Troubleshooting no ringback voice gateway

Hi all,
I'd like to ask about voice gateway that installed on my customer site. below is the topology:
/* Style Definitions */
table.MsoNormalTable
{mso-style-name:"Table Normal";
mso-tstyle-rowband-size:0;
mso-tstyle-colband-size:0;
mso-style-noshow:yes;
mso-style-priority:99;
mso-style-parent:"";
mso-padding-alt:0in 5.4pt 0in 5.4pt;
mso-para-margin:0in;
mso-para-margin-bottom:.0001pt;
mso-pagination:widow-orphan;
font-size:10.0pt;
font-family:"Times New Roman","serif";}
PSTN – Voice Gateway – C2960-48PST-L – Callmanager and IP Phone
                                                           |
                                                           |(FO)
Analog phone – PBX.A – Voice Gateway –––– Voice Gateway – PBX.B – Analog phone
the problem is that there's no ring back tone when calling to another site analog phone from ip phone/analog phone. The PBX (NEC) are using E1 connection to the router. Does the ringback provided by PBX or the Voice gateway?
Thanks in advance
Dias

Yes, we did find a solution. It ended up it was on the carrier's end. I believe that they were generating ring back inbound to calls we originated, but not outbound to calls originated from the carriers. They switched that so that they did not send us ring back, but did send carriers ring back for inbound calls and that appeared to solve the issue.
Here's my carrier's explanation in case that wasn't confusing enough: "We made in-band tone available to the SIP trunk via an egress profile change on the SONUS Gateway. Early-media is determined / initiated by
the called or far-end switch."

Similar Messages

  • Troubleshooting SS7 Interconnect for Voice Gateway

    Hi
    Recently we have implemeted SS7 Interconnect for Voice Gateway Solution in our network.
    We are using devices PGW2200, NAS 5400, SLT 2611, 4000 Series S/w and GK.
    Customers compalint that they are facing Echo, Clip in Voice and No voice related issues. Can you tell us the commands on PGW/GW/SLT to pinpoint. PGW we are using in Call Control Mode.
    Please help.
    Thanks in Advance.

    Hi,
    You can check in the /opt/CiscoMGC/etc directory properties.dat file which has the parameter *.EchoCanRequired which should be set to 1. This can binded by an individual trunk-group as well. Also make sure the voice-ports have the echo-cancel coverage and disable VAD to see if it makes it better.
    Regards
    -Kashif

  • VG202 Voice gateway

    I have multiple environments with Avaya Partner phone systems in them.  I used touse a multitech VOIP modem for connection back to Headquarters - this works perfectly fine.   Situation I am having, when the Multitech units die, I replace them with the Cisco VG202 unit.  When placed at the remote locations, the time in the windows of the Avaya phones never corrects it self...it always wrong.  It is correct on the Avaya system at the location, but the integration of the Cisco unit seems to make it go awry.  All other functionality is there, but the time display is a mess.  Any ideas as to where to troubleshoot this?   Manually establish the correct time from a console phone at the remote locations, and it goes awry within hours.                    

    Hi Peter,
    This forum is for discussions about specific known bugs. You might find better discussion in the IP Telephony forum: https://supportforums.cisco.com/community/netpro/collaboration-voice-video/ip-telephony
    Conceptually as an analog voice gateway the VG shouldn't generally provide any time information to an attached phone system. Is it possible the phone system is pointed to the VG as an NTP reference? In that situation the internal platform clock isn't terribly reliable and synch to an more reliable external NTP reference is recommended.  The only other possibility that comes to mind is delivery of a timestamp information along with callerid over the analog voice ports. In that situation the router again depends on internal clock. Configuring an NTP reference on the router will help to keep the router clock better synchronized.  These are all a bit of a stretch. The most likely, IMHO, is the pone system pointing to VG202 as NTP reference. In that case giving the VG202 a reliable NTP reference so the router can keep more accurate time would be my first approach.
    Hope this helps,
    Wes

  • Cisco 2911 Voice Gateway SIP PSTN Calls Fail

    Hello All,
        I am having trouble with outboud SIP PSTN calls through a Cisco 2911 Voice Gateway.  2911 VG terminates PSTN SIP Traffic and connects to Avaya CS1000M via QSIG PRI Trunks. When calls are attempted outbound fron the PBX the caller gets a fast busy.  Debug ISDN q931 shows the call hitting the 2911 properly, debug voip ccapi inout shows the call matching the correct dial peers and debug ccsip shows the invite to the PSTN Provider SBC, however within the invite the "from" address incorrectly shows the calling number with the provider SBC address (see below).  does anyone have any insight on how to correct this?  Attached are VG config and Debug isdn q931, voip ccapi inout, ccsip messages and ccsip call.  Thanks in advance for any help!!
    From: <sip:[email protected]>:tag=6166CDC4-882
    To: <sip:[email protected]>
    Shawn C. Smith

