Does anyone know what kind of performance impact the API has on ASR or ISR 44xx
routers running CUBE? I’ve used Securelogix with other SBC’s in large
Enterprise and Call Center deployments. The SBCs use ENUM to query Securelogix.
There is virtually no CPU impact on the SBC using ENUM even when
Is there a way to change the way Cisco Jabber clients (11.5) process SIP INFO?
I see the client sending SIP INFO related to RFC 5168 (XML Schema) and would
like to disable this. Thank you.
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https:
Yes. Using Oracle Internet Directory, which is based on OpenLDAP.
> On Apr 18, 2017, at 1:28 PM, Tim Frazee wrote:
>
> anyone using openldap or apacheDS for directories instead of the de-facto
> active directory?
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> ci
ill never get the same
> level of control via CUCM
>
> -Ankur
>
>
> On Tue, Dec 20, 2016 at 10:39 PM, Mark Holloway <mailto:m...@markholloway.com>> wrote:
> Hi all. Can SME fork SIP Invites to two different destinations? For example a
> call originates from t
Hi all. Can SME fork SIP Invites to two different destinations? For example a
call originates from the PSTN and SME forks the invite to CUCM and Skype.
Couldn’t find anything in SRND specific to SME. Only CUCM (Mobility) and Spark.
PSTN SIP TRUNK
|
|
CUBE
|
|
SME—— Skype
|
|
CUCM
lt;>
> From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of
> Norton, Mike
> Sent: Wednesday, October 12, 2016 6:50 PM
> To: Mark Holloway ; voip puck
>
> Subject: Re: [cisco-voip] Cisco and Lync/Skype with Media Bypass
>
>
>
> Are you usi
This is more of a Microsoft question but I’ve searched everywhere and cannot
find an answer. I’m currently working on a design to integrate CUCM and Skype
for Biz. The Skype client will have media bypass enabled. All the Lync/Skype
capacity calculators and TechNet articles talk about how many ca
te all copies.
>
>
> -Original Message-
> From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of
> Mark Holloway
> Sent: Thursday, March 03, 2016 4:12 PM
> To: cisco-voip@puck.nether.net
> Subject: [cisco-voip] Jabber and P-RTP-Stat
>
> Is it
Is it true the latest Jabber client does not support P-RTP-Stat in a SIP BYE
message? I searched (quickly, on my phone) but couldn't find the answer.
Sent from my iPhone
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Does anyone know if CUCM 11 still requires a fixed audio source to be
Multicast? The problem is when using an SBC other than CUBE for PSTN SIP
Trunking there is no way to convert the stream to Unicast that I’m aware of.
I’m open to suggestions.
Thanks,
Mark
_
After digging a bit more I think IM&P server is the right server to query using
REST.
> On Sep 16, 2015, at 8:46 AM, Mark Holloway wrote:
>
> Does anyone know if it’s possible to query CUCM over HTTP/HTTPS to see if a
> user is available to take a call? (ie. is the phone re
Does anyone know if it’s possible to query CUCM over HTTP/HTTPS to see if a
user is available to take a call? (ie. is the phone registered, is DND on or
off, are all lines occupied and the user is busy)
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e procname = "sqlfunctions"
>
> Thanks,
> Nick
>
> Disclaimer: IANADBA
>
> From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of
> Daniel Pagan
> Sent: Monday, September 14, 2015 10:07 AM
> To: Daniel Pagan; Mark Holloway; voip puck
> Subje
Hi all. When CUCM sends a SIP Invite to shared lines on SIP phones the user
portion contains some sort of (what appears to b) a randomly calculated
alpha-numeric string. Is there any information or documentation available on
how Cisco actually calculates what this string will be?
Here is an ex
Do you have a Wireshark capture of the SIP signaling for a failed call?
> On Jul 13, 2015, at 9:55 AM, Michael T. Voity wrote:
>
> Hello,
>
> Before we installed our Cisco CM 10.5.2 system everything here at the
> University is fed from a Nortel Avaya 81c / CS1000 system. The Telcom group
By default phones with SIP firmware using shared line appearances do not
register to CUCM for the shared lines, just the primary number. Is there a way
to force the shared lines to register? I’m trying to get remote users who
proxy through an SBC (using SIP) to register all lines (primary + sha
For call recording without analytics the Acme Packet/Oracle ISR (Interactive
Session Recorder) is the way to go. It has been deployed in some of the largest
CUCM/UCCE/CVP contact centers in the world. Currently it supports 24,000
concurrent calls in a single cluster in 8U of rack space. It suppo
Typically you can disable SIP INVITE AUTHENTICATION on PBX’s. What kind of PBX
is it?
