Hi
I would like to find out if anyone has the xml code for an external ldap
direcory search for Call Manager. I have mulitple LDAP directoies in my lab
and would like to get a phone service to point to them instead of its own
internal one.
Paul
___
Hi All
This is slightly off the CCIE topic but I am looking for information on
CallManager 7.0 and the trusted relay point. The is not much on CCO ro the
SRND on how to configure this.
Thanks
Loads of good info flying about. Mark, Are you saying that its not possible
to reserve channels using h323 with PRI. This would be the most useful way.
Paul
On Fri, Oct 31, 2008 at 4:11 AM, Mark Snow [EMAIL PROTECTED] wrote:
You are in fact correct.
I didn't have a router in front of me and
Technical Instructor
IPexpert, Inc.
Sent from my iPhone
On Oct 28, 2008, at 11:36 PM, Paul and Bobs [EMAIL PROTECTED]
wrote:
Thanks Mark
If I wanted to just use mgcp, is there a way to control which channels are
used. So I can reserve 10 channeles for outgoing and 10 for incoming
Hi All
I was wandering if anyone know of a way using both MGCP and H.323 to control
the channells on an E1/T1 circuit. For example - If I have a single E1
service with only 20 channels and I want to say reserve 5 for outgoing and
reserve 15 for incoming, is there a way on both protocols to do
Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and
CCIE Storage Lab Certifications.
--
On Oct 28, 2008, at 10:20 PM, Paul and Bobs wrote:
Hi All
I was wandering if anyone know of a way using both MGCP and H.323 to
control the channells on an E1/T1 circuit. For example - If I have
I am trying to get a working script from a production server working in the
lab. Both IPCC servers are running 4.0.4 but when I move the script across
and try look at it in the script editor it comes up with java. errors
any ideas
On Sun, Oct 12, 2008 at 2:25 PM, Jacob Owen [EMAIL PROTECTED] wrote:
Well, you will have the SRND's on the desktop when you lab. If you are just
wanting to find them to read, check out www.cisco.com/go/srnd
On Sat, Oct 11, 2008 at 10:16 PM, Paul and Bobs [EMAIL PROTECTED]
wrote:
HI
Does
HI
Does anyone know how to get to the SRND through the new documentation website??
Paul
Hi
Here is my BACD config
I am not getting it to load up. I have rebooted but still no luck.
When I issue the command sho call application sessions i get nothing.
When i issue the command call application voice load queue and the
the one above still get nothing.
application
service queue
HI All
Does anyone have a demo license for Unity with VPIM that I could use
to get it working in the lab.
Cheers
Paul
Hi All
Firstly checking to see whether site i back up and secondly have a question
on page 374 of the IPEXPERT book there is some cme ephone config for CUE GDM
I am trying to work out if this is needed if the AA script is not in use
What is the extra DN used for that is mapped to the key 4:20
Does anyone know of an easier way to find out what module is plugged into
what slot on the gateway besides
show diag and sho ver
If my system is setup with teh following DID number range
617 302 1XXX
and I only have DN setup for
617 302 10XX
when someone tries to call a number with my range but that has not been
allocated they get fast busy
617 302 1800
What I would like to try and do is create perhaps a CTI RP with
Thanks
On Sun, Sep 7, 2008 at 2:28 PM, Jonathan Charles [EMAIL PROTECTED] wrote:
AAR does not work and is not supported with IPCC.
Jonathan
On Sat, Sep 6, 2008 at 9:28 PM, Paul and Bobs [EMAIL PROTECTED]wrote:
Has anyone got IPCC working with AAR. When i call from BR1 to HQ IPCC
pilot
Has anyone got IPCC working with AAR. When i call from BR1 to HQ IPCC pilot
number i see the network congestion message and the isdn messages hit the
gateway but it doesnt seem to go through. Is it something to do with the
redirecting number in the isdn messages
Here is my CME config. I am not getting any musci on hold
The file exists in the flash.
Not sure what I am missing here
telephony-service
sdspfarm units 2
sdspfarm transcode sessions 2
sdspfarm tag 1 mtp0017e04a6999
max-ephones 30
max-dn 30
ip source-address 10.61.115.1 port 2000
system
] wrote:
You should not be able to call if layer 2 says TEI ASSIGNED...
Multiple Frame Established means the D channel is up
anything else means the D channel is down... how the calls work, I couldn't
tell you.
