Cisco CME v10 8831 conference Phones issues

We have a bug / issue with conference phones and Cisco Communications Manager Express (CME) v10.

The current Cisco conference phone is the 8831, the previous version was the 7937G. The 7937G is end of sale as of 31st March 2014 which means that the only available conference phone is the 8831 (unless you purchase refurbished / from distribution but this cannot be guaranteed due to stock levels). So realistically the option is the 8831.

The issue however, is that the 8831 is not supported by CME version 10, and CMEv10 is the current version, therefore companies upgrading to version 10 of CME had better be using older conference phones.

There is currently (28th March 2014) no work around for this, if you are running CMEv10 you either need to purchase older 7937G conference phones or wait for CME version 11 to be released, currently targeted for around July 2014.

Unified Attendant Console Receptionists at Multliple Sites

Something I was looking into as part of some recent work I was doing is the ability to have receptionists at multiple sites who can operate together as one team, effectively pooling their resources together so that call volume to a single site doesn't really matter. If there are too many calls coming into one site then receptionists at other sites will see this and can answer the calls as if they were local to that site.

This feature in the Cisco Unified Attendant Console is called remote attendant. And this it is further described in the design guide, see the below link.

Just one thing to be aware of though when considering the remote attendant feature is that there needs to be the ability to transcode across the WAN, I.E. you'll need an ISR at each site with receptionists and some DSP resources in the ISR.

References:
Unified Attendant Consoles Design Guides
http://www.cisco.com/en/US/products/ps7282/products_implementation_design_guides_list.html

Cisco WebEx Meetings Server: Scheduled Audio Meetings

Another quick little note.

Conferencing is a great feature and used pretty much by any company using a UC&C solution. With CUCM you get the ability to do Adhoc and Meet me conferences for up to 128 parties, however what you don't get and what a lot of people ask for, is scheduled audio conferencing, the ability do create invites from Outlook, send it to your parties and then have a conference at that specific time.

Scheduled audio conferencing would be best done using WebEx, it integrates brilliantly and you get all the features of full WebEx meetings, one of which is the scheduling. What if you don't want the full WebEx suite though? If you only have CUWL standard and just want to be able to Schedule audio conferences? Well for that you need WebEx meetings server.

References:
WebEx Meetings Server ordering Guide:
http://www.cisco.com/en/US/partner/prod/collateral/ps10352/ps10362/ps12732/ordering_guide_c07-719906.html

CCX Encrypted Calls

Just a quick note here.

A requirement came up recently to encrypt the calls stored recording on the CCX server. This is currently not something which is supported by Cisco, however there is a work around if you are brave:

https://supportforums.cisco.com/thread/2091243

CUCM Business Edition, More Scalable than you might think...

So an interesting question came up recently which I've always assumed the answer to be true, but I've never checked it, until now.

CUCM Business Edition 6000 (CUCMBE6K) is a low (ish) end UC&C solution from Cisco. The call processing element scales up to 1000 users and 2500 devices. But what do you do if you purchased and run CUCMBE6K, and need to expand to over 1000 users?

Well, Business Edition 6K is capped at 1000 users, and as I understand there is nothing you can do about that, but the licenses you have are essentially just CUWL standard, and I assume you are running this on a virtualised server? Even better if you purchased one of the BE6K hardware bundles. So you can migrate the CUWL business edition licenses to CUWL standard (because they are the same), which is a process with the GLO team at Cisco, and this entitles you to download the full version of CUCM. You can therefore just install the full blown CUCM, add your new CUWL standard licenses and scale up to your hearts delight, No more 1000 user cap.

Granted there will be some work here, you'll need to be reasonably savy with CUCM, or just utilise your integrator who sold you the solution in the first place. The other thing is if you purchased a CUCMBE6K hardware bundle you'll need to buy a copy of VMware vSphere (the version with the bundle is cut down for BE6K). But these things are still minimal compared to a whole new system.

I think this is a real unsung scalability feature, its such a minimal cost for such a flexible, scalable solution.

Cisco Voice Gateways with MS Lync

Cisco voice gateways can be used in a MS Lync environment. If local breakout to the PSTN is required then some kind of voice gateway is needed.

