[cisco-voip] Cisco mobile remote access

2017-11-19 Thread Leslie Meade via cisco-voip
With mobile remote access, the mobile number is query’s against the cucm. If I 
set the calling mask to be different to the cell number, let’s say client main 
line. The vxml is sent back to the Cisco cube and the call is release with the 
main line as the caller ID.

What I want to know, can anyone send me a vxml log/dump of a mobile device 
calling into MRA, and then send the call back out to the pstn with a different 
number.

Of course remove the numbers and put something else in, but I want to see the 
vxml debugs of this call flow.

I would do it, but I no longer have access to a working platform, and cube to 
pull the logs.

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[cisco-voip] CSR Information

2015-05-04 Thread Leslie Meade
I am exporting the Tomcast certs and i have checked it against a CSR checker to 
make sure that everything is good before i submit the CSR.

But I am getting a , in my Subject Alternative Name.


For example


Subject Alternative Name: , ca.forces.net


There is a comma in the subject feild and I do not know where to fix it. I have 
checked the web-security and there seems to be nothing funny there.


https://www.sslshopper.com/csr-decoder.html,



Leslie




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Re: [cisco-voip] CSR Information

2015-05-04 Thread Leslie Meade
Nope that's not it. i checked that as well



From: James Buchanan james.buchan...@gmail.com
Sent: Monday, May 4, 2015 7:15 AM
To: Leslie Meade
Cc: cisco-voip (cisco-voip@puck.nether.net)
Subject: Re: [cisco-voip] CSR Information

Check the parent domain field and see if a space has made it's way into it.

On Mon, May 4, 2015 at 10:13 AM, Leslie Meade 
leslie.me...@lvs1.commailto:leslie.me...@lvs1.com wrote:

I am exporting the Tomcat certs and i have checked it against a CSR checker to 
make sure that everything is good before i submit the CSR.

But I am getting a , in my Subject Alternative Name.


For example


Subject Alternative Name: , ca.forces.nethttp://ca.forces.net


There is a comma in the subject feild and I do not know where to fix it. I have 
checked the web-security and there seems to be nothing funny there.


https://www.sslshopper.com/csr-decoder.html,



Leslie





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Re: [cisco-voip] EMCC with SME

2015-04-21 Thread Leslie Meade
?Ok thanks.


From: bmead...@gmail.com bmead...@gmail.com on behalf of Brian Meade 
bmead...@vt.edu
Sent: Tuesday, April 21, 2015 2:02 PM
To: Leslie Meade
Cc: cisco-voip (cisco-voip@puck.nether.net)
Subject: Re: [cisco-voip] EMCC with SME

You'll want just a single dedicated EMCC PSTN access SIP Trunk here.  It will 
use that to communicate with the visiting cluster for PSTN calls.

On Tue, Apr 21, 2015 at 5:00 PM, Leslie Meade 
leslie.me...@lvs1.commailto:leslie.me...@lvs1.com wrote:

?Ok, by looking at the diagram again. Instead of creating new SIP trunks to 
point to the other cluster. I can reuse my existing SME Trunks for the task.



From: bmead...@gmail.commailto:bmead...@gmail.com 
bmead...@gmail.commailto:bmead...@gmail.com on behalf of Brian Meade 
bmead...@vt.edumailto:bmead...@vt.edu
Sent: Tuesday, April 21, 2015 1:48 PM
To: Leslie Meade
Cc: cisco-voip (cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net)
Subject: Re: [cisco-voip] EMCC with SME

The EMCC config file transfers is all done via TFTP.

The EMCC SIP Trunk only comes into play when trying to route calls back to the 
visiting cluster.

