hi All,
Any update on this?
On Wed, Oct 16, 2013 at 10:07 PM, sanity insanity
networksanitytoinsan...@gmail.com wrote:
Hi All,
I need your analysis based on the following Requirement and my
configuration. Please advise
Requirement :
---
On the HQ switch
Hi All,
I need your analysis based on the following Requirement and my
configuration. Please advise
Requirement :
---
On the HQ switch I need to configure the following...
1) TO ensure cos value of 5 mapped to dscp ef
2) On interface g1/0/4 make sure all incoming
Hello Guys,
I have 2 sites SB ( H323 gateway) and SC ( mgcp gateway) .
SB has the internal number of 3XXX and SC has the number range
of 4XXX.
SB has the voicemail pilot 2220 ( unity connection) and SC has
a voicemail pilot 4220 ( CUE )
I also have a HQ site with internal number range of
in SRST irrespective of what is
asked for?
-MJ
On Sun, Oct 6, 2013 at 11:35 PM, Bill whl...@gmail.com wrote:
when you are in SRST you should be able to send calls be 4 digit dialing
and this will address all your questions
Sent from my iPad
On Oct 6, 2013, at 12:11 PM, sanity insanity
Hello all,
I have configured presence and both softphone and deskphone modes , IM and
voicemail is working fine on the clients
However I have a question when I lift the handset of the phone ( hard phone
) that is assoicated
with the CUPC clients . I see that the presence status does not show
it was reiterated not
to use partial match, maybe in part because of the issue that I hit today.
Marty
On Tue, Sep 24, 2013 at 8:19 AM, sanity insanity
networksanitytoinsan...@gmail.com wrote:
Hi Guys ,
Thanks a lot for taking time out to reply to my question. It was really
helpful.
I
Resource-MVA and the call failed. But, after I copied my CUCM dial-peer
on the GW and changed the destination-pattern to 33$, it worked.
I hope this helps, and thanks for prompting me to stop procrastinating on
researching MVA!
Marty
On Wed, Sep 18, 2013 at 12:25 AM, sanity insanity
Hi Guys,
I have been trying to find the right way of configuring MVA. Below is my
configuration
Details:
=
My config is following
1) The dial-peers are set in the following way
dial-peer voice 102 voip
preference 2
destination-pattern 3300
session target ipv4:ip
Hello Guys,
Still waiting any update on this ?
On Mon, Sep 9, 2013 at 4:22 PM, sanity insanity
networksanitytoinsan...@gmail.com wrote:
hi Guys,
In the normal mode when wan is up I can call into the cue ( on site c )
through jtapi . However
when the wan link breaks and the when my
hi Guys,
In the normal mode when wan is up I can call into the cue ( on site c )
through jtapi . However
when the wan link breaks and the when my site c router and phones fall
into srst and then try placing calls to the cue using sip dial peer I
hear the following prompt - *I'm sorry*,
typo : point 2 :- 2) The ports and cti route points are correctly added
under application user . The same is mention in the cue cli
On Wed, Jul 31, 2013 at 9:55 PM, sanity insanity
networksanitytoinsan...@gmail.com wrote:
Hi All,
Need to know how to recover from this issue if it occurs
Hi All,
Need to know how to recover from this issue if it occurs in the lab.
1) CUE can ping callmanger . Also callmanger can ping cue .
2) The ports and route patterns are correctly added under application user
. The same is mention in the cue cli
3) The ctiports are mentioned in the cue
Hi Aman,
1) There have been no ephones created prior to going into srst
2) Also ephone-template and ephone-dn-template created before going into
srst.
3) still we see the above issue
4)Also the Xfer-to-VM line on the site C phone does not get downloaded to
srst mode when the phone goes into
hi All,
I have the following config for srst...
telephony-service
srst mode auto-provision all
srst dn line-mode octo
max-ephones 10
max-dn 10
ip source-address cme ip address port 2000
live-record 4700
voicemail 4220
max-conferences 8 gain -6
moh music-on-hold.au
transfer-system
Hello Guys,
I am just wondering if I can stop webpages on the callmanger from timing
out as I have to login again and again each time in logs out.
The 2 webpages I am interested in are :-
CUCM admin pages
OS page on CUCM
-MJ
___
For more
hi Guys,
In srst I use the following config...
telephony-service
srst mode auto-provision all
srst dn line-mode octo
1)Do I also need to configure srst dn template srst ephone template ?
2)what are best practices for setting up the cue in srst mode ? If possible
include details of..
-mwi
hi Guys,
Still waiting to hear back
Thanks again
On Sun, Jun 9, 2013 at 10:22 PM, sanity insanity
networksanitytoinsan...@gmail.com wrote:
hi Guys,
Thanks for your replies.
If the DB replication does go out of sync is there any troubleshooting
step that can be executed . Hope
hi Guys,
Thanks for your replies.
