I had some weirdness with the variant using auto-provision all
(not auto-provision none as per the blog article)
!
telephony-service
srst mode auto-provision all
!
In this case I expected CBarge and privacy-button to work out-of-the-box.
(I have disabled single-button-barge on CUCM and
: +1.810.454.0130
Mailto: vma...@ipexpert.com
On Apr 10, 2012, at 2:27 AM, Anthony Alba wrote:
I had some weirdness with the variant using auto-provision all
(not auto-provision none as per the blog article)
!
telephony-service
srst mode auto-provision all
!
In this case I expected CBarge
Hello Juan,
Try 8.0.2 for VPIM; it's included in the demo license.
Of course, you should still use 7.0 for all other tasks.
On Sat, Mar 10, 2012 at 12:13 AM, Juan Lopez lopez.hernandez.j...@gmail.com
wrote:
how do people train this part with their own HW if VPIM is not in the CUC
demo
I would like to build-up a step-by-step IPMA Proxy mode checklist and
verification.
If you configure the IPMA route point (with DN a superset of Managers' DNs
like 5XXX), configure the IPMA Service Parameters
on both Pub/Sub and restart he IPMA service, ought the IPMA route point
appear as
Does your call from CUCM to B-ACD allow G.711?
If not, then the suggested way is to use another incoming dial-peer with a
prefix e.g. 44#3500.
This matches G.729, strip the prefix 44#, then this should match dial-peer 3500
and invoke the transcoder.
I hit exactly the same problem and this was
Hi list,
I am referring to the Cisco sample script that uses Session steps on pg
17-2 of
Cisco Unified Contact Center Express Scripting and Development Series:
Volume 1, Getting Started with Scripts 7.0(1)
Task 3.1
configure campus qos
* egress media in the priority queue
* egress signal in the second queue
* do not permit any queue to exceed 25% of the total allowable bandwidth
Solution:
int f1/0/2
auto qos voip cisco-phone
!--- adjust auto qos
int f1/02
srr-queue bandw shape 4 4 4 4
The script populates the variable at runtime with Set Enterprise Call
Info step; this happens in the Select Resource step before you connect
the caller to the agent.
Now, much to my surprise, I actually managed to get this to work and I saw
the field get updated.
The question I want to ask the
have, especially on VM Ports and BR1
Phones? As well make sure there is no DN 5600 floating around without being
assigned to any device.
Can you paste the q931 message from BR1 GW and coming into HQ gateway too?
Cheers,
On Thu, Jan 19, 2012 at 12:15 AM, Anthony Alba ascanio.al
Hello, this issue has surfaced in the past but no one email seems to
summarize the exact requirements to get Voicemail to work during AAR. I'd
like to give a go and get your feedback:
Task: BR1, a H.323 GW, is in AAR, Voicemail must work
1. BR1 Ph2 dials Voicemail external PSTN DID directly:
...I just did a check: in Workbook 2 Lab 6, Tasks 7.1, 7.2 we are trusting
the phones+HWIC-4ESW on both BR1 BR2 , the class-map used is
class-map match-all wan-rtp
match dscp ef
etc. etc
...so as I thought, the DSG is not consistent here...
On Wed, Jan 18, 2012 at 12:52 PM, Anthony Alba
Hi Randall,
You cannot control which file is used for the options:it is hardcoded in
the TCL script and not exposed
as a param for us to change; i.e., there is no way to point the
configuration to use another options audio file.
To change the options menu you have to replace
Hi, the solution guide uses dummy (unregistered) CTI route points in many
tasks purely to forward calls (CFA) to Unity Connection, either mailboxes,
live record, call handlers, greetings administrator etc
Examples:
Lab 1: Dummy CTI route point at DN 1113 for MeetMe task
Why not just use a
I am configuring multicast paging on CME.
ephone-dn 8
number no-reg primary
name Sales Page
paging ip 239.3.10.1 port 2000
ephone XX
paging-dn 8
Two directly connected phones Ph1 Ph2 receive multicast paging and the RTP
stream shows to 239.3.10.1.
However, two phones Ph5 Ph6, not
Hello,
In TEHO tasks, do you automatically configure a local backup, if not
explicitly stated in the task?
