Chris,
For the lab purpose u need 2 GKs on HQ and pstn switch,(local zone ,remote
zone). However if I want GK on HQ, I need 2600+ router that will be
expensive. So I chose 1700 solution for HQ instead.and leave remote GK on
pstn switch.( just GK on pstn switch for local zone, no GK on HQ)
On T
Hello Team,
Question on VPIM:
I have an issue when I go to create a Delivery Location on Unity for
CUE it won't add it. If I create any other profile using other than
VPIM, such as SMTP it works.
I have already check my license and it does allow VPIM
Pardeep
Hello Team,
Is there any good procedure to setup NTP in CUCM because I tried the one
where you change the ntp.conf file and restart the NTP service but
doesn't seem to work for me.
This is what my ntp.conf file look like:
server 142.707.64.254
Thanks
Chris,
If u can ,use 256 mem on your 2610, (ito upgrade to 256 ram, should upgarde
to bootstarp ver 12.2.8r ), I use 128 mem and version is 12.4(17)
I use same router for GK function.
HTH
On Tue, Nov 11, 2008 at 3:00 PM, Chris Kagadis (kagadis.com) <
[EMAIL PROTECTED]> wrote:
> Thanks, Cyrus.
Sekchye,
u can set SRST reference of device pools on each CCM cluster to ip source
add of the SRST/CME router.
They will register with GW. u can verify process with debug ephone register.
HTH,
On Tue, Nov 11, 2008 at 2:30 PM, sekchye goh <[EMAIL PROTECTED]> wrote:
> Hi!
>
> Not sure wheth
Hi!
Not sure whether any voice experts in this forum can help me on this problem.
Due to some special requirements, we have a remote branch with two
groups of ip phones, each group registering to different call-manager
cluster.
There is only one H323 (not MGCP) SRST gateway in this branch.
Hi Vik,
Yes I have set the DP/Region for G711 and the incoming DP on CME for
G711 and the call still fails. Also when the DP/Region is set for
G729 with the incoming DP on CME set for G729 I have cleaned up the
dspfarm profile to only use g711u, g711a and g729r8 codecs. The call
still fails (5 s
I forgot to mention that, u should provide some sort of loop prevention
mechanism to prevent loop between intl1 and intl2 dialpeers ( Input and
output on following scenario)
If I were u, I will manipulate calling number in first translation rule and
then try to match that on last intl1 and intl2 d
Michael,
following setup will work, however it's rather complicated solution.
intl1 dial-peer
|
| intl1 dial-peer
| ---translation Rules intermediate dial-peer
---translation Rules |
intl2 dial-peer
|
| intl2 dial-p
Chris,
u can build up your pstn switch with 2610 with 1xNM-HD-2VE and 2xVIC-2MFT-T1
cards. Setup is very simple and just building up touting number plan by
bunch of dial peers.
it's possible by either ds0-group or PRI setup.
On Tue, Nov 11, 2008 at 1:41 PM, Chris Kagadis (kagadis.com) <
[EMA
Question... Let's say we asked to limit the number of international calls from
CME to 2 calls maximum at any time. If we have the only one international
dial-peer, it's simple - just use "max-conn" on the dial-peer. But what if we
have multiple dial-peers, which eventually falls under the defini
I would like to set up a PSTN switch in my home lab. I have three ISRs that
I would like phones in each "branch" to be able to call any other branch via
T1 PRI. Currently, each of my ISRs have a VWIC-1MFT-T1, and would like to
have the cards terminate to VWIC-1MFT-T1 cards on the PSTN switch. Can
s
ops, you have to upload the Prompt w/ Prompt Management Menu.
Sergio.> From: [EMAIL PROTECTED]> To: [EMAIL PROTECTED];
ccie_voice@onlinestudylist.com> Date: Tue, 11 Nov 2008 02:25:53 +> Subject:
Re: [OSL | CCIE_Voice] IPCC custom prompt> > James> You have to upload the
script with Script
James
You have to upload the script with Script management menu
Sergio
- Mensagem Original -
De: James Jung <[EMAIL PROTECTED]>
Enviada: segunda-feira, 10 de novembro de 2008 20:27
Para: ccie_voice@onlinestudylist.com
Assunto: [OSL | CCIE_Voice] IPCC custom prompt
Hi all
I want to use
By cleaning up the codecs- only have g711u, g711a and g729r8.
