ATA187, SIP trunks, CUBE

Hi,
What dial-peer fax statements would be advised in a case where the ITSP (SIP Trunking) only supports g711a pass though for fax?
We are having a hard time making outbound faxes work reliably from a ATA187.
How much 'relay' can/should the gateway do in this case (to my understanding there is no 'relay' in this case)?
Should we rely solely on the ATA187 configuration in CUCM and the settings on the fax machine ? (Speed, ECM, etc)
Gateway/CUBE 15.3, CUCM 9.1.1a
Regards,
Erik
Sent from Cisco Technical Support iPhone App

The following is an example config you can make use of with an SIP enviroment.
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
fax protocol pass-through g711alaw
h323
modem passthrough nse codec g711alaw
sip
  registrar server expires max 3600 min 3600
  asserted-id pai
  privacy pstn
  options-ping 60
  midcall-signaling passthru
dial-peer voice 200 voip
description -----  WYCHODZACE SIP-T ------
destination-pattern [1-9]........
modem passthrough nse codec g711alaw
session protocol sipv2
session target ipv4:SIP-SRV:5020
voice-class codec 23 
voice-class sip dtmf-relay force rtp-nte
voice-class sip early-offer forced
voice-class sip pass-thru headers unsupp
voice-class sip pass-thru content unsupp
voice-class sip pass-thru content sdp
dtmf-relay rtp-nte
req-qos controlled-load audio
fax-relay ecm disable
fax protocol pass-through g711alaw
no vad
dial-peer voice 101 voip
description **Outgoing Call to CUCM**
destination-pattern 895216...
session target ipv4:CCM1
voice-class codec 1 
voice-class h323 1
voice-class sip dtmf-relay force rtp-nte
voice-class sip early-offer forced
dtmf-relay h245-alphanumeric
fax protocol pass-through g711alaw

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    o=BroadWorks 316169737 1 IN IP4 10.111.111.254
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    a=fmtp:101 0-15
    a=rtpmap:100 X-NSE/8000
    a=fmtp:100 200-202
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    Sent: 
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    Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e72063a5aba5d
    From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
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    Date: Sun, 30 Nov 2014 11:44:29 GMT
    Call-ID: [email protected]
    CSeq: 101 INVITE
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    Contact: <sip:[email protected]:5060>
    Supported: sdp-anat
    Server: Cisco-SIPGateway/IOS-15.2.4.M5
    Content-Type: application/sdp
    Content-Disposition: session;handling=required
    Content-Length: 289
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 7965 2747 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
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    a=fmtp:101 0-15
    a=rtpmap:19 CN/8000
    a=ptime:20
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    Received: 
    SIP/2.0 180 Session Progress
    Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC72218
    From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
    To: <sip:[email protected]>;tag=71913148-1417348035284
    Call-ID: [email protected]
    CSeq: 101 INVITE
    Timestamp: 1417347869
    Supported: 
    Contact: <sip:[email protected]:5060;transport=udp>
    Session: Media
    Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
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    PM-HO-VG-01#4839-a234-4237-95e6-a7037322f0f4
    Content-Type: application/sdp
    Content-Length: 355
    v=0
    o=BroadWorks 316169737 1 IN IP4 10.111.111.254
    s=-
    c=IN IP4 10.111.111.254
    t=0 0
    m=audio 20074 RTP/AVP 18 101 100
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    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
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    X-BroadWorks-Correlation-Info: bbf94839-a234-4237-95e6-a7037322f0f4
    Content-Type: application/sdp
    Content-Length: 355
    v=0
    o=BroadWorks 316169737 1 IN IP4 10.111.111.254
    s=-
    c=IN IP4 10.111.111.254
    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
    Nov 30 11:44:31.630: //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 11:44:31.630: //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 11:44:31.630: //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=z9hG4bK3EC81D00
    From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
    To: <sip:[email protected]>;tag=71913148-1417348035284
    Date: Sun, 30 Nov 2014 11:44:29 GMT
    Call-ID: [email protected]
    Max-Forwards: 70
    CSeq: 101 ACK
    Allow-Events: telephone-event
    Content-Length: 0
    Nov 30 11:44:31.634: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Sent: 
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e72063a5aba5d
    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]
    CSeq: 101 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-Disposition: session;handling=required
    Content-Length: 289
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 7965 2747 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 11:44:31.726: //-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=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.