E.g Workbook 2 Lab 3
Previously we had configured 911/999, local, LD, international dialing at
all sites.
The TEHO task reads:
Configure Tail End Hopoff wherever possible throughout your
[Finally, got most of IPMA working after my earlier saga]
The only thing not working is Assistant's TransVM softkey and Transfer to
Voice Mail on Assistant Console.
I believe all my CSS/partitions are correct; also the Manager phone can use
the TrnsfVM.
Every other function of IPMA seems to be
Sorry for the noise: tracked the problem to proxy DN not having a vmbox
pilot; IPMA uses the proxy line VM pilot, it does not extract the VM pilot
from the Manager's line.
On Fri, Dec 30, 2011 at 7:51 PM, Anthony Alba ascanio.al...@gmail.comwrote:
[Finally, got most of IPMA working after
and working - whew -
what a complicated business IPMA proxy line is :-((
On Fri, Dec 30, 2011 at 8:34 PM, datucha123 datucha123 datucha...@gmail.com
wrote:
Does the Assistants Proxy Line has the VM Pilot Partition in its CSS?
On Fri, Dec 30, 2011 at 3:51 PM, Anthony Alba ascanio.al
the VM Profile
assigned to Assistants Proxy Line (Even the Managers DN did not had a VM
Profile assinged) and the TransfVM worked great.
On Fri, Dec 30, 2011 at 4:42 PM, Anthony Alba ascanio.al...@gmail.comwrote:
My error was the Proxy DN did not have a VM pilot set;
I assumed that IPMA
Gatekeeper license for 12.4-20T+ is very expensive.
HQ-RTR can run 12.4-15T without us students noticing any difference except
as you found out for the obvious cut-n-paste issues.
Interestingly, I accidentally misconfigured my home PSTN with 15.1 -
didn't notice any difference until I started
Hi,
What is the definitive requirement for having mls qos trust and
service-policy on an interface to work.
In the 12.44 command reference
Classification using a port trust state (for example, *mls qos trust* [*cos
* | *dscp* | *ip-precedence*] and a policy map (for example, *service-policy
Hello, I'm having a problem with the IPMA tomcat service;
during restart I'm getting the message
IPMA Application not started Servlet Name:Cisco IP Manager Assistant
Reason: Service failed to go active. Unable to create CTI provider App ID:
Cisco Tomcat Service
Any ideas?
I'm stumped as this
know if you are still facing problem.
Regards,
Mohammed Al Baqari
Sent from my iPhone
On Dec 26, 2011, at 6:27 PM, Anthony Alba ascanio.al...@gmail.com wrote:
Hello, I'm having a problem with the IPMA tomcat service;
during restart I'm getting the message
IPMA Application not started
Just curious:
1. You mean you RDP to the VM Lab PC and ssh from there to the servers?
Can you confirm you don't have routable/SSH access from the exam candidate
PC.
2. Does the candidate PC at least allow telnet access to the routers or
ONLY reverse telnet to the console server? I have been
easier to do it without
gateway called party transformation patterns (kind of defeats the object of
Called Party Transformation Patterns when you have to perform manipulations
on the RP or RL in combination with gw called party transformations).
Vik
On Dec 12, 2011, at 6:04 AM, Anthony Alba
Hi, in this task, the call from SA GK to remote PSTN GK is supposed to fail
(dialing 0119167) and we are supposed to use asn1 debugs to see the LRJ
(no route to destination).
BUT..
My call actually succeeded.
My question: is the un-bug in the initial PSTN config that is too liberal?
Should
configured.
Does your base config not include this command?
Vik Malhi – CCIE #13890
Managing Partner - IPexpert, Inc.
Telephone: +1.810.326.1444 ext 420
Fax: +1.810.454.0130
Mailto: vma...@ipexpert.com
On Dec 13, 2011, at 7:09 AM, Anthony Alba wrote:
Hi, in this task, the call from
Hi,
This is Lab 2 in the Five-Lab Handbook.
The the task: on SC (UK site +442077964XXX, MGCP gateway) the requirement
is to plus dial from
directory without EditDial +442077966596 but the phone display must show
To +442077966596
Normally globalized dial plan will not work; if I have one \+.!
