Overhead Paging with Cisco Berbee InformaCast - Caller ID when paging

Hello, I'm using Cisco Berbee InformaCast with a Bogen overhead paging system. I have Call Manager 6.1.4 and would like to know if there is a way to broadcast the name/number of the IP Phone from which the page originated. If anyone knows of a means of achieving this I would greatly appreciate your assistance.
Thanks,
Dave

Well, not sure why I haven't received any suggestions from anyone. I find it
hard to believe that there isn't anyone else who has the same need. After going from TAC to SingleWire Support here's what I have:
there is a way to have InformaCast display within message text the extension that sent a page.  The only catch we were running into with your current implementation was ReliCast.  If you're using a ReliCast dial pattern to page, then all InformaCast would display is the ReliCast CTI port that "called" InformaCast to make the page. It was my hope that there was a way to have the source of that trigger displayed in the reporting feature of InformaCast 7.  Unfortunately InformaCast would produce the same Information.  It would see the calling device as the ReliCast CTI port and log/report that information.
A possible solution to this issue, a panic button.  You could setup one of the line appearance buttons on a users phone as an emergency live all page.  Then, we can just add a bit of "script" to it that would display the time, day, and source device that made the page.
For example, you can create a new page type using a combination of Text and Live Audio.  In the Short Text section of the message you could place the following:
"Panic Button Triggered at ${time} on ${date} by ext ${senderInfo.directoryNumbers}"

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    fallback-dn 5428008
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    fallback-dn 5421462
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    time-format 24
    date-format dd-mm-yy
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    line 2
    no activation-character
    no exec
    transport preferred none
    transport input all
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    stopbits 1
    line 131
    no activation-character
    no exec
    transport preferred none
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    stopbits 1
    line vty 0 4
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    exec-timeout 60 0
    privilege level 15
    login local
    transport input all
    line vty 5 15
    session-timeout 60
    exec-timeout 60 0
    privilege level 15
    login local
    transport input all
    scheduler allocate 20000 1000
    ntp server 10.1.30.1
    end
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    =~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2014.03.03 17:57:44 =~=~=~=~=~=~=~=~=~=~=~=
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    Gateway Local Interface: GigabitEthernet0/0
    IPv4 Address: 10.198.2.9
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    Call Manager: 152.63.1.19, Port Number: 2000
    Priority: N/A, Version: 7.0, Identifier: 4
    Trustpoint: N/A
    Call Manager: 152.63.1.100, Port Number: 2000
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    Trustpoint: N/A
    Call Manager: 172.27.210.5, Port Number: 2000
    Priority: N/A, Version: 7.0, Identifier: 6
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    MTP Oper State: ACTIVE - Cause Code: NONE
    Active Call Manager: 152.63.1.19, Port Number: 2000
    TCP Link Status: CONNECTED, Profile Identifier: 1000
    Reported Max Streams: 400, Reported Max OOS Streams: 0
    Supported Codec: g711alaw, Maximum Packetization Period: 30
    Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
    Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30
    Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization Period: 30
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    Active Call Manager: 152.63.1.19, Port Number: 2000
    TCP Link Status: CONNECTED, Profile Identifier: 1001
    Reported Max Streams: 36, Reported Max OOS Streams: 0
    Supported Codec: ilbc, Maximum Packetization Period: 120
    Supported Codec: g722r64, Maximum Packetization Period: 30
    Supported Codec: g729br8, Maximum Packetization Period: 60
    Supported Codec: g729r8, Maximum Packetization Period: 60
    Supported Codec: gsmamr-nb, Maximum Packetization Period: 60
    Supported Codec: pass-thru, Maximum Packetization Period: N/A
    Supported Codec: g711ulaw, Maximum Packetization Period: 30
    Supported Codec: g711alaw, Maximum Packetization Period: 30
    Supported Codec: g729ar8, Maximum Packetization Period: 60
    Supported Codec: g729abr8, Maximum Packetization Period: 60
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    Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30
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    Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
    Supported Codec: g711ulaw, Maximum Packetization Period: 20
    Supported Codec: g711alaw, Maximum Packetization Period: 20
    Supported Codec: g729r8, Maximum Packetization Period: 220Supported Codec: g729ar8, Maximum Packetization Period: 220
    Supported Codec: g729br8, Maximum Packetization Period: 220
    Supported Codec: g729r8, Maximum Packetization Period: 220
    Supported Codec: ilbc, Maximum Packetization Period: 120
    Alg_Phone Oper State: ACTIVE - Cause Code: NONE
    Active Call Manager: 152.63.1.19, Port Number: 2000
    TCP Link Status: CONNECTED, Device Name: AN71FEF7F070081
    Reported Max Streams: 1, Reported Max OOS Streams: 0
    Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
    Supported Codec: g711ulaw, Maximum Packetization Period: 20
    Supported Codec: g711alaw, Maximum Packetization Period: 20
    Supported Codec: g729r8, Maximum Packetization Period: 220
    Supported Codec: g729ar8, Maximum Packetization Period: 220
    Supported Codec: g729br8, Maximum Packetization Period: 220
    Supported Codec: g729r8, Maximum Packetization Period: 220
    Supported Codec: ilbc, Maximum Packetization Period: 120
    Alg_Phone Oper State: ACTIVE - Cause Code: NONE
    Active Call Manager: 152.63.1.19, Port Number: 2000
    TCP Link Status: CONNECTED, Device Name: AN71FEF7F070082
    Reported Max Streams: 1, Reported Max OOS Streams: 0
    Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
    Supported Codec: g711ulaw, Maximum Packetization Period: 20Supported Codec: g711alaw, Maximum Packetization Period: 20
    Supported Codec: g729r8, Maximum Packetization Period: 220
    Supported Codec: g729ar8, Maximum Packetization Period: 220
    Supported Codec: g729br8, Maximum Packetization Period: 220
    Supported Codec: g729r8, Maximum Packetization Period: 220
    Supported Codec: ilbc, Maximum Packetization Period: 120
    Alg_Phone Oper State: ACTIVE - Cause Code: NONE
    Active Call Manager: 152.63.1.19, Port Number: 2000
    TCP Link Status: CONNECTED, Device Name: AN71FEF7F070083
    Reported Max Streams: 1, Reported Max OOS Streams: 0
    Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
    Supported Codec: g711ulaw, Maximum Packetization Period: 20
    Supported Codec: g711alaw, Maximum Packetization Period: 20
    Supported Codec: g729r8, Maximum Packetization Period: 220
    Supported Codec: g729ar8, Maximum Packetization Period: 220
    Supported Codec: g729br8, Maximum Packetization Period: 220
    Supported Codec: g729r8, Maximum Packetization Period: 220
    Supported Codec: ilbc, Maximum Packetization Period: 120
    eucamvgw01#

