Voice gateway Config issue

Hello everyone;
I do have an analog line that i want to affect to (04) four IP Phones, so that any one of these four ip phones users can use this same analog line, and when we call that analog line number, any one of these four ip phone users can answer.
Is this possible to do on the Voice gateway?
I think about configuring many connection plar under one voice port, like this: 
    voice port 1/0/0
    connection plar opx 1000
    connection plar opx 1001
    connection plar opx 1002
    connection plar opx 1003
And about, puting many answer adress under dial-peer voice this way:
     Dial-peer voice
     answer adress 1000
     answer adress 1001
     answer adress 1002
     answer adress 1003
but i don't know if it will work , and don't see how it could be done for the voice translation rule and the voice translation profile.
Ragards.
Camélia

This will be done in only specific Dial-peer and Voice port.
A voice Translation rule is not need because once the voice-port number is dialed for example 011-27272, the plar command forward it 2001 and the dial-peer 10 is matched, you only need to configure a hunt-pilot in CUCM.
Voice-Port 1/0/0
connection plar opx 2001
2.) You will need to match incoming calls from PSTN, you can do this by use the below commands to match all incoming calls from PSTN.
Dial-peer voice 1 Pots
Incoming called-number .
3.) Finally you will need to send calls to 2001 via CUCM to match the Hunt-Pilot.
Dial-peer voice 10 voip
destination-pattern 2001
session-target ipv4:<<CUCM-IP-Address>>
Please Rate.

Similar Messages

  • **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.

  • 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

  • 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 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

  • Changing CAS e&m-wink-start to a PRI on voice gateway

    Cisco IOS Software, C3900 Software (C3900-UNIVERSALK9-M), Version 15.1(4)M4, RELEASE SOFTWARE (fc1)
    I want to change a CAS  e&m-wink-start to a PRI on controller T1 0/1/0 using mgcp below. I am posting the new configuration below the current e&m-wink-start configuration on the voice gateway router.
    Current
    controller T1 0/1/0
    cablelength long 0db
    ds0-group 1 timeslots 1-24 type e&m-wink-start
    description
    New configuration
    Router(config)# no contoller T1 0/1/0
    Router(config)# no interface Serial0/1/0
    Router(config)#controller t1 0/1/0
    Router(config-controller)#cablelength long 0db
    Router(config-controller)#framing esf
    Router(config-controller)#linecode b8zs
    Router(config-controller)#clock source line 
    Router(config-controller)#pri-group timeslots 1-24 service mgcp
    Router(config-controller)#description circuit ID
    Router(config-if)# interface serial0/1/0
    Router(config-if)# no ip address
    Router(config-if)# encapsulation hdlc
    Router(config-if)# isdn switch-type primary-4ess
    Router(config-if)# isdn incoming-voice voice
    Router(config-if)# isdn bind-l3 ccm-manager

    Hi,
    You need additional configs for this to work..You also need to configure the isdn bind on the D-channel not as you have it..
    Router(config-if)# interface serial0/1/0:23
    Router(config-if)# no ip address
    Router(config-if)# encapsulation hdlc
    Router(config-if)# isdn switch-type primary-4ess
    Router(config-if)# isdn incoming-voice voice
    Router(config-if)# isdn bind-l3 ccm-manager
    Additional configs....change ip to suit your needs
    ccm-manager redundant-host 192.168.103.114
    ccm-manager mgcp
    ccm-manager music-on-hold bind xxx--put relevant interface
    mgcp
    mgcp call-agent 192.168.103.115 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 modem passthrough voip codec g711alaw
    mgcp modem passthrough voip redundancy
    mgcp ip qos dscp af31 media
    mgcp ip qos dscp cs3 signaling
    mgcp package-capability rtp-package
    mgcp package-capability sst-package
    mgcp package-capability pre-package
    mgcp default-package dtmf-package
    no mgcp package-capability res-package
    no mgcp package-capability fxr-package
    mgcp tse payload 100
    no mgcp timer receive-rtcp
    mgcp sdp simple
    mgcp fax t38 inhibit
    mgcp rtp payload-type g726r16 static
    mgcp bind control source-interfacexx -----------------------put relevant interfcae here
    mgcp bind media source-interface xxx--------------------same
    Please rate all useful posts
    "opportunity is a haughty goddess who waste no time with those who are unprepared"

  • VG 224 Voice Gateway using MCGP or fax

    Does anyone have a experience or a config for running MGCP on a VG224 voice gateway (using it for inbound/outbound fax)to Call Manager? I've set up SCCP and it worked initially but in production it drops calls and does not respond after reboot without re-enabling SCCP and STAPP

    I deployed 5 of these to service a mix of phones and faxes a few months ago using MGCP.
    I used fax passthrough; we've had 0 problems so far with the setup.... customer has had several power cuts and no startup problems.
    Regards
    Aaron
    Please rate helpful posts...