Suggest you use rack rentals to familiarize yourself with all the hardware and
do a few labs.
Then get some local hardware phones connecting to the pod remotely. You can
then decide whether to do it yourself as it is expensive to put everything
together. The rack rental guide shows all the hw
If we allow a participant to put a conference on hold with MOH, i.e.,
Suppress MOH to Conference Bridge == FALSE
does the Conference Bridge need a unicast-only MRGL to reflect MOH back
to all the other participants?
I find that if the device pool MRGL has a multicast-enabled MRG then
although a
Hello,
Mobile Connect (SNUR) issue:
** using E.164 for remote destination e.g. +12123941234
** using globalized dial plan with one route pattern \+.!
** using one translation pattern \+.! (for plus dialing from directory)
whose CSS sees the global route pattern.
I do not want the devices to see
Bonjour Nicholas
I am in the same position, RS trying to move to Voice, I have just passed
the written.
I have also built a lab and I use rack rentals to see the configurations
and deliberate 'bugs'.
My lessons learnt so far for home rack:
** use Intel for your VMware server; the versions of the
that Hit Dial peer with Voice class codec ,
it make sense as the router though that he can support Both codecs
I hope this clarify the issue you saw
Ash
On Wed, Nov 30, 2011 at 8:36 PM, Anthony Alba ascanio.al...@gmail.com wrote:
Very strange: I can now get both inbound and outbound
Hi, I'm working on Vol 2 Lab 2: H.323 GK controlled trunk between CUCM to
CME.
This is different from Vol 2 Lab 1 task where we used a CUBE between CUCM
and CME.
My problem: I cannot get inbound G.729 calls from CUCM to CME SIP phones to
work: it clearly is some sort of codec issue; when I
The IOS gateway cam also use H.225 back to CUCM to affect the Called Number
display.
Amy covers this in the audio: if you want to control the display on the
phone, then do manipulation at the RP, even though this DM may be trumped
by RL details. You also have to disable the H.225 notification on
. Instead you are putting g729a G729ab.
** **
Regards,
Mohammed Al Baqari
** **
*From:* ccie_voice-boun...@onlinestudylist.com [mailto:
ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Anthony Alba
*Sent:* Thursday, December 01, 2011 3:38 AM
*To:* CCIE Study
*Subject
Very strange: I can now get both inbound and outbound calls to CME SIP
working with transcoder invoked at BR2-RTR. I cannot use voice-class codec
1 under the dial-peer.
This surprises me: why would voice class codec hurt the task?
voice class codec 1
codec pref 1 g729r8
codec pref 2 g711ulaw
Hi, the demo license lasts for 90 days; it might be easiest to snapshot the
VM before integration - you will reset the 90 day clock each time you
revert to the pre-integration snapshot.
On Tue, Nov 15, 2011 at 11:56 AM, Cisco Nut rafayc...@gmail.com wrote:
Hello-
Any one knows how I can
Hi Bruno,
Alas Unity Connection 7 has a separate VPIM license SKU (aka UNITYCN7-VPIM)
which is not covered by the demo license.
The rack rentals will have the VPIM license installed.
8.x versions will have VPIM covered in the demo license so if you clone
VM/upgrade to 8.x you might be able to
Hi Bruno,
I can confirm that Unity Express 7 to Unity Connection 8.02c works with
VPIM using the demo license of CUC8.
I was able to do the Vol2 Lab2 Q8.3 VPIM task per the solution guide.
I did not notice any difference compared with the Proctor Labs rack (CUC7).
I saw this in the license
Hello Duncan,
In System -- Enterprise Parameters -- CRS Application Parameters do you
see
IPCC Express Installed = true
Auto Attendant Installed = true
?
This is necessary to get IPCC Extension to be assignable.
Sometimes CUCM and UCCX get out of sync on whether they have been
integrated.
This
I do that - I have set up a basic Pub Sub with all the System stuff (CM
servers, date/time, device pools etc) preconfigured (to save time to jump
into any lab) but no phones, route plan, users ('cept for uccx admin user)
or anything else. When replication is stable I shutdown down both VMs and
Hi,
Looking at Unity Connection System Call Handlers: when a system call
handler takes a message where does the message go to?
