Cisco Unified Call Studio 8.5

I need to write an app using
Cisco Unified Call Studiom this app will run every 30 secs do some DB query and make outbound calls, I see that the START of Call element is the only thing I have to Start a call, my application will not get any call, any ideas? thanks.
[email protected]
thanks.

Studio 8.0 installs on XP and Vista. Studio 8.5 installs on Windows 7, Vista, XP.
There are no real performance issues that Cisco could object to if the above operating systems were "guest O/S" on a VMware ESX server - it's not the same situation as the CVP servers, where they do have firm guidelines.
Studio works in this environment and I have customers who run Studio on a "guest O/S" - Cisco did not say specifically that this would be an issue.
But I don't have any documents that confirm that this is OK.
Regards,
Geoff

Similar Messages

Cisco Unified Call Studio 8.0 and above with VMWare image?

There is no official Cisco document which says "CallStudio" can run on VMware environment. However when the licensing was based on SystemID till 7.0 release, we have seen Cisco installing them in couple of customer "demo" location under VMware in region. Version 8.0 and above, the licensing mechanism for Call Studio uses a new FlexLM-based [IPlocking] license. None of the official documents like release notes /installation and upgrade guide / hardware configuration guide doesn't specify support of VM Ware support for Call Studio. Would installing Call Studio in VM Ware image would work above 8.0 with the new licensing in place and is it officially supported by Cisco?
Thanks!
-Sethu

Studio 8.0 installs on XP and Vista. Studio 8.5 installs on Windows 7, Vista, XP.
There are no real performance issues that Cisco could object to if the above operating systems were "guest O/S" on a VMware ESX server - it's not the same situation as the CVP servers, where they do have firm guidelines.
Studio works in this environment and I have customers who run Studio on a "guest O/S" - Cisco did not say specifically that this would be an issue.
But I don't have any documents that confirm that this is OK.
Regards,
Geoff

Cisco Unified Call Center Express on Wallboard

Hpw can I output 1 line of data from the Call Center Express software(specifically the Open Call datat) onto 2 wallboards in the call center?

Yes it is supported. Check this out http://docwiki.cisco.com/wiki/Unified_Communications_Virtualization_Sizing_Guidelines#Sizing_Examples
GP.
Pls rate helpful posts by clicking on stars below the post !!

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.

DOCUMENTATION - Cisco Unified Workforce Optimization - Call Recording

Hello,
I'm trying to sell the Cisco Unified Workforce Optimization - Call Recording option in a customer, but he is asking me to send him some documentation about capacities and functionalities. I'm not able to find this kind of information of the website. Can anybody help me? Thanks!

Hi,
Please check the following link. hope this helps!!!
http://www.cisco.com/en/US/products/ps8293/products_data_sheets_list.html
Thanks,
Dass
Please rate useful posts

Cisco Unified Incoming calls explanation needed

Hi,
This may be a vague question, if so, let me know what other info is needed. Basically, I was wondering how incoming calls were handled via Cisco Call Manager and Unity. We are using Unified Call Manager version 7.1.5. Most of our branches utilize Unity 8.0.3 System Call Handlers. So, when a caller calls into the branch, what exactly happens? I would think that the call comes into the router, gets redirected to the Unity server to play the System Greeting (we record our own), waits for an action from the caller, then redirects the call accordingly. Recently, one of the other guys tried tracing the call through the router. He saw the call come in, redirect out, come back in, then redirect out again. This, to us, seems like two too many redirects, unless we are misunderstanding how the call is supposed to be routed. Can anybody offer some insight? If more information or clarification is needed, I'll be happy to add what I can (within the limitations of our security policies). Thanks in advance!
Gary

Usual call flow is
PSTN - ISR - CUCM - Unity
In CUCM you have something that matches what a user dialed from PSTN and this sends calls to Unity.
The rest of the call handling completely depends on what you have configured on Unity/CUCM.
HTH
java
If this helps, please rate
www.cisco.com/go/pdihelpdesk

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.

Cisco Unified Contact Center Express 8.5 and Call Manager 8.5 on single ESXi

Does Cisco support 2 virtual machines (on ESXi), one for Cisco Unified Contact Center Express 8.5 and one for Call Manager 8.5?
I have blade ucs b200.
Thanks a lot, Svetomir

Yes it is supported. Check this out http://docwiki.cisco.com/wiki/Unified_Communications_Virtualization_Sizing_Guidelines#Sizing_Examples
GP.
Pls rate helpful posts by clicking on stars below the post !!