Here's a diagram of how the EMCC SIP Trunk is used- 
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/8_0_2/ccmfeat/fsgd-802-cm/fsemcc.html#wp1457098


On Tue, Apr 21, 2015 at 4:34 PM, Leslie Meade 
leslie.me...@lvs1.commailto:leslie.me...@lvs1.com wrote:

?So I know that in order to make this work you need SIP trunks between 
clusters.  And what i understand the SIP trunks only carry the communication 
for the EMCC config. Question is what if there is a SME between both clusters ? 
If the SIP trunks only carry EMCC information and nothing to do with call 
routing i believe it is ok to place sip trunks that terminate on each others 
Publisher and bypass the SME.



?


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Re: [cisco-voip] EMCC with SME

2015-04-21 Thread Leslie Meade
?Ok, by looking at the diagram again. Instead of creating new SIP trunks to 
point to the other cluster. I can reuse my existing SME Trunks for the task.



From: bmead...@gmail.com bmead...@gmail.com on behalf of Brian Meade 
bmead...@vt.edu
Sent: Tuesday, April 21, 2015 1:48 PM
To: Leslie Meade
Cc: cisco-voip (cisco-voip@puck.nether.net)
Subject: Re: [cisco-voip] EMCC with SME

The EMCC config file transfers is all done via TFTP.

The EMCC SIP Trunk only comes into play when trying to route calls back to the 
visiting cluster.

Here's a diagram of how the EMCC SIP Trunk is used- 
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/8_0_2/ccmfeat/fsgd-802-cm/fsemcc.html#wp1457098


On Tue, Apr 21, 2015 at 4:34 PM, Leslie Meade 
leslie.me...@lvs1.commailto:leslie.me...@lvs1.com wrote:

?So I know that in order to make this work you need SIP trunks between 
clusters.  And what i understand the SIP trunks only carry the communication 
for the EMCC config. Question is what if there is a SME between both clusters ? 
If the SIP trunks only carry EMCC information and nothing to do with call 
routing i believe it is ok to place sip trunks that terminate on each others 
Publisher and bypass the SME.



?


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[cisco-voip] EMCC with SME

2015-04-21 Thread Leslie Meade
?So I know that in order to make this work you need SIP trunks between 
clusters.  And what i understand the SIP trunks only carry the communication 
for the EMCC config. Question is what if there is a SME between both clusters ? 
If the SIP trunks only carry EMCC information and nothing to do with call 
routing i believe it is ok to place sip trunks that terminate on each others 
Publisher and bypass the SME.



?

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[cisco-voip] VG350 analog ports

2015-03-31 Thread Leslie Meade
Any one seen a problem where an analog phone gets battery/voltage but no ring 
tone. the phone will not work. However if you ring it everything works well. 
You are then able to lift the handset and get dial tone and make calls. Give 
the port X amount of time and you loose dial tone again.


Cheers


Leslie
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Re: [cisco-voip] VG350 analog ports

2015-03-31 Thread Leslie Meade
I think i have found the issue


https://supportforums.cisco.com/discussion/12286521/vg350-sccp-ports-losing-dial-tone?


Mind you I am running older code 15.1 but I think it is the same


Cheers



Leslie



From: cisco-voip cisco-voip-boun...@puck.nether.net on behalf of Leslie Meade 
leslie.me...@lvs1.com
Sent: Tuesday, March 31, 2015 4:45 AM
To: cisco-voip (cisco-voip@puck.nether.net)
Subject: [cisco-voip] VG350 analog ports


Any one seen a problem where an analog phone gets battery/voltage but no ring 
tone. the phone will not work. However if you ring it everything works well. 
You are then able to lift the handset and get dial tone and make calls. Give 
the port X amount of time and you loose dial tone again.


Cheers


Leslie
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[cisco-voip] Cucm 10.5.2 firmware

2015-02-02 Thread Leslie Meade
Can anyone tell me if there are firmware updates in this release ?
I looked at the release notes and I can not see it.



Sent from my iPad

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[cisco-voip] Mediasense 10.5 playback

2015-01-20 Thread Leslie Meade
I installed this and while 90% of the time I do not have issues I run into this 
error every so often.. It does not matter what version of browser I am using I 
some times get the error saying it cannot play it.. anyone seen this ?

