Re: [cisco-voip] SNR / RDP without a registered endpoint

[Roger Wiklund]

We are running this on all releases from 6 up to 11, it works fine.

It's very common in Sweden to only have a mobile phone (Mobile
Extension) and dial via the PBX.

On Tue, Dec 8, 2015 at 4:01 PM, Lelio Fulgenzi  wrote:
>
> I did this before in v7. Haven't confirmed in v9.
>
> Worked as expected. Without the registered phone you (obviously) lose the 
> ability to control the flow of calls (mobility soft keys), but you likely 
> don't need that.
>
> The instructions and documentation always have it paired with a hard phone so 
> that's likely the supported way. Which means, something might change in the 
> future that breaks an SNR without a hard phone setup.
>
>
> Sent from my iPhone
>
>> On Dec 8, 2015, at 9:44 AM, joel  wrote:
>>
>> Looking to setup an RDP solution for some users that do not have IP phones 
>> and curious if this could be done without having an actual registered IP 
>> phone.
>>
>>
>>
>> --
>> Joel
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Re: [cisco-voip] CUCM - MOH Silence

[Roger Wiklund]

Not sure about the trace but I would start with basic config
check/troubleshooting.

Are you running unicast or multicast MOH?
Is the IP Voice Media Streaming App running on all nodes? (If yes try
restarting the service)
Is the MOH resource in the MRG/MRGL?
Are all devices using that MRGL?
Is MOH selected for your codec? (System, Service Parameters, server, IP
Voice Media Streaming App)
Have you uploaded a new MOH file? If so try with the default.

I would start there, or try basic ip-phone to ip-phone calls, exclude voice
gateways etc and work my way forward.

On Thu, May 14, 2015 at 10:50 PM, Anthony Holloway 
avholloway+cisco-v...@gmail.com wrote:

 All,

 So, I'm a bit rusty this year on trace analysis and I need a second
 opinion.

 From the below screenshot snippets out of TranslatorX, it would appear as
 though the MOH_3 gets selected, then an AuConnectRequest gets issued, and
 not but a few seconds later, I see an AuDisconnectRequest.  The caller
 experience is simply silence on the line while the call is connected (or
 not connected) to MOH.

 If there was another key piece of information I could look for to help
 myself understand why it disconnected so quickly, what should I look for?
 Thanks.

 [image: Inline image 1]

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Re: [cisco-voip] Sip Trunk - CUCM and Third-party PBX

[Roger Wiklund]

Username and password does not necessarily mean REGISTER unless they
specifically said so of course.

Proxy authentication is another way to authenticate with
username/password. That you can configure under User Management -
SIP Realm

When you place an outbound call to the third-party PBX they will
respond with 407 (Proxy Authentication Required) and include a SIP
Realm.
CUCM will match that with configured SIP Realm and create a new INVITE
with configured credentials.

On Mon, May 11, 2015 at 3:33 PM, Claiton Campos claitoncam...@gmail.com wrote:
 I have a scenario where I need to create a SIP trunk between a CUCM 10.5 and
 a third-party PBX. The problem is that the third-party PBX prompts the trunk
 sip is authenticated through username and password should I register on the
 CUCM. Has anyone had an experience with this type of configuration on a SIP
 Trunk?

 Tks,

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Re: [cisco-voip] how codec transparent works?

[Roger Wiklund]

First of all, switch to SIP, then make a test call and collect logs in
CUCM, 2800 and Asterisk. Just compare the codec offered in the SDP.
Modify the CUCM/2800/Asterisk so they all have the same codec, I.E
g711alaw AND ulaw, 20ms sample rate etc. remove any other codecs not
used.

You want to avoid using MTP in CUCM as resources are limited. You can
use software MTP in 2800 but it just adds complexity to solve a simple
problem.