--
Vik Malhi CCIE #13890, CCSI #31584
Senior Technical Instructor - IPexpert, Inc.
Telephone: +1.810.326.1444
Fax: +1.810.454.0130
Mailto: [EMAIL PROTECTED]
Join our free online support and peer group communities:
http://www.IPexpe
Please confirm that this is not the transcoder.
Set the DP on CCM to use g711.
Set the inbound voip dial-peer to use g711.
Confirm the results of this test. If the call works try cleaning up the
codecs in your dspfarm profile.
--
Vik Malhi CCIE #13890, CCSI #31584
Senior Technical Instructor
This the 3rd day in a row that I have had this problem. It seems to be
a bug when using the ipipgw functionality of the CME. It happens
regardless of codec.
Has anyone been able to call from HQ or BR1 to the BACD script with
G729 across the WAN ?
Thx,
Mike Brooks
CCIE# 16027 (R&S)
On Sun, Nov 9
Hi all
I want to use custom prompt which is recorded and formatted correctly.
When I put it in the system prompt folder, I can use it. But I cannot
invoke the prompt if it is in a different folder(for example
c:\prompt\).
What is the proper script command for this using P[]?
JamesJ.
Perfection. Thanks.
From: Alex [mailto:[EMAIL PROTECTED]
Sent: Tuesday, 11 November 2008 10:17 a.m.
To: James Jung; Vik Malhi; Ryan Trauernicht;
ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] MOH Multicast failover to Unicast
JamesJ,
You c
Re: [OSL | CCIE_Voice] MOH Multicast failover to UnicastJamesJ,
You can configure unicast MOH server to be in a separate location. Then for
this location allow whatever x 24kbps BW for whatever number of unicast MOH
G729 streams you need to activate when multicast MOH server fails.
Rgds
Alex
Not possible.
--
Vik Malhi CCIE #13890, CCSI #31584
Senior Technical Instructor - IPexpert, Inc.
Telephone: +1.810.326.1444
Fax: +1.810.454.0130
Mailto: [EMAIL PROTECTED]
Join our free online support and peer group communities:
http://www.IPexpert.com/communities
IPexpert - The Global Leader
Thanks Alex
JamesJ
From: Alex [mailto:[EMAIL PROTECTED]
Sent: Tuesday, 11 November 2008 10:10 a.m.
To: James Jung; OSL CCIE Voice Lab Exam
Subject: Re: [OSL | CCIE_Voice] Unity MWI on SRST phone
Q. What is the support for voicemail integration with the Cisco
Q. What is the support for voicemail integration with the Cisco Unity server
through analog or DTMF?
A. SRST uses the same in-band analog or DTMF voicemail integration method that
Cisco Unified Communications Manager Express uses to allow call forward busy,
call forward no answer, or call forwar
Sorry, typo, 5 g729 calls.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of James Jung
Sent: Tuesday, 11 November 2008 9:46 a.m.
To: Vik Malhi; Ryan Trauernicht; ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] MOH Multicast failover
Hi
Questions about this moh.
If you want to allow 2 g729 calls and moh(first multicast and later
unicast), how much bandwidth do we need to configure?
With 48 Kbps in the location bandwidth, it allows 2 g729 calls and
multicast.
But when the multicast moh server is down, the moh cannot be h
The failover should happen transparently.
To test you should stop the IP Voice Media Streaming App service on the sub.
Unicast MOH is dependent on Locations CAC being available whereas multicast
is not. Check the DP of the PUB MOH server and make sure that there is
enough bandwidth to the BR1 sit
Hi,
Can anyone tell me how to configure to turn on mwi for phones in SRST
mode when someone left a message?
Is it possible?
JamesJ
Wow- this is news- I didn't know there was a limitation with the # of
softkeys...seems like a special "feature" to me.
