CUCM route calls diferents gateways/sip trunks

Hi at all, I have CUCM 6.1.1 and I want to route calls throughs diferents gateways or sip trunks.
I planned to do with route groups, but I can not add on a route group a H323 gateway and a SIP trunk at the same time.
How can route calls in different ways?
In the CUCM page "Route patterns" I want to make alternative routes, for example, the number 6666 is on route "666X" through a "gateway/route list", but if I can not contact by going this route I need to go through the alternative route "XXXX" through another "gateway/route list".
How can I make by going first to one pattern and then the other pattern?
Thanks!
Fran

Ok thanks but one question more.... if I have a MGCP Gateway? Can I do this from my MGCP Gateway? or I need an H323 Gateway.
And another possibility.... I dont know if it's right....
Can I do this with Partitions and CSS?
This is for example I 'll have a CSS "Global" with Partitions (Primary and Secondary);
It could go the route first to 666x Gateway with CSS "Global" and partitions (Primary and Secondary). This way I do not know if it is routed first through the Gateway of the partition as Primary and Secondary alternative partition that is served by the SIP Trunk.
Using the "Dial Number Analyzer" I get the second path XXXX (SIP Trunk) as an alternative route ...

Similar Messages

  • Controlling which CUCM server communicate over a SIP trunk

    We have 3 CUCM servers, two sub and one pub at two different physical locations.
    There are two SIP trunk servers (non Cisco), we wish to have the CUCM at the same physical location to communicate with the SIP trunk device at its location, instead of going over the WAN to communicate with the other one.
    Communications are initiated from the CUCM side through a route pattern that points to a RL/RG that contains the SIP trunk.
    The SIP trunk uses a DP that uses a CUCM group that only contains the local CUCM server.
    Can this be setup to reliably control which CUCM server talks to the SIP trunk server at its location?
    Do we need to configure something differently?

    Hi,
    If the CUCM's are located in different physical location then the communication should be over the IP-WAN and if it is about different CUCM Cluster then we can use H323 ICT Trunk between CUCM server for communication which is again WANLink.
    Still not clear with your query, However as you want to control the Gwy/Trunks using the Route Pattern. You can create two separate route pattern for internal & external.
    Wherein in Internal Route Pattern you can specify the ICT/SipTrunk/GWY which is connected within the cluster under Route List/RouteGroup to route the call.
    And for External Route Pattern specify the Gwy/SipTrunk which connects you to the outside world.
    Regards,
    Venkatesh

  • Unable to place call on calls on hold - SIP Trunk from CUCM to CUBE and from CUBE to ISTP