On Mon, May 11, 2015 at 5:16 PM, Brian Meade bmead...@vt.edu wrote:
 If you want us to be able to figure out your H.323 negotiation problem
 without codec transparent in place, you need to provide these debugs:
 debug h225 asn1
 debug h245 asn1

 On Mon, May 11, 2015 at 6:33 AM, s m sam.gh1...@gmail.com wrote:

 thank you Ryan,

 i have no problem with sip and it is ok (although i do not know it has MTP
 or not). i think transcoder may not needed because as i know, it translate
 two different codecs to each other but in my scenario, both side uses
 g711alaw. please let me know if i misunderstand it.


 On Mon, May 11, 2015 at 11:31 AM, Ryan Huff ryanh...@outlook.com wrote:

 The reason that is happening is due to media negotiation failure as you
 mention (both call legs are not offering the same codec capabilities). In
 that exact configuration, you would need a transcoder (which you could run
 on the router if you have enough DSP).

 Are you sold on h323 or can you do a full SIP trunk (with MTP) between
 cucm and asterisk?

 Thanks,

 Ryan



  Original Message 
 From: s m sam.gh1...@gmail.com
 Sent: Monday, May 11, 2015 12:25 AM
 To: Ryan Huff ryanh...@outlook.com
 Subject: Re: [cisco-voip] how codec transparent works?
 CC: cisco-voip@puck.nether.net

 hello Ryan,

 thank you for your reply. without codec transparent, my phone rings but
 when i answer i have no voice and it hangs up after 5 seconds. asterisk says
 no answer. this is so strange for me. i think media negotiation failed,
 right? is there any hint to have h323 trunk to asterisk with specific codec
 (not transparent one)???

 On Sun, May 10, 2015 at 4:32 PM, Ryan Huff ryanh...@outlook.com wrote:

 Codec transparent just passes sdp through to the other call leg without
 trying to do media negotiations.

 So without codec transparent,  what happens?
 Thanks,

 Ryan



  Original Message 
 From: s m sam.gh1...@gmail.com
 Sent: Sunday, May 10, 2015 01:19 AM
 To: cisco-voip@puck.nether.net
 Subject: [cisco-voip] how codec transparent works?

 hello everybody,

 anybody knows how codec transparent works?

 i have a strange problem. i want to set h323 trunk between asterisk and
 cisco 2800. it only works when i set codec transparent in dial-peer nodes.
 show commands in cisco shows that i have a call with g711alaw but if i set
 codec g711alaw in dial-peers, i do not have any success call. i know it is
 codec compatibility problem. is there any difference between g711 codecs
 which cisco and asterisk utilize? what happened when codec is set to
 transparent? dose anyone know anything about it?

 thanks is advance
 SAM




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Re: [cisco-voip] 3rd party conference phones

[Roger Wiklund]

We are using a couple of SNOM conference phones, they do the job just
fine. Not wireless but PoE.

https://www.snom.com/en/products/snom-complementary-line/snom-meetingpoint-sip-conference-phone/

On Tue, Apr 28, 2015 at 11:22 PM, Nick Thompson n...@imperial.org wrote:
 I just checked with a customer who has a bunch of the FLX2’s deployed and he 
 is seeing the same behavior.  He also tested with the standard SIP device and 
 advanced SIP device with no change.  He was going to call Revo and see if 
 there is something he is missing or if it is just a limitation of the device 
 firmware.

 Nick

 On Apr 28, 2015, at 10:09 AM, Ed Leatherman ealeather...@gmail.com wrote:

 Circling around on this thread,

 So we're trying out revolabs FLX2.

 They are pretty easy to use and have some nice features. The unit reminds 
 you to put everything back on the charging station after a call is over, 
 which is an interesting touch.

 Main concern for me right now is that you cannot invoke an ad hoc conference 
 that uses a cm controlled conference bridge (that I can tell) from the unit. 
 The conference function can merely join the two line appearances that the 
 unit has.

 We're still going to try it out in one of our rooms where the executive 
 absolutely had to have wireless but I don't see widely deploying it 
 elsewhere; people use the ad hoc bridges an aweful lot.

 Ed



 On Thu, Apr 9, 2015 at 10:05 AM, Nick Thompson n...@imperial.org wrote:

 Revolabs FLX2 - http://www.revolabs.com/flx2

 Revolabs is the OEM for the 8831 conference phone as well.