  • VG 224 Voice Gateway using MCGP for fax

    Does anyone have any experience or configs to set up a VG224 Voice Gateway to manage via Call Manager 4.1 using MGCP (for inbound/outbound fax)? SSCP is currently set up, but seems unstable.

    try this link for VG 224 support on MGCP
    http://www.cisco.com/en/US/tech/tk652/tk777/technologies_tech_note09186a0080159cf3.shtml

  • Voice gateway SPA8000 problem sending faxes.

    Hello, I am a representative of the corporation Kazakhmys using the whole range of your devices. The problem is as follows: Voice Gateway Cisco SPA 8000 does not send faxes to voice gateway Grandstream GXW4232. The problem occurs only if the two voice gateways work with each other. The call comes to the SPA 8000, but after lifting the handset, the session immediately reset. What settings I need to change?

    It seems you have RTP packet Size set to 30ms. It's not optimal even for voice call. You should not exceed 20ms for voice, for the fax I propose you to use 10ms.
    Note it may not solve your problem, but it increase reliability of fax session.
    But now back to the issue. If I understand correctly, the attached log has been captured on
    Cisco-SIPGateway/IOS-15.2.4.M7
    OK. It seem that most important line of such log is:
    068963: Feb 24 05:50:16.778: //6170583/D6D178518A93/CCAPI/cc_api_call_disconnect_done:
    Disposition=0, Interface=0x160DE9D8, Tag=0x0, Call Id=6170583,
    Call Entry(Disconnect Cause=3, Voice Class Cause Code=0, Retry Count=0)
    If I understand correctly the log, the call has been terminated by SIP Gateway.
    I see no apparent issue with ATA8000 (not counting the Packet Size value mentioned on top). There is a problem with SIP Gateway. Unfortunately, I'm not expert on the matter nor this forum is best forum for question related to SIP Gateway. You should ask on IP Telephony forum instead.

  • 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

  • How does a voice gateway handle a call received from CUCM?

    For example, I have a voice gateway configured via a SIP trunk as a device in CUCM. When our users dial the international pattern (8011!) and CUCM forwards that to the device(gateway) over the SIP trunk, how does the gateway handle that request in terms of matching dial-peers? Is it handled the same way a call incoming from the PSTN would be? By matching the destination pattern?
    Ultimately what I'm trying to do is figure out whether our international calling is configured properly but that's difficult to do If I don't know what the gateway is doing with the call once it receives it from the CUCM.
    Thanks in advance!

    Thanks, we have a conference call Monday and I'll try some debugging to see what's going on. I am a little more confused now if that's possible because my understanding is that they can make INTL calls but based off of the configs I've seen and what you've told me they should be able to. Below are my configs. There is a translation profile applied but it points to a blank rule. I also didn't see any transformations in the route pattern.
    mgcp profile default
    dial-peer voice 1 pots
     incoming called-number .
     direct-inward-dial
     port 0/0/0:23
    dial-peer voice 100 voip
     incoming called-number .
     dtmf-relay rtp-nte
     codec g711ulaw
     ip qos dscp cs3 signaling
     no vad
    dial-peer voice 101 voip
     destination-pattern 5...
     session protocol sipv2
     session target ipv4:x.x.x.x
     dtmf-relay rtp-nte
     codec g711ulaw
     ip qos dscp cs3 signaling
     no vad
    dial-peer voice 2 pots
     destination-pattern 911
     port 0/0/0:23
     forward-digits all
    dial-peer voice 3 pots
     destination-pattern 9[94]11
     port 0/0/0:23
     forward-digits 3
    dial-peer voice 5 pots
     translation-profile outgoing NATL
     destination-pattern 9[2-9]......
     port 0/0/0:23
     forward-digits 7
    dial-peer voice 6 pots
     translation-profile outgoing NATL
     destination-pattern 9707[2-9]......
     port 0/0/0:23
    dial-peer voice 7 pots
     translation-profile outgoing NATL
     destination-pattern 91[2-9]..[2-9]......
     port 0/0/0:23
     forward-digits 11
    dial-peer voice 8 pots
     translation-profile outgoing INTL
     destination-pattern 9011T
     port 0/0/0:23
     prefix 011
    voice translation-rule 10
     rule 1 // // type any international plan any unknown

  • 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

  • Could CUOM back up voice gateway configuration?