    Hi Cisco Community,
    I have a SIP Trunk setup between the CUCM and CUBE and another SIP Dial Peers from the CUBE to the ITSP. All incoming/outgoing calls, DTMF-Relay works well except one thing which is the ability to hold the call.
    On the SIP Trunk from the CUCM to the CUBE, I did not select “MTP” because when I do so, I am forced to select my preferred MTP codec which when selected G.729/G.729a, all my outgoing calls goes out using G.729r8. This codec works well for most of the calls until the ITSP replies back with G.729br8. When this condition occur, my call simply fails (this is very intermittent and only some random numbers).
    That said, I have some issues with DTMF Relay when I select MTP on the SIP Trunk. DTMF Relay only works if the call is G.729r8 all the way from the CUCM to the ITSP. If the ITSP replies back with G.729br8, the call might established but will simply be “voice-only”.
    The current setup is no MTP is selected and everything is working perfectly. I am happy with that until I place a call on hold, which when I do so, the call immediately terminate. Could you please help me understand why?
    I have all media resources configured such as G.729r8 MTP, G.729br8 MTP, G.711u MTP, Transcoding with all codecs, etc. All MRG and MRGL are configured on all devices and SIP Trunks.
    Below is an example of a call that is connected with the current setup:
    Note:
    IP: 10.18.81.2 (CUBE)
    IP: 10.18.81.11 (CUCM SUB)
    IP: 10.111.111.254 (ITSP SBC)
    PM-HO-VG-01#
    PM-HO-VG-01#
    Nov 30 11:44:29.938: //-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=z9hG4bK2e72063a5aba5d
    From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
    To: <sip:[email protected]>
    Date: Sun, 30 Nov 2014 11:44:29 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: 101 INVITE
    Expires: 180
    Allow-Events: presence, kpml
    Supported: X-cisco-srtp-fallback,X-cisco-original-called
    Call-Info: <sip:10.18.81.11:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
    Cisco-Guid: 1020645888-0000065536-0000124117-0189862410
    Session-Expires:  1800
    Contact: <sip:[email protected]:5060>
    Max-Forwards: 70
    Content-Length: 0
    Nov 30 11:44:29.942: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Sent: 
    SIP/2.0 100 Trying
    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]>
    Date: Sun, 30 Nov 2014 11:44:29 GMT
    Call-ID: [email protected]
    CSeq: 101 INVITE
    Allow-Events: telephone-event
    Server: Cisco-SIPGateway/IOS-15.2.4.M5
    Content-Length: 0
    Nov 30 11:44:29.946: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Sent: 
    INVITE sip:[email protected]:5060 SIP/2.0
    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]>
    Date: Sun, 30 Nov 2014 11:44:29 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: 101 INVITE
    Timestamp: 1417347869
    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
    Max-Forwards: 69
    Session-Expires:  1800
    Content-Type: application/sdp
    Content-Disposition: session;handling=required
    Content-Length: 301
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 7676 6958 IN IP4 10.18.81.2
    s=SIP Call
    c=I
    PM-HO-VG-01#N IP4 10.18.81.2
    t=0 0
    m=audio 22256 RTP/AVP 18 0 8 101
    c=IN IP4 10.18.81.2
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=yes
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    Nov 30 11:44:29.950: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Received: 
    SIP/2.0 100 Trying
    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]>
    Call-ID: [email protected]
    CSeq: 101 INVITE
    Timestamp: 1417347869
    Nov 30 11:44:30.658: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    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
    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 11:44:30.662: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    Sent: 
    SIP/2.0 183 Session Progress
    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: 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
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=rtpmap:19 CN/8000
    a=ptime:20
    Nov 30 11:44:31.226: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
    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
    X-BroadWorks-Correlation-Info: bbf9
    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
    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/Msg/ccsipDisplayMsg:
    Received: 
    SIP/2.0 200 OK
    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>
    Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
    Accept: application/media_control+xml,application/sdp,application/xml
    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.

  • 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.

  • Unable to perform call transfer & call park through SIP Trunk (SKYPE)

    The Scenario is:
    I have set up a SIP trunk to SKYPE and we are able to make outbound call to a number via SIP Trunk.
    After the call is established, when we tried to make call transfer, the call DROP and the phone at the other end shows error "Temp Fail".
    I tried to "enable MTP" in SIP Trunk and We are able to perform call-transfer but it limits the call session to 1.
    Anyone has facing the same issue?

    MTP is needed to invoke supplementary functions like hold, transfer etc. Make sure that the MTP is checked on SIP trunk, MTP is assigned to the MRGL of the device pool on SIP trunk and has sufficient resources.
    HTH
    Manish

  • 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

  • Changing external Caller ID over a SIP Trunk to SIP Provider

    I am working with a client and when they place calls out to any external user they have the wrong name showing on the external caller ID. 
    I have spoken with the SIP provider and apparently they want us to pass the CNAM, or rather they have it setup for us to do this.
    I opened a case with Cisco and the TAC engineer said the provider has to do this because it cannot be done from CUCM or the gateway.
    For example, it says right now "location A" for external calls and I want to change this to say "location B" . 
    Is this even possible?

    what is the call flow? did you check the caller name in SIP trunk configuration?