  • Ask the Expert : Call Recording with Cisco Unified Communication Manager (UCM)

    Welcome to the Cisco Support Community Ask the Expert conversation.  This is an opportunity to learn and ask questions about Cisco Unified CM call recording solution that provides the ability to record customer conversations for compliance purpose. This topic will cover an overview, configuration and troubleshooting of the call recording feature.
    Monday, January 19th, 2015 to Friday, January 30th, 2015
    Harmit Singh is a technical leader with the High Touch Technical Services (HTTS) and Technical Assistance Center (TAC) Unified Communications teams based in Bangalore. He has broad experience in Cisco Unified Communications infrastructure solutions. He has 10 years of experience working with large enterprise and service provider networks. He also holds CCIE certifications (#20012) in Voice and Collaboration as well as Red Hat and VMware certifications.
    Mohammed Noorulla Khan is a customer support engineer in High-Touch Technical Services (HTTS)  Unified Communications teams based in Bangalore. His areas of expertise include Cisco Unified Communications Manager, Gateways, and Jabber. He has over 6 years of industry experience working with large enterprise and service provider networks. He also holds CCIE certifications (#35741) in Voice and VMware certifications.
    ** Remember to use the rating system to let Mohammed and Harmit know if you've received an adequate response.  **
    Because of the volume expected during this event, the experts might not be able to answer every question. Remember that you can continue the conversation in the Collaboration, Voice and Video  community, subcommunity, IP Telephony, shortly after the event. This event lasts through January 30th 2015. Visit this forum often to view responses to your questions and those of other Cisco Support Community members.