Jonathan
On Wed, Sep 3, 2008 at 12:15 AM, Paul and Bobs [EMAIL PROTECTED]wrote
Hi
I am trying to block the CLI of internal calls to each other or change the
calling number. Is there a way to mask or block the cli for two phone in the
same internal partition
Paul
Hi
I am stugling with the FRST setup. This is the config from my BR1 router
interface Serial0/2/0
no ip address
encapsulation frame-relay
frame-relay interface-dlci 122 ppp Virtual-Template1
!
interface Virtual-Template1
bandwidth 64
ip address 192.168.122.2 255.255.255.252
ip
Hi
In Call Manager express, to get MWI working I believe you onyl have to
configure the following
ephone-dn 10
number 5000
mwi on
ephone-dn 11
number 5001
mwi off
I have found documenation that also configures the following but doest give
an explaination
dial-peer voice 5000 voip
Does anyone know of a way to get Unity to playback the calling number of in
incoming call.
Scernario: I have a analogue phone that i quickly need to find out what the
number is. It is attached to a VG224. I would like to be able to call a
number which forwards to Unity and the Unity can play out
that. CRS can
easily convert numbers into a prompt to be read back to the caller. I'm not
sure about Unity though.
On Tue, Aug 26, 2008 at 5:16 AM, Paul and Bobs [EMAIL PROTECTED]wrote:
Does anyone know of a way to get Unity to playback the calling number of
in incoming call.
Scernario: I have
Is there a way to flush the settings out of the BACD when making changes.
I have made a change to the param aa-hunt1 4200 in the queue service so
when users dial 1 they get sales. I had it set at param aa-hunt2 4000
Whenever i call in I cant dial 1 but can see from the debug that I am
getting
Applied this command
disc_pi_off
to the voice port and it solved the issue.
On Sun, Aug 24, 2008 at 2:01 PM, Jonathan Charles [EMAIL PROTECTED] wrote:
do a debug vpm signal and see if you get the power_denial
Jonathan
On Sat, Aug 23, 2008 at 7:41 PM, Paul and Bobs [EMAIL PROTECTED
Morning everyone
When I make a call from on-net to off net adn the off net party hangs up the
on-net party gets a re-order tone. I had found the PI command a while back
to cancel call when off-net hangs up but can remember it again. Any ideas.
Paul
When my site B goes into SRST and i try to call voicemail, it makes the call
over the pstn but the dtmf tone are not being recognised by unity.
I must have to relay them but in srst i dont have any voip dial-peers so not
sure. any ideas??
Thanks Guys
Service-Engine0/0.
SC-RTR#service-module service-Engine 0/0 sess
SC-RTR#service-module service-Engine 0/0 session
Trying 10.61.115.1, 2194 ... Open
***
***
On Tue, Aug 5, 2008 at 5:18 AM, Paul and Bobs [EMAIL PROTECTED]wrote:
Thanks
So do i reboot the module then hit ***. I have not done
Thanks.
I cannot access the module though. It says that is has failed and when I
session to the module it gives me a blank screen.
Paul
On Mon, Aug 4, 2008 at 3:50 AM, Preethi Thamina [EMAIL PROTECTED]
wrote:
Hi!
Try taking the module offline and restore it to factory default
Hi Guys
I have just upgraded my lab CUE and am now getting the following error. Any
ideas??
Service Module is Cisco Service-Engine0/1
Service Module supports session via TTY line 258
Service Module is trying to recover from error
Service Module heartbeat-reset is enabled
Service Module status is
Training Tools for the Cisco CCIE
RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and
CCIE Storage Lab Certifications.
--
*From:* [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED] *On Behalf Of *Paul and Bobs
*Sent:* Saturday, July 19, 2008 10
On Site B, which has the MGCp gateway, I have set the call manager to only
allow significatn digits of 4. yet when when I make a phone call from HQ to
site b the call fails. only when I set up a translation pattern that strips
the digits down to leave me with 4 digits, then the call goes through.
Hi Guys
Struggling on something, the Cue Module. I setup the CME with all the
setting required including the web admin system name admin password cisco
and have reset the cisco CUE to factory defaults setting the new account
details to admin and cisco in CUE. When I get to the first setup page on
Hi
If I have a 6608 blade and I am using say 3 ports for T1 PSTN lines and 1
for Conference and they are registered to one callmanager cluster, can i
register the other 5 ports for conference resources on a seperate cluster.
Paul
My config is pasted below. Fro some reson when I enter dspfarm this command
disappears. and when i enter dspfarm transcode maximum session ? i get
0-0.
Once I get these configured how can i check to see that the dsp are
correctly configured and is there a command to see how many dsp are left nad
My config is pasted below. Fro some reson when I enter dspfarm this command
disappears. and when i enter dspfarm transcode maximum session ? i get
0-0.