 On Apr 9, 2015, at 7:41 AM, Ed Leatherman ealeather...@gmail.com wrote:

 Hello

 I've had a few requests for Wireless conference phones lately for some 
 medium sized rooms. In these cases the executive wants to get any cables 
 etc off the table, so a wired base station w/ cordless module for the table 
 fits the bill as much as a WiFi device would. And because Wireless!

 I'm curious what other people might be using, I see that polycom has the 
 SoundStation2W that uses a POTS line. I've also had some folks using a 
 Jabra Speak510 paired with some of the bluetooth capable Cisco Phones, but 
 this only works well in small rooms.

 Thanks!



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Re: [cisco-voip] Cisco Finesse vs CAD

[Roger Wiklund]

We have depolyed it for one customer so far. It better and worse than
CAD at the same time.
It's like when Cisco moved from CUPC to Jabber. As it's based on new
tech it has potential but there's a huge backlog for feature parity
with CAD.

What do you say to customers who were used to basic functions like
chat, blind transfer and recent call list. The agent cant even see a
call history list!

I have given up hope with Cisco... maybe in UCCX 11.

On Tue, Apr 14, 2015 at 5:40 PM, Mathew Miller miller.mat...@gmail.com wrote:
 I would assume that is a fair assessment that CAD will be phased out.

 There is a decent white paper on the differences between CAD and Finesse
 here
 http://www.cisco.com/c/en/us/products/collateral/customer-collaboration/unified-contact-center-express/white-paper-c11-730883.pdf
 (It's slightly out of date for 10.6).




 On Mon, Apr 13, 2015 at 11:58 AM, JASON BURWELL
 jason.burw...@foundersfcu.com wrote:

 I am planning a UCCX upgrade and based on recent posts, am I correct to
 understand that CAD is now being phased out in favor of Finesse?



 Also, can anyone currently on Finesse share any information on how well it
 works? I initially heard some negative feedback about it but that was
 earlier in the lifecycle so I am keeping my fingers crossed the stability
 has improved.



 Can anyone point me to a link that has any screen shots, user training or
 any other information giving me a preview of what the end user experience
 will be like? For those who have migrated from CAD to Finesse, what was the
 experience with training and user acceptance on Finesse?



 Thanks

 Jason




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Re: [cisco-voip] (no subject)

[Roger Wiklund]

I have.

Went from 2500 to 7500 on CUCM 10.5(1).

You need to download the VMware Disk Size Reallocation COP file for
10.5. Worked like a charm.

http://www.cisco.com/web/software/282204704/18582/CleanupCommonCOPfilev1.3.pdf
http://www.cisco.com/web/software/282204704/18582/ciscocm.vmware_disk_size_reallocation_v1.0.pdf

On Fri, Mar 20, 2015 at 2:28 PM, Justin Steinberg jsteinb...@gmail.com wrote:
 Has anyone successfully expanded the virtual disk size of CUCM VMs without
 rebuild/DRS?



 I have an install where CM 10.5 is using the 2500 user template and we want
 to increase to 7500 users.  The 2500 OVA is 1 vCPU, 4GB, 1x80GB.The 7500
 OVA is 2vCPU, 6 GB, 1x110GB.In the past, the older 7500 user CM versions
 had two virtual 80 GB disks, however since 9.1 the 7500 user is a single 110
 GB disk.   It seems like with a single virtual disk it would be easier to
 expand an existing VM without rebuild.



 There are several bugs on the topic:

 https://tools.cisco.com/bugsearch/bug/CSCug63058

 https://tools.cisco.com/bugsearch/bug/CSCuc58936



 In older CM versions there was a COP file to assist with allowing the VM to
 use more disk when the vdisk was increased.  However, now I believe that it
 is just built in to CM to use more disk on reboots if it detects a vdisk
 change instead of needing to run the OVA.



 There is still conflicting documentation on the topic, so I will probably
 open a TAC case but curious if anyone has dealt with this before?


