SIP dialer call flow

Hello

can someone explain how to call flow for sip dialer? We have ucce 904 and ipivr 9

-Sylvie

Base officer, here are the basics

  1. Import executes and records are added to the table of list of the composition. This table is in the database of the Dialer
  2. Dialer component on the PG is administered the ITC monitors server records and campaign manager for an available agent in the Group of skills defined when setting up the campaign.
  3. Dialer identifies an available agent and sends a route request to the MRPG, which then forwards the request to the router where running a booking script
  4. The router detects an available agent and returns a label to the MRPG that sends the label to the Dialer. The Dialer places a reservation call the agent and once delivered, automatically place the call on hold through the CTI Server
  5. The dialer uses its ports configured in call Manager dialer to call out via voice gateway
  6. The dialer can use CPA to analyze RTP audio streams to detect when a human voice is linked
  7. Assuming that a voice live is detected, the dialer now transfers the call from its port Dialer to a reserved officer who appears as a second call to the agent. The dialers (dialers) use the CTI server to auto answer call and disconnects his booking, also put request to update the campaign manager
  8. At the end of the call, the dialer detail records are passed in the campaign to the Router Manager
  9. The router forwards these historical data to the recorder
  10. Historical data committed to the database of loggers
  11. Historical data are repliced to the HDS intermittently. Data are finally committed to the HDS thin repoting database

This is all assuming you have some basics of the Dialer, its installation process, the components and the MRPG (s) who is involved.

Tags: Cisco Support

Similar Questions

  • SIP Dialer with voice gateway using the FXO Ports

    Hello

    We have a laboratory UCCE consisting of an outgoing SIP dialer and a h323 2811 voice modem router with 2 FXO ports. The Dialer reserve an agent and sending the call to the bridge, but the call fails immediately with the bridge return a "404 not found" error to the Dialer. Call directly from a CUCM extension attempts to connect successfully through the FXO. Any thoughts?

    IOS Ver - T4 15.1 (3)

    UCCE worm - 8.5.4

    The gateway configuration:

    version 15.1
    horodateurs service debug datetime msec
    Log service timestamps datetime msec
    no password encryption service
    sequence numbers service
    !
    hostname LAB-RS-ING_VXML-GW
    !
    boot-start-marker
    boot-end-marker
    !
    !
    logging buffered 50000000
    no logging monitor