[cid:image001.png@01D034B0.E0F81EB0]
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[cisco-voip] Web directory from CUCM

2015-01-15 Thread Leslie Meade
Is there a away to pull the phone book from Callmanager 10.5 to a web page ? 
Client wants to searchable phonebook, that is pulling the data from CUCM.

Cheers

Leslie

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Re: [cisco-voip] Device pack installation methodology question

2015-01-06 Thread Leslie Meade
I'd also recommend this way as well. A whole lot easier

Leslie


From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
Matthew Collins
Sent: Monday, January 05, 2015 1:38 AM
To: Ryan Huff; cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Device pack installation methodology question

Hi Ryan,

I have just installed the device pack on the publisher first. Then changed the 
device defaults back to their originals, Then reboot the publisher, TFTP and 
subs in turn. No need to mess around with the CM groups.

One tip I got from the forum a couple of years ago when installing device pack 
was to open the device defaults page in IE, Then install the device pack using 
another browser Firefox or chrome, once the device pack is installed go back to 
the IE page without refreshing and select save, This will then revert all the 
firmware version back to the pre device pack state, Saves having to manually 
change each one back.


Regards

Matthew


From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Ryan 
Huff
Sent: 03 January 2015 20:12
To: cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net
Subject: [cisco-voip] Device pack installation methodology question

I need to install a device pack on a 2 node 9.1(2) CCM cluster to get support 
for some 88xx phones but I do not want to update the loads for anything else.

The approach I am going to use is:

Drop the publisher out of the CM Group, forcing all phones to the subscriber. 
Install the device pack on the publisher and reboot the publisher. Once the 
publisher is backup, set all the device defaults back to what I want them to be 
then add the publisher back to the CM Group. Then drop the subscriber from the 
CM Group forcing all the phones on the publisher and start the install process 
over for the subscriber. Once everything is back up add the subscriber back to 
the CM Group.

Does that sound reasonable or is there an easier way?

Thanks,

Ryan
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Re: [cisco-voip] Uplinx Question

2014-12-05 Thread Leslie Meade
It is currently in use where I am contracting at the moment..

They have get it setup so that the end users/help desk folks never log into 
callmanager/unity etc to do MAC's

They also use it to bulk import users, that makes their Unity, Jabber, Remote 
destination profile as well as extension mobility profiles.

I like it way better that using cisco application

Leslie


From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Sean 
Knight via cisco-voip
Sent: Friday, December 05, 2014 7:07 AM
To: 'cisco-voip@puck.nether.net'
Subject: [cisco-voip] Uplinx Question

Has anyone used this software to manage phone?  If so how are your thoughts on 
it?

http://www.uplinx.com/reporttool-usd.htm

Thanks,

Sean E. Knight

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Re: [cisco-voip] Uplinx Question

2014-12-05 Thread Leslie Meade
But that is a separate application from the uplinx device program

Leslie Meade


..
Mobile:778.228.4339 | Main: 604.676.5239
Email: leslie.me...@lvs1.commailto:leslie.me...@lvs1.com

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Heim, 
Dennis
Sent: Friday, December 05, 2014 7:26 AM
To: Kevin Przybylowski; Sean Knight
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Uplinx Question

I have been using it as a leave behind/as-built documentation. Not that most 
people read it, but rather put documentation on a scale and weight to determine 
how sufficient it is.

Dennis Heim | Emerging Technology Architect (Collaboration)
World Wide Technology, Inc. | +1 314-212-1814
[twitter]https://twitter.com/CollabSensei
[chat]xmpp:dennis.h...@wwt.com[Phone]tel:+13142121814[video]sip:dennis.h...@wwt.com
Innovation happens on project squared -- http://www.projectsquared.com


From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Kevin 
Przybylowski
Sent: Friday, December 05, 2014 9:23 AM
To: Sean Knight
Cc: cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Uplinx Question

Yep, it gets a nod from me as well.  The templates are easy to modify if you 
want to add logos or modify the format.  I would suggest you grab the demo, run 
it against your lab.  It's well worth it to have a comprehensive document on 
either your own cluster (as it has some management tools), or if you're looking 
to add to your turn-over documents.