Is there any option on Cisco Unified CM Administration 9.1.1 to show caller ID number when using MOBILITY option?

Dear All,
I am using Cisco Unified CM Administration 9.1.1 and MOBILITY works fine.
However when somebody calls from external phone number into internal user number and MOBILITY option is ON and obviously, forwards the call for the user's cell.....in this case user can see only main company number, but not a caller number.
So, is there any option to change it?
Thank you

Go to the GW and run some debugs to find out exactly what you're sending, chances are, your telco is overwriting the original called number as it's nor part of your DID range.
HTH
java
if this helps, please rate
www.cisco.com/go/pdihelpdesk

Ask the Expert: Upgrading Cisco Unified Communications Manager (CUCM) to Version 9.1 (Drive to 9)

Welcome to the Cisco Support Community Ask the Expert conversation. Learn from experts Vijay Rao and Amit Singh about simplified upgrade process and focused support from Cisco to migrate to version 9.1.
This is a continuation of the live Webcast
Drive to 9 is a comprehensive and holistic program designed to help you upgrade the current Cisco® Unified Communications Manager installed base to version 9.1 or higher. This upgrade will enable customers to have next-generation collaboration experiences.
During the live event, Cisco subject matter experts Vijay Rao and Amit Singh focussed on the simplified upgrade process and focused support from Cisco to migrate to version 9.1. They also talked about the changes made to the licensing model of User Connect Licensing and Cisco Unified Workspace Licensing.
Vijay Rao is a Network Consulting Engineer and is currently a unified communications (UC) consultant for Bank of America. He has been providing consulting assistance to the bank for the past 6 years. He helps design complex UC networks for large enterprise customers. He was previously part of Cisco IT in the Asia Pacific, Japan, and China (APJC) region and was instrumental in designing and implementing the Bangalore campus. He has been working with Cisco for 9 years and has 12 years of UC experience. He has a Cisco CCVP® certification.
Amit Singh is a customer support engineer at the Cisco Technical Assistance Center in Bangalore, India. He has 7 years of experience in his areas of expertise: wireless, Cisco Unified Communications Manager, multiservices, Cisco Unity®, and Cisco Unified Contact Center Express. He has been involved in various escalation requests from India, Singapore, and Australia and is currently working as a technical lead for the Voice team in Bangalore, India. He is a computer science graduate.
Remember to use the rating system to let Vijay and Amit know if you have received an adequate response.
Vijay and Amit might not be able to answer each question due to the volume expected during this event. Remember that you can continue the conversation on the Collaboration, Voice and Video sub-community forum shortly after the event. This event lasts through July 19, 2013. Visit this forum often to view responses to your questions and the questions of other community members.
Webcast related links:
Webcast Video
FAQ from the live webcast
Slides from the live webcast

Hello Robert,
Apologies for a delayed response, some days get very hectic.
In CallManager, we only define the SRST reference, and CUCM version and SRST version are independent of each other.
The only thing, which is related and will change with CUCM upgrade is Phone F/w version.
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/compat/ccmcompmatr1.pdf
You may just want to check your, phone f/w compatibility with the SRST version running on your ISR G1 Gateways:
http://www.cisco.com/en/US/products/sw/voicesw/ps2169/products_device_support_tables_list.html
For Example: SRST version 7.1
http://www.cisco.com/en/US/prod/collateral/voicesw/ps6788/vcallcon/ps2169/data_sheet_c78-520521.html
You may want to do some lab testing with CUCM 9.1 and an SRST supported f/w on your phones.
If you decide to run the old Phone/F/w to support the SRST version, you may not be able to take advantage of new features.
Also, you can try and upgrade your phones(Wih CUCM 9.1) and test them with your SRST version.
It should work fine, but from a troubleshooting perspective, TAC may request you to come into a Cisco Supported combination.
Please, let me know if this clarifies your doubt or we can have a quick phone call.
Regards
Amit Singh