    i have same problem my cucm ip is 192.168.200.53
    my Voice Gateway is SIP by ip 192.168.200.86 for internal
    and 172.29.7.94
    and my SIP Server is 10.208.9.69
    if its oky can yuo take a look at my problem please
    this is the syslog from debug
    May 30 20:19:34.284: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    INVITE sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 192.168.200.53:5060;branch=z9hG4bK3bd451bf17e0
    From: "Aysar Mohamed" <sip:[email protected]>;tag=37693~244641b0-36ac-434c-91c1-823f25a68b28-18299026
    To: <sip:[email protected]>
    Date: Fri, 30 May 2014 20:19:34 GMT
    Call-ID: [email protected]
    Supported: timer,resource-priority,replaces
    Min-SE:  1800
    User-Agent: Cisco-CUCM8.6
    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
    Supported: Geolocation
    Call-Info: <sip:192.168.200.53:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
    Cisco-Guid: 3047462016-0000065536-0000004549-0902342848
    Session-Expires:  1800
    P-Asserted-Identity: "Aysar Mohamed" <sip:[email protected]>
    Remote-Party-ID: "Aysar Mohamed" <sip:[email protected]>;party=calling;screen=yes;privacy=off
    Contact: <sip:[email protected]:5060>
    Max-Forwards: 70
    Content-Length: 0
    May 30 20:19:34.284: //-1/B5A494800000/CCAPI/cc_api_display_ie_subfields:
       cc_api_call_setup_ind_common:
       cisco-username=2217156
       ----- ccCallInfo IE subfields -----
       cisco-ani=2217156
       cisco-anitype=0
       cisco-aniplan=0
       cisco-anipi=0
       cisco-anisi=1
       dest=90555769123
       cisco-desttype=0
       cisco-destplan=0
       cisco-rdie=FFFFFFFF
       cisco-rdn=
       cisco-rdntype=0
       cisco-rdnplan=0
       cisco-rdnpi=-1
       cisco-rdnsi=-1
       cisco-redirectreason=-1   fwd_final_type =0
       final_redirectNumber =
       hunt_group_timeout =0
    May 30 20:19:34.288: //-1/B5A494800000/CCAPI/cc_api_call_setup_ind_common:
       Interface=0x30CF41D4, Call Info(
       Calling Number=2217156,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
       Called Number=90555769123(TON=Unknown, NPI=Unknown),
       Calling Translated=FALSE, Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE,
       Incoming Dial-peer=0, Progress Indication=NULL(0), Calling IE Present=TRUE,
       Source Trkgrp Route Label=, Target Trkgrp Route Label=, CLID Transparent=FALSE), Call Id=465
    May 30 20:19:34.288: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
    May 30 20:19:34.288: :cc_get_feature_vsa malloc success
    May 30 20:19:34.288: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
    May 30 20:19:34.288:  cc_get_feature_vsa count is 1
    May 30 20:19:34.288: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
    May 30 20:19:34.288: :FEATURE_VSA attributes are: feature_name:0,feature_time:832953048,feature_id:85
    May 30 20:19:34.288: //465/B5A494800000/CCAPI/cc_api_call_setup_ind_common:
       Set Up Event Sent;
       Call Info(Calling Number=2217156(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
       Called Number=90555769123(TON=Unknown, NPI=Unknown))
    May 30 20:19:34.288: //465/B5A494800000/CCAPI/cc_process_call_setup_ind:
       Event=0x2B82D890
    May 30 20:19:34.288: //-1/xxxxxxxxxxxx/CCAPI/cc_setupind_match_search:
       Try with the demoted called number 90555769123
    May 30 20:19:34.288: //465/B5A494800000/CCAPI/ccCallSetContext:
       Context=0x2ABC2E44
    May 30 20:19:34.288: //465/B5A494800000/CCAPI/cc_process_call_setup_ind:
       >>>>CCAPI handed cid 465 with tag 0 to app "_ManagedAppProcess_Default"
    May 30 20:19:34.288: //465/B5A494800000/CCAPI/ccCallProceeding:
       Progress Indication=NULL(0)
    May 30 20:19:34.288: //465/B5A494800000/CCAPI/ccCallSetupRequest:
       Destination=, Calling IE Present=TRUE, Mode=0,
       Outgoing Dial-peer=802, Params=0x2ABC19D4, Progress Indication=NULL(0)
    May 30 20:19:34.288: //465/B5A494800000/CCAPI/ccCheckClipClir:
       In: Calling Number=2217156(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed)
    May 30 20:19:34.288: //465/B5A494800000/CCAPI/ccCheckClipClir:
       Out: Calling Number=2217156(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed)
    May 30 20:19:34.288: //465/B5A494800000/CCAPI/ccCallSetupRequest:
       Destination Pattern=9T, Called Number=0555769123, Digit Strip=FALSE
    May 30 20:19:34.288: //465/B5A494800000/CCAPI/ccCallSetupRequest:
       Calling Number=2217156(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
       Called Number=0555769123(TON=Unknown, NPI=Unknown),
       Redirect Number=, Display Info=Aysar Mohamed
       Account Number=2217156, Final Destination Flag=TRUE,
       Guid=B5A49480-0001-0000-0000-11C535C8A8C0, Outgoing Dial-peer=802
    May 30 20:19:34.288: //465/B5A494800000/CCAPI/cc_api_display_ie_subfields:
       ccCallSetupRequest:
       cisco-username=2217156
       ----- ccCallInfo IE subfields -----
       cisco-ani=2217156
       cisco-anitype=0
       cisco-aniplan=0
       cisco-anipi=0
       cisco-anisi=1
       dest=0555769123
       cisco-desttype=0
       cisco-destplan=0
       cisco-rdie=FFFFFFFF
       cisco-rdn=
       cisco-rdntype=0
       cisco-rdnplan=0
       cisco-rdnpi=-1
       cisco-rdnsi=-1
       cisco-redirectreason=-1   fwd_final_type =0
       final_redirectNumber =
       hunt_group_timeout =0
    May 30 20:19:34.288: //465/B5A494800000/CCAPI/ccIFCallSetupRequestPrivate:
       Interface=0x30CF41D4, Interface Type=3, Destination=, Mode=0x0,
       Call Params(Calling Number=2217156,(Calling Name=Aysar Mohamed)(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
       Called Number=0555769123(TON=Unknown, NPI=Unknown), Calling Translated=FALSE,
       Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE, Outgoing Dial-peer=802, Call Count On=FALSE,
       Source Trkgrp Route Label=, Target Trkgrp Route Label=, tg_label_flag=0, Application Call Id=)
    May 30 20:19:34.288: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
    May 30 20:19:34.288: :cc_get_feature_vsa malloc success
    May 30 20:19:34.288: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
    May 30 20:19:34.288:  cc_get_feature_vsa count is 2
    May 30 20:19:34.288: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
    May 30 20:19:34.288: :FEATURE_VSA attributes are: feature_name:0,feature_time:832952824,feature_id:86
    May 30 20:19:34.292: //466/B5A494800000/CCAPI/ccIFCallSetupRequestPrivate:
       SPI Call Setup Request Is Success; Interface Type=3, FlowMode=1
    May 30 20:19:34.292: //466/B5A494800000/CCAPI/ccCallSetContext:
       Context=0x2ABC1984
    May 30 20:19:34.292: //465/B5A494800000/CCAPI/ccSaveDialpeerTag:
       Outgoing Dial-peer=802
    May 30 20:19:34.292: //466/B5A494800000/CCAPI/cc_api_call_proceeding:
       Interface=0x30CF41D4, Progress Indication=NULL(0)
    May 30 20:19:34.292: //465/B5A494800000/SIP/Msg/ccsipDisplayMsg:
    Sent:
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 192.168.200.53:5060;branch=z9hG4bK3bd451bf17e0