It's a no brainer and they seem to release updates very quickly to keep up with 
new releases of CUCM .


From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Sean 
Knight via cisco-voip
Sent: Friday, December 5, 2014 10:07 AM
To: 'cisco-voip@puck.nether.net'
Subject: [cisco-voip] Uplinx Question

Has anyone used this software to manage phone?  If so how are your thoughts on 
it?

http://www.uplinx.com/reporttool-usd.htm

Thanks,

Sean E. Knight

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[cisco-voip] Extension Mobility login is unavailable (23)

2014-11-27 Thread Leslie Meade
Client is getting this error when they try to use EM.
The error tells me that it is to do with EMCC and they are not configured for 
this, any ideas ?

I have restarted the EM service in the cluster and it was working before hand.

Cheers

Leslie

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Re: [cisco-voip] Extension Mobility login is unavailable (23)

2014-11-27 Thread Leslie Meade
Yes to all. It is not just one site. I have tried other phones in the cluster 
and it is also happening to them as well.



From: Ryan Huff [mailto:ryanh...@outlook.com]
Sent: Thursday, November 27, 2014 8:31 AM
To: Leslie Meade; cisco-voip cisco-voip@puck.nether.net
Subject: RE: [cisco-voip] Extension Mobility login is unavailable (23)

Is the phone subscribed to the Extension Mobility Service?
Is Extension Mobility capabilities enabled on the phone that the user is trying 
to log into?

From: leslie.me...@lvs1.commailto:leslie.me...@lvs1.com
To: cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net
Date: Thu, 27 Nov 2014 16:21:59 +
Subject: [cisco-voip] Extension Mobility login is unavailable (23)
Client is getting this error when they try to use EM.
The error tells me that it is to do with EMCC and they are not configured for 
this, any ideas ?

I have restarted the EM service in the cluster and it was working before hand.

Cheers

Leslie


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Re: [cisco-voip] Unity sending the *

2014-11-23 Thread Leslie Meade
I know that works well, but the need is to have the end user press * and 
ring the extension from a call handler



Leslie Meade


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Mobile:778.228.4339 | Main: 604.676.5239
Email: leslie.me...@lvs1.commailto:leslie.me...@lvs1.com

From: Mark Drucker [mailto:drucker.m...@gmail.com]
Sent: Sunday, November 23, 2014 11:19 AM
To: Leslie Meade
Cc: cisco-voip (cisco-voip@puck.nether.net)
Subject: Re: [cisco-voip] Unity sending the *


Hi Leslie,

You can send it through a translation pattern that will prepend the *.


Mark

On Sun, Nov 23, 2014 at 9:55 AM, Leslie Meade 
leslie.me...@lvs1.commailto:leslie.me...@lvs1.com wrote:
Can unity send a * ?

I have a route pattern that needs a * to be entered for proper routing. While 
this is working fine from a phone, I am having a hard time getting Unity to 
send it.
I would imagine that I cannot due to the fact that * is a regex number and it 
will not send it.


Cheers

Leslie


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--
Mark Drucker CCIE #39606
Sr. Architect - Collaboration
(925) 321-5791
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Re: [cisco-voip] Jabber contact disappears consistently

2014-11-22 Thread Leslie Meade
I also have seen the same issue, and put everyone onto the same server

Leslie Meade


..
Mobile:778.228.4339 | Main: 604.676.5239
Email: leslie.me...@lvs1.commailto:leslie.me...@lvs1.com

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Jason 
Aarons (AM)
Sent: Saturday, November 22, 2014 11:33 AM
To: Josh Warcop; Pavan K; cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Jabber contact disappears consistently

I complete agree, put all Jabberusers on single server.  Saw the same bugs in 
10x.