    !
    No aaa new-model
    no location network-clock-participate 1
    !
    voice-card 0
    !
    voice-card 1
    dspfarm
    !
    dot11 syslog
    IP source-route
    !
    !
    IP cef
    !
    !
    !
    IP host mediaserver 10.15.80.83
    No ipv6 cef
    Authenticated MultiLink bundle-name Panel
    !
    !
    !
    !
    !
    !
    Circuits-port FXO group
    hunting-sequential diagram
    !
    !
    !
    voip phone service
    No IP trust to authenticate
    h323 connections allow h323
    allow connections h323 to SIP
    allow connections sip h323
    allow sip to sip connections
    no service additional sip refer
    service additional media - renegotiate
    signs before any
    H323
    Modem passthrough codec g711ulaw nse
    SIP
    interface FastEthernet0/0 source control binding
    bind media source interface FastEthernet0/0
    header-passage
    !
    voice class codec 1
    g711ulaw codec preference 1
    preferably 5 codec g729r8
    !
    vocal h323 class 1
    H225 timeout tcp establish 3
    !
    !
    !
    !
    translation-article 99 of the voice
    rule 1 / ^ 4 / /334/
    !
    !
    voice translation-profile PROFILE_TO_CVP
    translate called 99
    !
    !
    Crypto pki token removal timeout default 0
    !
    !
    !
    !
    Archives
    The config log
    hidekeys
    !
    !
    !
    !
    !
    !
    !
    interface Loopback0
    255.255.255.255 IP address 172.20.1.1
    !
    interface FastEthernet0/0
    IP 255.255.255.0
    automatic duplex
    automatic speed
    !
    interface FastEthernet0/1
    IP 255.255.255.0
    Shutdown
    automatic duplex
    automatic speed
    !
    !
    Router eigrp 15
    10.0.0.0 network
    Auto-resume
    !
    IP forward-Protocol ND
    !
    no ip address of the http server
    no ip http secure server
    !
    IP route 0.0.0.0 0.0.0.0 10.x.x.x
    !
    !
    !
    !
    control plan
    !
    !
    voice-port 0/0/0
    Ports FXO 1 trunk group
    surveillance cut dualtone Mid-communication
    output attenuation - 3
    no echo - cancel enable
    No non-linear
    No vad
    broadcast-maximum: 250
    broadcast / nominal 200
    minimum of playout / high-delay
    broadcast-delay mode fixed
    call waiting times - disconnect 5
    timeouts wait-version 5
    connection ÉRA 4930
    4930 description
    !
    voice-port 1/0/0
    2 ports FXO trunk-group
    surveillance cut dualtone Mid-communication
    output attenuation - 3
    no echo - cancel enable
    No non-linear
    No vad
    broadcast-maximum: 250
    broadcast / nominal 200
    minimum of playout / high-delay
    broadcast-delay mode fixed
    call waiting times - disconnect 5
    timeouts wait-version 5
    connection ÉRA 4198
    4198 description
    !
    voice-port 0/0/2
    !
    voice-port 0/0/3
    !
    !
    !
    profile MGCP default
    !
    !
    voice POTS dial-peer 1
    incoming called-number.
    direct line to inside
    !
    Dial-peer voice 2 voip
    session protocol sipv2
    incoming called-number.
    codec voice-class 1
    no class voice sip reset timer expires 183
    DTMF-relay h245 alphanumeric
    No vad
    !
    Dial-peer voice voip 4198
    destination-model 4198
    session target ipv4:10.15.80.81
    codec voice-class 1
    h323 voice-class 1
    voice-class sip rel1xx taken in charge "100rel.
    no class voice sip reset timer expires 183
    DTMF-relay h245 alphanumeric
    No vad
    !
    voice pots Dial-peer 301
    trunkgroup FXO Ports
    Description * MIA local calls *.
    destination-model 305 [2-9]...
    alert progress_ind activate 8
    progress_ind enable progress 8
    direct line to inside
    prefix 305
    !
    Dial-peer voice voip 4930
    destination-model 4930
    session target ipv4:10.15.80.81
    codec voice-class 1
    h323 voice-class 1
    voice-class sip rel1xx taken in charge "100rel.
    no class voice sip reset timer expires 183
    DTMF-relay h245 alphanumeric
    No vad
    !
    Dial-peer voice voip 7000
    Description of Agents
    destination-model 7...
    session protocol sipv2
    session target ipv4:10.15.80.81
    codec voice-class 1
    voice-class sip rel1xx taken in charge "100rel.
    no class voice sip reset timer expires 183
    !
    Dial-peer voice voip 9999
    Phone agent test description
    destination-model 9999
    session protocol sipv2
    session target ipv4:10.15.80.81
    codec voice-class 1
    voice-class sip rel1xx taken in charge "100rel.
    !
    Dial-peer voice voip 1444
    Phone agent test description
    destination-model 1444
    session protocol sipv2
    session target ipv4:10.15.80.81
    codec voice-class 1
    voice-class sip rel1xx taken in charge "100rel.
    !
    voice pots Dial-peer 303
    trunkgroup FXO Ports
    Description * LD calls *.
    destination-pattern 1 [2-9]...
    alert progress_ind activate 8
    progress_ind enable progress 8
    direct line to inside
    Forward-digit 11
    1 prefix
    !
    voice pots Dial-peer 302
    trunkgroup FXO Ports
    Description * MIA local calls *.
    destination-model 786 [2-9]...
    alert progress_ind activate 8
    progress_ind enable progress 8
    direct line to inside
    prefix 786
    !
    voice pots Dial-peer 9
    destination-model 91 [2-9]...
    alert progress_ind activate 8
    progress_ind enable progress 8
    direct line to inside
    port 0/0/0
    Forward-digit 11
    !
    !
    Dial-peer voice 1400 voip
    Description of Agents
    destination-model 14...
    session protocol sipv2
    session target ipv4:10.15.80.81
    codec voice-class 1
    voice-class sip rel1xx taken in charge "100rel.
    !
    !
    !
    !
    !
    Line con 0
    line to 0
    line vty 0 4
    opening of session
    transport of entry all
    !
    No Scheduler allocate
    end