  • Callcentric SIP Trunk (ITSP -- 2811 CUBE -- CUCM 8.6

    I have a SIP trunk from call centric that goes into my lab gear - they appear to be a good sip service due to cost but I'm having some trouble getting calls to route correctly. The call flow is Callcentric.com ITSP (SIP) --> 2811 (acting as cube) -->SIP Trunk --> CUCM 8.6. Phones are registered to CUCM.
    I have the sip trunk registered and calls come in to the router (I see them in ccsip message/call debugs) The 2811 running  15.1(4)M7). Callcentric sends the username of the customer in the sip Invite instead of the called number, the called number is in the TO field. I have several DID’s from Callcentric (18452055544, 18452055545, 18452055546) for my lab. There are a few configs on here for CME where the customer number (17772253754) is simply translated to their phone DN - which is fine if you only have 1 DN with callcentric but more than 1 and thats not feasible since every inbound did will be matched to that 17772253754 translation/phone dn.
    I’m using the a guide from http://tblog.cisco.be/2011/02/17/cube-conditional-sip-profiles/ using the Copy function as described http://www.cisco.com/c/en/us/products/collateral/ios-nx-os-software/ios-software-release-15-1-3-t/product_bulletin_c25-635704.html
    I haven’t been able to find anything where they actually explain all the header fields so Its mostly trial and error.. so far mostly error.  I think I’m close.. but who knows. Any assistance would be greatly appreciated
    voice class sip-profiles 1
    request INVITE peer-header sip TO copy ".sip:(.*)@." u01
    request INVITE sip-header SIP-Req-URI modify ".*@(.*)" "INVITE sip:\u01@\1"
    CUCM (single/pub)- 192.168.1.200
    2811 acting as cube - 192.168.1.203
    Calling Number - 18165297500
    Called Number - 18452055544
    vrtr1#show  sip register status
    Line                             peer       expires(sec) registered P-Associ-URI
    ================================ ========== ============ ========== ============
    17772253754                      -1         20           yes
    vrtr1#
    The Call Setup Information is:
    Call Control Block (CCB) : 0x49646C28
    State of The Call        : STATE_DEAD
    TCP Sockets Used         : NO
    Calling Number           : 18165297500
    Called Number            : 17772253754 (my customer number not called number)
    Source IP Address (Sig  ): 192.168.1.203 (my 2811 router)
    Destn SIP Req Addr:Port  : 204.11.192.159:5080
    Destn SIP Resp Addr:Port : 204.11.192.159:5080
    Destination Name         : 204.11.192.159
    Feb 14 11:20:53.303: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    INVITE sip:[email protected]:5060 SIP/2.0
    v: SIP/2.0/UDP 204.11.192.159:5080;branch=z9hG4bK-805ff2443b18502ff96181045b62dd74
    f: <sip:[email protected]>;tag=3601387252-874282
    t: <sip:[email protected]>
    i: [email protected]
    CSeq: 1 INVITE
    Max-Forwards: 8
    m: <sip:[email protected]:5080;transport=udp>
    Supported: timer
    c: application/sdp
    l: 350
    v=0
    o=NexTone-MSW 2147483647 2147483647 IN IP4 204.11.192.159
    s=sip call
    c=IN IP4 204.11.192.159
    t=0 0
    m=audio 61094 RTP/AVP 18 0 8 101
    a=fmtp:18 annexb=no
    a=fmtp:101 0-15
    a=rtpmap:101 telephone-event/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:18 G729/8000