    Hi Maheshwar,
    Thank you for your query. Please find my response below:
    1> Do we support recording with HCS environment and which 3rd party vendors are validated with HCS based call control 10.1.1?
    Answer: Whether you use a standalone UCM cluster, UCCE or HCS, call recording would be supported across the board in the same manner.
    Please refer to the following link:
    http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cust_contact/contact_center/hcs-cc/10_0_1/Install_and_Config/CHCS_BK_ICC270D0_00_installing-and-configuring-cisco-hcs/CHCS_BK_ICC270D0_00_installing-and-configuring-cisco-hcs_chapter_011.html#CHCS_RF_T1105284_00
    Option
    Notes
    Recording
    All Recording applications that are supported by Unified CCE are supported on HCS for CC. For details, see Recording section in Agent and Supervisor Capabilities.
    With respect to which 3rd party vendors have been validated, marketplace.cisco.com is a good place to crosscheck this info. You will find a Cisco Compatible Logo against the partners listed there. The logo is used to signify that the PARTNER product has undergone technical interoperability testing with the Cisco product specified. The interoperability testing is conducted by a third party laboratory based on testing criteria set forth by Cisco. PARTNER is solely responsible for the support and warranty of its product. Placement of the PARTNER product or information pertaining thereto, on the Cisco Marketplace website does not constitute an offer to sell the PARTNER product in any way. For further information on the PARTNER products, please visit the PARTNER company website.
    Please refer to the following link and use the search field under Collaboration Technology:
    https://marketplace.cisco.com/catalog/search?utf8=%E2%9C%93&search[q]=&search[technology_category_ids]=23%2C24%2C197%2C1940%2C1941%2C1921%2C1576%2C1897%2C1983%2C2418%2C26%2C198%2C1904&search[order]=tier&per_page=20&_=1421663854257&ts=1421663855441
    2> Which end points are supported for recording via HCS call control?
    Answer: The following link should help clarify this:
    http://solutionpartner.cisco.com/web/sip/wiki/-/wiki/Main/Unified+CM+Silent+Monitoring+Recording+Supported+Device+Matrix
    Please let us know if you have any follow up questions. Hope this is helpful.
    Regards,
    Harmit Singh.

  • Is it possible to integrate Nokia E62 with Cisco Call Manager 4.x?

    The Nokia E-61 can be integrated with Cisco Call manager 4.x using a Nokia Call Connect 1.0 client.
    1.Is it possible to do the same with Nokia E62.If it can be done,what would be the requirements?
    2.Is Nokia E-62 PDA a Cisco Compatible Extension device?

    Thanks a lot for the prompt reply.You are right in saying that nokia and cisco are increasing the number of models supported.Infact the E65 has already joined the club.But my point of interest is the nokia E62 in particular.This Nokia E62 device
    1) does not support WLAN interface(while E61,E65 does).
    2) It is not Cisco Commpatible extension device.(I did not find E62 in the list of Cisco Compatible extensions.chk out the attachments for more info).
    3)It is not compatible with Nokia Intellisync CAll Connect 1.0(while E61 and E65 are compatible.chk out the attachment for more info).
    I found this rather weird and i wondered if Nokia E62 can be integrated with Cisco Call manager ,especially a device which is not Cisco compatible.The only possibility i can see from your answer is that the Nokia E62 might not be SIP enabled(correct me if i am wrong here!)
    My objective is to send a message from a CUAE script to a nokia E62 device.Please excuse me if my posting is not in the right place.i would be highly grateful to you if you could give me pointers to an appropiate location where i can get my queries answered.

  • Any one using/deployed Telematrix phones with Cisco Call Manager

    Any one using/deployed Telematrix phones with Cisco Call Manager

    Yes, it does:
    Cisco Unified IP Phone Series 7900
    http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/compat/ccmcompmatr.html#wp51474
    Config on CUCM itself is no different from any other phones neither other requirements like DLUs.
    Of course you should already have APs for connectivity.
    HTH
    java
    if this helps, please rate

  • Integration with Cisco Call Manager?

    We have Cisco VOIP phones and we would like to continue managing calls with Cisco Call Manager.  We currently have voicemail on Cisco Unity.  We currently have Exchange 2010, but plan to start using Exchange 2013 soon.
    However, we would like to migrate voicemail service from Cisco Unity to Exchange Unified Messaging and also want to take advantage of most of the voicemail-related features of UM such as voice to text sent to email, voice mail waiting indicator light on
    the phones and sending voice mail audio attachments to email. 
    Is Exchange 2013 Unified Messaging compatible Cisco phones that are managed by Cisco Call Manager?
    We have 500 users and 2 Exchange 2010 servers (1 CAS and 1 Mailbox Server).  We now have a few Exchange 2013 licenses available in addition to our existing 2010 licenses that are already used on our existing 2 servers.  Can we keep the 2 existing
    Exchange 2010 servers as they are and create a new Unified Messaging server on  Exchange 2013 in the same environment or do we need to upgrade all the 2010 servers to 2013 if we use Unified Messaging on 2013?

    Hi Sachin,
    We do run a hybrid Callmanager to Nortel Meridian setup (works well).
    Here are some great docs;
    Cisco Unified CallManager
    Case Study: Nortel 61C PBX to Cisco IP Telephony Migration
    From this good doc;
    http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_white_paper09186a00801115e0.shtml
    Nortel Meridian PBX and Cisco CallManager Integration
    http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_configuration_example09186a008011888c.shtml
    Cisco Unified CallManager System Guide, Release 4.2(1)
    Cisco DPA Integration
    http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_administration_guide_chapter09186a008055cd53.html
    Hope this helps! Let me know if you need additional info.
    Rob

  • Cisco Unified WFO - Call Recording and Quality Management with Extension Mobility agents