Ask the Cisco VIP: Troubleshooting SIP in Cisco Unified communications

Troubleshooting SIP in Cisco Unified communications deployments with Cisco VIP Ayodeji Okanlawon
This is a Q&A Ask the Expert Session continuation from the Live Webcast
Ask your questions on Session Initiation Protocol (SIP) and how it is redefining our UC world.The Session Initiation Protocol (SIP) is a signaling communications protocol, widely used for controlling multimedia communication sessions such as voice and video calls over Internet Protocol (IP) networks.
Featured Expert
Ayodeji Okanlawon, a Cisco Designated VIP, is the Lead Consultant Engineer for Global Solutions Design and Engineering at Verizon Business. In his past, he has worked at Intact IS, NCS Global, and Schlumberger Information Solutions. His experience includes development of design and deployment of large scale IP telephony projects on Cisco Call Manager platforms, Cisco Voice gateways, Cisco Jabber cloud and on premise solution. His expertise includes SIP solutions, CUBE design and Deployment, Troubleshooting: Voice gateways, CUCM, Unity connection, CUPS. Deji has been awarded the Cisco Designated VIP in 2013 and 2014. Deji holds a Bachelor of Science (BS), Electrical and Electronics Engineering, Second Class Upper from Obafemi Awolowo University.
According to Deji, “If you want to advance your career, if you’re serious about your skill sets, you’ve got to be in the forums.” (Read the Interview >>)
We look forward to your participation. This event is open to all, including partners.
* * Remember to use the rating system to let Deji know if you have received an adequate response. * *
Deji might not be able to answer each question due to the high volume expected during this event. This event runs January 13 through January 23, 2015. Visit this forum often to view responses to your questions and the questions of other community members.

Derrick,
RFC 3261defines ways to provide increased security for a SIP session.
The following describes areas in SIP that provides security for the protocol
1. Authenticating users.
We need to authenticate a user to ensure that the sender of the message is who he claims to be.
To achieve this SIP uses digest authentication between a UAC, proxy and a UAS. This provides the most basic level of authentication challenge between a client, proxy and a server.
2. Secure SIP signalling
The next area we can secure is SIP signalling itself. For this we use SSL/TLS. This is similar to using https in web browsers. With TLS before our any signalling is exchange X.509 certificates are used create a secure TLS channel. All our SIP messages are then transported within the secure channel.
NB: The digest authentication mentioned above for authenticating a user agent is just authentication. The messages are not protected from reading or modification hence it is recommended that these messages are carried inside a secure TLS channel for better security.
3. Privacy and Identification
Additional security features in SIP provides means where any user can choose to either reveal or conceal his identity.
4.Secure RTP
SIP also provides the ability to secure the media channel. It is not enough to secure signalling while anyone can listen to the media. RFC3830 discusses how the encryption should be done.
5. S/MIME
S/MIME encapsulation is used to protect sip headers making it impossible for any one in between the sender and receiver to modify the sip headers
Regards

Ask the Expert: Cisco Unified Contact Center Express (UCCX) Version 10.0 - Upgrade, Migration, and New Features Overview

With Abhiram Kramadhati
Welcome to the Cisco Support Community Ask the Expert conversation. This is an opportunity to learn and ask questions about the upgrade, migration methods, and new features of the latest released Version 10.0 of Cisco Unified Contact Center Express (UCCX) with Cisco expert Abhiram Kramadhati.
Abhiram will address the following on the latest release of Cisco UCCX Version 10.0:
Installation
Upgrade from previous versions - both Linux and Windows
Migration from MCS to Cisco UCS environment - Different methods and best practices
New features - Overview and limitations
This discussion will center on install and upgrade best practices, changes in hardware support, and migration methods from MCS to Cisco UCS. He can also briefly discuss the new features introduced in 10.0. The discussion focuses the latest versions, but queries about general Cisco UCCX topics can be addressed too if time allows.
Abhiram Kramadhati is an engineer with the Contact Center Backbone group. He has been working with Cisco UCCX since he joined Cisco. During two years at Cisco, he has built his expertise around Cisco UCCX telephony applications, Java Telephony API (JTAPI) integration, Cisco UCCX system behavior, LDAP components, and Cisco UCCX as IP interactive voice response in Unified Contact Center Enterprise (UCCE) environments. He also works on other technologies, including Unified Communications Manager and UCCE. He has been involved in many technical escalations in the Asia Pacific region. Abhiram also holds a CCIE in voice (40065).
For more details about this topic, refer to the recently published Tech-Talk Video and Blog.
Remember to use the rating system to let Abhiram know if you have received an adequate response.
Abhiram might not be able to answer each question due to the volume expected during this event. Remember that you can continue the conversation in the Voice, Video, and Collaboration community, sub-community, Contact Center discussion forum shortly after the event. This event lasts through January 31, 2014. Visit this forum often to view responses to your questions and the questions of other community members.