    a=ptime:20
    a=sendrecv
    a=silenceSupp:off - - - -
    a=setup:actpass
    Feb 14 11:20:53.327: //936/310B294680AD/SIP/Msg/ccsipDisplayMsg:
    Sent:
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 204.11.192.159:5080;branch=z9hG4bK-805ff2443b18502ff96181045b62dd74
    From: <sip:[email protected]>;tag=3601387252-874282
    To: <sip:[email protected]>
    Date: Fri, 14 Feb 2014 17:20:53 GMT
    Call-ID: [email protected]
    CSeq: 1 INVITE
    Allow-Events: telephone-event
    Server: Cisco-SIPGateway/IOS-12.x
    Content-Length: 0
    Feb 14 11:20:53.327: //936/310B294680AD/SIP/Msg/ccsipDisplayMsg:
    Sent:
    SIP/2.0 404 Not Found
    Via: SIP/2.0/UDP 204.11.192.159:5080;branch=z9hG4bK-805ff2443b18502ff96181045b62dd74
    From: <sip:[email protected]>;tag=3601387252-874282
    To: <sip:[email protected]>;tag=35399D8-63
    Date: Fri, 14 Feb 2014 17:20:53 GMT
    Call-ID: [email protected]
    CSeq: 1 INVITE
    Allow-Events: telephone-event
    Server: Cisco-SIPGateway/IOS-12.x
    Reason: Q.850;cause=1
    Content-Length: 0
    Feb 14 11:20:53.419: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    ACK sip:[email protected]:5060 SIP/2.0
    v: SIP/2.0/UDP 204.11.192.159:5080;branch=z9hG4bK-805ff2443b18502ff96181045b62dd74
    f: <sip:[email protected]>;tag=3601387252-874282
    t: <sip:[email protected]>;tag=35399D8-63
    i: [email protected]
    CSeq: 1 ACK
    Max-Forwards: 10
    l: 0
    u all
    Feb 14 11:20:57.067: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    ACK sip:18452055544;cic=0288;rn=6465471001;[email protected]:5070 SIP/2.0
    v: SIP/2.0/UDP 204.11.192.159:5080;branch=z9hG4bK-6bceae47efe9f53b4234698a32ac8beb
    f: <sip:[email protected]>;tag=3601387252-874282
    t: <sip:[email protected]>;tag=35399D8-63
    i: [email protected]
    CSeq: 1 ACK
    Max-Forwards: 8
    l: 0
    ************************** Running Config **************************
    sh run
    vrtr1#sh running-config
    Building configuration...
    Current configuration : 4189 bytes
    ! Last configuration change at 00:34:03 CST Fri Feb 14 2014
    ! NVRAM config last updated at 20:26:58 CST Thu Feb 13 2014
    ! NVRAM config last updated at 20:26:58 CST Thu Feb 13 2014
    version 15.1
    service timestamps debug datetime msec localtime
    service timestamps log datetime msec localtime
    no service password-encryption
    hostname vrtr1
    boot-start-marker
    boot system flash:
    boot system flash flash:c2800nm-ipvoicek9-mz.151-4.M7.bin
    boot-end-marker
    card type t1 0 0
    logging buffered 4096 notifications
    enable password cisco
    no aaa new-model
    memory-size iomem 5
    clock timezone CST -6 0
    clock summer-time CST recurring
    no network-clock-participate wic 0
    dot11 syslog
    ip source-route
    ip cef
    ip name-server 192.168.1.9
    no ipv6 cef
    multilink bundle-name authenticated
    voice service voip
    ip address trusted list
      ipv4 192.168.1.0 255.255.255.0
      ipv4 204.11.192.0 255.255.255.0
    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 sip moved-temporarily
    no supplementary-service sip refer
    sip
      bind control source-interface FastEthernet0/0
      bind media source-interface FastEthernet0/0
      registrar server expires max 1800 min 1800
      localhost dns:callcentric.com