    Do you have installation trunks in call to the dialer Manager?

  • At the CCA IVR call flow

    Can someone please answer what is the call flow from the phone of the customer to the agent. What servers will be that he hit. The switch believers PSTN telephony server call customers phone hits. Telephony server has several processes server like call center server etc., application server has ACD server etc. What is the speed of the incoming CCA a call.

    The message flow is as follows:

    1 call comes in the trunk (no VoIP, so VoIP trunk goto step 3)
    2. call gateway Hits
    3 gateway send invite to the redirect server
    4 forwarding server load balances calls and returns a message moved (302)
    5 SIP call arrives on the call as a new call center
    6 call Center informs the new call CTI Server
    7 CTI server transferred call new message to the Instant Messaging server
    8. the appellant interacts with IVR/campaign
    9 IVR sends appeal to Workgroup/queue
    10. call integrated ACD queue
    11 call Center sends the message to the Instant Messaging server to be considered as an appeal of the CDA
    12 call Center queue time of ACD Server queries
    13. the customer listening to hold music, promotional messages, etc., while waiting to be connected with an agent
    14 ACD Server informed Call Center as an Agent is available
    15 ACD call from queue
    16 call Center sends to the Agent via CTI Server
    17 agent accepts the new interaction, the call is filled
    18 call Center cancels ring timeout

  • SCp call flow between ipphone and CME messages.

    Hello world.

    I'm looking for some good link on sccp call flow between cisco ip phone and a MAN messages.

    I'd appreciate your help if you could direct me to a good link.

    Thank you and have a nice day.

    Hello

    Take a look at this: http://flylib.com/books/en/2.10.1.156/1/

    Is an excerpt chapter of Cisco IP Communications Express: CallManager Express with Cisco Unity Express: http://www.ciscopress.com/store/cisco-ip-communications-express-callmanager-express-9781587142918

    Hoping to help, please don't forget to rate the post if it isn't.

    Kind regards!

    -Adrian.

  • Webex Cloud CMR SIP video calls disconnect after 20 seconds

    Hello

    I am running version 9.1 with Exressway-C CUCM / 8.6 Expressway-E.  I have no problem doing business SIP video calls.  However when I make video calls on a site of WebEx CMR, the call becomes still cut off after 20 seconds.  It is audio and video for the first 20 seconds and then disconnects the call.

    I was able to recreate this situation of 3 of my clients running CUCM version 9.

    I have 2 clients running CUCM v10.5 and the issue is not the case.  I don't know if this question is 100% related to the CUCM version, but looks like a correlation obviouse.

    I watched the SIP trace and it seems that the endpoint CUCM is end the call with normal call clearing.  I tried 8945 phones, jabber for iphones and a C40 and I get the same results.

    If anyone has any ideas I would appreciate it.

    Thank you.

    Hello

    Can you receive calls B2B? During the call set up, Webex will try to set up another session TCP inbound to your E Expressway on port 5060 and 5061, do you have this open to outside within the TCP sessions? (see page 5 of the CMR-Cloud deployment guide: http://www.cisco.com/c/dam/en/us/td/docs/telepresence/infrastructure/sol...)

    If they are open, the TCP session is configured for an incoming invitation with complete SDP of the TPS Cloud, you can check your sip max message size is located at 11000 - if you went from CUCM ~8.6 5000 was the default size and the default continues for an upgrade.  Check under system-> Service settings-> CUCM service-> show click on the button to display all options, and then check the size of incoming messages SIP advanced Max.