How do you retrieve the message if the System Call Handler is used as,
e.g., an AA?
Thanks
Anthony
___
For more
For pure text files there is the well documented tcl trick
tclsh
puts [open flash:myfile.txt a+] {
!paste text file
}
tclquit
On 5 Nov 2011, at 01:57, John Smith cci...@yahoo.com wrote:
Forgive my ignorance, but if you needed to transfer a file from a PC to a
router and had no
I have encountered this too
Are you able ( does iOS allow it) to put the compress rtp config in the parent
shaper class?
On 4 Nov 2011, at 21:42, Nicolaers Luk luk.nicola...@quentris-gdfsuez.be
wrote:
Hi,
I'm trying to setup CB traffic shaping with CRTP.
This is the config of the policy
Current firmware 9.1(1)SR1 and later does plus dial for 7965G (and all other
type B phones). Press * for 1 second.
This is not the lab version, though it should be compatible with CUCM 7, so
you can test it out.
Problem: in Vol2 Lab7 DISA Dialing tasks 3.1 and 3.2 I can get either of
them to work separately but not together.
Any thoughts?
Task 3.1: when the Remote Destination calls in, show the CLID as the Remote
Destination# and not the Mobility User 4D extension.
Solution: Configure a special
– CCIE #13890
Managing Partner - IPexpert, Inc.
Telephone: +1.810.326.1444 ext 420
Fax: +1.810.454.0130
Mailto: vma...@ipexpert.com
On Oct 24, 2011, at 10:18 PM, Anthony Alba wrote:
Hello,
When I try to use MVA with MGCP hairpin I cannot make calls.
When I try to make a call I get
, Anthony Alba ascanio.al...@gmail.comwrote:
Hi Vik
Thank you for your reply. I am getting closer...I can now place calls via
MVA to
HQ locations G.711 but WAN calls do not connect media strems.
HQ-RTR#show call active voice compact
callID A/O FAX Tsec Codec typePeer Address
Hello,
When I try to use MVA with MGCP hairpin I cannot make calls.
When I try to make a call I get the IVR menu again
Dial 3945010 get IVR menu
Enter PIN #
Enter 1 1002# ( to make a call)
...instead of being connected I get back to the IVR menu.
I seem to be trapped in some sort of loop.
How do you allow a user to get to the
ccmuser Cisco Unified CM User Options page from Unity Connection (*not*
CUCM)?
https://10.10.210.13:8443/ccmuser/showHome.do
The username/password is accepted but I just get bounced back to the login
page.
All Feature Services/Network Services are running.
That's what I thought too...but once in a while in previous labbing by accident
I managed to get a login.
I want to make it reproducible...but can't recall the magic words!
On 23 Oct 2011, at 00:53, Inder Singh ising...@gmail.com wrote:
Hi Anthony,
I don't think it is possible to get to
If you use an anchored regex ^ it may not consider the 9 as explicitly matched
digit.
What happens if you use 9[2-9]..$ ?
On 21 Oct 2011, at 18:59, Ccie Voice v.c...@yahoo.com wrote:
Hi all,
I have very strange problem, and I need someone to help me to understand why?
I am
Slightly OT: is the US shifting to allowing 10D or 1+10D for all calls or is it
still highly telco specific? Is 7D still used? I thought that many cities
require 10D even for local calls.
Obviously-not-living-in-the-US Anthony
On 16 Oct 2011, at 16:38, Brian btmulg...@gmail.com wrote:
Hi Vik
Within the CME cluster everything functions: I am using the firmware
SIP41.8-4-1S.
3001 SCCP / 7961G
3005 SIP / 7691G
3002 SCCP / CIPC
1. CME can transfer between all 3XXX DNs/SIP,SCCP for an active call to
CUCM
2. CME SCCP can be holder, be held, and transferred within CUCM
3. CME SIP
in the SIP phone.
For (4) can you confirm the H323 Trunk in UCM has the following settings:
inbound FastStart is enabled, Wait for far end H245 TCS is disabled and MTP
Required is enabled. Also use g711 end to end (g711-DP/Region) and it should
work.
On Thu, Oct 13, 2011 at 10:20 AM, Anthony
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