Cisco and Microsoft have an established history of cooperating across technologies to provide customers with innovative business solutions. From Cisco's standpoint we support voice gateways connected to anything that complies with the standards, including MS Lync. From MS's standpoint, I believe they do certification tests with vendors' equipment before they say they support it. 

As far as I know, deploying Cisco ISR G2 in integration with Microsoft Lync has no issues except for a couple of caveat which is Cisco does not support Microsoft proprietary codecs (ie RTAudio and RTVideo).

The below document is a good reference for an ISR voice gateway terminating a SIP trunk in a Lync enironment:

http://www.cisco.com/en/US/solutions/collateral/ns340/ns414/ns728/ns833/1197443.pdf

Just an update to the above post, I've since found that there are companies, such as Sonus, who build Lync SBAs which are affectively SRST boxes which also have the voice gateway features, you specify the interfaces and they can build the device, I believe it's just a server with the appropriate cards, but this is certainly something worth bearing in mind.

Cisco Prime and IP Phones

Within Cisco Prime Infrastructure IP phones are not counted with regards to licensing. This is because Cisco Prime Infrastructure does not manage IP phones. Here is a link which details the devices which can be managed with Prime Infrastructure: http://www.cisco.com/en/US/products/ps12239/products_device_support_tables_list.html.

I am told there is a limited amount of management which can be done to IP phones within Prime infrastructure if you know what you are doing but for a comprehensive UC management solution see below:

In order to manage IP phones, you will need to have Cisco Prime Collaboration, which does count the IP phones for licenses. For more information regarding Prime Collaboration licensing, please refer to http://www.cisco.com/en/US/prod/collateral/netmgtsw/ps6491/ps12363/guide_c07-728239.html.

ISDN Q.931 = QSIG

So here's something I didn't know... ISDN Q.931 signalling is often abbreviated as QSIG.

Sufficise to say if you need an interface on an ISRG2 to connect to an ISDN circuit or another device using Q.931 signalling then this is the kind of card you need: VWIC3-1MFT-T1/E1
 There is no information on the datasheet that I can see but here is a support forums reference:

https://supportforums.cisco.com/message/3778089#3778089

A quick post on SIP trunking

SIP trunks are not the circuits themselves. a SIP trunk is the voice channels being delivered over an IP circuit rather than something else like ISDN.

The physical media the SIP trunk is delivered on can vary and is normally specified by the SP, it could be Ethernet, ADSL, a leased line of some kind, or perhaps delivered into an MPLS cloud and then onwards from there.

SIP trunks need to be terminated by a SBC for example Cisco CUBE.

That'll do for now. I'll add later if I find out more.

Here's a link to my previous post:
http://twhittle1.blogspot.co.uk/2013/04/cisco-cube-and-sip-trunks.html?view=mosaic

Cisco UC on UCS

So the way things are going Cisco is looking to have all UC applications on UCS servers rather than the existing MCS boxes. This is not new and makes complete sense, it's just a little more complex to size.

I've previously written about BE6K (which is fantastic!):
http://twhittle1.blogspot.co.uk/2013/05/cucmbe6k-ucs-bundle.html

However what do you do if you need something bigger? The answer is Cisco TRC - Trusted Reference Configurations.

Here is the link which details what the TRCs actually are, including a Bill of Materials (BoM):
http://docwiki.cisco.com/wiki/UC_Virtualization_Supported_Hardware#UC_on_UCS_Tested_Reference_Configurations
http://docwiki.cisco.com/wiki/UC_Virtualization_Supported_Hardware

And how do you know if your application is supported on these TRCs? Well use these links:
http://docwiki.cisco.com/wiki/Unified_Communications_in_a_Virtualized_Environment
http://docwiki.cisco.com/wiki/Unified_Communications_Virtualization_Supported_Applications
They have the same information on them, I've just included them both for reference.

And how many applications will be supported on your TRC hardware? Well this information can be found here:
http://docwiki.cisco.com/wiki/Unified_Communications_Virtualization_Downloads_(including_OVA/OVF_Templates)

For example, the OVA Capacity for CUCM 7,500 users requires (2) vCPU (vCPU is equal to core on the processor) and it will use 12gig of RAM, (6 being used and 6 as a reserve), and 110Gb of hard drive space.

Looking at TRC 1 it includes (2) quad core processor, 64Gb RAM, and more than 2Tb of hard Drive.