 Justin


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Re: [cisco-voip] CUBE across VRFs

[Roger Wiklund]

Most providers are using SBCs facing the customer. Verizon for example
has two /25, first for SIP and second for RTP. Just point a static
route to them on the outside interface, then the default route towards
the inside interface.

On Wed, Feb 4, 2015 at 11:16 PM, Matthew Loraditch
mloradi...@heliontechnologies.com wrote:
 I’ve seen it come two ways, riding your same MPLS circuit, in which case if
 you have a dedicated VG you just default route that to your MPLS router and
 there you go.



 The other way is like you say and I’ve done that with att and I didn’t have
 to route with them, they NAT’d everything on their side to me. So I just
 routed their couple SBC IPs/Subnets across that handoff and my default still
 goes into my LAN.



 I’m sure there are other ways as well.



 Matthew G. Loraditch – CCNP-Voice, CCNA-RS, CCDA
 Network Engineer
 Direct Voice: 443.541.1518

 Facebook | Twitter | LinkedIn | G+



 From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of
 Norton, Mike
 Sent: Wednesday, February 04, 2015 5:05 PM
 To: Erick
 Cc: cisco-voip@puck.nether.net


 Subject: Re: [cisco-voip] CUBE across VRFs



 What I’m failing to understand is... if I set the CUBE’s default route to be
 my router on my network, then how will CUBE be able to reach the SIP
 provider’s call servers on the SIP provider’s network? It seems like I will
 need a routing protocol on whichever side of the CUBE doesn’t get a default
 route. Is that a normal requirement?



 Just to back up a bit, I have been assuming CUBE would have two interfaces –
 one on my network, one on the SIP provider’s network. I’ve always assumed
 that this was the normal way of deploying CUBE but maybe I’m off base there
 and getting myself confused.



 -mn





 From: Erick [mailto:erick...@gmail.com]
 Sent: February-03-15 6:50 PM
 To: Norton, Mike
 Cc: Jason Aarons (AM); cisco-voip@puck.nether.net
 Subject: Re: [cisco-voip] CUBE across VRFs



 Only one voice vrf can be defined in IOS. Global under voice service voip.



 Cube-SP lets you do multiple vrf's but is EoL and way different
 configuration.



 If you plop a cube off your router and router interface is in a vrf and your
 separate cube is on that network then it should be fine as the cube is just
 a host then  with default route to router.


 Sent from my iPhone


 On Feb 3, 2015, at 6:08 PM, Norton, Mike mikenor...@pwsd76.ab.ca wrote:

 Doesn’t have to be two VRFs, could be one VRF and the global route table, if
 that makes a difference. This idea is no connectivity between them, other
 than the application-layer connectivity provided by CUBE. This is
 hypothetical – I’m just trying to understand how/if this would work. I’m
 looking to plop a CUBE between my network and a SIP provider’s network
 without having to participate in routing protocol on either side.



 -mn



 From: Jason Aarons (AM) [mailto:jason.aar...@dimensiondata.com]
 Sent: February-03-15 5:02 PM
 To: Norton, Mike; cisco-voip@puck.nether.net
 Subject: RE: CUBE across VRFs



 You have two VRFs, do they have connectivity between them?



 From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of
 Norton, Mike
 Sent: Tuesday, February 3, 2015 4:36 PM
 To: cisco-voip@puck.nether.net
 Subject: [cisco-voip] CUBE across VRFs





 Can CUBE sit across two separate VRFs? I’ve never used it, but I’m
 envisioning an ISR having a VRF-Lite with default route pointed at my
 network, and a VRF-Lite with default route pointed at the SIP provider’s
 network. I’m thinking this would be the preferred way to do it, but maybe
 I’m missing something?



 My Googling is dredging up a lot of really old info that I’m not sure is
 still relevant.