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Josh 
Warcop
Sent: Friday, November 21, 2014 3:48 PM
To: Pavan K; cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Jabber contact disappears consistently


Do to the bugs I've seen I have been pinning all users to a single node in the 
cluster and just having HA failover.


Date: Fri, 21 Nov 2014 11:31:39 -0600
From: pav.c...@gmail.commailto:pav.c...@gmail.com
To: cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net
Subject: [cisco-voip] Jabber contact disappears consistently

We have an interesting problem on a new jabber deployment that has us stumped. 
Wonder if anybody else saw this.

Two jabber servers with ha enabled and balanced users. Using 10.5su1 for jabber 
windows and ucm/imp.

Leveraging sip directory uri as the IM scheme due to a multi forest environment 
with duplicate Samaccountnames across domains.

UserA contact list has userB on it. Folks can im each other without any 
problem. If we move userB from his imp server to another server in the same 
subcluster, userB disappears from userA's contact list.

Repeatable across multiple users with different userA and userB and happens 
every time regardless of moving them from server1 to server2 or vice versa.

Using router to router communication between nodes and default jabber-config.

Any ideas ?
Have a TAC case open but its going nowhere.


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itevomcid
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[cisco-voip] Reporting packages

2014-10-16 Thread Leslie Meade
I am looking for a call reporting application that will also take into account 
abandoned calls. The client is not running UCCX and wants to get reports on 
this.
Now I know there are lots of applications that will do in, out and how long 
they were on a call for (that's just normal call accounting), but they have 
asked for abandoned calls

Cheers


Leslie
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Re: [cisco-voip] Reporting packages

2014-10-16 Thread Leslie Meade
I should add this

The client wants a  list of calls though the day, or any given time for any 
given number or group for example

ext 1234 had 30 inbound calls, 10 outbound calls, 29 abandoned average time on 
call was 4 minutes



From: Leslie Meade
Sent: Thursday, October 16, 2014 7:42 AM
To: 'Brian Meade'
Cc: cisco-voip (cisco-voip@puck.nether.net)
Subject: RE: [cisco-voip] Reporting packages

That’s the point they do not want any call queueing, I had it configured for 
hunt groups but that was too limiting, and I just reverted back to directly 
calling the end user.
I see  in  7.1 there is sections in the CCDR Admin guide that they talk about 
CDR abandoned calls

http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/service/7_0_1/cdr-defs/cdradmin/cdrex.html#wpmkr1201032

But cannot find anything in the 10.X CDR admin guide


From: bmead...@gmail.commailto:bmead...@gmail.com [mailto:bmead...@gmail.com] 
On Behalf Of Brian Meade
Sent: Thursday, October 16, 2014 7:37 AM
To: Leslie Meade
Cc: cisco-voip (cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net)
Subject: Re: [cisco-voip] Reporting packages

What's your definition of an abandoned call without contact center?  Are you 
doing call queuing on the hunt pilots?

Brian

On Thu, Oct 16, 2014 at 10:30 AM, Leslie Meade 
leslie.me...@lvs1.commailto:leslie.me...@lvs1.com wrote:
I am looking for a call reporting application that will also take into account 
abandoned calls. The client is not running UCCX and wants to get reports on 
this.
Now I know there are lots of applications that will do in, out and how long 
they were on a call for (that’s just normal call accounting), but they have 
asked for abandoned calls

Cheers


Leslie

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[cisco-voip] Jabber/ Phone DND

2014-09-11 Thread Leslie Meade

Is there a reason why DND on an IP Phone does not show as DND on a jabber 
client ?

All other states are working except for DND

Cheers

Leslie


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Re: [cisco-voip] Jabber/ Phone DND

2014-09-11 Thread Leslie Meade
I have two jabber end points with ip phones as well.