    -Jonathan

  • "Temp fail" with the BLF speed dial + Call Pickup

    Hello all,.

    I am running version 8.6.2 CUCM and many phones IP 7962 G.

    I have a Manager configured with extension 1001 and his PA with extension 1002. The two extensions are configured in the same group of pick-up call.

    On the phone IP of PA, the second button is configured as a BLF Speed Dial with Call Pickup for the Manager.

    When someone calls the Manager, she can see that the line manager sounds but when she push the button to identify which called, she hears a busy tone and the IP phone displays a message that says "Temp Fail". I checked everything on IP phones and everything seems to be OK. Am I missing something?

    When I changed the line manager of 1001 to 1003, in the Group of pick-up even call it's works fine. She is able to pick up calls to 1003. But the server of CUCM, there is no difference between 1001 and 1003 extension.

    Please notify.

    Best regards,

    J. Mr. Kabundi.

    No matter what PickUp group they belong, do not need to be the same for the BLF pick-up function.

    But if they are in the same group, as you say: can the normal collection through the display key PickUp?

    If only change the number to work, I think you may have a problem with the DNs/Partitions. Never renamed the partition? Try to remove the Manager DN, delete the number not assigned to your server and add it again.

    If this still does not work... I will re-start the server if possible.

    Kind regards

    Sven

  • [SPA3102] SIP recording every hour with the 401 error and directly 12 OK

    Location: INET-ADSL modem in bridge mode-SPA3102.
    Problem: not really, everything seems to work, can dial in and out.
    But...
    Because I'm curious, I have a logserver of installation and checked what happened every hour of registration.

    What I see in the syslog hourly the 3102 made re - enroll by the SIP provider, but first I think 401 Unauthorized een 12 error and measured, I see an OK message.

    Seems weird to me.

    The same thing happens when I compose, firstly a 401 that OK.

    Can someone explain why the first attempt gives an error and how to avoid this?

    Some details of syslog:

    I replaced a text in the syslog:

    x.x.x.x is real My outside IP address.
    yyyyyyyyy = my local phone number including the area code, such as 31201234567, 31 = NL, 20 = codeZone for Amsterdam
    username = username by my SIP provider

    message 1:
    REGISTER SIP:SIP.poivy.com SIP/2.0
    Via: SIP/2.0/UDP x.x.x.x:5060; direction = z9hG4bK-2283ef7b
    From: + 31yyyyyyyyy ; tag = c85fff819484d288o0
    To: + 31yyyyyyyyy
    Call ID: [email protected]

    CSeq: 6104 REGISTRY
    Max-Forwards: 70
    Authorization: Digest username = "username", realm = "sip.poivy.com", nonce = "1663445546", uri = "sip:sip.poivy.com", algorithm = MD5 response = "c8c5b94c384559bb490b59b72be1c674"
    Contact: + 31yyyyyyyyy ; expires = 3600
    User-Agent: Linksys/SPA3102-3.3.6(GW)
    Content-Length: 0
    Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFICATION OPTIONS, see
    Support: x-sipura

    Message 2:
    SIP/2.0 401 Unauthorized
    Via: SIP/2.0/UDP x.x.x.x:5060; direction = z9hG4bK-2283ef7b
    From: + 31yyyyyyyyy ; tag = c85fff819484d288o0
    To: + 31yyyyyyyyy
    Contact: sip:x.x.x.x:5060
    Call ID: [email protected]
    CSeq: 6104 REGISTRY
    Server: (very nice Sip Registrar/Proxy Server)
    Allow: ACK, BYE, CANCEL, INVITE, REGISTER, OPTIONS, INFO, MESSAGE
    WWW-Authenticate: Digest realm = "sip.poivy.com", nonce = "1667015687", algorithm = MD5
    Content-Length: 0