  • CUCM 8.6 call busy between SIP phones and thirdparty phones

    Hi Everybody,
    I have the following error on my logs:
    Invalid Disconnect Cause(cause=47), No Reason Header Appended
    14:46:50.091 |//SIP/SIPCdpc(3,74,281707)/ci=144600614/ccbId=35257792/scbId=0/getXCiscoViPRFallbackIDAndDTMFKey: Device type 8, Pstn Fallback is not enabled|3,100,63,1.36454459^192.168.0.15^*
    14:46:50.091 |//SIP/SIPCdpc(3,74,281707)/ci=144600614/ccbId=35257792/scbId=0/getDefCcRegister: Secure status=1, mSrtpPresent=0|3,100,63,1.36454459^192.168.0.15^*
    14:46:50.091 |//SIP/SIPCdpc(3,74,281707)/ci=144600614/ccbId=35257792/scbId=0/compareAndUpdateMedia: sdpStatus=0, CMEndPointSDP role=1, SIPEndPointSdpRole=2|3,100,63,1.36454459^192.168.0.15^*
    14:46:50.091 |//SIP/SIPCdpc(3,74,281707)/ci=144600614/ccbId=35257792/scbId=0/getDefCcRegister: Secure status=1, mSrtpPresent=0|3,100,63,1.36454459^192.168.0.15^*
    14:46:50.091 |//SIP/SIPCdpc(3,74,281707)/ci=144600614/ccbId=35257792/scbId=0/handleSIPUACSessionExpires: isMidCall[0], response[200], method[102]|3,100,63,1.36454459^192.168.0.15^*
    14:46:50.091 |//SIP/SIPCdpc(3,74,281707)/ci=144600614/ccbId=35257792/scbId=0/parseSessionExpires: refresh_interval[1800], refresher[uas]|3,100,63,1.36454459^192.168.0.15^*
    14:46:50.091 |//SIP/SIPCdpc(3,74,281707)/ci=144600614/ccbId=35257792/scbId=0/setSIPSessionExpiresTimer: interval[1768] secs|3,100,63,1.36454459^192.168.0.15^*
    14:46:50.091 |//SIP/SIPCdpc(3,74,281707)/ci=144600614/ccbId=35257792/scbId=0/handleSecureRec: enforce srtp flag: 0, remote end srtp support: 0|3,100,63,1.36454459^192.168.0.15^*
    14:46:50.091 |//SIP/SIPCdpc(3,74,281707)/ci=144600614/ccbId=35257792/scbId=0/updateCNToCC: identityCngFlag[0x1f], isConnInfoInd[1], ccContactHeader[]|3,100,63,1.36454459^192.168.0.15^*
    14:46:50.091 |//SIP/SIPCdpc(3,74,281707)/ci=144600614/ccbId=35257792/scbId=0/handleSIPConnectInd: Exit with state = outCall_200Rcvd|3,100,63,1.36454459^192.168.0.15^*
    14:46:50.091 |//SIP/SIPCdpc(3,74,281707)/ci=144600614/ccbId=35257792/scbId=0/handleSIPConnectInd: Exit with state = outCall_200Rcvd|3,100,63,1.36454459^192.168.0.15^*
    14:46:50.091 |//SIP/SIPCdpc(3,74,281707)/ci=144600614/ccbId=35257792/scbId=0/setSIPUpdateFlags: mIsUpdateForSignalingAllowed = 1  mIsUpdateForMediaAllowed = 1  mPendingOutgoingUpdate = 0|3,100,63,1.36454459^192.168.0.15^*
    14:46:50.098 |//SIP/SIPCdpc(3,74,281707)/ci=144600614/ccbId=35257792/scbId=0/appendReasonHdr: appendReasonHdr - Invalid Disconnect Cause(cause=47), No Reason Header Appended|3,100,63,1.36454459^192.168.0.15^*
    14:46:50.098 |//SIP/SIPCdpc(3,74,281707)/ci=144600614/ccbId=35257792/scbId=0/addTransparencyInfo: attaching transparency object|3,100,63,1.36454459^192.168.0.15^*
    14:46:50.098 |//SIP/SIPCdpc(3,74,281707)/ci=144600614/ccbId=35257792/scbId=0/appendRPIDHdrForOriginalCalledParty: SIP device does not Support Orig Dialled Phone nego: 0|3,100,63,1.36454459^192.168.0.15^*
    14:46:50.098 |//SIP/SIPCdpc(3,74,281707)/ci=144600614/ccbId=35257792/scbId=0/getDefAe: SIPCdpc=281707, nodeId=3, processNumber=73  ci=144600614, branch=0|3,100,63,1.36454459^192.168.0.15^*
    14:46:50.098 |MRM::waiting_MrmDeallocateMtpResourceReq- Deallocate received for CI=53993831 count=0|3,100,63,1.36454459^192.168.0.15^*
    14:46:50.098 |MRM::waiting_DeallocateMtpResourceReq- ERROR  Deallocate received for an unknown Call Identifier  Ci = 53993831|3,100,63,1.36454459^192.168.0.15^*
    14:46:50.098 |//SIP/SIPHandler/ccbId=0/scbId=0/sip_stop_timer: type=SIP_TIMER_DISCONNECT value=500 retries=10|3,100,63,1.36454459^192.168.0.15^*
    14:46:50.098 |//SIP/SIPHandler/ccbId=0/scbId=0/sip_start_timer: type=SIP_TIMER_DISCONNECT value=500 retries=10|3,100,63,1.36454459^192.168.0.15^*
    14:46:50.098 |//SIP/SIPUdp/wait_SdlSPISignal: Outgoing SIP UDP message to 192.168.0.15:[5060]:
    the calling number is 34967850938
    the callied number is 19026