--
Vik Malhi CCIE #13890, CCSI #31584
Senior Technical Instructor - IPexpert, Inc.
Telephone: +1.810.326.1444
Fax: +1.810.454.0130
Mailto: [EMAIL PROTECTED]
Join our free onl
Is it possible to have an agent logged into a phone on the CME router?
Since the phone is not registered to CCM, you can't associate the phone
to the user agent. But could you associate a CTI route point to the
agent user and then have the CTI route point forward all to the CME?
Bartosz,
Try using 2 sets of dialpeers on IPIPGW: one set hardcoded to G711 with
"max-conn 1" and second set hardcoded to G729 with "max-conn 1"+"priority
1". Please report your results if/when you have a chance of labbing this
up:-)
Cheers
Alex
- Original Message -
From: "Bartosz So
Hi,
I'm trying to call BR2 CME user "0113313283003"
>From PSTN-WAN Router traces(below) I can see the called number but when I
checked from BR2 traces (as attaached to the e-mail)
Called Number=33132433313 is changing.
I attached BR2 config (this time E1 is up state)
What can be the problem?
Hello Alex,
> IPIPGW won't negotiate the codec with HQR, it has to be statically
> nailed down for all 4 legs. If you wish to have G729 call from CCM to
> CME via IPIPGW, then it should be 4 dialpeers with default G729 codecs
> (2 with "incoming called-number", 2 with "destination-pattern"):
That
Pardeep,
What u get from sh call active voice history brief?
What is the reason for call disconnect in the output?
On Tue, Nov 11, 2008 at 4:40 AM, Pardeep Singh (pardsing) <
[EMAIL PROTECTED]> wrote:
> Team,
>
> I am having an issues when I call from my HQ site to CME using Gatekeeper
>
Can the moderator of this list please remove me?. I've requested this before
but still continue to receive emails. Please remove me a.s.a.p. Thank you.
Hi Robert
Thank you for your comment.
And than you all for the help.
JJ
-Original Message-
From: Robert Schuknecht [mailto:[EMAIL PROTECTED]
Sent: Monday, 10 November 2008 9:11 p.m.
To: James Jung; OSL CCIE Voice Lab Exam
Subject: Antw: [OSL | CCIE_Voice] H323 GW PSTN -> BR1 call.
James,
Bartosz,
IPIPGW won't negotiate the codec with HQR, it has to be statically nailed
down for all 4 legs. If you wish to have G729 call from CCM to CME via
IPIPGW, then it should be 4 dialpeers with default G729 codecs (2 with
"incoming called-number", 2 with "destination-pattern"):
dial-peer vo
Hello,
> Not sure I understand your question, but I think you are asking why you
> can't get more than 1 G711 call to the remote zone?
No, the question is - why G729 calls do not work with IPIPGW but they do
work if GK without IPIPGW is used.
Or in other words - why G711 calls work with IPIPGW bu
Team,
I am having an issues when I call from my HQ site to CME using
Gatekeeper Trunk over to CUE pilot number.
Here is the call flow:
HQ Phone-->>>dial 4001GK Trunk>CME = Call is successful
HQ Phone--->>>dial 4111GK Trunk>CMECUE PILOT NUMBER = Call
give this error back onc
Hello,
Not sure I understand your question, but I think you are asking why you
can't get more than 1 G711 call to the remote zone?
You have:
bandwidth remote 144
This will allow only one G711 call to your remote zone. Each G711 call
uses 128K of bandwidth (64K in each direction)
To have m
Hello,
I have problem with calls from CCM to BR2 via GK and IPIPGW.
If I configure HQ-RTR GK without IPIPGW (no invia/outvia) like this:
!
!
gatekeeper
zone local HQ-RTR sldx.lab 172.1.100.1
zone local VGK sldx.lab
zone remote PSTN-WAN sldx.lab 10.1.200.2
zone prefix PSTN-WAN 011*
bandwidth r
Robert,
Have you seen this link
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note09186a008097cd93.shtml ?