User A puts his desk phone into DND, this presence state is not showing on his 
or other users Jabber clients.
However if User A puts their Jabber into DND the desk phone will go into DND 
and since jabber was put into DND presence is shown.

I think it would have to so with shared lines and DND. If you have shared lines 
and place one into DND the other still ring and not show that one is in DND


From: bmead...@gmail.com [mailto:bmead...@gmail.com] On Behalf Of Brian Meade
Sent: Thursday, September 11, 2014 9:03 AM
To: Leslie Meade
Cc: cisco-voip (cisco-voip@puck.nether.net)
Subject: Re: [cisco-voip] Jabber/ Phone DND

Are you saying that a shared line on Jabber isn't changing to DND when the IP 
Phone goes DND?  Or User B on Jabber observing User A doesn't see User A go 
Busy when User A's IP Phone is set to DND?

On Thu, Sep 11, 2014 at 11:35 AM, Leslie Meade 
leslie.me...@lvs1.commailto:leslie.me...@lvs1.com wrote:

Is there a reason why DND on an IP Phone does not show as DND on a jabber 
client ?

All other states are working except for DND

Cheers

Leslie



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[cisco-voip] Jabber deployment Contact list

2014-08-18 Thread Leslie Meade
I am trying to find out if there is a bulk import function in Jabber to where a 
pre populated list of contacts can be installed ?
As far as I can tell that there is not. Each user will have to manually import 
from a XML file.

There is no bulk way of doing this ?

Cheers



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Re: [cisco-voip] ATA 190?

2014-08-14 Thread Leslie Meade
They look exactly the same including the gui, and you have to install a COP 
file to get them to run. Even 10.5 does not have them included by default.



From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Brian 
Meade
Sent: Thursday, August 14, 2014 9:03 AM
To: Matthew Loraditch
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] ATA 190?

I think it was a cost savings thing.  The new ATA 190 runs on the same hardware 
as the SPA112 I believe.

On Thu, Aug 14, 2014 at 11:45 AM, Matthew Loraditch 
mloradi...@heliontechnologies.commailto:mloradi...@heliontechnologies.com 
wrote:
So apparently this is the new ATA model and 187s are End of Sale this year. Are 
there any actual improvements between the 187 and this? Was it just a component 
shortage, perhaps?

Thanks!


Matthew G. Loraditch – CCNP-Voice, CCNA-RS, CCDA
1965 Greenspring Drive
Timonium, MD 21093

direct voice. 443.541.1518tel:443.541.1518
fax.  410.252.9284tel:410.252.9284

Twitterhttp://twitter.com/heliontech  |  
Facebookhttp://www.facebook.com/#!/pages/Helion/252157915296  | 
Websitehttp://www.heliontechnologies.com/  |  Email 
Supportmailto:supp...@heliontechnologies.com?subject=Technical%20Support%20Request
Support Phone. 410.252.8830tel:410.252.8830



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Re: [cisco-voip] Message when call forward all to vociemail

2014-08-08 Thread Leslie Meade
That's the strange part there is nothing coming back in the route plat report 
or Extension report

Leslie Meade


..
Mobile:778.228.4339 | Main: 604.676.5239
Email: leslie.me...@lvs1.commailto:leslie.me...@lvs1.com

From: Chris Ward (chrward) [mailto:chrw...@cisco.com]
Sent: Friday, August 08, 2014 10:04 AM
To: Leslie Meade; cisco-voip (cisco-voip@puck.nether.net)
Subject: RE: Message when call forward all to vociemail

This looks like a CUCM call routing issue. I doubt CUC logs or queries will 
help.

The DN 5000, can you run a route plan report and see where it shows up? In a 
Call Forward scenario including CFA, you can specify a different CSS than a 
normal would use so it's possible that call flow is different for a CFA vs a 
normal call even when dialing the same number.