    From: "Aysar Mohamed" <sip:[email protected]>;tag=37693~244641b0-36ac-434c-91c1-823f25a68b28-18299026
    To: <sip:[email protected]>
    Date: Fri, 30 May 2014 20:19:34 GMT
    Call-ID: [email protected]
    CSeq: 101 INVITE
    Allow-Events: telephone-event
    Server: Cisco-SIPGateway/IOS-12.x
    Content-Length: 0
    May 30 20:19:34.292: //466/B5A494800000/SIP/Msg/ccsipDisplayMsg:
    Sent:
    INVITE sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 172.29.7.94:5060;branch=z9hG4bK461C
    Remote-Party-ID: "Aysar Mohamed" <sip:[email protected]>;party=calling;screen=yes;privacy=off
    From: "Aysar Mohamed" <sip:[email protected]>;tag=7394E4-1898
    To: <sip:[email protected]>
    Date: Fri, 30 May 2014 20:19:34 GMT
    Call-ID: [email protected]
    Supported: timer,resource-priority,replaces,sdp-anat
    Min-SE:  1800
    Cisco-Guid: 3047462016-0000065536-0000004549-0902342848
    User-Agent: Cisco-SIPGateway/IOS-12.x
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    CSeq: 101 INVITE
    Timestamp: 1401481174
    Contact: <sip:[email protected]:5060>
    Call-Info: <sip:172.29.7.94:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
    Expires: 180
    Allow-Events: kpml, telephone-event
    Max-Forwards: 69
    Session-Expires:  1800
    Content-Length: 0
    May 30 20:19:34.300: //466/B5A494800000/SIP/Msg/ccsipDisplayMsg:
    Received:
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 172.29.7.94:5060;branch=z9hG4bK461C
    Call-ID: [email protected]
    From: "Aysar Mohamed"<sip:[email protected]>;tag=7394E4-1898
    To: <sip:[email protected]>
    CSeq: 101 INVITE
    Content-Length: 0
    May 30 20:19:34.612: //466/B5A494800000/SIP/Msg/ccsipDisplayMsg:
    Received:
    SIP/2.0 183 Session Progress
    Via: SIP/2.0/UDP 172.29.7.94:5060;branch=z9hG4bK461C
    Record-Route: <sip:10.208.9.69:5060;transport=udp;lr>
    Call-ID: [email protected]
    From: "Aysar Mohamed"<sip:[email protected]>;tag=7394E4-1898
    To: <sip:[email protected]>;tag=sbc0806eppk5yip-CC-57
    CSeq: 101 INVITE
    Contact: <sip:[email protected]:5060;user=phone>
    Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER
    Content-Length: 328
    Content-Type: application/sdp
    v=0
    o=- 17192647 17192647 IN IP4 10.208.9.69
    s=SBC call
    c=IN IP4 10.208.9.69
    t=0 0
    m=audio 39910 RTP/AVP 8 0 102 102 18 116
    a=rtpmap:8 PCMA/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:102 AMR/8000
    a=rtpmap:102 AMR/8000
    a=rtpmap:18 G729/8000
    a=rtpmap:116 telephone-event/8000
    a=ptime:5
    a=fmtp:116 0-15
    a=fmtp:18 annexb=yes
    May 30 20:19:34.612: %SIP-3-UNSUPPORTED: Unsupported ptime value
    May 30 20:19:34.612: //466/B5A494800000/CCAPI/cc_api_caps_ind:
       Destination Interface=0x0, Destination Call Id=-1, Source Call Id=466,
       Caps(Codec=0x2, Fax Rate=0x2, Vad=0x1,
       Modem=0x0, Codec Bytes=160, Signal Type=2)
    May 30 20:19:34.612: //466/B5A494800000/CCAPI/cc_api_caps_ind:
       Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
       Playout Max=1000(ms), Fax Nom=300(ms))
    May 30 20:19:34.612: //465/B5A494800000/CCAPI/cc_api_caps_ack:
       Destination Interface=0x0, Destination Call Id=-1, Source Call Id=465,
       Caps(Codec=g729r8(0x4), Fax Rate=FAX_RATE_VOICE(0x2), Vad=ON(0x2),
       Modem=OFF(0x0), Codec Bytes=20, Signal Type=2, Seq Num Start=3882)
    May 30 20:19:34.612: //465/B5A494800000/CCAPI/cc_api_caps_ack:
       Destination Interface=0x0, Destination Call Id=-1, Source Call Id=465,
       Caps(Codec=g729r8(0x4), Fax Rate=FAX_RATE_VOICE(0x2), Vad=ON(0x2),
       Modem=OFF(0x0), Codec Bytes=20, Signal Type=2, Seq Num Start=3882)
    May 30 20:19:34.612: //466/B5A494800000/CCAPI/cc_api_event_indication:
       Event=170, Call Id=466
    May 30 20:19:34.612: //466/B5A494800000/CCAPI/cc_api_event_indication:
       Event Is Sent To Conferenced SPI(s) Directly
    May 30 20:19:34.612: //466/B5A494800000/CCAPI/cc_api_event_indication:
       Event=98, Call Id=466
    May 30 20:19:34.612: //466/B5A494800000/CCAPI/cc_api_event_indication:
       Event Is Sent To Conferenced SPI(s) Directly
    May 30 20:19:34.612: //466/B5A494800000/CCAPI/cc_api_call_cut_progress:
       Interface=0x30CF41D4, Progress Indication=INBAND(8), Signal Indication=SIGNAL RINGBACK(1),
       Cause Value=0
    May 30 20:19:34.612: //466/B5A494800000/CCAPI/cc_api_call_cut_progress:
       Call Entry(Responsed=TRUE)
    May 30 20:19:34.612: //465/B5A494800000/CCAPI/ccCallCutProgress:
       Progress Indication=INBAND(8), Signal Indication=SIGNAL RINGBACK(1), Cause Value=0
       Voice Call Send Alert=FALSE, Call Entry(Alert Sent=FALSE)
    May 30 20:19:34.612: //465/B5A494800000/CCAPI/ccCallCutProgress:
       Call Entry(Responsed=TRUE)
    May 30 20:19:34.612: //465/B5A494800000/CCAPI/ccConferenceCreate:
       (confID=0x30C11410, callID1=0x1D1, gcid=8C9E3127-E76E11E3-8274BE8C-EC3B12A0, tag=0x0)
    May 30 20:19:34.616: //466/B5A494800000/CCAPI/ccConferenceCreate:
       (confID=0x30C11410, callID2=0x1D2, gcid=8C9E3127-E76E11E3-8274BE8C-EC3B12A0, tag=0x0)
    May 30 20:19:34.616: //465/B5A494800000/CCAPI/ccConferenceCreate:
       Conference Id=0x30C11410, Call Id1=465, Call Id2=466, Tag=0x0
    May 30 20:19:34.616: //465/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
    May 30 20:19:34.616: cc_api_get_xcode_stream : 4702
    May 30 20:19:34.616: //466/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
    May 30 20:19:34.616: cc_api_get_xcode_stream : 4702
    May 30 20:19:34.616: //465/B5A494800000/CCAPI/ccConferenceCreate:
    May 30 20:19:34.616: ccConferenceCreate: ret1=0, codecMask1=2, bytes1=160, negot1=0, dtmf1=0
                        ret2=0, codecMask2=2, bytes2=160, negot2=1, dtmf2=6,
                        tx_dynamic_pt1=0, rx_dynamic_pt1=0, codec_mode1=0, params_bitmap1 =0
                        tx_dynamic_pt2=8, rx_dynamic_pt2=8, codec_mode2=0, params_bitmap2 =0
    May 30 20:19:34.616: //465/B5A494800000/CCAPI/ccConferenceCreate:
       delay media to slow start case, codec negotation is not done
    May 30 20:19:34.616: //465/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
    May 30 20:19:34.616: cc_api_get_xcode_stream : 4702
    May 30 20:19:34.616: //465/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
    May 30 20:19:34.616: cc_api_get_xcode_stream : 4702
    May 30 20:19:34.616: //465/B5A494800000/CCAPI/cc_api_bridge_done:
       Conference Id=0x16, Source Interface=0x30CF41D4, Source Call Id=465,
       Destination Call Id=466, Disposition=0x0, Tag=0x0
    May 30 20:19:34.616: //466/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
    May 30 20:19:34.616: cc_api_get_xcode_stream : 4702
    May 30 20:19:34.616: //466/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
    May 30 20:19:34.616: cc_api_get_xcode_stream : 4702
    May 30 20:19:34.616: //466/B5A494800000/CCAPI/cc_api_bridge_done:
       Conference Id=0x16, Source Interface=0x30CF41D4, Source Call Id=466,