    Julien,
    Please use the link be low to collect cucm traces and use the advanced editor on the forum (located on top right hand corner of the discussion widnow) to attach the trace
    https://supportforums.cisco.com/docs/DOC-29901
    Ensure you collect the trace from the folowing
    1. the server that the phone is registered to
    2. If this server is different from the server in the cucm group of the sip trunk, then you need to also collect traces from the server (s) in the cucm group assiged to the sip trunk that connects to the 3rd party cluster...
    NB: If you have three servers in the cucm group of the sip trunk, you have to collect the trace from all three servers. This is because calls are dsitributed in a round robin fashion to servers in a sip trunk...
    FInally before you send the trace over, please ensure the calling and called numbers are present. Also include the time of the test call
    Please rate all useful posts
    "opportunity is a haughty goddess who waste no time with those who are unprepared"

  • PMF to allow outgoing calls through SIP Trunk Without Registering

    Hello,
    I have an intermitant issue with one of our UC320W's running 2.3.2(6) firmware.  The customers VOIP SIP trunk becomes unregistered for periods of time, stopping incoming and outgoing calls.  Once unregistered it takes quite a while to rergister.  Our service provider has informed us that the re-register period is the cause and we should try and shorten it, so first question is there a way to do this, also what is the re-register retry window in the first place?
    I have an analogue line that can receive calls only so I have made this the fallover number with the VOIP provider, that gives a little releife for incoming calls, but not outgoing.  I beleive in other phone systems a SIP trunk does not need to be registered to make an outgoing call, and it is usually an option to say only make outgoing calls if the SIP trunk is registered.  I cannot find that option anywhere to deselect it, is there a PMF I could apply to allow outgoing calls without registering?
    Thank you,
    Tony

    Hi Tony,
    Please install the SIP_Trunk_Register_Timer.pmf at status->Devices->Alter PMFs in configure utility. Please remember to apply the configuration afterwards. This PMF can let user to select the re-register period. You can find the PMF at https://supportforums.cisco.com/docs/DOC-16301
    Regards,
    Wendy Yang

  • Confused by basic SIP Trunk configuration.

    I've went through a few basic SIP trunk configurations and Youtube videos the last couple days but can't figure out what I'm doing wrong.
    I've set up H323 and MGCP no problem, but I can't figure out the SIP trunk set up. I'm guessing there are some concepts I'm not understanding yet.
    I've got a CUCM lab set up. A 2851 PSTN Simulator, 2851 H323 Gateway at the Main site with a 9.0 CUCM setup in that site and a Branch site that I'm trying to set up as a SIP trunk to connect two phones.
    CUCM is on the 192.168.5.x/24 subnet. 172.16.0.x/24 is the subnet connecting the serial(internet) cable between the two gateways in which I'm trying to establish the trunk between.
    The Branch phones are still registering with the CUCM at the main site. The Route Pattern is looking to the Branch Route List which has the SIP Trunk listed. I'm just getting a fast busy when trying to place a call from the branch site to the main site.
    The most frustrating thing I'm not understanding, is that the debug ccsip and call debugs on my SIP Branch gateway shows absolutely nothing.  I've tried registering the branch phones with the SIP Trunk, but stopped when I figured that shouldn't be necessary.
    If someone can make some sense of this, I'd truly appreciate it!