I would imagine "DNS A record" method is the simplest - it all boils down to
configuring 2 IP addresses in DNS for a given name+short TTL.
If you want IP phones using s
Robert,
I put "ccie." as my full FQDN in unity AD setup be4, so I set it up based on
that. I tried FQDNs like voice.com but, no difference.
What I'm not understand here is "Queued mail for delivery" at the end of the
trace means?
it means it transfered the message? or still in router's queue?
If
Is this a real-life question or a lab task?
Anyway, I think it can be achieved using Private VLANs to separate
Server+Data+Voice traffic. Make the router port promiscuous+configure
secondary IP addresses on the router port if you are using a separate subnet
per VLAN
http://www.cisco.com/en/US/
You only need to add the
H323-gateway voip bind srcaddr ip-addr
This command is for H323 gateway, not gatekeeper.
Pete Olson
[EMAIL PROTECTED]
425-965-2577
From: James Jung [mailto:[EMAIL PROTECTED]
Sent: Sunday, November 09, 2008 8:29 PM
To: OSL CCI
Erick,
i did not try the following, but what if you make the MOH Audio Source, of the
IPCC CTI Ports to play a *.wav file with Ring-Tone. Search on Publisher/IPCCx
under: c:\program files\wfavvid..., for *.wav files i think you will find some
adequat soundfiles.
HTH
/Robert
>>> Erick Pineda<
Hi List,
i have only a few days left until my first real LAB-Exam (its on Friday). And i
don´t get the IP.Phone Services run in a redundant fashion.
I would like to use DNS SRV Records, for services like IPMA, EM,FastDials,
Addressbook...etc. But i can´t get it to work.
Could anybody please exp
Then try mucic on hold.. file as ring back tone...
--- On Mon, 10/11/08, Erick Pineda <[EMAIL PROTECTED]> wrote:
From: Erick Pineda <[EMAIL PROTECTED]>
Subject: [OSL | CCIE_Voice] IPCCX ring back tone to caller
To: "OSL CCIE Voice Lab Exam"
Date: Monday, 10 November, 2008, 6:57 PM
an ipccx
Jeremy,
if i understand your debug right, then are not using Fully Qualified Domain
Names. In the trace i saw the following line: "netw smtp 5 RCPT TO [EMAIL
PROTECTED]" . Are you using ccie as FQDN? If yes, try to change it to ccie.lab
or any other Top-Level Domain which your DNS Server is us
an ipccx agent gets a call, when he makes the tranfer the callers hears a ring back tone.does any boby has an idea how to do it, because right now when i make the tranfer the caller hear mucic on hold..RegardsErick
Hi,
I setup vpmi between cue and unity and I get this trace output. I cannot
find msg which was sent by ipphone in exchange 2000 message tracking
center.
and of course mwi would not turned on.
Here is trace out put.
I wasted 3 hours and still cannot understand why exchange does not get the
msg.
You seem to have forgotten your frame-relay interface type which would be DCE
on the FR switch side. (frame-relay intf-type dce)
Please see the following link for more info on setting up FR switch and setting
DCE via the interface type:
http://www.cisco.com/en/US/docs/ios/12_2/wan/configuration/
The question is not to use any trunk between the router and switch.
If I have only 2 vlans on HQ than this can be achieved easily. But if I have
more than 2 vlans e.g server,data and voice, how can I achieve this without
using trunk.
Thanks
CONFIDENTIALITY - The information contained in this el
Many thanks Scott, it did it!!! I forgot the AAR Field and the Ext Num Mask on
the Hunt-Pilot.
RObert
>>> Hardesty, Scott<[EMAIL PROTECTED]> schrieb am Montag, 10. November 2008 um
01:54 in Nachricht f6c9a5ae0fe45e837f5e2d8c6feeccd6:
> Check your voicemail hunt pilot configuration. At the VERY
James,
in my opinion, you have two options:
1) Configure your H323 Gateway, with the IP-Address of the Interface pointing
to Callmanager, in the Gteway Config Page on Callmanager
2) Don´t issue the "gateway" command on the RTR. Without the "gateway" command
the RTR won´t register to any gateke
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