Once I get these configured how can i check to see that the dsp are
correctly configured and is there a command to see how many dsp are left nad
Thanks
I have corrected this but it makes no difference.
On Sun, May 18, 2008 at 4:33 PM, senthil natarajan [EMAIL PROTECTED]
wrote:
I see some typo,
param handoff-string a
This has to be
param handoff-string aa
-senthil
On Sun, May 18, 2008 at 12:49 AM, Paul and Bobs [EMAIL
to apply service-policy with cRTP on both outbound/egress and
inbound/ingress interfaces at the same time.
Rgds
Alex
- Original Message -
*From:* Paul and Bobs [EMAIL PROTECTED]
*To:* ccie_voice@onlinestudylist.com
*Sent:* Saturday, May 17, 2008 2:35 AM
*Subject:* [OSL | CCIE_Voice
in the script, example:
param aa-hunt1 1001 --- means option 1 goes to
hunt group 1001
param aa-hunt2 2001 --- means option 2 goes to
hunt group 2001
and so on.
Remember to reload the scripts after you make a
change.
- Original Message -
From: Paul and Bobs
I am running the following config on my lab for compressed RTP. when I apply
it to the serial interface outbound I loose the audio stream in that
direction. call remains up and one way audio is there. When I remove the
service policy output the audio come back.
Any ideas
Thanks
class-map
3 and a 2 Port fxs is installed you only have 3 left and can not
allocate them.
Ed
- Original Message
From: Paul and Bobs [EMAIL PROTECTED]
To: ccie_voice@onlinestudylist.com
Sent: Saturday, May 10, 2008 11:25:37 PM
Subject: [OSL | CCIE_Voice] DSP Resources
Hi
What
Hi
What is the minimum DSP's you need in order to configure the minimum amount
of transcoding resaources. I have a PVDM2-8. I am using 3 to terminate the
G711 calls on the controller E1 and was wandering if there was any way of
using the rest \as a transcoding resource.
Thanks
paul
PM, David L. Blair [EMAIL PROTECTED]
wrote:
I think this might be a codec issue. If the scenrio is the following, the
call completes then is immediately terminated is a codec mismatch
issue. Remember the default codec on routers is G.729.
David
- Original Message -
*From:* Paul
and route filters?
It might be that you've got 2 seperate route patterns, one with the # and
one without. Check to make sure they're pushing to the same route
list/group and that they're in the same partition, etc.
LH
Paul and Bobs wrote:
HI Guys
Not sure if im looking in the right place
HI Guys
Not sure if im looking in the right place yet. What is the timer used to
shorten the delay once a digits have been enetered. I know the inter-digit
timer is the t302 but im having a issue where if i dial my international
site from HQ and put a # at the end the call goes through. But if if
.
Regards,
Anup
*From:* [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED] *On Behalf Of *Vik Malhi
*Sent:* Friday, April 11, 2008 8:25 PM
*To:* 'Paul and Bobs'
*Cc:* ccie_voice@onlinestudylist.com; 'Jacob Owen'
*Subject:* Re: [OSL | CCIE_Voice] MGCP and SRST
you should also block port 2427 (udp
Is there any command to see which application currently holds access to the
isdn interface.
On Fri, Apr 11, 2008 at 2:37 PM, Paul and Bobs [EMAIL PROTECTED]
wrote:
Thanks Jacob
I tried that and it changed the command
to
application
global
service alternate DEFAULT
] [mailto:
[EMAIL PROTECTED] *On Behalf Of *Paul and Bobs
*Sent:* Thursday, April 10, 2008 11:34 PM
*To:* Jacob Owen
*Cc:* ccie_voice@onlinestudylist.com
*Subject:* Re: [OSL | CCIE_Voice] MGCP and SRST
Is there any command to see which application currently holds access to
the isdn interface
RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and
CCIE Storage Lab Certifications.
--
*From:* Paul and Bobs [mailto:[EMAIL PROTECTED]
*Sent:* Friday, April 11, 2008 4:18 PM
*To:* [EMAIL PROTECTED]
*Cc:* Jacob Owen; ccie_voice
Certification Training Tools for the Cisco CCIE
RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and
CCIE Storage Lab Certifications.
--
*From:* Paul and Bobs [mailto:[EMAIL PROTECTED]
*Sent:* Friday, April 11, 2008 6:07 PM
*To:* [EMAIL
-back
call application alternate DEFAULT
http://www.cisco.com/warp/public/788/AVVID/mgcpfallback.html
On Fri, Apr 11, 2008 at 11:06 PM, Paul and Bobs [EMAIL PROTECTED]
wrote:
BR1 Config
BR1-RTR#sho run
Building configuration
What is the first port on VG248 used for. Port 0??