    Message 3:
    REGISTER SIP:SIP.poivy.com SIP/2.0
    Via: SIP/2.0/UDP x.x.x.x:5060; direction = z9hG4bK-4d7052c
    From: + 31yyyyyyyyy ; tag = c85fff819484d288o0
    To: + 31yyyyyyyyy
    Call ID: [email protected]

    CSeq: 6105 REGISTRY
    Max-Forwards: 70
    Authorization: Digest username = "username", realm = "sip.poivy.com", nonce = "1667015687", uri = "sip:sip.poivy.com", algorithm = MD5 response = "46f2176652c0ad8d27f8d3ad1cf72c24"
    Contact: + 31yyyyyyyyy ; expires = 3600
    User-Agent: Linksys/SPA3102-3.3.6(GW)
    Content-Length: 0
    Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFICATION OPTIONS, see
    Support: x-sipura

    Message 4:
    SIP/2.0 200 Ok
    Via: SIP/2.0/UDP x.x.x.x:5060; direction = z9hG4bK-4d7052c
    From: + 31yyyyyyyyy ; tag = c85fff819484d288o0
    To: + 31yyyyyyyyy
    Contact: + 31yyyyyyyyy ; expires = 3600
    Call ID: [email protected]

    CSeq: 6105 REGISTRY
    Server: (very nice Sip Registrar/Proxy Server)
    Allow: ACK, BYE, CANCEL, INVITE, REGISTER, OPTIONS, INFO, MESSAGE
    Content-Length: 0

    @hw: thank you for your tip and your right on the spot!

    Never, ever, thought of this way of "logging" in a system and use delberatley an error response. With your tip, I thought that allows to read the RFC and found RFC 3665. This RFC describes the Protocol SIP basic call flow. And there he was, almost at the beginning of the real story on page 5! The protocol uses a command register which gives a message of 401 error back just to be sure to avoid security problems "man in the middle" (if I understand correctly). He present a challenge and you the answer to a totalizer new order including your answer on the challenge. Which will lead to an OK return the message.

    To resume: nothing weird, it is as expected. My curiosity is frankly satisfied. Another day with what I've learned something. Thank you.

  • Endpoint collaboration - Protocol call by default Auto - does not seek after failing with SIP H323

    Hi all

    Hoping someone in the community may be able to explain how the "auto" setting for the default calling Protocol?

    My understanding is that, when together, he should try to SIP and H323 - however, it seems to just try SIP then stop.

    I've seen CSCum27379, but could not know if it was fixed in the end.

    Context:

    We are currently move to SIP from H323, and have all items on the same VCS cluster endpoint (Polycom HDX & Cisco) double (first H323 / by default). The problem is that some people dial still using IP addresses, however our new external gateway and bridge services cannot be connected to the use of SIP (non-interoperable).

    The Polycom HDXs are ok, they'll just try a protocol after the other.

    The Cisco seems to be stuck on one, so either not will have access to the new services (except preconfigured in the address book), or will have all their IP address interoperability of point to point calls.

    Is there a way to direct address IP SIP work while composition on the VCS?

    I was also hoping to create a search rule that blackholed IP SIP addressed calls would force the reversal, but did not: (.)

    As always, appreciate any guidance you can provide.

    Rich

    This bug link you to, CSCum27379, there is no solution for it that its status is "completed".

    There is a specific order in which the endpoint will attempt to establish a call and depends on what protocol endpoint is saved for use, pg 84 SX20 Administrator's Guide :

    Allows the automatic selection of the memorandum of Appeal based on what protocols are available. If several protocols are available, is of the order of priority: 1) SIP; (2) H323; (3) H320. If the system is unable to register, self-selection chooses H323

    Endpoint won't try the first Protocol in the list and then switch to the other, he tries only one he can and that's all.  If you subscribed using SIP, but you need to make a call to a registered endpoint H323, you must set the Protocol to call to serve as a H323 or allow interoperability on the VCS to allow SIP-H323 calls.

  • How a dial-peer SIP Trunk using a Registrar?