    Hello Aditya and thanks for the consideration!
    I do have a direct IP connection, but I want to set up a SIP trunk and use it just to know how to do it before I do it in production. 
    I did end up deleting the phones from CUCM so they can register with the 2851 CME that I'm setting up as a SIP trunk. So it is registering there, and I set the allow connections and bind sip commands.
    I am now getting Debugs and calls from the SIP Trunk router going to CUCM, but the error message is No Codec, and I Get the fast busy after the call rings on the CUCM Main Site side. So looks like the negotiation is failing. Here is my CLI for the SIP Trunk now after the changes have been made and phones registered to the SIP Branch site as well as the Debug when I tried to place a call to extension "5000":
    Note: I did try to change the codecs in the dial-peers to g729r8 instead of 711 and same fast busy after answering.
    ==============================================
    Branch_SIP#show run
    Building configuration...
    Current configuration : 3529 bytes
    ! Last configuration change at 03:15:11 UTC Thu Apr 2 2015
    version 15.1
    service timestamps debug datetime msec
    service timestamps log datetime msec
    no service password-encryption
    hostname Branch_SIP
    boot-start-marker
    boot-end-marker
    ! card type command needed for slot/vwic-slot 0/2
    enable secret 5 $1$hOXF$gvfmWW1ZIQE0mAMVg.u1c/
    no aaa new-model
    dot11 syslog
    ip source-route
    ip cef
    ip dhcp excluded-address 10.0.10.1 10.0.10.10
    ip dhcp excluded-address 10.0.30.1 10.0.30.10
    ip dhcp pool Data
     network 10.0.10.0 255.255.255.0
     default-router 10.0.10.254
     option 150 ip 192.168.5.250
     dns-server 192.168.5.200
    ip dhcp pool Voice
     network 10.0.30.0 255.255.255.0
     default-router 10.0.30.254
     dns-server 192.168.5.200
     option 150 ip 172.16.0.1
    ip dhcp pool data
     option 150 ip 172.16.0.2
    no ipv6 cef
    multilink bundle-name authenticated
    voice service voip
     allow-connections sip to sip
     sip
      bind media source-interface Loopback1
    voice-card 0
    crypto pki token default removal timeout 0
    license udi pid CISCO2851 sn FTX1031A2FM
    redundancy
    interface Loopback1
     ip address 2.2.2.2 255.255.255.255
    interface GigabitEthernet0/0
     no ip address
     duplex auto
     speed auto
    interface GigabitEthernet0/0.10
     encapsulation dot1Q 10
     ip address 10.0.10.254 255.255.255.0
    interface GigabitEthernet0/0.30
     encapsulation dot1Q 30
     ip address 10.0.30.254 255.255.255.0
    interface GigabitEthernet0/1
     no ip address
     shutdown
     duplex auto
     speed auto
    interface Serial0/3/0
     no ip address
     shutdown
     clock rate 2000000
    interface Serial0/3/1
     ip address 172.16.0.1 255.255.255.0
     clock rate 250000
    interface Internal-Service-Module0/0
     no ip address
     shutdown
     !Application: CUE Running on AIM2
     hold-queue 512 out
    router eigrp 1
     network 0.0.0.0
     network 2.2.2.2 0.0.0.0
     network 10.0.0.0
     network 10.0.10.0 0.0.0.255