Taking the given System requirements for CUCM 9.0 for 7500 users, that leaves us 6Core, 52Gb of RAM, and at least 2Tb of hard drive on the UCS Server, this could then be used for other UC Applications.

Brilliant!

Other Links:
Datasheet:
http://www.cisco.com/en/US/prod/collateral/voicesw/ps6790/ps5748/ps378/solution_overview_c22-597556.html

CUCMBE6K UCS Bundle

So, CUCM business edition 6000 is great! It's reasonably simple and easy to quote, so any UC opportunities for sub 1000 should always be BE6K.

There are 2 main elements, the licensing and the hardware, I'll ignore the phone here because you can add them on later depending on the customer environment.

BE6K Licensing Top level Part - R-CBE6K-K9
BE6K UCS Bundle Top level part - BE6K-ST-BDL-K9=

Within the licensing you select the licensing type. UCL or UWL. UWL is usually often preferred due to the included features for the price, see my previous post:
http://twhittle1.blogspot.co.uk/2013/04/webex-audio-entitlement-with-cuwl-pro.html?view=mosaic

One note on the licensing is that the version of UWL which can be selected is Business edition UWL. Despite having a different name it is functionally the same as Standard UWL, and can be upgraded in the same way as standard to Pro if needed. Professional cannot be purchased from the BE6K top level part.

With the hardware you only need to select the power cable (although consider an extra PSU), the bundle includes all the hardware you need, and specifically it includes all the hardware required to run up to 5 UC applications, CUCM, Unity, CUCM in presence mode (for Jabber), Unified Attendant console (on top of an instance of windows server) and CCX. Attendant console needs to be purchased separately and so does CCX licensing but you get the idea, that's a lot of UC application in one place and one package.


here's a link to a later blog entry which I've written which talks more on it's scalability beyond the 1000 user limit:
http://twhittle1.blogspot.co.uk/2013/10/cucm-business-edition-more-scalable.html

Remote access to voicemail using Unity Connection

Remote access to voicemail can be provided in Unity Connection by the users dialling into the voicemail system, from anywhere internal or external, then entering the user extension and pin.

The following user guide shows this interaction from the users point of view:

http://www.cisco.com/en/US/docs/voice_ip_comm/connection/9x/user/guide/phone/b_9xcucugphone_chapter_010.html#task_4CDEBFCC4EE1426497B58E261FA57F7A

Databases in Cisco UC applications

An obscure question came up in a piece of work I was doing so I'm recording it here for future reference just in case it comes back. I was trying to find out what were the databases used by UC applications and what formats were they.

CUCM and Unity Connection both have relational databases (using the SQL language) however they are accessed via IBM Informix. CWMS uses oracle 11g and Attendant console uses MS SQL Server 2005 or 2008

E1 ISDN G703 Cisco VIC cards

So this is something which has always confused me, which interface to choose when connecting to a ISDN30 being used for voice and what the difference is. Here is the information I've picked up from researching this:

PRI ISDN is a service which operates on top of an E1 circuit.

VWIC3-1MFT-G703 vs VWIC3-1MFT-T1/E1
The difference here is essentially if you want framed or unframed E1.
Unframed (G.703) allows you to use the full bandwidth of the circuit (2048kbps), this mode is not often given and is more costly.
Framed E1 gives you 1984kbps bandwidth because timeslot 0 is reserved for signalling and sync

The E1 connector is either 2 pairs (RX and TX) on a single RJ48 interface or 2 x BNC coax.

Sources:
http://www.farsite.com/cable_standards/G.703_E1_ANSI_T1.403_T1.shtml

CUCM UCL License types

The license types available for UCL are the following:

Essential
Basic
Enhanced
Enhanced Plus
Public Space
Telepresence

The different licenses allow different things as shown on this page:
http://www.cisco.com/en/US/prod/collateral/voicesw/ps6882/ps9156/product_solution_overview0900aecd806cc7a4.html

However Just a couple of notes:
Public space licenses are exactly that, for use on public space devices such as conference phones and other public phones. There is no user profile attached to a public space phone, if you need an employee extension on the device you need the enhanced license.
Public Space phones are going away in UCL 9.0 and onwards, to be replaced with enhanced licenses.