      outbound-proxy dns:callcentric.com
    voice class codec 1
    codec preference 1 g729r8
    codec preference 2 g711ulaw
    voice class sip-profiles 1
    request INVITE peer-header sip TO copy ".sip:(.*)@." u01
    request INVITE sip-header SIP-Req-URI modify ".*@(.*)" "INVITE sip:\u01@\1"
    voice-card 0
    crypto pki token default removal timeout 0
    license udi pid CISCO2811 sn FTX1133A4QR
    controller T1 0/0/0
    cablelength long 0db
    interface FastEthernet0/0
    description ** LAN **
    ip address 192.168.1.203 255.255.255.0
    duplex auto
    speed auto
    h323-gateway voip interface
    h323-gateway voip bind srcaddr 192.168.1.203
    interface FastEthernet0/1
    no ip address
    shutdown
    duplex auto
    speed auto
    ip forward-protocol nd
    no ip http server
    no ip http secure-server
    ip route 0.0.0.0 0.0.0.0 192.168.1.1
    snmp mib persist circuit
    control-plane
    voice-port 0/1/0
    voice-port 0/1/1
    voice-port 0/1/2
    voice-port 0/1/3
    ccm-manager mgcp
    no ccm-manager fax protocol cisco
    ccm-manager music-on-hold
    ccm-manager config server 192.168.1.200 
    ccm-manager config
    mgcp
    mgcp call-agent 192.168.1.200 2427 service-type mgcp version 0.1
    mgcp dtmf-relay voip codec all mode out-of-band
    mgcp rtp unreachable timeout 1000 action notify
    mgcp modem passthrough voip mode nse
    mgcp package-capability rtp-package
    mgcp package-capability sst-package
    mgcp package-capability pre-package
    no mgcp package-capability res-package
    no mgcp package-capability fxr-package
    no mgcp timer receive-rtcp
    mgcp sdp simple
    mgcp fax t38 inhibit
    mgcp rtp payload-type g726r16 static
    mgcp bind control source-interface FastEthernet0/0
    mgcp bind media source-interface FastEthernet0/0
    mgcp profile default
    dial-peer voice 999100 pots
    service mgcpapp
    port 0/1/0
    dial-peer voice 999101 pots
    service mgcpapp
    port 0/1/1
    dial-peer voice 999102 pots
    service mgcpapp
    port 0/1/2
    dial-peer voice 999103 pots
    service mgcpapp
    port 0/1/3
    dial-peer voice 999010 pots
    service mgcpapp
    port 0/1/0
    dial-peer voice 6 voip
    description ## INBOUND DID to CUCM ##
    session protocol sipv2
    session target ipv4:192.168.1.200
    incoming called-number 17772253754
    voice-class sip profiles 1
    dtmf-relay h245-alphanumeric
    no vad
    dial-peer voice 7 voip
    description ## INBOUND DID to CUCM ##
    session protocol sipv2
    session target ipv4:192.168.1.200
    incoming called-number 1845205554[4-5]
    voice-class sip profiles 1
    dtmf-relay h245-alphanumeric
    no vad
    sip-ua
    credentials username 17772253754 password 7 106C1B49111F17194D realm callcentric.com
    authentication username 17772253754 password 7 08035E1E1D11000553 realm callcentric.com
    no remote-party-id
    retry invite 2
    retry register 10
    timers connect 100
    mwi-server dns:callcentric.com expires 3600 port 5060 transport udp
    registrar dns:callcentric.com expires 3600
    sip-server dns:callcentric.com
    host-registrar
    line con 0
    line aux 0
    line vty 0 4
    password cisco
    login
    transport input all
    scheduler allocate 20000 1000
    ntp server 199.102.46.72
    ntp server 23.227.162.123 prefer
    end
    exit