     network 10.0.30.0 0.0.0.255
     network 172.16.0.0
    ip forward-protocol nd
    no ip http server
    no ip http secure-server
    ip route 0.0.0.0 0.0.0.0 172.16.0.2
    tftp-server flash:term45.default.loads
    tftp-server flash:jar45sccp.8-5-3TH1-6.sbn
    tftp-server flash:cnu45.8-5-3TH1-6.sbn
    tftp-server flash:apps45.8-5-3TH1-6.sbn
    tftp-server flash:dsp45.8-5-3TH1-6.sbn
    tftp-server flash:cvm45sccp.8-5-3TH1-6.sbn
    control-plane
    voice-port 0/0/0
    voice-port 0/0/1
    mgcp profile default
    dial-peer voice 1 voip
     description **Incoming Call from SIP Trunk**
     session protocol sipv2
     session target sip-server
     codec g711ulaw
    dial-peer voice 2 voip
     description **Outgoing Call to SIP Trunk**
     destination-pattern 5...
     session protocol sipv2
     session target sip-server
     codec g711ulaw
    sip-ua
     sip-server ipv4:192.168.5.250
    telephony-service
     codec g711ulaw
     max-ephones 24
     max-dn 48
     ip source-address 172.16.0.1 port 2000
     system message SIP Branch Site
     cnf-file location flash:
     load 7960-7940 P00308010200.bin
     max-conferences 8 gain -6
     transfer-system full-consult
    ephone-dn  1
     number 4008
    ephone-dn  2
     number 4005
    ephone  1
     device-security-mode none
     mac-address 001D.A21A.2065
     button  1:1
    line con 0
     exec-timeout 0 0
    line aux 0
    line 194
     no activation-character
     no exec
     transport preferred none
     transport input all
     transport output lat pad telnet rlogin lapb-ta mop udptn v120 ssh
     stopbits 1
     speed 115200
    line vty 0 4
     password cisco
     login
     transport input all
    line vty 5 15
     password cisco
     login
     transport input all
    scheduler allocate 20000 1000
    end
    Branch_SIP#show debug
    TFTP:
      TFTP Event debugging is on
    CCSIP SPI: SIP Call Statistics tracing is enabled       (filter is OFF)
    Branch_SIP#
    *Apr  2 03:20:32.351: //25/0C496935804A/SIP/Call/sipSPICallInfo:
    The Call Setup Information is:
    Call Control Block (CCB) : 0x4B6C5C28
    State of The Call        : STATE_DEAD
    TCP Sockets Used         : NO
    Calling Number           : 4008
    Called Number            : 5005
    Source IP Address (Sig  ): 172.16.0.1
    Destn SIP Req Addr:Port  : 192.168.5.250:5060
    Destn SIP Resp Addr:Port : 192.168.5.250:5060
    Destination Name         : 192.168.5.250
    *Apr  2 03:20:32.351: //25/0C496935804A/SIP/Call/sipSPIMediaCallInfo:
    Number of Media Streams: 1
    Media Stream             : 1
    Negotiated Codec         : No Codec
    Negotiated Codec Bytes   : 0
    Nego. Codec payload      : 255 (tx), 255 (rx)
    Negotiated Dtmf-relay    : 0
    Dtmf-relay Payload       : 0 (tx), 0 (rx)
    Source IP Address (Media): 2.2.2.2
    Source IP Port    (Media): 19472
    Destn  IP Address (Media):  -
    Destn  IP Port    (Media): 0
    Orig Destn IP Address:Port (Media): [ - ]:0
    *Apr  2 03:20:32.351: //25/0C496935804A/SIP/Call/sipSPICallInfo:
    Disconnect Cause (CC)    : 63
    Disconnect Cause (SIP)   : 503
    Branch_SIP#