> On May 14, 2015, at 1:44 AM, Tim Smith wrote:
>
> Hi Claiton,
>
> I don’t think this has changed recently.
> You can’t do a SIP REGISTER from CUCM directly on a trunk.
>
> You need to have something i
Too bad, because Perimeta is terrible.
>
> HCS-CC requires an underlying infrastructure that is pretty extensive and
> requires A2Q.
>
> You have to have a Perimeta
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suppose there
is any way to use a router to convert the stream to Unicast before it hits the
SBC?
> On Apr 13, 2015, at 4:42 PM, Brian Meade wrote:
>
> Are you just talking about for fixed audio source?
>
> On Mon, Apr 13, 2015 at 4:07 PM, Mark Holloway <mailto:m...@markho
Are there any plans for Cisco to allow (bring back) Unicast MoH on 10.5? If
trunking CUCM to a non-Cisco device there are cases where MoH breaks because
not every endpoint will support Multicast.
Thanks,
Mark
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I don’t think CUBE licensing is perpetual. You should call Cisco to get a
definite answer but the folks I know had to rip and replace.
> On Apr 10, 2015, at 4:28 PM, Countryman, Edward
> wrote:
>
> Does anyone know if I would be allowed to move SUBE session licensing from my
> 3925 router to
Just go with the Acme Packet SBC. ;) It’s the most widely deployed SBC, HA
works flawlessly, it’s certified (and used) by all the major carriers and it’s
the only non-Cisco SBC certified by Cisco for CUCM/UCCE/CVP etc.. It doesn’t
hurt that the web interface has built in ladder diagrams which i
Has anyone deployed CUCM as an application server in a VoLTE/IMS architecture?
Looks like CUCM officially supports this using ISC trunks facing an S-CSCF.
I’ve found very little documentation on this architecture from Cisco. I would
be interested if anyone can point me in a useful direction.
that perhaps there is a way this
can be accommodated within CUCM.
On May 30, 2014, at 12:20 PM, Brian Meade wrote:
> Can you send a CallManager SDI/SDL trace for one of these calls?
>
>
> On Fri, May 30, 2014 at 12:14 PM, Mark Holloway wrote:
> Yep, it’s TLS. Certific
> On Fri, May 30, 2014 at 11:41 AM, Mark Holloway wrote:
> I’ve got a non-Cisco SIP device sending SIP Invites to CUCM (SIP Trunk). The
> SDP from my device includes RTP and sRTP in the SIP Invite. Reading Cisco
> docs it looks like the way Cisco expects sRTP to work is the SIP Invi
I’ve got a non-Cisco SIP device sending SIP Invites to CUCM (SIP Trunk). The
SDP from my device includes RTP and sRTP in the SIP Invite. Reading Cisco docs
it looks like the way Cisco expects sRTP to work is the SIP Invite should only
include sRTP assuming if the call should be encrypted. If bo
The Acme Packet SBC has a very advanced SIP Header Manipulation regex based
language. It also supports LUA. You can write “plugins” for the SBC using LUA
to extend it’s functionality. Pretty cool for an edge appliance!
On May 5, 2014, at 5:12 AM, Tim Smith wrote:
> Hi guys,
>
> I love this c
I’ve seen instances where the SIP SDP contains either a=send or a=receive for
Blind Transfer on CUCM and that causes one way audio. On one instance this was
between CUCM and Lync. Fortunately the customer had an Acme Packet SBC and we
used SIP Header Manipulation Rules to fix it.
Can you get a
This is pretty simple. Put an Acme Packet SBC on the edge of the customer
premise to sit between CUCM and Lync SIP Trunks. It’s Lync certified and very
robust. It can also double as an SBC for PSTN SIP Trunks.
On Feb 3, 2014, at 11:19 AM, Michel L. M. B. Perez
wrote:
> Hello Guys,
>
> I ha
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