Hi Anurag,
Thanks for your questions.
1:Is there change in DB architecture as CUIC is the only option as compared to previous linux version UCCX ?
I assume this is from the tables regarding historical data. The database schema essentially remains the same since UCCX 9.0 had CUIC too and we had a seperate DB Space for CUIC and we still continue with that. The traditional historical tables remain and the replication process remains the same too.
2:Is there any version change for Linux OS used as VOS,
The Linux version is Red Hat Linux 5. To be precise:
[root@uccx10pub /]# cat etc/redhat-release
Red Hat Enterprise Linux Server release 5.7 (Tikanga)
3:Is there any API architecture change in UCCX 10 from previous releases ?
I can answer this more of an overview. The only enhancement in the API side is the introduction of REST API step in the script editor. You can now make REST calls from the script and this ofcourse opens up a whole new world of possiblities.
4:Since from UCCX 10 , we can only use either CAD or Finnesse at one time, whats the impact of changing this after some time in production, let say , i used CAD for 2 months and then i decided to move to Finesse, whats the impact ? or is it a smooth change as switching CUIC and HRC in previoius release ?
For the scenario you mentioned, there is absolutely no problem. The point to note is that the Finesse services are activated/deactivated but the CAD desktop services are ALWAYS running. The only condition to keep in mind is that you can use ONLY ONE type of agent desktop at any time. Also if Finesse is not used and CAD operations are used extensively, it is advisable to shutdown the Finesse service.
5:Is 3rd Party UCS hardware supported by UCCX 10 instead of using Cisco manufactured UCS , can i use HP hardware for Virtualisation ?
Yes, it can be used. This is something called as "Third party specs based specification". The most important things seen for compatibility are:
Inter CPU Model
It it is on thVMWare Hardware Compatibilty List
You can get more information about this on the "Can I use this server?" section of UC Virtualized Hardware page:
http://docwiki.cisco.com/wiki/UC_Virtualization_Supported_Hardware#.22Can_I_use_this_server.3F.22
6:Is Host name change supported?
Yes, the hostname change is supported. The prcocedure is documented in the UCCX 10.0 Administration Guide:
http://www.cisco.com/en/US/docs/voice_ip_comm/cust_contact/contact_center/crs/express_10_0/configuration/guide/UCCX_BK_W1AF9DDD_00_uccx-admin-guide-10.0.pdf (Pg 168)
Cheers,
Abhiram Kramadhati

CVP call studio and default audio

Not sure if this should get posted here or on the development forum but it's getting posted here anyways...
We have 4 combo boxes in our CVP deployment so we have 4 media servers. I'm trying to figure out what to configure for the Default Audio Path URI in Call Studio when creating a project .It appears that I have to specify either a single media server to pull the audio from or pull it from flash.
I know our gateways will pull down the wav files from the media servers but if I specify flash in the studio application, do I need to load the wav file directly onto flash on the router or can it access the media servers and pull it down? We run most of our ICM scripts off of microapps, not custom vxml apps. Most of the documentation and the partner class I took doesn't look at multi server deployments and combo boxes like as most standard deployments are like this now. I don'ts ee anything in the Cisco docs for CVP that specify this information. Any help would be appreciated.
We are using SIP and CSS if that makes a difference.
ICM 7.5.9
CVP 7.0.2
UCM 7.1.3
thanks
Paul

The normal way is to use the VIP you built on your CSS. Let's say this is a.b.c.d and it manages the IP address of the media servers as a "service", providing load balancing and resilience.
For each application in Studio you should make a matching default audio path and put all the media files under that. Makes it easy - even if there is repetition of system files (1.wav, 2.wav etc). You can control the lifetime (expiration) on IIS more easily.
Let's assume your application is called "foobar" and your locale is "en-us".
So set the path to
http://a.b.c.d/en-us/app/foobar/
Don't forget the trailing slash. As you go through the Studio program, you just need to specify an audio item as "mywavefile.wav" and that makes it really easy to look at the Prompt Dictionary, as they have simple wav file names and no path information.
Others may have different views, of course. I'm prepared to vigorously defend mine.
On IIS you will have wwwroot\en-us\app\foobar with a bunch of files.
Regards,
Geoff

Ask the Expert: Deployment and Troubleshooting Cisco Unified Contact Center Express (UCCX) Deployments