  • ITSP SIP Trunk audio issue

    Hi Guys,
    call flow:
    external caller > service provider SIP Trunk >CUBE VG>CUCM>User ip phone.
    no firewall between
    we are not facing this audio issue for all the calls but also for few calls , i can say 3 out of 10 calls.
    under VG bind media and control command recently added by TAC guys instruction but no use.
    recently we changed our office but no changes for device or configuration
    also attached debug log for the issue call.
    ONE THING I NOTICE 2 HOUR TIME DIFFERENCE IN VOICE GATEWAY than  actual time.
    Voice gateway show run: ---------
    aaa session-id common
    memory-size iomem 10
    clock timezone CET 1 0
    clock summer-time CEST recurring last Sun Mar 2:00 last Sun Oct 2:00
    network-clock-participate wic 0
    dot11 syslog
    ip source-route
    ip traffic-export profile tac mode capture
    ip traffic-export profile sniffer mode capture
      bidirectional
    ip traffic-export profile Test mode capture
      bidirectional
    ip cef
    no ip dhcp use vrf connected
    ip dhcp excluded-address 172.18.122.1 172.18.122.50
    ip dhcp pool PHONES
       network 172.18.122.0 255.255.255.0
       domain-name ldhenergy.net
       option 150 ip 172.18.122.10
       default-router 172.18.122.8
    no ip domain lookup
    ip domain name ldhenergy.com
    ip host ld-lsn-cm-01 172.18.122.10
    no ipv6 cef
    multilink bundle-name authenticated
    isdn switch-type primary-net5
    voice call send-alert
    voice call convert-discpi-to-prog
    voice call carrier capacity active
    voice rtp send-recv
    voice service voip
    ip address trusted list
      ipv4 172.18.122.11 255.255.255.255
    dtmf-interworking rtp-nte
    allow-connections h323 to h323
    allow-connections h323 to sip
    allow-connections sip to h323
    allow-connections sip to sip
    redirect ip2ip
    h323
    sip
      bind control source-interface FastEthernet0/0
      bind media source-interface FastEthernet0/0
    voice class codec 1
    codec preference 1 g711alaw
    codec preference 2 g711ulaw
    voice translation-rule 20
    rule 1 /044578\(....\)$/ /\1/ type any unknown plan any unknown
    voice translation-rule 30
    rule 1 /021343\(....\)$/ /\1/ type any unknown plan any unknown
    voice translation-rule 40
    rule 1 /^\(.*\)/ /0\1/
    voice translation-profile SIPIN
    translate called 30
    voice-card 0
    dspfarm
    dsp services dspfarm
    crypto pki token default removal timeout 0
    controller E1 0/0/0
    interface FastEthernet0/0
    ip address 172.18.122.3 255.255.255.0
    ip helper-address 193.73.102.255
    duplex auto
    speed auto
    interface FastEthernet0/1
    ip address 10.128.18.9 255.255.255.0
    duplex auto
    speed auto
    interface Integrated-Service-Engine1/0
    ip unnumbered FastEthernet0/0
    service-module ip address 172.18.122.11 255.255.255.0
    !Application: CUE Running on NME
    service-module ip default-gateway 172.18.122.8
    no keepalive
    router ospf 1005
    network 172.18.122.0 0.0.0.255 area 0.0.0.1
    ip forward-protocol nd
    no ip http server
    no ip http secure-server
    ip route 0.0.0.0 0.0.0.0 172.18.122.8
    ip route 10.20.0.0 255.255.0.0 172.18.122.8
    ip route 172.18.122.11 255.255.255.255 Integrated-Service-Engine1/0
    ip tacacs source-interface FastEthernet0/0
    control-plane
    ccm-manager fallback-mgcp
    ccm-manager mgcp
    no ccm-manager fax protocol cisco
    ccm-manager music-on-hold
    ccm-manager config server 172.18.122.10 
    ccm-manager config
    mgcp call-agent 172.18.122.10 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 profile default
    sccp local FastEthernet0/0
    sccp ccm 172.18.122.10 identifier 1 version 5.0.1
    sccp
    sccp ccm group 1
    associate ccm 1 priority 1
    associate profile 10 register HW-MTP
    associate profile 20 register TRANSCODE
    dspfarm profile 20 transcode 
    codec g711ulaw
    codec g711alaw
    codec g729ar8
    codec g729abr8
    maximum sessions 4
    associate application SCCP
    dspfarm profile 10 mtp 
    codec g711alaw
    maximum sessions hardware 24
    associate application SCCP
    dial-peer voice 343 voip
    translation-profile incoming SIPIN
    session protocol sipv2
    incoming called-number .
    voice-class sip bind control source-interface FastEthernet0/0
    voice-class sip bind media source-interface FastEthernet0/0
    dtmf-relay rtp-nte sip-notify sip-kpml cisco-rtp h245-signal h245-alphanumeric
    codec g711alaw
    no vad
    dial-peer voice 344 voip
    destination-pattern 0T
    session protocol sipv2
    session target ipv4:62.2.46.4
    voice-class sip bind control source-interface FastEthernet0/0
    voice-class sip bind media source-interface FastEthernet0/0
    dtmf-relay rtp-nte sip-notify sip-kpml cisco-rtp h245-signal h245-alphanumeric
    codec g711alaw
    no vad
    dial-peer voice 1600 voip
    destination-pattern 16..
    session protocol sipv2
    session target ipv4:172.18.122.10
    voice-class sip bind control source-interface FastEthernet0/0
    voice-class sip bind media source-interface FastEthernet0/0
    dtmf-relay sip-notify
    codec g711alaw
    no vad
    dial-peer voice 1616 voip
    destination-pattern 1616
    session protocol sipv2
    session target ipv4:172.18.122.10
    dtmf-relay rtp-nte sip-notify sip-kpml cisco-rtp h245-signal h245-alphanumeric
    codec g711alaw
    no vad
    dial-peer voice 1699 voip
    destination-pattern 1699
    session protocol sipv2
    session target ipv4:172.18.122.10
    voice-class sip bind control source-interface FastEthernet0/0
    voice-class sip bind media source-interface FastEthernet0/0
    dtmf-relay sip-notify
    codec g711alaw
    no vad
    sip-ua
    call-manager-fallback
    max-conferences 4 gain -6
    transfer-system full-consult
    ip source-address 172.18.122.3 port 2000
    max-ephones 42
    max-dn 144
    Regards
    Vigeesh

