 10.1.10.1 source bvi100
    Type escape sequence to abort.
    Sending 5, 100-byte ICMP Echos to 10.1.10.1, timeout is 2 seconds:
    Packet sent with a source address of 10.1.1.1
    Success rate is 100 percent (5/5), round-trip min/avg/max = 1/1/1 ms
    I have bind the interface of cme ( 10.1.1.1) but the call fails again
    Attached is the trace
    Anything to advice?

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