    Hi all

    I'll put up a GUY using two records.  With the generous help of people on this forum, I got records in doubles to work.  But now I need to know how to configure the dial-peer to use registration information.  For example, I have this set up:

    SIP - ua

    Password 08114342101A0A1A43 7 authentication username 5555555555

    ...

    IPv4:11.11.11.11:6034 at the office 1 expires 3600

    Registrar 2 ipv4:22.22.22.22:6035 expires 3600

    How to configure a dial-peer to send traffic to one of the recordings?  I tried this, and it does not work:

    Dial-peer voice voip 105

    Description * outgoing SIP Trunk call *.

    translation-profile outgoing PSTN_Outgoing

    destination-model 91%...

    session protocol sipv2

    Registrar of target session? WHAT SHOULD I USE HERE?

    codec voice-class 2

    DTMF-relay rtp - nte

    No vad

    Thanks in advance.

    Hi Tod,

    Please go through this post, hope that it answers your question:

    http://tekcert.com/blog/2011/02/03/CME-configuration-example-SIP-trunks-ViaTalk-and-voipms

    Specify the article accordingly.

    Kind regards

    Kevin

  • Call the flow of VCS-E and VCS - C

    Hi all

    I'm looking for some documents describing the call flow of VCS-E and VCS - C in H.323 as the jpg I have attached.

    (Especially when endpoint dials on internet to the intranet)

    Are there documents like tthat?

    Best regards

    Kotaro Hashimoto

    Call the stream while VCS - E uses the SANCTION for traversal Protocol (almost identical even use H.460.18 to signal flow).

    Please note that this call flow only not understand key messages each Exchange of messages between VCS & endpoint as well as VCS-E & VCS - C.

  • How to answer voice calls using built-in modem and dialer Windows in Vista Ultimate?

    How can I answer incoming voice calls using Dialer Windows in Vista Ultimate.  I can make outgoing UST fine voice using Dialer calls, but I don't see any option to answer incoming calls.

    Hi Dobrodaddy,

    The option to receive calls through dialer.exe is not available in Windows Vista and 7, as it was in Windows XP. You can use a third-party software for this task.

    Note: Using third-party software, including hardware drivers can cause serious problems that may prevent your computer from starting properly. Microsoft cannot guarantee that problems resulting from the use of third-party software can be solved. Software using third party is at your own risk.

    Thanks and greetings

    Ajay K

    Microsoft Answers Support Engineer
    ***************************************************************************
    Visit our Microsoft answers feedback Forum and let us know what you think.

  • Typical call for CM flow / unit / Fax without obtaining additional?

    Configuration:

    Centralize the VMO unit / Call Manager / fax server in the data center. Routers SRST supporting all offices. Of Fax Server supported for the unit, such as RightFax or others on the approved list.

    What would the typical call flow if we wish to have the DID numbers subscribed to do double-duty as their fax number (so that we don't need to get all the new lines DID new business cards, etc.)

    Is it still possible? How would an incoming fax call get routed to the fax server automatically rather than as an incomprehensible series of fax tones in the voice mail box?

    What happens if the user is in fact on their phone and hear the fax tone? Is it possible to Call Manager program to generate a soft button on the 7960 that says 'Send fax'?

    Thank you.

    Short story is there is no way to achieve smooth - for reasons that you kind of point in your question (i.e., what to do if the user is on their phone?). Correctly do gateways would have generated fax tone detection (in some cases, the sender Server/fax won't that until he hears a fax tone on the side response - even if it has a hole in it) and spend an event upstairs that is supplied to the unit via CM - us would then is a fax call and send it to the fax server as well as information on the phone number and number composed as the fax ended up being routed to the right area.

    There was people who tried to do things moderately awkward IVR on gateways (i.e. the caller hears "if it is a voice call, press 1" which has then passes the call via as usual or he will send the call to a port on the fax)-not an ideal interface at all and the location of is not possible among other problems.

    Is really the only way to go about it.