    I suggest  do a network capture or enable debug ccsip mesages.
    look for conneciion ip address inside sdp field and check that are recheacble.
    regards

  • No ringing back tone from PSTN (SIP trunk) via CUBE

    Hello,
    I have an issue about ringing back tone when I call from outside --> PSTN (SIP trunk) --> CUBE --> UCCX --> redirect call to extension. I hear IVR and can do DTMF. then press extension, no ringing back tone.
    However when I call from PSTN (SIP trunk) --> CUBE --> DID (direct to IP Phone). I heard ringing back tone.
    Call from inside to outside, I heard ringing back tone.
    I connect cucm to cube by create H.323 gateway.
    cucm 10.x
    uccx 10.x
    cube (cisco 2901) Version 15.2(4)M5
    Please help
    Thank you

    Can you try changing theg Service Parameter"Send H225 User Info Msg" parameter and set it to "Use ANN for ringback" and see if it helps pls?
    Also make sure you have Annunciators registered and available in the MRGL assigned to H.323 Gateway.
    It is clusterwide parameter and hence applies to all node in the cluster.

  • Why do we need MTP in the SIP trunk for CVP warm transfers

    Hi All,
    Why do we need to enable MTP in SIP trunk between CUCM and CVP for CVP based trasnfers???
    Thanks in advance!!
    Regards,
    Thammaya Gupta K.

    I saw also in the CDR logs that the IP Phone media transport going to CUBE is in G711.And as well in the wireshark capture of the IP communicator that the CUCM invoke to use the g711 codec but as per ITSP logs they are now in the g729.
    @ Jamie If I un-tick the MTP point required in SIP trunk will make the call leg from IP Phone to CUBE g729 (w/o hw resource), I have also tried to use g729 preferred originating codec, but still the IP Phone is using g711.
    I have seen a documentation states:
    " To configure G.729 codecs for use with a SIP trunk, you must use a hardware MTP or transcoder that supports the G.729 codec." - I read this on the CUCM help page under configuring SIP trunk setting.
    Our ultimate goal is to use g729 without using HW MTP/ transcoder.
    IP Phone ->CUCM SIP Trunk ->CUBE-> ITSP

  • Keep alive a SIP Trunk?

    Hi!
    I'm trying to register a SIP Trunk to a SIP server. The trunk registration is done, but not keep alive. The trunk register with SIP server when an outgoing call starts, but when this call ends, the SIP trunk closes the connection with SIP server. Then, the
    outgoing calls work OK, but the incoming calls doesn't work because the SIP Trunk is unregistered while no active outgoing calls.
    Then, can i keep alive the SIP Trunk registration with SIP Server?
    Thanks!!!

    You need to talk to the SIP provider and get them enable OPTIONS on the SIP Trunk and enable OPTIONS on the PSTN Gateway. Check the registration interval of the SIP trunk on the Gateway and try increasing it to a higher value.
    http://thamaraw.com

  • 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

Maybe you are looking for

  • Error occurred while installing the updates (iTunes)

    Error occurred while installing the updates (iTunes) Applies To: Windows | Windows XP | Programs I am running Windows XP and trying to update to iTunes version 10.5 (from version 10.3.1.55) and I keep getting the following error message, 'Error occur

  • Setting color of a specific line?

    okay so in swing is it possible to set a portion of the text in a text field to a different color? Like in my chat program(below) can I make it so all incoming chat(instring) is blue and all outgoing text(string) is red? Am I using the wrong text com

  • Video/audio out to HDTV-what leads do I need?

    Hi, I've got the iMac most recently released (just before the Intel one just announced). Basically, I've got some HD content on here which I can play with VLC media player, and I own a HDTV. I want to mirror what's on my iMacs screen on the HDTV, and

  • Incorrect Formatting when converting to .PDF

    Hi All, I am experiencing an issue when converting a file to pdf when using pegasus opera program. This has been working fine for ages until the last week or so. My colleague is also using the same setup and it is working fine for her. When convertin

  • Trouble with VRF traffic on ASR9000

    Hi We have a trouble to established  neighbord and pass tables from Cisco 7600 to ASR9000 when configure a vrf between both devices. the configuration on cisco 7600 is: router bgp 2006 bgp router-id 172.16.110.97 bgp log-neighbor-changes bgp graceful