 --

 Mike Norton



 itevomcid

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Re: [cisco-voip] SBC/SIP Trunk Design queries

[Roger Wiklund]

We use Acme 3820 with our UCaas platform. Behind it we serve Cisco,
Mitel, Avaya and Lync PBX:es.
You can't go wrong with Acme, especially when it comes to SIP
manipulation, there's nothing you can't do. For me that's the number
one selling point.
HA is awesome with hitless failover. You can upgrade outside of
service windows if you have to.
Flow through/flow around is more flexible in Acme with media release
based on same-IP for example.
If you run Enterprise version or Service Provider version 7.2 you get
a web GUI which is very helpful when troubleshooting.
http://www.markholloway.com/blog/wp-content/uploads/2012/08/Screen-Shot-2012-08-08-at-3.00.55-PM.png

The only downside with Acme for me is Oracle. I miss the days when it
was just Acme Packet, the support was awesome. Now, not so much. It
feels like all the talented engineers jumped ship.

With that said we do use CUBE but for smaller on-prem solutions. It
does the job and it's easy to configure.

Sonus is another player that i've heard some buzz about.

On Thu, Mar 12, 2015 at 1:34 AM, David Lin david@msn.com wrote:
 I think one of important things is the capacity you are looking for.
 ACME does give you better scalability and better troubleshooting capability,
 but if you are only looking for couple hundreds of concurrent calls, you
 probably can live with CUBE to keep your cost lower.

 D.
 
 From: tim.sm...@enject.com.au
 To: terry.che...@gmail.com; cisco-voip@puck.nether.net
 Date: Wed, 11 Mar 2015 04:19:22 +
 Subject: Re: [cisco-voip] SBC/SIP Trunk Design queries


 Hi Terry,



 I do quite a bit of CUBE, and have done a bit of Acme as well.



 There were some recent partner sessions that talk about some interesting
 things coming for CUBE, so it’s worth making sure you are getting latest
 roadmap info.



 My main comparison points..



 # HA



 In enterprise there was HA on CUBE, and it was improving in each release
 (but there are caveats with it)

 Have found Acme HA to be seamless and rock solid.



 # Deployment



 Cisco has some great interop guides – if you go with a carrier that has
 spent the money, a lot of the hard work has been done for you in terms of
 testing (as you know SIP can be implemented and configured in many different
 ways – if someone hasn’t done a lot of testing up front, you do sometimes
 end up adding SIP profiles and tweaks as you discover issues)



 Acme has some very thorough guides – I’m not sure if they have interop
 testing with carriers – given they are in SP’s a lot, there is a good chance
 they do. I’d look into it that with the Acme SE. Talk to prospective ITSP’s
 about their testing, and supported SBC’s.



 # Ops



 CUBE enterprise is great, IOS, most people are familiar. You will most
 likely need to train people on Acme

 I find troubleshooting a bit of a let down with CUBE. Basically log to
 buffer, copy to file, or packet captures. Wireshark with ladders or
 TranslatorX are great, but it’s getting the files there that bugs me.

 Alternatively, there did seem to be a few 3rd party tools out there, but you
 are probably looking at $$$



 Acme has web interface, list of calls and then ability to drill down with
 ladder diagrams, messaging capture etc. You should see this before making
 decision.



 Some good knowledge on Acme forums

 Acme has very flexible manipulation – CUBE is quite good too (and they have
 great profile testing tool) – plus you can also use CUCM LUA on the SIP
 trunk



 # On your other notes



 Centralised – this is great for flexibility DR etc, standard stuff be aware
 of the call volumes over the WAN, caller ID considerations for emergency and
 local pizza shop type services



 WAN – we terminate on existing equipment, and Acme is in a VLAN, I think
 this is most flexible.. you have a very flexible set up in Acme in regard to
 networking, lots of zones, interface options etc.



 Transcoding – I think you could still utilise CUCM registered transcoders
 for the ASR scenario..



 Virtual - We use virtual Acme, it had some teething problems in very first
 versions (and a clunky license on USB stick thing going on) but it seems to
 be good now

 We don’t have transcoding / media resources in the virtual
 edition



 Flow through / around – a lot of designs the carrier doesn’t have
 connectivity into the rest of the network, so flow through is quite typical.