    Hi All,
    We're considering Cisco Unified WFO - Call Recording and Quality Management for a customer running UCCX 8.0, agents on multiple WAN sites, all agents using extension mobility.
    The documentation I've been able to find describes three different recording methods:
    Using Desktop Recording service (Endpoint) to record from an agent’s desktop.
    Server Recording - Uses SPAN (not so good for remote sites)
    Network Recording - Uses CUCM recording service / SIP trunk / phone's built in bridge.
    Network recording or Desktop recording should be suitable for the customer but it seems that Extension mobility is not supported.  Extension Mobility is not mentioned in the 8.5 installation guide, it is mentioned as ‘not supported’ in the 8.0 guide as follows:
    'Server Recording and Network Recording have the following limitations:
    • Extension mobility is not supported.'
    Neither version's documentation specifically mention extension mobility in relation to the desktop recording method, though I realise this is a similar approach to the 'server recording' method.
    So the question I have is:  Is extension mobility supported in any way on version 8.0, or version 8.5 for recording?  And if so which recording method(s) are supported?
    Thanks,
    Jonathan

    Hi,
    I had more luck asking questions over at the Calabrio forum - they make the software and Cisco re-brand a version of it - there is some good info on their portal (http://portal.calabrio.com), you have to register but it's fairly painless.  The answer I got was:
    "QM Desktop recording has always supported extention mobility as it determines the recorded user by the desktop user's login. Extention mobility was not supported for Server and Network recording until the Calabrio QM 8.6.2 release in April 2011 and will be added to Cisco QM starting with QM 8.5.2 in June 2011"
    Regards,
    Jonathan

  • Cisco Unified WFO - Call Recording and Quality Management stops recording with conferenced translator

    I'm having an issue whenever one of our employee's conferences in an external translator, as soon as they bridge the customer into the call with the translator QM stops recording.  I can hear the intial conversation with the customer, and then the employee put the customer on hold and call the translator service.  Only when the two are bridged together the call always stops recording. 
    We are performing all recording on the server side, and are not using Quality Manager Desktop agents.
    Any help would be much appreciated.  Thanks!
    -Chris                   

    So I found the following information listed below.  I don't manage the Cisco Unified CM portion of our telco system.  Can we limited the ourselves to a single Codec, and would this even resolve the issue.  Does this cause other issues if we didn't limit the devices that are recording to a single codec?
    Recording IssuesThis topic explains how to diagnose and resolve problems that occur with contact
    recordings.
    Calls for devices configured for Network Recording drop when you try to conference
    or transfer a call.
    Symptom. When you try to transfer or conference a call and one or more devices on
    the call configured for network recording, the transfer or conference fails and
    parties drop off of the call.
    Cause. Cisco Unified CM does not support codec changes for devices that are
    configured for call recording. The codec must remain the same throughout the life
    of the call. For conference calls, the conference bridge must support the codec
    used before the conference completes.
    Solution. Update the Cisco Unified CM configuration to ensure that no codec
    changes occur for network recorded devices.

  • Click to Call with Cisco WIM 4.4?

    Hi eGain Experts,
    Could anybody confirm that Click to Call with Cisco WIM 4.4 will work with UCCE? I understand Web Callback will work, however not too sure about Click to call. I know eGain CallTrack or ClickToCall can achieve this.  Is anybody able to confirm if this feature is configurable with the Cisco WIM licensing arrangements? if so could somebody provide sample configuration on how to achieve would be very useful.
    Thank you very much.
    Regards,
    Yavuz

    Short answer no. All evidence points to UCCX only.
    Sent from Cisco Technical Support iPad App

  • ATA-186 with overhead paging systems

    Since the ATA-186 supplies 600 Ohms, would I need a seperate device such as a Bogen TAM-B to supplie the audio for overhead paging, or could I just use the ATA-186?

    ATA-186 can be used as an audio source.

  • Message from Berbee Informacast

    Hi,
    We are currently using Cisco Call Manager 5.1.2 and Berbee Informacast 5.1.1 [server:Windows 2003 R2 SP2]system. Once in a while all our IP phones display following broadcast message " Telephone Service Working " [ this message is on Informacast Administration Message page]. What is causing this broadcast? How can I prevent this broadcast? Do I need to take any further action to resolve this issue? Please let me know.
    Thanks in advance.
    Alex Thazhayil

    Informacast has been successfully tested with thousands of phones. The actual messages are broadcast via multicast while phone control is sent via unicast. The multicast message is made available on a stream on the network and must remain available until the last phone on the network has received the message. In an installation of thousands of phones the stream must remain available on the network until the last phone is activated via unicast phone control. This means that the first phones that are activated are forced to get the message for the entire time the stream is available, thus you get some phones that repeat the message.

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