       Destination Call Id=465, Disposition=0x0, Tag=0x0
    May 30 20:19:34.616: //465/B5A494800000/CCAPI/cc_generic_bridge_done:
       Conference Id=0x16, Source Interface=0x30CF41D4, Source Call Id=466,
       Destination Call Id=465, Disposition=0x0, Tag=0x0
    May 30 20:19:34.616: //465/B5A494800000/CCAPI/ccConferenceCreate:
       Call Entry(Conference Id=0x16, Destination Call Id=466)
    May 30 20:19:34.616: //466/B5A494800000/CCAPI/ccConferenceCreate:
       Call Entry(Conference Id=0x16, Destination Call Id=465)
    May 30 20:19:34.616: //465/B5A494800000/CCAPI/cc_process_notify_bridge_done:
       Conference Id=0x16, Call Id1=465, Call Id2=466
    May 30 20:19:34.616: //465/B5A494800000/SIP/Msg/ccsipDisplayMsg:
    Sent:
    SIP/2.0 183 Session Progress
    Via: SIP/2.0/UDP 192.168.200.53:5060;branch=z9hG4bK3bd451bf17e0
    From: "Aysar Mohamed" <sip:[email protected]>;tag=37693~244641b0-36ac-434c-91c1-823f25a68b28-18299026
    To: <sip:[email protected]>;tag=739628-1BDB
    Date: Fri, 30 May 2014 20:19:34 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
    Remote-Party-ID: <sip:[email protected]>;party=called;screen=yes;privacy=off
    Contact: <sip:[email protected]:5060>
    Supported: sdp-anat
    Server: Cisco-SIPGateway/IOS-12.x
    Content-Type: application/sdp
    Content-Disposition: session;handling=required
    Content-Length: 233
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 2639 5276 IN IP4 192.168.200.86
    s=SIP Call
    c=IN IP4 192.168.200.86
    t=0 0
    m=audio 18288 RTP/AVP 8 0 19
    c=IN IP4 192.168.200.86
    a=rtpmap:8 PCMA/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:19 CN/8000
    May 30 20:19:34.680: //466/B5A494800000/SIP/Msg/ccsipDisplayMsg:
    Received:
    SIP/2.0 500 Server Internal Error
    Via: SIP/2.0/UDP 172.29.7.94:5060;branch=z9hG4bK461C
    Record-Route: <sip:10.208.9.69:5060;transport=udp;lr>
    Call-ID: [email protected]
    From: "Aysar Mohamed"<sip:[email protected]>;tag=7394E4-1898
    To: <sip:[email protected]>;tag=sbc0806eppk5yip-CC-57
    CSeq: 101 INVITE
    Reason: Q.850;cause=127;text="interworking unspecified"
    Warning: 399 - "SoftX3000 R601-CCU Rel POS:[3103] Release from CR"
    Content-Length: 0
    May 30 20:19:34.680: //466/B5A494800000/CCAPI/cc_api_call_disconnected:
       Cause Value=41, Interface=0x30CF41D4, Call Id=466
    May 30 20:19:34.680: //466/B5A494800000/CCAPI/cc_api_call_disconnected:
       Call Entry(Responsed=TRUE, Cause Value=41, Retry Count=0)
    May 30 20:19:34.680: //465/B5A494800000/CCAPI/ccCallReleaseResources:
       release reserved xcoding resource.
    May 30 20:19:34.680: //466/B5A494800000/CCAPI/ccCallSetAAA_Accounting:
       Accounting=0, Call Id=466
    May 30 20:19:34.680: //465/B5A494800000/CCAPI/ccConferenceDestroy:
       Conference Id=0x16, Tag=0x0
    May 30 20:19:34.680: //465/B5A494800000/CCAPI/cc_api_bridge_drop_done:
       Conference Id=0x16, Source Interface=0x30CF41D4, Source Call Id=465,
       Destination Call Id=466, Disposition=0x0, Tag=0x0
    May 30 20:19:34.680: //466/B5A494800000/CCAPI/cc_api_bridge_drop_done:
       Conference Id=0x16, Source Interface=0x30CF41D4, Source Call Id=466,
       Destination Call Id=465, Disposition=0x0, Tag=0x0
    May 30 20:19:34.680: //465/B5A494800000/CCAPI/cc_generic_bridge_done:
       Conference Id=0x16, Source Interface=0x30CF41D4, Source Call Id=466,
       Destination Call Id=465, Disposition=0x0, Tag=0x0
    May 30 20:19:34.680: //466/B5A494800000/SIP/Msg/ccsipDisplayMsg:
    Sent:
    ACK sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 172.29.7.94:5060;branch=z9hG4bK461C
    From: "Aysar Mohamed" <sip:[email protected]>;tag=7394E4-1898
    To: <sip:[email protected]>;tag=sbc0806eppk5yip-CC-57
    Date: Fri, 30 May 2014 20:19:34 GMT
    Call-ID: [email protected]
    Max-Forwards: 70
    CSeq: 101 ACK
    Allow-Events: kpml, telephone-event
    Content-Length: 0
    May 30 20:19:34.684: //466/B5A494800000/CCAPI/ccCallDisconnect:
       Cause Value=41, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=41)
    May 30 20:19:34.684: //466/B5A494800000/CCAPI/ccCallDisconnect:
       Cause Value=41, Call Entry(Responsed=TRUE, Cause Value=41)
    May 30 20:19:34.684: //466/B5A494800000/CCAPI/cc_api_call_disconnect_done:
       Disposition=0, Interface=0x30CF41D4, Tag=0x0, Call Id=466,
       Call Entry(Disconnect Cause=41, Voice Class Cause Code=0, Retry Count=0)
    May 30 20:19:34.684: //466/B5A494800000/CCAPI/cc_api_call_disconnect_done:
       Call Disconnect Event Sent
    May 30 20:19:34.684: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
    May 30 20:19:34.684: :cc_free_feature_vsa freeing 31A5D9F0
    May 30 20:19:34.684: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
    May 30 20:19:34.684:  vsacount in free is 1
    May 30 20:19:34.684: //465/B5A494800000/CCAPI/ccCallDisconnect:
       Cause Value=41, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=0)
    May 30 20:19:34.684: //465/B5A494800000/CCAPI/ccCallDisconnect:
       Cause Value=41, Call Entry(Responsed=TRUE, Cause Value=41)
    May 30 20:19:34.684: //465/B5A494800000/SIP/Msg/ccsipDisplayMsg:
    Sent:
    SIP/2.0 503 Service Unavailable
    Via: SIP/2.0/UDP 192.168.200.53:5060;branch=z9hG4bK3bd451bf17e0
    From: "Aysar Mohamed" <sip:[email protected]>;tag=37693~244641b0-36ac-434c-91c1-823f25a68b28-18299026
    To: <sip:[email protected]>;tag=739628-1BDB
    Date: Fri, 30 May 2014 20:19:34 GMT
    Call-ID: [email protected]
    CSeq: 101 INVITE
    Allow-Events: telephone-event
    Server: Cisco-SIPGateway/IOS-12.x
    Reason: Q.850;cause=41
    Content-Length: 0
    May 30 20:19:34.684: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    ACK sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 192.168.200.53:5060;branch=z9hG4bK3bd451bf17e0
    From: "Aysar Mohamed" <sip:[email protected]>;tag=37693~244641b0-36ac-434c-91c1-823f25a68b28-18299026
    To: <sip:[email protected]>;tag=739628-1BDB
    Date: Fri, 30 May 2014 20:19:34 GMT
    Call-ID: [email protected]
    Max-Forwards: 70
    CSeq: 101 ACK
    Allow-Events: presence, kpml
    Content-Length: 0
    May 30 20:19:34.688: //465/B5A494800000/CCAPI/cc_api_call_disconnect_done:
       Disposition=0, Interface=0x30CF41D4, Tag=0x0, Call Id=465,
       Call Entry(Disconnect Cause=41, Voice Class Cause Code=0, Retry Count=0)
    May 30 20:19:34.688: //465/B5A494800000/CCAPI/cc_api_call_disconnect_done:
       Call Disconnect Event Sent
    May 30 20:19:34.688: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
    May 30 20:19:34.688: :cc_free_feature_vsa freeing 31A5DAD0
    May 30 20:19:34.688: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
    May 30 20:19:34.688:  vsacount in free is 0
    May 30 20:19:36.044: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    OPTIONS sip:172.29.7.94:5060 SIP/2.0
    Via: SIP/2.0/UDP 10.208.9.69:5060;branch=z9hG4bKmisco3ykfiooegpygsphkocp1T20326
    Call-ID: isbcfemyk1p1mkteets1tcmi53eeehfhikcp@SoftX3000
    From: <sip:172.29.7.94:5060>;tag=sbc0803k1pyk51o