    Hello all,
    I wonder if CUOM could back up voice gateway configuration.
    And, I assume Cisco works back up router/VGW config in most of cases,  is it correct?
    Thank you in advance,
    master001

    By design, the iphone will sync itunes content with ONE computer at a time. If you attempt to sync such content with a second computer, ALL itunes content will first be erased from your phone & then replaced with the content from the second computer. This is a design feature & cannot be overridden. Because you formatted your computer, the iphone will see your computer as a "new" computer.
    1.First, disable auto sync when an ipod/iphone is connected, in preferences, in itunes, which in windows is in the edit menu.
    2. Put one contact & one event in whatever programs you use for that purpose on your computer, they can be fake, doesn't matter, the important point is to have one entry in each.
    3. Connect your phone, itunes running, DO NOT SYNC.
    4. Go up to Store>Authorize this Computer.
    5. Go up to File>Transfer Purchases. All of the purchased itunes content on your phone will be transferred, music you ripped on your own will not be transferred. You will have to use third party software to first extract that music from your phone BEFORE YOU DO ANYTHING. Same for photos not in your camera roll, the photo sync is one way...computer to phone. This is one example:
    http://www.wideanglesoftware.com/touchcopy/index.php
    6. Right click the phone in the device pane & select Reset Warnings.
    7. Right click the phone, again, & select Backup.
    8. Right click the phone, again, & select Restore from Backup. Select the backup you just made. Voice Memos are included in the iphone's backup.
    9. This MUST be followed by a sync to restore your itunes content, which you select as before from the various tabs.
    You'll get a popup regarding your contacts & calendars asking to merge or replace, select merge.
    This article details what's included in the backup you'll make & restore from:
    http://support.apple.com/kb/HT1766

  • 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."

  • Call Manager register fxs port with voice gateway- problem

    I have a CUCM 6 and a Voice Gateway V224. I've configured the voice gateway's voice FXS ports as MGCP.
    I have a Voip connected and registered to the CUCM and a Pots phone connected to the Voice Gateway.
    If i dial from the Voip to the Pots phone it rings. The problem is that i cannot ring from the Pots to the Voip phone.
    I have no dial tone.
    If i write no shut down on the  voice port i have a tone. If i configure mgcp on the voice port i have a busy ringtone.
    I've entered no mgcp and mgcp commands and i've reset the voice gateway.
    How can i call from the pots to the voip phone?
    The ios version on the voice gateway is Version 12.4(22)T4.
    Here is an outghtput from the Voice gateway.
    ccm-manager mgcp
    ccm-manager fax protocol cisco
    ccm-manager music-on-hold
    ccm-manager config server 10.1.1.33
    ccm-manager config
    mgcp
    mgcp call-agent CCMIOSS 2427 service-type mgcp version 0.1
    mgcp rtp unreachable timeout 1000 action notify
    mgcp modem passthrough voip mode nse
    mgcp package-capability rtp-package
    mgcp package-capability sst-package
    no mgcp package-capability res-package
    no mgcp timer receive-rtcp
    mgcp sdp simple
    mgcp validate domain-name
    mgcp rtp payload-type g726r16 static
    mgcp profile default
    timeout tone busy 600
    timeout tone dial 600
    dial-peer voice 999223 pots
    service mgcpapp
    port 2/23
    dial-peer voice 999222 pots
    service mgcpapp
    port 2/22
    dial-peer voice 999888 pots
    service mgcpapp
    port 2/23
    The CUCM 6 is registered with the voice gateway.

    Is your campaign using CPA? If so, what's the behavior if CPA is not enabled? 
    I think the best thing to do is to run a trace...
    Call Manager > Cisco Unified Serviceability > Trace > Configurations
    Select a CUCM server - any subscriber would work. 
    Service Group - CM Services
    Cisco CallManager (Inactive)
    Enable SIP Stack Trace and apply to all nodes. Download and install RTMT
    Make a bunch of outbound tests and then open RTMT > Trace & Log Central > Collect Files > Check "All Servers" for Cisco CallManager > Next > Next > Relative Range if you made the test calls within the last X minutes, otherwise you can set a From and To datetime. Click Finish and go through your SDL logs and see what errors you find and post them here. 
    Also, make sure your phone is in the correct CSS in Call Manager

Maybe you are looking for

  • DVR-110 External FireWire Case Recommendation??

    Hi everyone, I'm running a MDD G4 with an original DVR-104 DVD burner. I've bought a DVR-110 DVD Burner and want to connect via FireWire to be able to choose between the two burners when creating DVDs. Can anyone recommend a Fire Wire DVD drive enclo

  • Blackberry Desktop Software not responding

    Blackberry Desktop Software Not Responding - receiving this message even after uninstall, reinstall. My device is Curve 8320. Thanks.

  • Lightroom 4.1 Slideshow Skipping Issue

    I have tried creating a slideshow in Lightroom 4.1 and followed all of the necessary steps Adobe has provided. On computer the slideshow seems fine but ufortunately, when I burn the slideshow to a DVD for Television, the slides skip. I think it may h

  • ACL and sequence numbers

    I had the first two lines in the access list and all was well, I then added the 3rd. From what I need to put the 3rd entry (deny host 10.1.30.51) after the second entry and before the permit any. Even though I created sequence numbers in order of the

  • Backup drive not mounting.

    Here is the scenario. I was trying to install Leopard the other night when I ran into the HFS volume issue. or whatever. So, I already had my Tiger system backed up to the old 80 gig hard drive that was in my Macbook before I upgraded to a 200 gig mo