  • Method call as default activity in ADF task flows

    Hello

    I have a workflow to execute a method on the page loading to set cookie values.
    I added a control method of data as the default activity in the workflow. But it is not called immediately, it is executed only if it is called from another view.

    Any help will be appreciated.

    WebCenter Portal App 11.1.1.6


    Code workflow task:

    <? XML version = "1.0" encoding = "UTF-8"? >
    < adfc-config xmlns = "http://xmlns.oracle.com/adf/controller" version = "1.2".
    ID = "___5" >
    < task-flow-definition = id "cookie-task-flow" >
    < default activity id = "__17" > addEmpNoCookie < / default activity >
    < transaction id = "__38" >
    < new-transaction / >
    < / transaction >
    < data-control-scope id = "__39" >
    < shared / >
    < / data-control-scope >
    < managed-bean id = "__1" >
    < id managed-bean-name = "__4" > cookieBean < / managed-bean-name >
    < managed-bean-class id = "__3" > tr.com.signum.roketsan.utils.CookieBean < / managed-bean-class >
    < managed-bean-scope id = "__2" > pageFlow < / managed-bean-scope >
    < / managed-bean >
    <-l' call the method id = "addEmpNoCookie" >
    < method id '__8' = > #{bindings.addEmpNoCookie.execute} < / method >
    < result id = "__16" >
    < id fixed-result = "__7" > addEmpNoCookie < / fixed-results >
    < / results >
    < / method >
    < use-page-fragments / >
    < / task-flow-definition >
    < / adfc-config >

    Def of the activity of the Middle page of

    <? XML version = "1.0" encoding = "UTF-8"? >
    < pageDefinition xmlns = "http://xmlns.oracle.com/adfm/uimodel."
    version = "11.1.1.61.92".
    ID = "cookie_task_flow_cookie_task_flow_addEmpNoCookiePageDef".
    Package = "TR.com.Signum.roketsan.pageDefs" SkipValidation = "true" > "
    < Settings / >
    < executables / >
    < links >
    < methodAction id = "addEmpNoCookie" RequiresUpdateModel = "true".
    Action = 'invokeMethod' MethodName = "addEmpNoCookie."
    IsViewObjectMethod = 'false' DataControl = "CookieBean."
    InstanceName = "CookieBean.dataProvider" > "
    < NamedData NDName = 'cookieValue.
    NDValue = "#{webCenterProfile [securityContext.userName] .employeeNumber} '"
    NDType = "java.lang.String" NDOption = "3" / > "
    < / methodAction >
    < / links >
    < / pageDefinition >

    Hello

    It contains only the activity of the method, and if so, why is he launched with the help of a new transaction? Note that navigation always is an ID of the view and the workflow will not run if there is no view to navigate to. So either the calling flow calls this workflow and workflow returns immediately after execution of the activity of default method then it refers to a display of flow tasks calling or you add a view. Note that cookies are set on the response of a call and it is not issued before coming to a view. For this reason, using a method call activity could be a bad approach to try this so a phase listener is better suited

    Frank

  • No Audio with call transfer to the CUE Script

    Hello

    I have a CUE script set up to dial several extensions and transfer them to a conference MeetMe hosted on the same 2911 router where the CUE ISM is installed. The call flow is:

    CUCM 2911 <---> <--->CUE

    with calls to the script being initiated by phones registered with the CUCM.

    Dial the pilot script, calls to the specified extensions are undertaken and transferred to the DN MeetMe (7070) successfully, but there is no sound on calls. The script seems to complete successfully as well, because all of the extensions as well as the original one is transferred to the DN MeetMe. The command "display the compact active voice call" shows all participants connected to 7070 and the output of 'see the ephone-dn conf' displays the number of active sessions to 7070

    Direct calls to the DN MeetMe (without going through the script) work very well.

    Logging in the atrace.log seems to show that reactivate the QUEUE calls will fail. I tested with all three SIP call transfer methods CUE without result.

    CUE and 2911 configs as well as the atrace.log CUE file is attached.