    To: <sip:172.29.7.94>
    CSeq: 1 OPTIONS
    Max-Forwards: 70
    Content-Length: 0
    May 30 20:19:36.048: //467/8DAABF6C8278/SIP/Msg/ccsipDisplayMsg:
    Sent:
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 10.208.9.69:5060;branch=z9hG4bKmisco3ykfiooegpygsphkocp1T20326
    From: <sip:172.29.7.94:5060>;tag=sbc0803k1pyk51o
    To: <sip:172.29.7.94>;tag=739BBC-1CE2
    Date: Fri, 30 May 2014 20:19:36 GMT
    Call-ID: isbcfemyk1p1mkteets1tcmi53eeehfhikcp@SoftX3000
    Server: Cisco-SIPGateway/IOS-12.x
    CSeq: 1 OPTIONS
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    Allow-Events: telephone-event
    Accept: application/sdp
    Supported: 100rel,timer,resource-priority,replaces,sdp-anat
    Content-Type: application/sdp
    Content-Length: 446
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 3496 1601 IN IP4 172.29.7.94
    s=SIP Call
    c=IN IP4 172.29.7.94
    t=0 0
    m=audio 0 RTP/AVP 18 0 8 9 4 2 15
    c=IN IP4 172.29.7.94
    m=image 0 udptl t38
    c=IN IP4 172.29.7.94
    a=T38FaxVersion:0
    a=T38MaxBitRate:9600
    a=T38FaxFillBitRemoval:0
    a=T38FaxTranscodingMMR:0
    a=T38FaxTranscodingJBIG:0
    a=T38FaxRateManagement:transferredTCF
    a=T38FaxMaxBuffer:200
    a=T38FaxMaxDatagram:320
    a=T38FaxUdpEC:t38UDPRedundancy
    My SIP GW internal ip address is 192.168.200.86
    and the Public IP is : 172.29.7.94
    My CUCM is 192.168.200.53
    my GW Config is :
    voice service voip
     allow-connections h323 to h323
     allow-connections h323 to sip
     allow-connections sip to h323
     allow-connections sip to sip
     fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
     sip
      registrar server
    voice class codec 1
     codec preference 1 g711alaw
     codec preference 2 g711ulaw
     codec preference 3 g729r8
     codec preference 4 g729br8
    voice translation-rule 3
     rule 1 /^9\(\)/ /\1/
    voice translation-rule 4
     rule 4 /^22217/ /7/
     rule 5 /^2217/ /7/
     rule 6 /^022217/ /7/
     rule 7 /^0122217/ /7/
    voice translation-rule 5
     rule 1 /^5/ /905/
     rule 2 /^1/ /901/
     rule 3 /^2/ /902/
     rule 4 /^3/ /903/
     rule 5 /^4/ /904/
     rule 6 /^6/ /906/
     rule 7 /^7/ /907/
     rule 8 /^8/ /908/
     rule 10 /^00/ /900/
     rule 11 /'+'/ /900/
    voice translation-profile OUT
     translate called 3
    voice translation-profile REDIAL
     translate calling 5
    voice translation-profile SIP-NEW
     translate called 4
    application
     service mva http://192.168.200.53:8080/ccmivr/pages/IVRMainpage.vxml
     service ccm http://192.168.200.53:8080/ccmivr/pages/IVRMainpage.vxml
    license udi pid CISCO2921/K9 sn FCZ164960G0
    hw-module pvdm 0/0
    hw-module pvdm 0/1
    interface Embedded-Service-Engine0/0
     no ip address
     shutdown
    interface GigabitEthernet0/0
     description $ETH-LAN$$ETH-SW-LAUNCH$$INTF-INFO-GE 0/0$
     ip address 192.168.200.86 255.255.255.0
     duplex auto
     speed auto
    interface GigabitEthernet0/1
     ip address 172.29.7.94 255.255.255.252
     duplex auto
     speed auto
    ip http server
    ip http access-class 23
    ip http authentication local
    no ip http secure-server
    ip http timeout-policy idle 60 life 86400 requests 10000
    ip route 0.0.0.0 0.0.0.0 192.168.200.1
    ip route 10.208.9.0 255.255.255.0 172.29.7.93
    access-list 23 permit 10.10.10.0 0.0.0.7
    control-plane
    mgcp profile default
    sccp local GigabitEthernet0/0
    sccp ccm 192.168.200.53 identifier 1 priority 1 version 7.0
    sccp
    sccp ccm group 1
     associate ccm 1 priority 1
     associate profile 2 register NAGHI-MTP
    dspfarm profile 2 mtp
     codec g711alaw
     maximum sessions hardware 25
     associate application SCCP
    dial-peer voice 802 voip
     description ** SIP TO STC **
     translation-profile outgoing OUT
     destination-pattern 9T
     session protocol sipv2
     session target ipv4:10.208.9.69:5060
     session transport udp
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay sip-notify rtp-nte sip-kpml
     no vad
    dial-peer voice 811 voip
     description ** SIP INCOMING FROM STC **
     translation-profile incoming SIP-NEW
     translation-profile outgoing REDIAL
     destination-pattern 7...
     session protocol sipv2
     session target ipv4:192.168.200.53
     incoming called-number 022217...$
     dtmf-relay sip-notify rtp-nte sip-kpml
     codec g711alaw
    dial-peer voice 812 voip
     description ** SIP INCOMING FROM STC **
     translation-profile incoming SIP-NEW
     translation-profile outgoing REDIAL
     destination-pattern 7...
     session protocol sipv2
     session target ipv4:192.168.200.53
     incoming called-number 22217...$
     dtmf-relay sip-notify rtp-nte sip-kpml
     codec g711alaw
    dial-peer voice 813 voip
     description ** SIP INCOMING FROM STC **
     translation-profile incoming SIP-NEW
     translation-profile outgoing REDIAL
     destination-pattern 7...
     session protocol sipv2
     session target ipv4:192.168.200.53
     incoming called-number 2217...$
     dtmf-relay sip-notify rtp-nte sip-kpml
     codec g711alaw
    dial-peer voice 814 voip
     description ** SIP INCOMING FROM STC **
     translation-profile incoming SIP-NEW
     translation-profile outgoing REDIAL
     preference 1
     destination-pattern 7...
     session protocol sipv2
     session target ipv4:192.168.200.63
     incoming called-number 022217...$
     dtmf-relay sip-notify rtp-nte sip-kpml
     codec g711alaw
    dial-peer voice 815 voip
     description ** SIP INCOMING FROM STC **
     translation-profile incoming SIP-NEW
     translation-profile outgoing REDIAL
     preference 1
     destination-pattern 7...
     session protocol sipv2
     session target ipv4:192.168.200.63
     incoming called-number 22217...$
     dtmf-relay sip-notify rtp-nte sip-kpml
     codec g711alaw
    dial-peer voice 816 voip
     description ** SIP INCOMING FROM STC **
     translation-profile incoming SIP-NEW
     translation-profile outgoing REDIAL
     preference 1
     destination-pattern 7...
     session protocol sipv2
     session target ipv4:192.168.200.63
     incoming called-number 2217...$
     dtmf-relay sip-notify rtp-nte sip-kpml
     codec g711alaw
    dial-peer voice 817 voip
     description ** SIP INCOMING FROM STC **
     translation-profile incoming SIP-NEW
     translation-profile outgoing REDIAL
     destination-pattern 7...
     session protocol sipv2
     session target ipv4:192.168.200.53
     incoming called-number 0122217...$
     dtmf-relay sip-notify rtp-nte sip-kpml
     codec g711alaw
    dial-peer voice 818 voip
     description ** SIP INCOMING FROM STC **
     translation-profile incoming SIP-NEW
     translation-profile outgoing REDIAL
     preference 1
     destination-pattern 7...
     session protocol sipv2
     session target ipv4:192.168.200.63
     incoming called-number 0122217...$
     dtmf-relay sip-notify rtp-nte sip-kpml
     codec g711alaw
    Please i need ur help ASAP