I cant seem to get my calls to route over my PSTN interface when site BR1 is
in SRST. Debug voip dialpeer all displays that the correct dial-peer is
being matched as the outbound dial-peer but the interface is not being
engaged at all. Running debug isdn q931 produces no results even when i
remove
Ed
- Original Message
From: Paul and Bobs [EMAIL PROTECTED]
To: ccie_voice@onlinestudylist.com
Sent: Wednesday, April 9, 2008 3:00:11 AM
Subject: [OSL | CCIE_Voice] Calling number manipulation on MGCP gateways
Guys
Trying to manipulate the calling number on MDCP gateway. I can
... not sure if there is a reasonable
solution tho...
Jonathan
On Wed, Apr 9, 2008 at 2:00 AM, Paul and Bobs [EMAIL PROTECTED]
wrote:
Guys
Trying to manipulate the calling number on MDCP gateway. I can do it n
h323
with voice translation patterns but wandering if there was similar
Trying to get the 6608 gateway to register.
I have enable the interface in 6500 and can see that it has ip address and
address of tftp server.
I have then added it to the ccm as a 6000 t1 gateway with PRI.
I can seem to get it or the resources to register.
Paul
HI all
If anyone has had much luck with reistering the 6608 gateway to CCM in the
proctorlabs I would really like to hear from you. Interested in what steps
you took to get it registered.
Thanks
HI Guys
Got a teaser that bugging me now. I have in my lab HQ and Br1 connected with
E1 crossover to simulate PSTN as best I can. I have Br1 connected to CCM
with MGCP and HQ configured as H323 gateway in CCM. I have respective route
paterns for the remote sites configured to point to the
]
wrote:
on hq rtr, make sure you bind h323 to interface for the IP address that
you are using in ccmadmin to identify the interface. otherwise, ccmadmin
doesn't recognize the gateway properly and by default won't connect the
call.
On Tue, Apr 8, 2008 at 11:18 PM, Paul and Bobs [EMAIL
be CCM, 2# could be CME, etc.
Adam W. Moore
CCIE-Voice #18462
CCVP, CCNP, CIPTSS, UCSE
CIPTOS, CIPCCXS, CCNA, CCDA
MCSE(2K/2K3), MCSA(2K/2K3), MCP
Verizon Network Integration
500 Technology Drive
South Charleston, WV 25309
304-746-1011(W), 304-545-5261(C)
*Paul and Bobs [EMAIL
into 'any other method on the gatekeeper...'
You could just assign the 10-digit numbers as the extensions.
Jonathan
On Sun, Apr 6, 2008 at 1:04 AM, Paul and Bobs [EMAIL PROTECTED]
wrote:
Im struggling a bit with this diaplan in CME adn gatekeeper routing. I
am
trying to get CUE
Great idea
Thanks
On Mon, Apr 7, 2008 at 7:53 AM, Jonathan Charles [EMAIL PROTECTED] wrote:
Someone else replied, directly to me, said to put the full 10 digit as
a secondary on the ephone-dn
Jonathan
On Sun, Apr 6, 2008 at 3:23 PM, Paul and Bobs [EMAIL PROTECTED]
wrote:
Thanks
HI
With FRTS (not the FRF12 way) is it best to apply the LLQ config to the
map-class FRTS then apply the service-policy FRTS to the virtual interface
or can you apply the service-policy FRTS to the virtual-template with the
LLQ and then on the sub-interface serial 0/0.1 apply the service-policy
Guys
Got a silly question thats bugging me. Do you have to explicitly block
patterns you do not want users to be able to dial. Let me tell you what im
thinking.
'PhoneA' in in PAR-INT with CSS-ALL
'PhoneB' in in PAR-INT with CSS-INT
1800XXX is in PAR-FREE
and only CSS-ALL contains
Howdy
When creating hunt groups and haveing the hunt group hunt to phone 1 on say
a second line and then phone 2 on the second line and finally forwarding to
voice mail. Would the following config work
telephony-service
voicemail
!
ephone-hunt 1 sequential
pilot 3111
list 3100,3101
final
Hi
I am having an issue with my CUE. Whenever I setup a new mailbox for a user
and try to access the mailbox it is saying that I can only access it from
the primary extension for security reasons thens hangs up.
This happens on all new mailboxes.