With Anirudh Ramachandran and Abhiram Kramadhati
Welcome to the Cisco Support Community Ask the Expert conversation. This is an opportunity to learn and ask questions about the latest advancements in Cisco UCCX (such as the integration of Cisco Social Miner to provide agent chat and better reporting using the Cisco Unified Intelligence Center), as well as the existing features of Historical Reporting, custom reporting using the historical database, Agent Email services, JTAPI integration with CUCM, and the HA over WAN cluster mechanism.
Anirudh Ramachandran is a customer support engineer at the Cisco Backbone Technical Assistance Center in Bangalore, India. Working in the Asia-Pacific time zone for the last two years, he focuses on Cisco Unified Contact Center Express issues and specializes in Linux, JTAPI/CTI integration, and UCCX system and database issues. He holds the CCNP Voice and UCCX Specialist certifications, and is also a Red Hat Certified Engineer. Anirudh writes tools and automates bug workarounds for UCCX in addition to working on TAC service requests, and currently has authored and co-authored seven such tools. Anirudh graduated from the National Institute of Technology Karnataka with a Bachelor of Technology in Computer Engineering.
Abhiram Kramadhati is an engineer with the Contact Center Backbone team in the Asia Pacific timezone. He has been working with UCCX since he started with Cisco 2 years ago. During his time at Cisco, he has built his expertise around UCCX Telephony applications, JTAPI integration, UCCX system behaviour, LDAP components and also UCCX as IPIVR in UCCE environments. He also works on other technologies including Unified Communications Manager and UCCE. He has been involved in many technical escalations in the region. Abhiram is a Telecommunications engineer from Bangalore, India.
Remember to use the rating system to let Anirudh and Abhiram know if you have received an adequate response.
They might not be able to answer each question due to the volume expected during this event. Remember that you can continue the conversation on the Collaboration, Voice and Video Contact Center subcommunity discussion forum shortly after the event. This event lasts through May 3, 2013. Visit this forum often to view responses to your questions and the questions of other Cisco Support Community members.