  • Where can I learn the structure of Voice gateway ?

    I'm making a essay about Structure of Voice gateway: hardware and software construction. I can not find any books or any manual deal with it . So,can someone give me more information about structure of voice gateway or give me some useful resources please?
    I'm waiting for your replying.

    Cisco doesn't make public the software structure of their products; to a certain extent it can be inferred from documentation and familiarity with the product, but probably not enough for an in-deep analysis.
    For your task, I suggest you focus on some open-source development of GW, that has no secrets.
    Hope this helps, please rate post if it does!

  • How to create a Global Contacts in our CM or Voice Gateway

    Hi
    we have a UCM6.1.2 and a H.323 voice gateway
    we get many calls from different vendors and so on
    i want to some how assign a contact or a name to the calls that come in often
    for example
    if we get 100 calls from 973-333-3333 i want that number to show up and also the customer's name for example Customer XYZ
    can i do this any where?
    is it possible?
    any help will be appreciated.
    Thanks
    Regards

    You can configure Caller ID which is an analog service offered by a central office, which supplies calling party information to subscribers.
    Caller ID Name Delivery Issues on Cisco IOS Gateways:
    http://www.cisco.com/en/US/tech/tk652/tk653/technologies_configuration_example09186a00800a9a49.shtml
    Caller ID:
    http://www.cisco.com/en/US/docs/ios/12_3/vvf_c/cisco_ios_voice_configuration_library_glossary/vclclid.html

  • Cucm 10.1version - training videos and hand guides on understanding voice gateway h323 and SIP? thanks

    cucm 10.1version - any free training videos and hand guides on understanding voice gateway h323 and SIP and how to configure one?  thanks

    Learncisco gives a very good introduction to CUCM - I recommend you start there.

  • Setup of 2 B-channel transfer (TBCT) on Cisco Voice Gateway

    Hi all,
    In my scenario, I have several remote locations, each with their own T1-PRI connection to the PSTN via Cisco voice gateways.
    The call flow has the caller calling a PSTN number at the remote location.  This call hits a CUCM translation pattern and is immediately hairpinned/tromboned out to a PSTN number located at the UCCE central location.
    The customer would like to perform a 2 b-channel transfer back up to the CO switch, to free up B-channels to the remote location.
    My question is this:  Assuming my switch type is supported and the PSTN provider supports 2 b-channel transfers, do I simply need to add "isdn supp-service tbct" to my serial interface configuration?
    I have seen discussions that it is as simple as this, to needing some tcl/xml code, to also requiring aaa to be setup.
    As there are over 100 locations being considered for this enhancement, the customer would like to know if it is 15 minutes or 2 hours per location to implement.
    Thanks,
    Joe

    Thanks Paolo,
    That was my expectation.
    They have this going on the VG's utilized by CVP, so I will ask them to review the XML in use there for tweaking for the remote locations.
    Thanks again!
    Joe