    Thank you for the reply. I've updated the dial-peers as sugested. I'm now seeing an invite go out to my CUCM however the call fails with a 403 (forbidden) which appears to come from the ITSP (Callcentric). I've included a new set of ccsip message debugs and the dial-peers as adjusted. Please let me know what you think.
    dial-peer voice 6 voip
    description ## INBOUND CALL from ITSP ##
    session protocol sipv2
    session target sip-server
    incoming called-number 17772253754
    voice-class sip profiles 1
    dtmf-relay rtp-nte
    no vad
    dial-peer voice 100 voip
    description ## INBOUND DID to CUCM ##
    destination-pattern 17772253754
    session protocol sipv2
    session target ipv4:192.168.1.200
    voice-class sip profiles 1
    dtmf-relay rtp-nte
    no vad
    dial-peer voice 7 voip
    description ## INBOUND DID to CUCM ##
    session protocol sipv2
    session target ipv4:192.168.1.200
    incoming called-number 1845205554[4-5]
    voice-class sip profiles 1
    dtmf-relay rtp-nte
    no vad
    Feb 15 10:18:11.424: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    INVITE sip:[email protected]:5060 SIP/2.0
    v: SIP/2.0/UDP 204.11.192.164:5080;branch=z9hG4bK-d78945b444598bc22c8509d069f4789d
    f: ;tag=3601469891-655
    t: [email protected]>
    i: [email protected]
    CSeq: 1 INVITE
    Max-Forwards: 8
    m:
    Supported: timer
    c: application/sdp
    l: 350
    v=0
    o=NexTone-MSW 2147483647 2147483647 IN IP4 204.11.192.164
    s=sip call
    c=IN IP4 204.11.192.164
    t=0 0
    m=audio 61782 RTP/AVP 18 0 8 101
    a=fmtp:18 annexb=no
    a=fmtp:101 0-15
    a=rtpmap:101 telephone-event/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:18 G729/8000
    a=ptime:20
    a=sendrecv
    a=silenceSupp:off - - - -
    a=setup:actpass
    Feb 15 10:18:11.456: //2419/9933162D820E/SIP/Msg/ccsipDisplayMsg:
    Sent:
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 204.11.192.164:5080;branch=z9hG4bK-d78945b444598bc22c8509d069f4789d
    From: ;tag=3601469891-655
    To: [email protected]>
    Date: Sat, 15 Feb 2014 16:18:11 GMT
    Call-ID: [email protected]
    CSeq: 1 INVITE
    Allow-Events: telephone-event
    Server: Cisco-SIPGateway/IOS-12.x
    Content-Length: 0
    Feb 15 10:18:11.460: //2420/9933162D820E/SIP/Msg/ccsipDisplayMsg:
    Sent:
    INVITE sip:@192.168.1.200:5060 SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bK9A91F35
    From: [email protected]>;tag=8408644-12C8
    To:
    Date: Sat, 15 Feb 2014 16:18:11 GMT
    Call-ID: [email protected]
    Supported: 100rel,timer,resource-priority,replaces,sdp-anat
    Min-SE:  1800
    Cisco-Guid: 2570262061-2509443555-2182021079-2501285341
    User-Agent: Cisco-SIPGateway/IOS-12.x
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    CSeq: 101 INVITE
    Timestamp: 1392481091
    Contact:
    Expires: 180
    Allow-Events: telephone-event
    Max-Forwards: 7
    Content-Type: application/sdp
    Content-Disposition: session;handling=required
    Content-Length: 273
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 2786 1511 IN IP4 192.168.1.203
    s=SIP Call
    c=IN IP4 192.168.1.203
    t=0 0
    m=audio 18168 RTP/AVP 18 101
    c=IN IP4 192.168.1.203
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=no
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=ptime:20
    Feb 15 10:18:11.552: //2420/9933162D820E/SIP/Msg/ccsipDisplayMsg:
    Received:
    SIP/2.0 407 Proxy Authentication Required
    v: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bK9A91F35;rport=57100;received=24.123.98.94
    f: [email protected]>;tag=8408644-12C8
    t:
    i: [email protected]
    CSeq: 101 INVITE
