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.

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  • 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
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    Contact: <sip:[email protected]:5060>
    Supported: sdp-anat
    Server: Cisco-SIPGateway/IOS-15.2.4.M5
    Content-Type: application/sdp
    Content-Disposition: session;handling=required
    Content-Length: 289
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 7965 2747 IN IP4 10.18.81.2
    s=SIP Call
    c=IN IP4 10.18.81.2
    t=0 0
    m=audio 22350 RTP/AVP 18 101 19
    c=IN IP4 10.18.81.2
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=yes
    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.

  • 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

  • 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 ...

  • 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?

  • 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

  • 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.

  • Issue with instant ringback when using sip trunk to SP

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

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

  • Incoming Calls via CUBE to IVR- intermittent garbled audio

        Connecting via an ITSP and incoming calls are:
    ITSP--SIP Trunk-- MyCUBE--SIP Trunk--CUCM--IVR(CTI Ports and RPs)
                                                                            |
                                                                     7900 Type A sccp Handsets  
    INbound calls to sccp phones direct sound OK.
    Outbound calls sound OK
    Inbound calls to IVR intermittent voice quality issues (outside caller hears garbled message)
    QoS reservations are adequate and no issues with that (after hours with v low traffic and closed call centrestill have this issue)
    CUBE is 3825 and tried 15.1M6, 7 and 2.
    All devices and trunk in same region= HQ (tried SIP trunk in Hub_none with same results). intra and inter Region has 64k per call
    CUBE has XCode, Soft and Hard MTPs, Conf resources registered with CUCM 8.6
    When incoming call comes in via ISDN to CUBE, no issues.
    Using G711alaw and G729 r8 in CUBE dial peers. DTMF works fine.No VAD.
    Also, when some calls present to agent via IVR, sometimes the agent with 7900 hears silence whilst external caller hears the agent fine. Inbound calls direct to agent DID work 100%. I have noticed on the dead air calls to agent, the ptime = 0 on the handset for Rx... CUBE inbound/oubound Dial peers are matching g711a so I am not sure what is issue.

    The problem was solved with the following commands:
    network-clock-participate wic 0
    network-clock-select 1 E1 0/0/0
    modem country v12 belgium
    The connections are stable now.
    Best regards
    Thomas

  • Unity Connection SIP trunk integration vs alerting name

    Hello,
    I have been implementing a Unity Connecion 8.6 with a CUCM cluster 8.6 via a SIP trunk.
    The previous implemention of the Unity server was set up using voicemail ports;
    When users used to call the voicemail pilot number, we were able to configure an alerting name on the vm ports, saying (to VoiceMail) for example.
    I am looking for a solution for the new implementation via the SIP trunk. I just want users to see 'to VoiceMail' on their phone when they call the Unity Connection system.
    Thanks for the help!
    Best regards,
    Antoine

    I have found the solution:
    If you want to dispaly a name such as Voicemail, etc you can change that in Unity Connection under Port Group --> Advanced Settings --> "Remote-Party-ID"
    cheers!

  • Modify calling number in SIP invite on CUCM 10.5

    Hello,
    I am working at a customer with CUCM 10.5 who uses MGCP gateways to access the PSTN via T1 PRI ISDN.
    They use four digit DNs internally and need to prefix these with 713657 to make the outbound CLID work ok - i.e. a call to the PSTN from extension 1000 needs to send 7136571000 to the ISDN provider.
    I configured this using Calling Party Transformations and this works fine e.g.
    A Calling Party Transformation for 1XXX would prefix 713657.
    The problem I have is that the customer has a NICE active recording system which communicates with the CUCM cluster using a SIP trunk.
    The invites that CUCM sends via the SIP trunk show the full ten digits rather than the four digit extension which will not work according to company deploying the recording system.
    If I remove the Calling Party Transformation then the SIP invite shows four digits and the call recording works but the outbound CLID does not work.
    Can anyone suggest a way to fix this? The customer does not want to change the gateway protocol from MGCP to H323 which would be my favoured choice. Any change of calling party setting on CUCM (e.g. ticking the use external mask for calling party on route pattern) affects the SIP invite.
    Ideally I need a way to modify the number in the SIP invite but I cannot find any example of how to do this.
    Any suggestions are welcome.
    Thanks

    Hi, thanks for your response.
    The Calling Party Transformation CSS is not applied to the SIP trunk but is applied to the T1 port of the MGCP gateway.
    The Transformations are still applied to the SIP Invite messages via the trunk so I guess this is a quirk of the calling recording profile setup on CUCM.
    I did try creating another Calling Party Transformation setup which stripped the unwanted digits and applied it to the SIP trunk but it had no effect.

  • 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

  • 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

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