 However, we do have carriers here that have SBC’s on your
 WAN, so flow through can be nice here – it also then makes CUBE HA less
 important, i.e. if call is set up, media is from end point to carrier SBC
 already (if no xcoding involved)



 So I won’t say one way or the other, just my thoughts on things you can
 consider.

 I like both, and will continue to work on both!



 Cheers,



 Tim





 From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of
 Terry Cheema
 Sent: Wednesday, 11 March 2015 1:10 PM
 To: cisco-voip voyp 

Re: [cisco-voip] CUBE across VRFs

[Roger Wiklund]

Use Acme Packet instead of CUBE if you need to handle multiple
customers/overlapping IPs.

To bad they are Oracle now


On Wed, Feb 4, 2015 at 2:49 AM, Erick erick...@gmail.com wrote:
 Only one voice vrf can be defined in IOS. Global under voice service voip.

 Cube-SP lets you do multiple vrf's but is EoL and way different
 configuration.

 If you plop a cube off your router and router interface is in a vrf and your
 separate cube is on that network then it should be fine as the cube is just
 a host then  with default route to router.

 Sent from my iPhone

 On Feb 3, 2015, at 6:08 PM, Norton, Mike mikenor...@pwsd76.ab.ca wrote:

 Doesn’t have to be two VRFs, could be one VRF and the global route table, if
 that makes a difference. This idea is no connectivity between them, other
 than the application-layer connectivity provided by CUBE. This is
 hypothetical – I’m just trying to understand how/if this would work. I’m
 looking to plop a CUBE between my network and a SIP provider’s network
 without having to participate in routing protocol on either side.



 -mn



 From: Jason Aarons (AM) [mailto:jason.aar...@dimensiondata.com]
 Sent: February-03-15 5:02 PM
 To: Norton, Mike; cisco-voip@puck.nether.net
 Subject: RE: CUBE across VRFs



 You have two VRFs, do they have connectivity between them?



 From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of
 Norton, Mike
 Sent: Tuesday, February 3, 2015 4:36 PM
 To: cisco-voip@puck.nether.net
 Subject: [cisco-voip] CUBE across VRFs





 Can CUBE sit across two separate VRFs? I’ve never used it, but I’m
 envisioning an ISR having a VRF-Lite with default route pointed at my
 network, and a VRF-Lite with default route pointed at the SIP provider’s
 network. I’m thinking this would be the preferred way to do it, but maybe
 I’m missing something?



 My Googling is dredging up a lot of really old info that I’m not sure is
 still relevant.



 --

 Mike Norton



 itevomcid

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Re: [cisco-voip] licensing turnaround time 8.6 to 10.5

[Roger Wiklund]

I second the hahahahaha@24 hours. More like 30-40 days of dealing with
the robots at licensing.
It's literally like talking to a robot. They can never help you unless
you want a demo license. They have their stupid script that they
follow and anything outside of it they need the product manager. (17
days now and no answer from PM)

Hope you have a great day sir...bullshit!





On Sat, Dec 13, 2014 at 6:04 AM, Erick erick...@gmail.com wrote:
 I can confirm unity connection stops :)  a reboot gets it going for another
 24 hours unless you fix the violation.

 Sent from my iPhone

 On Dec 12, 2014, at 10:34 PM, Lelio Fulgenzi le...@uoguelph.ca wrote:

 I also believe what happens during the grace period and after also differs.

 For example, I don't think callmanager ever stops working, but connection
 might.

 http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/connection/9x/administration/guide/9xcucsagx/9xcucsag310.html#pgfId-1092685

 If the required number of licenses are not installed on the ELM server, the
 license status becomes “Violation”. However, you can still use the licensed
 features on Unity Connection for 60 days, which is the grace period. During
 this grace period, you are required to obtain and install the required
 number of licenses or reduce the usage of the licensed features in order to
 avoid license violation. However, if you do not take the required action
 during the grace period, then license status becomes “Expire”.