  • **ERROR**: CCPMSG_InCall: ChannelID IE invalid 100(0x64) in SIP voice gateway

    Hi all,
    I am configuring a SIP voice gateway. After finsihing the config, the outbound calls are working properly. But I met some issue for inbound calls.
    There is no debug log for show ccsip message and show voice dialpeer command. Only q931 has the debug log.
    Can any one have any idea for this issue?
    below is the debug shows for show isdn q931.
       Jun  1 05:27:49.435: ISDN Se0/0/0:15 Q931: RX <- SETUP pd = 8  callref = 0x03A3
    Sending Complete
    Bearer Capability i = 0x8090A3
    Standard = CCITT
    Transfer Capability = Speech
    Transfer Mode = Circuit
    Transfer Rate = 64 kbit/s
    Channel ID i = 0xE1858384
    Preferred, Interface 5, Channel 4
    Calling Party Number i = 0x2181, '18600586101'
    Plan:ISDN, Type:National
    Called Party Number i = 0xC1, '82197910'
    Plan:ISDN, Type:Subscriber(local)
    High Layer Compat i = 0x9181
    High Layer Compat i = 0x9181
    Jun  1 05:27:49.439: ISDN Se0/0/0:15 Q931: Received SETUP  callref = 0x83A3 callID = 0x0004 switch = primary-net5 interface = User
    d8w-sr-2811f20a-vpn#
    Jun  1 05:27:49.439: ISDN Se0/0/0:15 **ERROR**: CCPMSG_InCall: ChannelID IE invalid 100(0x64) rejecting call
    Jun  1 05:27:49.443: ISDN Se0/0/0:15 Q931: TX -> RELEASE_COMP pd = 8  callref = 0x83A3
    Cause i = 0x80E418 - Invalid information element contents
    下午01:28:17: Zhuliang: Jun  1 05:27:49.439: ISDN Se0/0/0:15 **ERROR**: CCPMSG_InCall: ChannelID IE invalid 100(0x64) rejecting call
    Jun  1 05:27:49.443: ISDN Se0/0/0:15 Q931: TX -> RELEASE_COMP pd = 8  callref = 0x83A3
    Cause i = 0x80E418 - Invalid information element contents  

    Channel ID i = 0xE1858384 Preferred, Interface 5, Channel 4 Jun  1 05:27:49.443: ISDN Se0/0/0:15 Q931: TX -> RELEASE_COMP pd = 8  callref = 0x83A3 Cause i = 0x80E418 - Invalid information element contents下午01:28:17: Zhuliang: Jun  1 05:27:49.439: ISDN Se0/0/0:15 **ERROR**: CCPMSG_InCall: ChannelID IE invalid 100(0x64) rejecting callJun  1 05:27:49.443: ISDN Se0/0/0:15 Q931: TX -> RELEASE_COMP pd = 8  callref = 0x83A3 Cause i = 0x80E418 - Invalid information element contents  
    To what are you connecting? Call comes with explicit interface indicator IE, that is not normal for ISDN E1.

  • ISR as CUBE and Voice Gateway

    Can I set an ISR 2951 as CUBE to receive SIP trunks and configure the same box as voice gateway to deliver TDM E1 voice channels to an enterprise PBx?
    Thanks
    Sent from Cisco Technical Support iPhone App

    Yes You can...
    Please rate all useful posts
    "The essence of christianity is not the enthronement but the obliteration of self --William Barclay"

  • Configure E1 card on voice gateway

    Hi
    I have CUCM 9 and Cisco 2801 voice gateway
    There is 2 card on cisco 2801 : VIC2-4FXO and VWIC3-1MFT-T1/E1
    I would like to know how I will configure the gateway in the CUCM (H323, MGCP ...)
    Should I add H323 or MGCP gateway or something else ?
    I also need to know how I will configure E1 card on the Cisco 2801 to handle CUCM PSTN incoming and outgoing call
    I will apreciate if you help with some how to document
    Thanks in advance
    Regards

    Which protocol to use, is up to you, you should know the requirements of your customer to define that.

  • How to Save Global Contacts in our CM or Voice Gateway

    Hi
    we have a UCM6.1.2 and a H.323 voice gateway
    we get many calls from different vendors and so on
    i want to some how assign a contact or a name to the calls that come in often
    for example
    if we get 100 calls from 973-333-3333 i want that number to show up and also the customer's name for example Customer XYZ
    can i do this any where?
    is it possible?
    any help will be appreciated.
    Thanks
    Regards

    You can configure Caller ID which is an analog service offered by a central office, which supplies calling party information to subscribers.
    Caller ID Name Delivery Issues on Cisco IOS Gateways:
    http://www.cisco.com/en/US/tech/tk652/tk653/technologies_configuration_example09186a00800a9a49.shtml
    Caller ID:
    http://www.cisco.com/en/US/docs/ios/12_3/vvf_c/cisco_ios_voice_configuration_library_glossary/vclclid.html

  • How to install permenant UC License on 2951 voice gateway

    Hi All,
    I have C2951 Voice gateway and I want to install Permenant UC license and I don't know how to get it and how to install on it and how to activate it,so I would like to request all of you that if you have any idea please explain me in details with step by step, i will be appreciate all of you and hop I will receive response form you very soon.
    Thanks
    Rizwan

    Hi Rizwan,
    a. u need to send the mail to [email protected] along with PAK code and serial number of 2951 router.this u can get by running command show version or show license udi
    b. after u recive the license file[uck9_license.lic], upload this file in router flash using command copy tftp flash:
    c. u can check the same by running command show flash:
    d. now u need to install the license file . run the command license install flash:uck9_license.lic
    e. now run the command license boot module c2900 technology-package uck9 and accept :yes
    f. reboot the router
    g. after the router reboots, run command show license feature /sh license all.
    regds,
    aman

  • How to verify voice channels on voice gateway

    Hello,
    I am in the testing phase of deploying a 2921 voice gateway with 2 PRIs. After connecting the T1s, how do I verify if all the channels are up and ready for use? Also, how do I verify if clocking on the T1 interfaces are configured correctly?
    This 2921 is a replacement for a 2811 currently being used. When I login to the 2811 via SSH, I see this:
    560240: Aug 13 20:37:24.368: %ISDN-6-CONNECT: Interface Serial0/2/0:3 is now connected to 3854143418 N/A
    560241: Aug 13 20:37:24.368: %ISDN-6-CONNECT: Interface Serial0/2/0:3 is now connected to 3854143418 N/A
    560242: Aug 13 20:37:24.544: %ISDN-6-CONNECT: Interface Serial0/2/0:5 is now connected to 7753428028 N/A
    560243: Aug 13 20:37:24.544: %ISDN-6-CONNECT: Interface Serial0/2/0:5 is now connected to 7753428028 N/A
    560244: Aug 13 20:37:27.128: %ISDN-6-CONNECT: Interface Serial0/2/0:4 is now connected to 7753296300 N/A
    560245: Aug 13 20:37:27.128: %ISDN-6-CONNECT: Interface Serial0/2/0:4 is now connected to 7753296300 N/A
    560246: Aug 13 20:37:27.128: %ISDN-6-CONNECT: Interface Serial0/2/0:22 is now connected to 3296300 N/A
    560247: Aug 13 20:37:27.128: %ISDN-6-CONNECT: Interface Serial0/2/0:22 is now connected to 3296300 N/A
    560248: Aug 13 20:37:27.652: %ISDN-6-CONNECT: Interface Serial0/2/0:20 is now connected to 3525101 N/A
    560249: Aug 13 20:37:27.652: %ISDN-6-CONNECT: Interface Serial0/2/0:20 is now connected to 3525101 N/A
    560250: Aug 13 20:37:30.320: %ISDN-6-CONNECT: Interface Serial0/2/1:22 is now connected to 3525101 N/A
    560251: Aug 13 20:37:30.320: %ISDN-6-CONNECT: Interface Serial0/2/1:22 is now connected to 3525101 N/A
    How do I do that with the 2921 in an SSH session? I've enabled "logging monitor" and "term mon" but I still can't see messages like that.
    I thank you very much for your time.