    Any ideas would be very appreciated.

    Hi Miroslav,

    You can check the method of transfer in CUE - please define Bye-also and without h450 service of voip telephony services in the CME.

    HTH,

    Alex

  • Understand the flow of appeal through VCS c/e

    Hey Geeks,

    I write this to understand "how to work things. Here's the design.

    I have a VCSC configured with the name of the domain example.com SIP (we have internal DNS server to resolve)

    I have a VCSe configured with the name of the cisco.com SIP domain (we have external DNS server to make globally routable)

    I create a link bw VCSC and VCSe course.

    How the call will flow between the Ep A and B Ep

    Scenario a.

    My Ep A is [email protected] / * / dials Ep B Jabber client (user) recorded on VCSe [email protected] / * /

    How call flows; I understand that the flow of the beginning IE Ex 90 will send a register message to VCSC etc etc.

    Scenario B:

    If Ep A [email protected] / * / call a 3rd party (distinct) = (inter appeal cases) End point ie [email protected] / * /

    How runs the call.

    Please excuse me for asking layman explanation:

    Thanks in advance

    Vikram

    Hi Vikram,

    First thing to note is in most of the customer scenario prefer same sip domain on highways and control so that they can avoid transform and simplify the numbering plan.

    Happens to your scenario.

    Scenario a.

    EP A VCS-contrl<--traversal-->VCS - Ex <--SIP--><--sip-->Ep B (points of termination assumptions made using the SIP protocol)

    EP began with the sending of a guest for VCS-cntrol SIP message, which gets transmitted to VCS - by VCS-cntrl exp, hope you're installation rules research properly on vcs control.

    VCS - exp send this appeal to Ep B and call connect. You can google for SIP call flow, so nothing different happens in this case.

    Scenario B

    EP A VCS-cntrl<--traversal-->VCS - Exp<-->public cloud<--SIP--><-->Ep B (dell.com)

    A DNS zone, which uses the dns configured on exp do query srv records to the external field like "dell.com" is now required to apply for external part VCS - exp

    in this scenario again Ep began with the sending of a message to the VCS-cntrl invitation gets sent to the SCV - exp rules-based research. VCS - exp begins looking for new address based on the rule of the research and since it does not find the URI ending with exp and "dell.com" starts to send query record srv for the domain "dell.com". DNS configured on exp in the SRV company A sends question and get an answer for this domain with the company B VCS - exp - ip address, and then vcs - exp in company A starting configuration of the call to the remote ip address.

    now in the present, you can have several scenarios and I recommend you consult the guide deployment for VCS-control and traversal solution VCS - exp.

    Rgds,

    Alok

Maybe you are looking for

  • iPhone battery percentage 6 iOS 9.3.1 showing incorrect

    I have an iPhone with iOS 9.3.1 6.  In recent weeks, I noticed the battery percentage does not display correctly.  It shows 100% for hours after disconnecting the charger.  He dies when the percentage of battery indicates 25 to 28%.  How can this be

  • Satellite B3204-L35W-Click2 screen touch/HID USB driver is missing.

    I really just need to know where I can re - download the driver, because it seems that it just does not exist on my computer more.If I go into 'Tablet PC Settings', which I can only do via a shortcut on the advice of one of the guides, I found, I cre

  • How to select a specific channel in another file?

    Hello I have several tdv-files (~ 100) I want to analyze for a specific channel. All files have the same structure with the same channels. With tiara, I can load all the files at once to see the files in my dataportal. What I want to do is to select

  • Accuracy of the Variables window

    Hello I thought to remember that it is possible to define the number of digits displayed in the variable view. At the present time, see double with five decimal places... where can I change this setting to display numbers 'everything '? Or is this fe

  • Computer hangs at startup. Error quota.dll

    I'm not very technically advanced with computers.  Both my desktop computer (which I use rarely) and my laptop (which I use all the time) suddenly started to freeze at startup.   I can use it in SafeMode only. I get the following error messages: 1) L