+Chris
TME - MediaSense and Unity Connection

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
leslie.me...@lvs1.commailto:leslie.me...@lvs1.com
Sent: Friday, August 08, 2014 12:45 PM
To: cisco-voip (cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net)
Subject: [cisco-voip] Message when call forward all to vociemail

I am trying to track down if this is an actual loop or not. I have gone through 
all the voicemail ports and the configuration as well as Unity and cannot see 
where the issue lies.
The call goes to voicemail fine and there seem to be no issues.

Would there be a SQL command I could run to help me find anything this is 
related to the extension ?


Cheers

Leslie

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Re: [cisco-voip] Selfdeployment

2014-07-11 Thread Leslie Meade
Tired that there was a SCCP template that I deleted but still no go.


Leslie Meade


..
Mobile:778.228.4339 | Main: 604.676.5239
Email: leslie.me...@lvs1.commailto:leslie.me...@lvs1.com

From: Ryan Ratliff (rratliff) [mailto:rratl...@cisco.com]
Sent: Friday, July 11, 2014 11:10 AM
To: Leslie Meade
Cc: Bill Riley; cisco-voip voyp list
Subject: Re: [cisco-voip] Selfdeployment

Look and see if a phone button template for that phone is hanging around.

This sql query may be handy:
select pt.name from phonetemplate pt where pt.name matches '*-Individual 
Template' and not exists (select 1 from device where fkphonetemplate = pt.pkid)

-Ryan

On Jul 11, 2014, at 1:53 PM, Leslie Meade 
leslie.me...@lvs1.commailto:leslie.me...@lvs1.com wrote:

Yea the profile is not the default one but the system still tells me that it 
cannot be attached to the profile.
I delete the phone and set it to auto register again and still get the same 
thing



Leslie Meade


..
Mobile:778.228.4339 | Main: 604.676.5239
Email: leslie.me...@lvs1.commailto:leslie.me...@lvs1.com

From: Bill Riley [mailto:b...@hitechconnection.net]
Sent: Friday, July 11, 2014 10:46 AM
To: Leslie Meade; cisco-voip 
(cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net)
Subject: Re: [cisco-voip] Selfdeployment

Sorry, you should assign a user profile to the user that is not the system 
default.


On 7/11/2014 12:34 PM, Bill Riley wrote:
Does the phone auto-register again?

Also I believe the work around is to not use a profile that is not the default 
and assign it to the user?


On 7/11/2014 11:33 AM, Leslie Meade wrote:
While I wait on a response back form TAC.

CUCM 10.5 with self-deployment running. I got one phone registered and it 
worked well. Then I decided to delete that phone and test it again.
Now I get the error of This device could not be associated to your account. 
Please contact the System administrator to complete provisioning.

I have used this doc.. and made sure that the UDT is on the user profiles.

I am also aware of the bug that states that the standard/default profiles can 
be broken and the work around is to manual create them or assign them to the 
user.

I look at the debug and see the devices that I am trying to register report 
that SEP005056911998 ~ 1033/2 ~  inside isAlreadyProvisioned
I assume that this is what is causing it to fault.

Any ideas ?







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[cisco-voip] Sage CRM and uccx

2014-07-03 Thread Leslie Meade
Does sage crm support integration into uccx ?


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Re: [cisco-voip] Cisco Call Recording Solution

2014-05-28 Thread Leslie Meade
While the mediasense app is easy to get running the gui for play back is woeful 
at best and no reporting.

Mediasence is a oem of calabrio.




Sent from Samsung Mobile


 Original message 
From: Chris Ward (chrward)
Date:05/28/2014 10:15 (GMT-08:00)
To: Ken Rhodes ,cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Cisco Call Recording Solution

As the TME for MediaSense, I am obligated to tell you to take a look at Cisco 
MediaSense. MediaSense provides audio recording via the phone's built-in bridge 
or CUBE media-forking (which can be dial-peer controlled or CUCM controlled as 
of 10.0). We are also offering video call recording in our 10.5 release using 
CUBE dial-peer forking.