    Hello,
    To check clocking you can use the "show network-clock" command. For example:
    SiteA-RTR#show network-clock
    Network Clock Configuration
    Priority Clock Source Clock State Clock Type
    1 T1 0/0/0 GOOD T1
    10 Backplane GOOD PLL
    Current Primary Clock Source
    Priority Clock Source Clock State Clock Type
    1 T1 0/0/0 GOOD T1
    The above is an example of a "good" configuration, assuming that T1 0/0/0 is the controller terminating the line from your carrier. In your case, it would be T1 0/2/0 or 0/2/1. You can also look at "show controller t1 [interface]" (e.g. show controller t1 0/2/0) and pay particular attention to "Line Code Violations", "Slip Secs", and "Errored Secs" counters. Note that the "Clock Source Line" status in the "show controller t1" command is not enough to verify you have proper clocking. 
    Example of a "bad" clock scenario:
    SiteA-RTR(config-voiceport)#do sh controller t1 0/0/0
    T1 0/0/0 is up.
    Applique type is Channelized T1
    Cablelength is long 0db
    No alarms detected.
    alarm-trigger is not set
    Soaking time: 3, Clearance time: 10
    AIS State:Clear LOS State:Clear LOF State:Clear
    Version info Firmware: 20080918, FPGA: 13, spm_count = 0
    Framing is ESF, Line Code is B8ZS, Clock Source is Line.
    CRC Threshold is 320. Reported from firmware is 320.
    Data in current interval (483 seconds elapsed):
    3 Line Code Violations, 4 Path Code Violations
    207 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins
    206 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 109 Unavail Secs
    To make sure the clock is set correctly, look at the "network-clock-participate" and the "network-clock-select" commands.
    For the ISDN status information, you can use "show isdn status". For example:
    SiteA-RTR(config)#do sh isdn stat
    Global ISDN Switchtype = primary-ni
    %Q.931 is backhauled to CCM MANAGER 0x0003 on DSL 0. Layer 3 output may not apply
    ISDN Serial0/0/0:23 interface
    dsl 0, interface ISDN Switchtype = primary-ni
    L2 Protocol = Q.921 0x0000 L3 Protocol(s) = CCM MANAGER 0x0003
    Layer 1 Status:
    ACTIVE
    Layer 2 Status:
    TEI = 0, Ces = 1, SAPI = 0, State = MULTIPLE_FRAME_ESTABLISHED
    Layer 3 Status:
    0 Active Layer 3 Call(s)
    Active dsl 0 CCBs = 0
    The Free Channel Mask: 0x800000FF
    Number of L2 Discards = 0, L2 Session ID = 15
    *Note: The above is output from a gateway provisioned to use MGCP.
    For all gateways, just make sure that Layer 1 is "ACTIVE" and that Layer 2 has "MULTIPLE_FRAME_ESTABLISHED".
    You can also use the "show isdn service" command to look at status for individual B-channels.
    SiteA-Rtr#sh isdn service
    PRI Channel Statistics:
    %Q.931 is backhauled to CCM MANAGER 0x0003 on DSL 0. Layer 3 output may not appl
    ISDN Se0/0/0:23, Channel [1-24]
    Configured Isdn Interface (dsl) 0
    Channel State (0=Idle 1=Proposed 2=Busy 3=Reserved 4=Restart 5=Maint_Pend)
    Channel : 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4
    State : 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 3
    Service State (0=Inservice 1=Maint 2=Outofservice 8=MaintPend 9=OOSPend)
    Channel : 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4
    State : 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 2
    Use the "Channel State" and "Service State" to verify things are what you expect.
    HTH.
    -Bill (@ucguerrilla)
    http://ucguerrilla.com

  • How to make CUCM as a TFTP server , then copy files to Voice Gateway ?

    how to make CUCM as a TFTP server , then copy files to Voice Gateway ? anyone knows?

    Hi,
    Please check the following link
    http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/6_1_1/ccmsys/cucm-accm-611/a02tftp.html#wp1023004
    Understanding How Devices Access the TFTP Server
    You can enable the IP phones and gateways to discover the TFTP server IP address in one or more of the following ways, depending on the device type:
    •Gateways and phones can use DHCP custom option 150.
    Cisco recommends this method. With this method, you configure the TFTP server IP address as the option value.
    •Gateways and phones can use DHCP option 066.
    You may configure either the host name or IP address of the TFTP server as the option value.
    •Gateways and phones can query CiscoCM1.
    Ensure the Domain Name System (DNS) can resolve this name to the IP address of the TFTP server. Cisco does not recommend this option because it does not scale.
    •You can configure phones with the IP address of the TFTP server. If DHCP is enabled on the phone, you can still configure an alternate TFTP server IP address locally on the phone that will override the TFTP address that was obtained through DHCP.
    •Gateways and phones also accept the DHCP Optional Server Name (sname) parameter.
    •The phone or gateway can use the value of Next-Server in the boot processes (siaddr).
    Devices save the TFTP server address in nonvolatile memory. If one of the preceding methods was available at least once, but is not currently available, the device uses the address that is saved in memory.
    You can configure the TFTP service on the first node or a subsequent node, but usually you should configure it on the first node. For small systems, the TFTP server can coexist with a Cisco Unified Communications Manager on the same server.
    HTH
    Manish

Maybe you are looking for

  • Dump for Shopper On Behalf of Field

    Hi Srm Gurus, My requirement was to restrict the search help for Shopper on Behalf in SRM. I Implemented the BADI 'BBP_F4_READ_ON_ENTRY' for method 'GET_BOBUSER'. In testing i got the expected result in the Search Help. When i try to select the entry

  • Can I redeem gift card purchase at Australia on China Apple store?

    i was wondering if I could use the gift card I buy from Australia by redeem it on China Apple Store ?

  • Acrobat looses windows focus after showing a wxWidgets modal dialog

    Here's a code sample:      AVWindow dlgWindow = AVWindowNewFromPlatformThing(AVWLmodal, NULL, NULL, gExtensionID, dlg->GetHWND());     AVAppBeginModal(dlgWindow);     dlg->ShowModal();                                   // dlg is a wxDialog*     AVWin

  • Third Party Tool-Open Hub Destination

    dear all,           can any one guide me as to how data(master data in particular)  from bw can be transferred to oracle data base using open hub destiantion.is rfc connection between bw and third part tool enough,do we have to maintain the same stru

  • Remove data of specific sales organization

    Hi guys, Anyone know how can I remove/delete all data specific to a sales organization/company? Thanks