    Proxy-Authenticate: Digest realm="callcentric.com", domain="sip:callcentric.com", nonce="8ae6b7b1cea74cf401e8a26fd3c7371b", opaque="", stale=TRUE, algorithm=MD5
    l: 0
    Feb 15 10:18:11.560: //2420/9933162D820E/SIP/Msg/ccsipDisplayMsg:
    Sent:
    ACK sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bK9A91F35
    From: [email protected]>;tag=8408644-12C8
    To:
    Date: Sat, 15 Feb 2014 16:18:11 GMT
    Call-ID: [email protected]
    Max-Forwards: 70
    CSeq: 101 ACK
    Allow-Events: telephone-event
    Content-Length: 0
    Feb 15 10:18:11.560: //2420/9933162D820E/SIP/Msg/ccsipDisplayMsg:
    Sent:
    INVITE sip:@192.168.1.200:5060 SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bK9AA1BA3
    From: [email protected]>;tag=8408644-12C8
    To:
    Date: Sat, 15 Feb 2014 16:18:11 GMT
    Call-ID: [email protected]
    Supported: 100rel,timer,resource-priority,replaces,sdp-anat
    Min-SE:  1800
    Cisco-Guid: 2570262061-2509443555-2182021079-2501285341
    User-Agent: Cisco-SIPGateway/IOS-12.x
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    CSeq: 102 INVITE
    Timestamp: 1392481091
    Contact:
    Expires: 180
    Allow-Events: telephone-event
    Proxy-Authorization: Digest username="17772253754",realm="callcentric.com",uri="sip:[email protected]:5060",response="a381f10fbbfbd255b444569fef0dddfe",nonce="8ae6b7b1cea74cf401e8a26fd3c7371b",opaque="",algorithm=MD5
    Max-Forwards: 7
    Content-Type: application/sdp
    Content-Disposition: session;handling=required
    Content-Length: 273
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 2786 1511 IN IP4 192.168.1.203
    s=SIP Call
    c=IN IP4 192.168.1.203
    t=0 0
    m=audio 18168 RTP/AVP 18 101
    c=IN IP4 192.168.1.203
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=no
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=ptime:20
    Feb 15 10:18:11.648: //2420/9933162D820E/SIP/Msg/ccsipDisplayMsg:
    Received:
    SIP/2.0 403 Incorrect Authentication
    v: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bK9AA1BA3;rport=57100;received=24.123.98.94
    f: [email protected]>;tag=8408644-12C8
    t:
    i: [email protected]
    CSeq: 102 INVITE
    l: 0
    Feb 15 10:18:11.660: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Sent:
    ACK sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bK9AA1BA3
    From: [email protected]>;tag=8408644-12C8
    To:
    Date: Sat, 15 Feb 2014 16:18:11 GMT
    Call-ID: [email protected]
    Max-Forwards: 70
    CSeq: 102 ACK
    Allow-Events: telephone-event
    Content-Length: 0
    Feb 15 10:18:11.660: //2419/9933162D820E/SIP/Msg/ccsipDisplayMsg:
    Sent:
    SIP/2.0 403 Forbidden
    Via: SIP/2.0/UDP 204.11.192.164:5080;branch=z9hG4bK-d78945b444598bc22c8509d069f4789d
    From: ;tag=3601469891-655
    To: [email protected]>;tag=8408714-B60
    Date: Sat, 15 Feb 2014 16:18:11 GMT
    Call-ID: [email protected]
    CSeq: 1 INVITE
    Allow-Events: telephone-event
    Server: Cisco-SIPGateway/IOS-12.x
    Reason: Q.850;cause=57
    Content-Length: 0
    Feb 15 10:18:11.752: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    apsc-vrtr1#ACK sip:[email protected]:5060 SIP/2.0
    v: SIP/2.0/UDP 204.11.192.164:5080;branch=z9hG4bK-d78945b444598bc22c8509d069f4789d
    f: ;tag=3601469891-655
    t: [email protected]>;tag=8408714-B60
    i: [email protected]
    CSeq: 1 ACK
    Max-Forwards: 10
    l: 0
    vrtr1#u al
    Feb 15 10:18:14.776: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    ACK sip:18452055544;cic=0288;rn=6465471001;[email protected]:5070 SIP/2.0
    v: SIP/2.0/UDP 204.11.192.164:5080;branch=z9hG4bK-e437c2c5cac5f1a6e147c1cd7c98aad7
    f: ;tag=3601469891-655
    t: [email protected]>;tag=8408714-B60
    i: [email protected]
    CSeq: 1 ACK
    Max-Forwards: 8
    l: 0