Below is a good graphic which shows the licenses for each phone in UCL version 9.0 onwards

Meetingplace 8.5 Scheduling, Call recording

I've done a previous port about Cisco Meetingplace 8.5 Licensing which lists a couple of different types of servers, here's the link for reference:
http://twhittle1.blogspot.co.uk/2013/04/meetingplace-85-licensing.html

But I've got a number of extra bits of information I want to get down so here is a follow up.

A web scheduling server is used to schedule meetings / calls through outlook or through a web interface. If this functionality is not required and only phone scheduling is needed then the web scheduler server isn't needed. A gotcha here is recordings, despite being a scheduler server it is required for recording meetings. 

With regards to the recordings, the Application (A/V) Server can store up to 1000 hours of audio recording or 160 hours of video recordings, and is limited to 100 simultaneous meetings being recorded.  Also note that each recording will consume an audio port so this will effect overall system capacity.

Cisco Unified MeetingPlace audio and video meeting recordings are initially stored only on the Application Server. Shortly after each recorded meeting ends, the Replication Service copies the meeting recording from the Application Server to the Web Server, where the recording is converted and stored for user playback.

Every day at 2 a.m. (local server time), the system deletes all recordings on the Application Server that are older than 24 hours. To display the available disk space for recordings (/mpx-record directory) on the Application Server, sign in to the CLI and enter df -k.

By default, the Web Server stores all recordings for meetings held on the server on a local disk. You can change the storage configuration to copy these items to an external backup location (such as a shared network drive on a dedicated storage server, a network-attached storage device, or a storage area network).  If the customer will have a large number of recordings, or would like to keep recordings for an extended period of time, using external storage will be needed.

If you want to record calls then the Web Server is practically mandatory. Using CUCM to initiate the recordings is definitely not a recommended option.  First of all, CUCM can initiate call recording but it doesn't "do" the call recording.  You would still need to add a 3rd party recording application.  Also, recording in CUCM pre-9.0 is not done on-demand, it is done via admin config or CTI invocation (MP is not CTI), and would turn on recording for all phones.  This means that all calls would be recorded, so you would have "x" number of copies of that meeting where "x" is the number of meeting attendees.  

Also, regardless of whether or not you have a version of CUCM that supports on-demand recording, there wouldn't be a good way of differentiating the calls that are an MP meeting from just normal calls in the recording database, so it would be difficult to manage and retrieve the recordings.  All calls would just look like a phone call that "phone x" made at this date/time, so you'd have to know which exact calls are an MP meeting.

WebEx audio entitlement with CUWL Pro subscription

If you look at the list of included items within CUWL pro you'll notice that WebEx social and WebEx meetings in in there, the following table is UC Licensing 9.0:

<--- You can see here WebEx listed.














Ignore WebEx Social for the time being because this the old Cisco Quad (social networking for the enterprise) very very good but not the subject of this post.

When you think of WebEx you normally think of a collaborative online meeting environment, which is audio, video, chat, slide show and an interactive white board. But what do you get with CUWL Pro? The quick answer is most of it, the longer answer is below...

First of all there are 2 options for WebEx with CUWL, Hosted (WebEx Meetings) or on-premises (WebEx Meeting Server). The Hosted version gives you a subscription to WebEx Meetings for 1 year (which can be increased to 3 or 5 years). One big note here is you get 1 port for every 10 CUWL users. So if you are deploying 100 CUWL users you will get 10 ports. A port allows 1 user to participate in a WebEx meeting, this can either be hosting a meeting or a participant. Ports don't have a bearing on the number of meetings you can have, so for the 10 port example this could be 1 meeting of 10 or 5 meetings of 2 etc. This service must be activated within 90 days otherwise you risk loosing days of the service.

The Hosted version requires that you have a dedicated server, virtual or otherwise, to install the WebEx Meeting Server software on. You don't have the concept of ports but rather "Meeting Server users" of which you get 1 Meeting Server user per CUWL Pro User, allowing everyone to take part in meetings. This is obviously better but you will have to host and maintain your own WebEx Meeting Server.