 Once the license status of Unity Connection software changes to “Expire”,
 the software will stop functioning. The Unity Connection server will not
 answer any calls to leave or retrieve voicemails. However, you can still
 add, modify or delete configuration data on the server. You are required to
 obtain and install the required number of licenses or reduce the usage of
 the licensed features to avoid license violation. After taking the required
 action, the license status changes from “Expire” to “Compliance” and you can
 use the licensed features on Unity Connection again.



 Sent from my iPad

 On Dec 12, 2014, at 9:00 PM, Steve Rubin s...@layer42.net wrote:


 FWIW, the grace period on a 10.5 install is 180 days :)   I have 174

 days left on one I installed last week.



 The Callmanager grace period is 180 days, but the Unity Connection grace
 period is shorter (90 or 120 days, I can¹t remember which).  I found this
 out the hard way.


 --
 Steve Rubin
  s...@layer42.net
 Layer42 Networks
 http://www.layer42.net/





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[cisco-voip] Finesse roadmap?

[Roger Wiklund]

Hi

I'm deploying CUCM 10.5(1) and UCCX 10.5(1) for a new customer.
I decided to go with Finesse instead of CAD because CAD is ye olde.

So far so good, been playing around with the desktop layout and the
sample gadgets etc.

However I came across this page and noticed there are a few things
missing compared to CAD.

http://www.cisco.com/c/en/us/products/collateral/customer-collaboration/unified-contact-center-express/white-paper-c11-730883.html

Because Unified CCX 10.0 is the first release of Finesse for Unified
CCX, some key features in Cisco Agent Desktop are not yet available in
Finesse. These features include:

● Direct preview outbound support
● Multiple-line handling
● Agent email
● Localization
● Agent-to-agent chat
● Some workflow events and actions
● Blind transfer
● Recent call list
● On-demand call recording

I believe 1,2 and 3 are added to 10.5(1).

However agent-to-agent chat, blind transfer and on-demand call
recording is a must, and I'm wondering when to expect finesse to be on
par with CAD.

Also I don't want to use MediaSense just to store recordings, that
might be fine with UCCE but not UCCX, it should be able to handle it
on the server as it did with CAD.

Thanks

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Re: [cisco-voip] Jabber as Softphone and Lync for Chat and Presence

[Roger Wiklund]

You could go Jabber Phone Mode + Lync or CUCILync

It's basically the same experience, a separate client in both cases.

On Fri, Aug 15, 2014 at 9:36 AM, Reto Gassmann v...@mrga.ch wrote:
 Hello Group

 we run a CUCM 9.1.2 with IMP and RCC to Lync.
 Now we want to deploy Jabber as a softphone only. So we want to use Lync as
 IM, Presence and RCC and Jabber as Softphone.

 How can we install and configure Jabber to coexist with Lync on the same
 Workstation?

 I found documentation with install parameters (eg CLICK2X) and Jabber xml
 configfile, but cannot put that together.

 Thanks for input

 Regards Reto

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[cisco-voip] CUBE dial-peer with b2bua command

[Roger Wiklund]

Hi

I've seen a couple of CUBEs configured with the following:

dial-peer voice x voip
 b2bua

I'm confused by this as the CUBE is per definition a B2BUA without that command.

The only thing I can find in the documentation on b2bua is regarding CME.

b2bua

To configure a dial peer associated with an individual Session
Initiation Protocol (SIP) phone in Cisco Unified CME or a group of
phones in a Cisco Unified SIP Survivable Remote Site Telephony (SRST)
environment to point to Cisco Unity Express, use the b2bua command in
dial-peer configuration mode. To disable B2BUA call flow on the dial
peer, use the no form of this command.

So I'm assuming it's redundant to configure it unless you run CME?

Thanks
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[cisco-voip] CUCM Letters in Translation Pattern?

[Roger Wiklund]

Hi

I'm trying to match a called number from a CUBE that's C990134T.

I created a TP that matches C990134. predot. DNA show match and proper
strip etc.

However it does not work. If I strip the C in the CUBE and send 990134T it
works, so it looks like there's a problem with letters.

The syntax on TP says A-D is fine, what am I missing? Anything special like
the backslash with + (\+)

Thanks
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