  • CME required CUBE license for ISP SIP Trunk

    Hi,
    We have IP telephony setup with CME 9.1 with 2921 ISRG2 router.
    Client would like to take ISP SIP trunk.
    Do we require CUBE license for the same.Because I tried in the 2851 router without the CUBE  license and its working fine.
    How I can check CUBE license installed my CME.
    When I check license in CME I can see that UC K9 and IPBasek9 license are permanent.
    Do this enough for the SIP trunk configuration.
    Thanks & Regards
    Nithin Louis.

    Just to add to George answer - CUBE licenses are somekind "honor" based right now which means you can configure everything and it will work but you need license paper for proving that you have adequate number of CUBE licenses for your system.
    Maybe in future Cisco will improve this with option that you will need to install license and only then it will work - like many other licensing right now...
    BR,
    Dragan

  • 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

  • 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

  • Issue with instant ringback when using sip trunk to SP

    Hi all,
    We use CUCM 8.0.2.
    We have a SIP trunk to a SP connected via one of our Cisco 2911 routers configured as a CUBE.
    Cisco IOS Software, C2900 Software (C2900-UNIVERSALK9-M), Version 15.0(1)M3, RELEASE SOFTWARE (fc2)
    c2900-universalk9-mz.SPA.150-1.M3.bin
    Cisco CISCO2911/K9 (revision 1.0)
    Technology Package License Information for Module:'c2900'
    Technology Technology-package
                      Current       Type
    ipbase        ipbasek9      Permanent
    security      securityk9    Permanent
    uc              uck9            Permanent
    data           None            None
    We also have several ISDN lines that run out via various Cisco routers configured as H323 gateways.
    We use 7945 and CIPC for our phones.
    We're having an issue with calls going via the SIP trunk where we hear ringing instantly after dialling - but before the actual device at the other end starts ringing (considerable difference).
    Using the SIP trunk: If I make a call to my mobile phone - I hear ringing instantly - about 3 rings before my mobile phone actually starts ringing - undesireable.
    Using H323 gateway: If I make a call to my mobile phone - I hear silence for a bit - then ringing when the mobile starts ringing - desired.
    Using SIP trunk: If I make a call to a landline that is ready - it rings instantly for at least 1 ring - before the actual phone I'm calling starts ringing - undesireable.
    Using H323 gateway: There is a momentary pause before hearing ringing on my phone and the phone I dialled - desired.
    Using SIP trunk: If I make a call to a landline that is off-hook (with no call-waiting/etc.) - it rings once and then returns the busy signal (the worst issue) - undesireable.
    Using H323 gateway: There is a momentary pause before hearing busy signal - desired.
    Phone to phone internally (same network): Operates as expected (instantly rings locally and on the phone I'm calling). Between phones that utilise the SIP trunk and phones that utilise the H323 gateways within the same network - communication is instant and as expected.
    Any ideas why this happens and how to stop it?
    I want it to not ring until the situation is known and that it can provide the appropriate feedback (ringing/busy/etc.).
    Some possibly relevant config (note that there is a known bug with this IOS that meant I had to declare the codec in each dial-peer as the voice class would not work):
    voice service voip
    address-hiding
    mode border-element
    allow-connections sip to sip
    sip
      bind control source-interface GigabitEthernet0/0
      bind media source-interface GigabitEthernet0/0
      header-passing error-passthru
      early-offer forced
      midcall-signaling passthru
    interface GigabitEthernet0/0
    ip address x.x.x.x 255.255.255.252
    ip access-group acl.SIP-IN in
    no ip redirects
    no ip unreachables
    ip verify unicast reverse-path
    ip virtual-reassembly
    duplex full
    speed 100
    no cdp enable
    gateway
    timer receive-rtp 1200
    sip-ua
    connection-reuse
    gatekeeper
    shutdown
    dial-peer voice 1 voip
    description *** INBOUND CALLS FROM CARRIER ***
    translation-profile incoming SIPTRUNK-INCOMING
    session protocol sipv2
    incoming called-number #blah blah#
    dtmf-relay rtp-nte
    codec g711alaw
    ip qos dscp cs5 media
    no vad
    dial-peer voice 61 voip
    description **** WA, SA AND NT NUMBERS ****
    destination-pattern 0[8]........
    session protocol sipv2
    session target ipv4:<MY SP's SIP SERVER>
    incoming called-number 0[8]........
    dtmf-relay rtp-nte
    codec g711alaw
    ip qos dscp cs5 media
    no vad
    dial-peer voice 81 voip
    description **** MOBILE NUMBERS ****
    destination-pattern 0[4]........
    session protocol sipv2
    session target ipv4:<MY SP's SIP SERVER>
    incoming called-number 0[4]........
    dtmf-relay rtp-nte
    codec g711alaw
    ip qos dscp cs5 media
    no vad
    dial-peer voice 500 voip
    description *** INBOUND SIP TRUNK TO CUCM PUB ***
    translation-profile outgoing SIPTRUNK-CALLING-ADD-0
    preference 1
    destination-pattern 5[12]..
    session protocol sipv2
    session target ipv4:<OUR CUCM PUBLISHER IP>
    dtmf-relay rtp-nte
    codec g711alaw
    ip qos dscp cs5 media
    no vad
    Any help or a point in the right direction would be greatly appreciated.
    Cheers,
    Brett