Paul
On undefined, Paul and Bobs [EMAIL PROTECTED] wrote:
Hi
I am having an issue with my CUE. Whenever I setup a new mailbox for a
user
and try to access the mailbox it is saying that I can only access it
from
the primary extension for security reasons thens hangs up.
This happens
to
route a call over from CME to UCM using means such as a Tech Prefix, or
using the 'alias static' command.
HTH
Mark SnowSr Technical Instructor
IPexpert, Inc.
Sent from my iPhone
On Mar 30, 2008, at 6:05 PM, Paul and Bobs [EMAIL PROTECTED]
wrote:
Morning Guys
Still strugling a bit
Whats the difference between setting the audio bandwidth to 80 in the region
and setting the location bandwidth to 80 for bandwidth management over the
WAN
as an example.
John.
*From:* [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED] *On Behalf Of *Paul and Bobs
*Sent:* Thursday, March 27, 2008 1:34 PM
*To:* Mark Snow
*Cc:* ccie_voice@onlinestudylist.com; ccievoice1
*Subject:* Re: [OSL | CCIE_Voice] Regions and Locations
Thanks very much
ON gatekeeper, what is the signifcance of the RAS ip address in the zone
local command. If its is left out what does it affect?
Paul
PROTECTED] wrote:
yeah you should be able to do a show gatekeeper endpoints and see it
registered. If you debug gatekeeper main 10 you can see as it is selected
when you have the invia and outvia commands setup.
On Thu, Mar 27, 2008 at 8:54 PM, Paul and Bobs [EMAIL PROTECTED]
wrote
This is my full config. Cant get the IPIPGW to register to Gatekeeper. Any
ideas
HQ-RTR#wr t
Building configuration...
Current configuration : 3696 bytes
!
! Last configuration change at 13:39:37 AEDT Fri Mar 28 2008 by admin
! NVRAM config last updated at 13:37:46 AEDT Fri Mar 28 2008 by admin
Is there any way on CUE to monitor what the ANI is of an incoming call like
you can with call viewer in UNity. I have a problem with my CUE that just
after initial setup I have a mailbox setup for 3001 , and when I dial CUE
from ext 3001 i get prompted for a userid. I put in 3001 and get a
HI
Can someone explain to me the reason on a 1544 Kbps link the FRTS cir is set
to 1466800 and bc set to 14668
Paul
Hi
Question 12.5
Frame relay interface is set to 768 kbps and the cir is 384
why is the sample config given in the proctor guide using 364800 which is
around 96% fo the cir
it also uses 3648 as the shape average bc
is there any standard or guidance on this.
Cheers
Paul
In Section 11 when seting up the IPMA, the setup of the line DN are not very
clear.
Is it saying that you need to have 2 DN of 2003 set on the phone and if so
what partition should the second one go into.
And how is a proxy line different to a normal line.
Cheers
Paul
?
Jonathan
On Sun, Mar 23, 2008 at 8:29 PM, Paul and Bobs [EMAIL PROTECTED]
wrote:
Hi
I have the followign version of IPCC.
4.0(4)_Build140
I am trying to configure the skills for my agents but the option is not
there. I am using the free 5seat license that comes with CallManager
PROTECTED] [mailto:
[EMAIL PROTECTED] *On Behalf Of *Paul and Bobs
*Sent:* Thursday, March 20, 2008 9:12 PM
*To:* ccie_voice@onlinestudylist.com
*Subject:* Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 25, Issue 102
Hi guys
I know the NDA on the CCIE prevents everyione from telking to much
Hi guys
I know the NDA on the CCIE prevents everyione from telking to much but the
Proctor lab, which one is closest to how the real lab feals. I am busy
working my way through the proctor guide and work book and will be attmpting
the multi protocol challenges from end of next week. Is it the
*Trying to get the outbound faxing working. I am getting the following NDR
as a returned email.*
The fax you attempted to send could not be successfully delivered.
Detailed error information:
The e-mail account does not exist at the organization this message was sent
to. Check the e-mail
Thanks guys
On Thu, Mar 20, 2008 at 10:23 AM, Devildoc [EMAIL PROTECTED] wrote:
Jonathan,
First, I'd like to apologize for my rambling. It happened to me on a few
occassions especially after a long day of studying. For some good reasons,
my brains mis-fired and i ended up making a simple
Hi again. Thanks for the great forum.
IN chapter 6, the dial plan , I would like some clarification on the call
flow.
If user in HQ make call 2XXX
This is matched by RL_HQ_BR1_LOC
If HQ is unavailable it then routes call to RG_BR1 and prefixes 1212
What is the trailing 1 in the 1212 for. I
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