Hi Anthony,
Thanks for the question.
This is an interesting requirement, since the UCCX trigger's configuration is translated only to the Call Forward Busy External setting on the CUCM.
Trigger creation:
144768: Apr 22 21:54:23.789 IST %MADM-ADM_CFG-7-UNK:JTAPITriggerServlet.updateNewTrigger() - Creating a new Trigger :1234
144876: Apr 22 21:54:23.884 IST %MADM-ADM_CFG-7-UNK:JTAPITriggerServlet routePoint = 1234
144877: Apr 22 21:54:23.884 IST %MADM-ADM_CFG-7-UNK:JTAPITriggerServlet description = testt
144878: Apr 22 21:54:23.884 IST %MADM-ADM_CFG-7-UNK:JTAPITriggerServlet deviceName = testt
144879: Apr 22 21:54:23.884 IST %MADM-ADM_CFG-7-UNK:JTAPITriggerServlet devicePool = {1B1B9EB6-7803-11D3-BDF0-00108302EAD1}
144880: Apr 22 21:54:23.884 IST %MADM-ADM_CFG-7-UNK:JTAPITriggerServlet devicePoolName = Default
144881: Apr 22 21:54:23.884 IST %MADM-ADM_CFG-7-UNK:JTAPITriggerServlet callingSearchSpace =
144882: Apr 22 21:54:23.884 IST %MADM-ADM_CFG-7-UNK:JTAPITriggerServlet callingSearchSpaceName = None
144883: Apr 22 21:54:23.884 IST %MADM-ADM_CFG-7-UNK:JTAPITriggerServlet redirectCSS = default
144884: Apr 22 21:54:23.884 IST %MADM-ADM_CFG-7-UNK:JTAPITriggerServlet location = {29C5C1C4-8871-4D1E-8394-0B9181E8C54D}
144885: Apr 22 21:54:23.884 IST %MADM-ADM_CFG-7-UNK:JTAPITriggerServlet locationName = Hub_None
144886: Apr 22 21:54:23.884 IST %MADM-ADM_CFG-7-UNK:JTAPITriggerServlet partition =
144887: Apr 22 21:54:23.884 IST %MADM-ADM_CFG-7-UNK:JTAPITriggerServlet partitionName = None
144888: Apr 22 21:54:23.884 IST %MADM-ADM_CFG-7-UNK:JTAPITriggerServlet voiceMailProfile =
144889: Apr 22 21:54:23.884 IST %MADM-ADM_CFG-7-UNK:JTAPITriggerServlet voiceMailProfileName = None
144890: Apr 22 21:54:23.884 IST %MADM-ADM_CFG-7-UNK:JTAPITriggerServlet forwardBusyVM =
144891: Apr 22 21:54:23.884 IST %MADM-ADM_CFG-7-UNK:JTAPITriggerServlet forwardBusyDestination =
144892: Apr 22 21:54:23.884 IST %MADM-ADM_CFG-7-UNK:JTAPITriggerServlet forwardBusyCSS =
144893: Apr 22 21:54:23.884 IST %MADM-ADM_CFG-7-UNK:JTAPITriggerServlet forwardBusyCSSName = None
144953: Apr 22 21:54:23.913 IST %MADM-LIB_AXL-7-UNK:AXL-ExecutionCmd-569.CCMLineSOAPAdmin: try makeRequest() on AXL: 10.106.113.142, AXLUser: axl, AXLPassword: XXXXXX
144954: Apr 22 21:54:23.913 IST %MADM-LIB_AXL-7-UNK:CCMVersionSOAPAdmin.getAXLVersion():7.1
144955: Apr 22 21:54:23.913 IST %MADM-LIB_AXL-7-UNK:AXL-ExecutionCmd-569.CCMLineSOAPAdmin: makeRequest() - Start REQUEST ====================
144956: Apr 22 21:54:23.913 IST %MADM-LIB_AXL-7-UNK:POST /axl/ HTTP/1.1
Connection: keep-alive
Host: 10.106.113.142:8443
Authorization: Basic YXhsOmF4bA==
SOAPAction: "CUCM:DB ver=7.1"
Accept: text/*
Content-type: text/xml; charset="utf-8"
Cache-Control: no-cache
Pragma: no-cache
Content-length: 440
http://schemas.xmlsoap.org/soap/envelope/">MADM_5691234CRS Line descriptionCallPark
144957: Apr 22 21:54:23.913 IST %MADM-LIB_AXL-7-UNK:AXL-ExecutionCmd-569.CCMLineSOAPAdmin: makeRequest() - End REQUEST ==================
144958: Apr 22 21:54:23.914 IST %MADM-LIB_AXL-7-UNK:AXL-ExecutionCmd-569.CCMLineSOAPAdmin: getSocket: MADM_LIB_AXL_AXL_SOCKET_POOL-0-79[TLS_RSA_WITH_AES_128_CBC_SHA: Socket[addr=10.106.113.142,port=8443,localport=44913]]
144987: Apr 22 21:54:24.195 IST %MADM-LIB_AXL-7-UNK:AXL-ExecutionCmd-570.CCMCTIRoutePointSOAPAdmin: makeRequest() - Start REQUEST ====================
144988: Apr 22 21:54:24.195 IST %MADM-LIB_AXL-7-UNK:POST /axl/ HTTP/1.1
Connection: keep-alive
Host: 10.106.113.142:8443
Authorization: Basic YXhsOmF4bA==
SOAPAction: "CUCM:DB ver=7.1"
Accept: text/*
Content-type: text/xml; charset="utf-8"
Cache-Control: no-cache
Pragma: no-cache
Content-length: 839
http://schemas.xmlsoap.org/soap/envelope/">MADM_570testttesttCTI Route PointCTI Route PointCTI Route PointSCCPUserRing1000010000
144989: Apr 22 21:54:24.195 IST %MADM-LIB_AXL-7-UNK:AXL-ExecutionCmd-570.CCMCTIRoutePointSOAPAdmin: makeRequest() - End REQUEST ==================
145014: Apr 22 21:54:24.647 IST %MADM-ADM_CFG-7-UNK:JTAPITriggerUtil.createRPAndLineOnCCM() - CTI RP created.
145015: Apr 22 21:54:24.647 IST %MADM-ADM_CFG-7-UNK:JTAPITriggerUtil.createRPAndLineOnCCM() - Created a Route Point = 1234
As you would aready know, the UCCX will send an AXL request (within the SOAP envelope) to the CUCM to create this RP. Looking at the existing code, there does not seem to be a method where we are differentiating between CFB_internal and CFB_external while sending this request.
We have taken this as an enhancement request and also spoken to the business unit about the same. It has been added to the roadmap, we will reach out to you offline to understand the business case so that the process can be expedited if needed.
Keep the questions coming
Cheers,
Abhiram Kramadhati

Cisco Unified Call Studio 8.5

Similar Messages

Maybe you are looking for