One other very important note is the audio minutes. Within the WebEx Meetings (both hosted and on-premise ) functionality you get unlimited VoIP minutes, meaning that all the users can join a WebEx meeting, user their USB headset / computer speakers and participate in the conference. WebEx also has the to dial in on the phone for free however someone has to front the bill for this and Cisco sells you this service in the form of audio minutes. If you do not pay for WebEx audio minutes your participants will only be able to participate using the computer audio. If you want a freephone number and particpants to be able to dial in you must purchase audio minutes, which are sold in minutes per month, here is an example top level part: L-WBX-AUDIO-5K. Audio license subscriptions must run in line with the WebEx data subscription.

References:
Cisco UC Licensing version 9.0
http://www.cisco.com/en/US/prod/collateral/voicesw/ps6882/ps9156/product_solution_overview0900aecd806cc7a4.html

Cisco CUBE and SIP Trunks

SIP trunking is becoming more common place nowadays so here is a little post containing information and some specific references to CUBE - The Cisco Unified Border Element.

First off SIP trunks can be used for a number of different applications but a popular use, and the one I'm describing here, is to replace traditional connectivity to the PSTN. A SIP trunk can be provided by a ITSP (Internet Telephony Service Provider), the ITSP then provides the connection to the PSTN. The primary advantages are cost and scalability, a SIP trunk often connects over Ethernet (Gigabit or Fast Ethernet) and so has a much higher scalability for calls compared to traditional circuits.

The 3 basic components would be
1) The SIP Trunk from the ITSP
2) A Border Element which is essentially a gateway between the IP PBX and the ITSP
3) A PBX, IP or Otherwise

In a Cisco environment element 2 is a router with CUBE licensing on it, often just referred to as a CUBE. A CUBE router requires the relevant UC licenses, such as the UC package license for the ISRG2's, DSPs for translating between calls with different codecs, CUBE licenses and appropriate interface, most commonly Ethernet but check with the ITSP.

CUBE Licenses are cumulative and licensed based on a number of sessions. An example is:
FL-CUBE-25

CUBE is defined as an IPIPGW (i.e. H.323-SIP, SIP-SIP or H.323-H.323 dial-peer connections, using the "voice service voip > allow-connections" CLI). Gatekeeper is defined as the H.323 GK functionality (and "gatekeeper" CLI). Meaning the SIP trunk to Provider network requires a CUBE license.

You do not need a CUBE license for a SIP trunk built directly between CUCM clusters. For example:
|<-----------------------signaling--------------------------------->|
IP Phone----CUCM Cluster 1-----------IP WAN--------------------CUCM Cluster 2-----TDM GW--------External Phone

You do need a SIP trunk license when you use a gateway to terminate the CUCM signalling (and often media as well) and re-generate the signalling to the SIP trunk provider or third party PBX. For example:
|<---signaling--------->|<-----------signaling---------->|
IP Phone-----CUCM Cluster------------------29xxGW-----------------------SIP provider/third party IP PBX-------External Phone 


References:
What is SIP Trunking?
http://www.siptrunk.org/whatissiptrunking.php

CUBE FAQ (2011)
https://supportforums.cisco.com/docs/DOC-17964

CUBE v8.8 datasheet
http://www.cisco.com/en/US/prod/collateral/voicesw/ps6790/gatecont/ps5640/product_data_sheet09186a00801da698.html

Power Cube 4 wont fit in 7900 series IP Phones

Here's an interesting item I've just discovered and worth remembering:

Is CP-PWR-CUBE-4= compatible with the 7900 series phones?

A: No this power cube is for the new 8900/9900 phones, compatible with Communications Manager. The power tip is different size, won't fit in other phones.

So now we know!

Cisco UCL vs UWL (CUWL)

Cisco UCL and UWL are two types of licensing for Cisco IP Telephony deployments. The top level difference between them is that UCL is licenced per user per application, where as UWL is just licensed per user. CUWL licensing covers many more applications than UCLand because of this costs more is you compare a set of UCL licenses with a set of UWL licenses, however UCL only covers one application so if you are doing more than call processing UCL starts getting expensive.

For example if you want call processing and voicemail you can either buy one CUWL license per user or 2 UCL licenses per user (one for call processing and the other for voicemail).

This is a great quick reference:
http://www.cisco.com/en/US/prod/collateral/voicesw/ps6882/ps9156/product_solution_overview0900aecd806cc7a4.html