    I ended up resolving this issue as follows:
    In CUCM, under Device > Device Settings > SIP Profile.
    I modifed the profile relevant to my SIP trunk, under the "Trunk Specific Configuration", I set "SIP Rel1XX Options" from "Disabled" to "Send PRACK if 1xx Contains SDP".
    Now, I get the expected delay before hearing ringback.
    Solved!

  • Best way to implement SIP Options Pings on a SIP Trunk

    I wanted to see if anyone had suggestions on the best way to configure SIP Options Pings.
    Typically I would configure them per dial-peer.  However, I really want to do it per destination IP address.  I do not want a SIP Options ping for every single dial-peer being sent out every X seconds.  
     Example:
    In my case I have 4 SIP trunks in the same CUBE.  Each pointing to a different destination IP.  There are 6 dial-peers per SIP trunk.  I really do not want 24 option pings going out every X seconds.  I guess I've never actually did a debug to see how many pings are going out at a time but I am assuming it sends one for each dial-peer or does it?
    If I am correct in my assumption, is there way to only send one ping per destination IP and if that single IP goes unresponsive then all 6 dial-peers go down?

    Hello,
    The OOD option ping is sent per dial-peer to the destination.
    Restrictions
    •The Cisco Unified Border Element OOD Options ping feature can only be configured at the VoIP Dial-peer level.
    •All dial peers start in an active (not busied out) state on a router boot or reboot.
    •If a dial-peer has both an outbound proxy and a session target configured, the OOD options ping is sent to the outbound proxy address first.
    •Though multiple dial-peers may point to the same SIP server IP address, an independent OOD options ping is sent for each dial-peer.
    •If a SIP server is configured as a DNS hostname, OOD Options pings are sent to all the returned addresses until a response is received.
    •Configuration for Cisco Unified Border Element OOD and TDM Gateway OOD are different, but can co-exist.
    //Suresh
    Please rate all the useful posts.

  • SIP trunking monitoring and usage report

    Hello all,
    Recently the IP telephony system has migrated to SIP trunking from PRI.   SIP trunking is running on CUBE 2900 serials.
    Could anyone advise the following?    much appreciate it.
    .  SIP trunking monitoring :  
       If MIB needed,   any brief steps to configure it on monitored device and monitoring system?
    . SIP trunking usage report :  
       Any software/application could generate the usage report?   and show the cocurrent calls?
    Thanks again,
    master002

    Thanks Nadeem for the reply,  looks Variphy Insight is the option?
    I have installed RTMT and was able to get the real time data instead of the usage report.  I also have CUOM -- operation manager, I haven't customized it to monitor SIP trunking yet.
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