We have a fairly extensive set of APIs available as well which allows our 
partner applications to integrate with MediaSense to provide additional or 
supplemental functionality.

Also, maybe one of the most attractive parts of MediaSense is that it is 
licensed on a concurrent port basis whereas most other recording products are 
licensed on a per user basis which can be more costly.

Let me know if you have any questions. If you are interested in more detail, 
you should reach out to your Cisco account team. Or, if you have access, I just 
did a MediaSense presentation at CiscoLive that should be online for viewing.

+Chris
TME - MediaSense and Unity Connection

-Original Message-
From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Ken 
Rhodes
Sent: Wednesday, May 28, 2014 1:02 PM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] Cisco Call Recording Solution

Does anyone have a good suggestion about a CUCM call recording solution? We 
currently have Zoom, however the sales and customer service has been lacking so 
we are looking at other providers.

Thanks

Sent from my iPhone
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[cisco-voip] Recording Proiles

2014-03-27 Thread Leslie Meade
I have a client that is using the 88XX model of phone with KEM's. They also 
have a recording profile configured and working with no issues.
However when there is a speed dial configured(either on the device or the KEM) 
the call is not being recorded. When we place a packet capture on the Nice 
software we do not see and packets hitting it.
Use a line and we see traffic being sent to the server. Now is it due to we can 
not a recording profile on a speed dial that is causing this ? or the BIB is 
not firing up when the speed dial is selected. I have not checked on the server 
as of yet.


If we span the recording the call gets recorded.

Leslie

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Re: [cisco-voip] Recording Proiles

2014-03-27 Thread Leslie Meade
Every line has a recording profile on it. If I select the speed dial it will 
select the primary line.
If I make a call from the primary line it records that call. It's only speed 
dial that's not functional.



Leslie Meade


..
Mobile:778.228.4339 | Main: 604.676.5239
Email: leslie.me...@lvs1.commailto:leslie.me...@lvs1.com

From: Pavan K [mailto:pav.c...@gmail.com]
Sent: Thursday, March 27, 2014 5:29 PM
To: Erick
Cc: Leslie Meade; cisco-voip (cisco-voip@puck.nether.net)
Subject: Re: [cisco-voip] Recording Proiles


Leslie,
When the speed dial is depressed, which line is being used to make the call ?
Perhaps that line does not have a. Recording profile. What happens when you 
dial the same number manually on the same line ?

I am reasonably sure that speed dials do not have anything to do with recording 
profiles.
On Mar 27, 2014 7:18 PM, Erick 
erick...@gmail.commailto:erick...@gmail.com wrote:
Is there only one line (dn) on the phone or multiple? If multiple maybe the 
speed dial is using a line without a recording profile.

I don't have a 88xx handy but will try from a speed dial on a 99xx if I get a 
chance tomorrow.

Sent from my iPhone

On Mar 27, 2014, at 5:33 PM, Leslie Meade 
leslie.me...@lvs1.commailto:leslie.me...@lvs1.com wrote:
I have a client that is using the 88XX model of phone with KEM's. They also 
have a recording profile configured and working with no issues.
However when there is a speed dial configured(either on the device or the KEM) 
the call is not being recorded. When we place a packet capture on the Nice 
software we do not see and packets hitting it.
Use a line and we see traffic being sent to the server. Now is it due to we can 
not a recording profile on a speed dial that is causing this ? or the BIB is 
not firing up when the speed dial is selected. I have not checked on the server 
as of yet.


If we span the recording the call gets recorded.

Leslie

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[cisco-voip] CUCM 10 Phones

2014-03-05 Thread Leslie Meade
I know that the 1st gen phones are end of life, but they should still connect 
to a callmanager 10 environment ?
I have a client that wants to upgrade the system, but at this time not willing 
to upgrade the handsets.
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