If the DB replication does go out of sync is there any troubleshooting step
that can be executed . Hope it does not take too much time to sync since
we would then loose the time to complete other tasks in the process.
1) Also does restart of NTP on
Hi All,
1) How long is a ccie written ( Passed ) exam valid?
2) If you fail a lab attempt while your ccie written is valid does your
ccie written exam validity get extended?
3) Where can I check the validity of my written exam ( passed) status? I
checked the Cisco certification website but it
hi Guys,
Still waiting for a reply.
Please assist.
-MJ
On Wed, Jun 5, 2013 at 6:34 AM, sanity insanity
networksanitytoinsan...@gmail.com wrote:
Hello All,
I have the following questions
1) which is the best way to bind the service module to the router is it
requried to bind
Hello All,
I have the following questions
1) which is the best way to bind the service module to the router is it
requried to bind it to the loopback of the router or the interface voice
vlan? Also is there any other routing related problems that we need to
take account before doing this?
Hello All,
I have a workbook question on How do I get this ringlist available at the
line
level of the phone . For example..
Line 2001
* Cisco Techno
* Classic Ring
How do I modify the Ringlist.xml file to achieve the above in the shortest
way?
-MJ
hi All,
I have the following config
ephone-dn 3
number 4001
label 4001
description +85224044001
name +85224044001
call-forward busy 4220
call-forward noan 4220 timeout 20
!
ephone-dn 5
number *4001
call-forward all 4220
!
ephone 2
mac-address 1089.CF01.7C99
ephone-template 1
and
drop-through-option
to send the call to param aa-hunt1.
en_bacd_music-on-hold.au - moh should be configured under telephony
service. Else the call will drop after All agents are currently busy...
--
Anirudh M Mavilakandy
On Sun, May 19, 2013 at 6:44 AM, sanity insanity
hi Guys,
Any update?
On Wed, May 15, 2013 at 11:15 AM, sanity insanity
networksanitytoinsan...@gmail.com wrote:
Hi Guys,
If there is a requirement using BACD that If agent did not answer the
call in 10 s , the call should be routed to next
agent. If agents are busy they should hear All
Hi Guys,
If there is a requirement using BACD that If agent did not answer the call
in 10 s , the call should be routed to next
agent. If agents are busy they should hear All agents are currently
busy...
The following prompts are available on the tftp serer ( ip address
X.Y.X.X) and this needs
hi All,
I have a few questions in relation to H323 troubleshooting
We are required to provision a proof of concept (POC) H323 trunk between
CUCM and voip service
provider . The cucm should send H323 traffic to service provider on ip
address XX.XX.XX.XX (public Ip address) . This call should
hi Guys,
Have not heard back on this...
On Thu, Apr 11, 2013 at 9:29 PM, sanity insanity
networksanitytoinsan...@gmail.com wrote:
Hi Guys,
I have a MGCP gateway setup when subscriber fail publisher will work
Call 911 while the call is active shut down the cucm sub services. Make
sure
Hi Guys,
I have a MGCP gateway setup when subscriber fail publisher will work
Call 911 while the call is active shut down the cucm sub services. Make
sure when
Complete to bring up the subscriber services so you can continue with the
tasks on the lab.
Capture the following
-- Backup call agent
should give you guidance here.
Given the way you presented the question, I would adjust retries to 0 and
then it should meet expected requirements.
-Bill
--
William Bell
blog: http://ucguerrilla.com
twitter: @ucguerrilla
On Mar 26, 2013, at 6:26 AM, sanity insanity wrote:
hi guys
hi guys,
Any update ? I Don't have this working...
-MJ
On Mon, Mar 25, 2013 at 3:02 PM, sanity insanity
networksanitytoinsan...@gmail.com wrote:
hi William,
Thanks for your reply.
I have now added the following for step (b) and added (c) ,(d),(e)
-start
-accept
-play prompt
Step (b) doesn't make sense to me. If you successfully redirect the
contact then the script logic shouldn't go to the queueLoop. You should
terminate.
--
William Bell
blog: http://ucguerrilla.com
twitter: @ucguerrilla
On Mar 21, 2013, at 1:03 PM, sanity insanity wrote:
Hi All
Hi All,
Need your help.
I am configuring DNs 4101 4102 ( both DNs are uccx agent extensions).
Calls to 4000 should here a greeting Press 1 to be transferred to priority
agent or stay online for next available agent . If the caller presses 1 ,
calls should be transferred to 4001.
Otherwise it
Hello All,
I have been trying this config for MVA for close to 2 weeks now and it
does not work . Here are the details
The Issue :
==
I am trying to Intiate a Call from PSTN phone to site B gateway (H323)
3033300 it should ask for
authentication once authenticated press 1
hi Guys,
I have to Configure IP Phones and gateways in such as way that all calls
within same site
should use G711 Codec. Also, all calls between the sites to remote IP
phones and
gateways should use G729 Codec.
RSVP Call Admission Control (CAC) between HQ and branch site based on
bandwidth
meet the requirements of
the task. Remember you have to meet the demands of calling and caller info
in SRST as well.
Sent from my iPad
On Mar 2, 2013, at 11:47 PM, sanity insanity
networksanitytoinsan...@gmail.com wrote:
Hello All,
I have route patterns set as the following on the CUCM
Hello All,
I have route patterns set as the following on the CUCM ( callmanger)
for my site B ( h323 gateway) for emergency , local , long dist and
International as the following
911 local route group
9.[2-9]XX--- local route group ( strip predot)
hi all,
what is an easy way to script this : -
Incoming calls from +44 number will be serviced by 2102 phone(line 2 of HQ
phone - IPCC extn) only and all other calls will
be routed to the normal queue.
If the call hits the queue, the user should hear the defaults prompt, all
are agents are
)
On Sun, Jan 20, 2013 at 7:22 PM, sanity insanity
networksanitytoinsan...@gmail.com wrote:
What is the easiest way of getting a customized background image and
configuring it for cme ip phones . Given that there is no TFTP server , we
just need to use the callmanger . Please illustrate
hi Guys,
Just need to know the following
1) Is it required for every END USER to be associated with their
corresponding username on Directory number level of the phone or is it
enough if we can just do this on the END USER page of the callmanger by
associating the MAC address of the
Hi Guys,
I am disabling the corporate directory on my phone ( ip phone 7975) using
the following procedure...
1)enter a false URL for the directory in enterprise parameter
2)using common phone profile on the phone and set it to external URL
3) Now I press the directory button and it says host
Hi Guys,
I am disabling the corporate directory on my phone ( ip phone 7975) using
the following procedure...
1)enter a false url for the directory in enterprise parameter
2)using common phone profile on the phone and set it to external url
3) Now I press the directory button and it says host
qos to do this and once you know this you can use
auto qos or just do it manually for FRF12.
On Sun, Dec 30, 2012 at 6:44 AM, sanity insanity
networksanitytoinsan...@gmail.com wrote:
Hi Bill,
Thanks for your reply. I shall be therefore using the following config
without using
auto qos
to config a class-map, then policy-map, then your nested
policy-map with the shaping, then create your map-class and finally apply
that
Bill
On Dec 31, 2012, at 6:54 AM, sanity insanity
networksanitytoinsan...@gmail.com wrote:
hi Bill and All,
Thanks for your reply,
1) I checked
tried using the Cisco QOS SRND?
http://www.cisco.com/en/US/docs/solutions/Enterprise/WAN_and_MAN/QoS_SRND/QoS-SRND-Book.html
Look around at 3-27
On Mon, Dec 24, 2012 at 8:44 PM, sanity insanity
networksanitytoinsan...@gmail.com wrote:
hi Guys,
Can you explain as to how we can achieve
[mailto:
ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *sanity insanity
*Sent:* Πέμπτη, 27 Δεκεμβρίου 2012 2:52 μμ
*To:* ccie_voice@onlinestudylist.com
*Subject:* [OSL | CCIE_Voice] In cme -srst mode the calls to CUE do not
work
** **
hi Guys ,
In the cme -srst mode the calls
://www.cisco.com/en/US/docs/solutions/Enterprise/WAN_and_MAN/QoS_SRND/QoS-SRND-Book.html
Look around at 3-27
On Mon, Dec 24, 2012 at 8:44 PM, sanity insanity
networksanitytoinsan...@gmail.com wrote:
hi Guys,
Can you explain as to how we can achieve the following by ...
Connecting HQ and a branch
On Dec 5, 2012, at 8:22 AM, Tanner Ezell tanner.ez...@gmail.com wrote:
Setup after call work timer (configured on the CSQ configuration page) to
give your user time after the call has ended. This will apply to everyone
who is taking calls from that queue.
On Wed, Dec 5, 2012 at 2:21 AM, sanity
HI Guys,
I am trying to write a UCCX script for the following...
Trigger 24044000 called from PSTN or from internally to 4000, it should be
directed to SC Phone 1 4101 or 4102 SC Phone 2 depending on longest idle
time.
Configure ip phone service for one button login for these agents
Write
guides:
http://www.cisco.com/en/US/products/sw/custcosw/ps1846/products_programming_reference_guides_list.html
2012/11/1 sanity insanity networksanitytoinsan...@gmail.com
hi Guys,
I really need your help to understand UCCX scripts ...How they are made?
and how they work?
Please help guys
Hi ALL,
Isn't there a easy way of learning QOS . I am just not able to analyse and
problem / question related to QOS.
Can you suggest an easy way to learn switch qos and WAN QOS ? also can I
practice QOS on any simulator/ tool like packet tracer .
Your inputs on this.
-J
On Sat, Sep
Hello,
I am trying to configure a script that says THank you for calling if you
dialed this by mistake press one to contact operator
else someone will be with you shortly. Press 1 should go to an internal
extension
How do I get this done on UCCx?
-PD
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