Hi,
I've done the new IPEXPERT volume 6 workbook labs 1 and 2. I am having
problems with my CIPC running SIP mode. In the workbook I needed to convert
it from a SCCP to a SIP phone via phone templates. I've done that and since
then I cannot use the CIPC anymore.
For example if I configure the
Dear experts,
Recently I’ve purchased the IPexpert voice BLS. Since I am totally new to
voice I have a lot to learn. I have a beginners problem, but it’s getting
frustrating since I cannot continue with my studies. I have already spend 2x
a proctorlabs vrack session and still cannot fix it.
Hi all,
UPDATE:
I am using the CIPC 7.0.1 (as specified on the IPexpert page). I noticed
that if I migrated the CIPC to SIP I could not get it back to SCCP.
I found a way to make it work again. In the installation package of CIPC
7.0.1 there is a CIPC.Locales.zip file, that's needs to be uploaded
.. but still it does not work. the status screen of the CIPC shows
“TFTP timeout: SEP00059A3C7800.cnf.xml not found”
So any thoughts are welcome!
Thanks.
Jb
2009/6/12 ccie voice cci...@gmail.com
Hi all,
UPDATE:
I am using the CIPC 7.0.1 (as specified on the IPexpert page). I noticed
Hi,
When I do lab 3 or lab 4 of the new volume 6 workbook. I noticed the
following
BR2 Phone 2 = IPBLUE SCCP DN=3002
BR2 Phone 4 = X-Lite SIP DN=3006
When I call from SIP to SCCP.. My phone rings and I can connect, but when I
call from my SCCP phone to the SIP phone, my SIP phone is not
Hi all,
I am doing lab 4 for the second time now and loaded the proctorlab lab 4
initial configs. I noticed that the answers in the workbook do not match.
There is a lot that needs to be configured like telephony-service details,
voice register global, etc.. before I can really start. I had
Hi,
I am using IP BLUE and using regedit files to change between phone
configurations.
When I start IP BLUE the phone registers correctly, but it takes about 30
seconds. What I mean is that program startup is slow.
I noticed that in the IPEXPERT video's their IPBLUE is starting up very
fast. Does
.
Take care.
Thks
2009/6/19 ccie voice cci...@gmail.com
Hi all,
I am doing lab 4 for the second time now and loaded the proctorlab lab 4
initial configs. I noticed that the answers in the workbook do not match.
There is a lot that needs to be configured like telephony-service details
Hi all,
I am getting a fast busy tone from Unity Connection over my CUCM -- CUC
SCCP Integration. I CAN press the messages button and log into my mailbox,
but when another user gets forwarded to voicemail by way of CFNA, I get a
fast busy tone. Any ideas? Thanks for the help!
of Unity Connection..
Sent via BlackBerry from T-Mobile
-Original Message-
From: CCIE VOICE ccievoiced...@gmail.com
Date: Tue, 23 Mar 2010 18:07:33
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Fast Busy (Reorder) CUC SCCP Integration
Is it possible to run the CUE/CME initialization wizard through the command
line? I am trying to prepare without using the GUI (thinking that it may
not be there in the lab). Any help with this question and other CUE command
line tips are greatly appreciated!
Everyone, this has been a thorn in my side for quite some time now. Here is
the scenario:
HQ-PHONES --- GK-TRUNK HQ-GK - c...@br2 BR2-PHONES (SIP)
1. HQ Phone dials a BR2 Phone and BR2 answers
2. BR2 Phone puts HQ on hold --- successful
3. BR2 Phone takes call off hold --
Written*
*Gmail:* ciscovoiceguru
*Skype:* ciscovoiceguru
*Twitter:* ciscovoiceguru
*1st Lab Attempt: *Aug 16, 2010
On 4/5/2010 6:59 PM, CCIE VOICE wrote:
Everyone, this has been a thorn in my side for quite some time now. Here
is the scenario:
HQ-PHONES --- GK-TRUNK HQ-GK - c
Hi,
Sorry for the incomplete email earlier.
Has anyone tried cBarge in SRST mode? I have my conference bridge registered
to telephony service when it goes into SRST. I have used priority 3 as the
telephony-service address. When my phones go in SRST mode, I can see they
have shared line and when I
Thanks Angel.
I am at work but will give it a go today.
I am just wondering that everytime when phones will go in SRST..do I have to
go into ephones and add the button? this does not look practical.
I will give it a go anyways.
vc
On Mon, Jun 7, 2010 at 9:39 AM, Angel Perez
will still stay there...I have not tested this but
this one colleague of mine had inbound calls issue after phones registred
back to CUCM as the call was going to the ehpones instead of taking the voip
dial-peer. Have you tested this?
vc
On Mon, Jun 7, 2010 at 10:00 AM, ccie voice cci
@Amp
So you choose a lab location based on lunch?
On Thu, Jun 10, 2010 at 1:14 PM, Amp amccar...@cciequest.com wrote:
I live here in the RTP area but have decided to take the lab in San Jose.
Here are my reasons:
1. Later Start Time
2. Longer Lunch
3. Better Weather
4. Just have a gut
Hello Angel,
Yes I made it work..its been quite few days now..
I just explicitly included privacy off commands under ephones and it
worked.
There is no need for srst auto prov all and dialpeer hunt 3 etc...
hth
On Mon, Jun 14, 2010 at 3:20 PM, Angel Perez gorr...@hotmail.com wrote:
Hi:
Did
Hi all,
I was going through the different ways we configure FRF.12 and MLPP on
serial interfaces and came across few findings.
When we are configuring FRF.12, we will use the command auto qos voip
trust under the DLCI. While when we are configuring MLPP then we will be
using auto qos voip trust
Hey everyone...I have NO IDEA what is causing my issue and I was hoping for
your assistance. I am currently working on Volume 2, Lab 1, Task 4.2 with
no success. The goal is to dial 3XXX from HQ or BR1 and route the call from
CUCM--GK--CUBE--BR2-RTR. I am getting the *Viazone gateway selection
Yep, I wanted to test registering another gateway in the viazone to try and
segment the problem.
From: kobel [mailto:findko...@gmail.com]
Sent: Monday, June 21, 2010 3:35 AM
To: CCIE VOICE
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Gatekeeper CUBE - Pulling My Hair
Turns out the dial-peer on the BR2 side was still using the default G729
codec. I am able to answer the call now.
From: CCIE VOICE [mailto:ccievoiced...@gmail.com]
Sent: Monday, June 21, 2010 12:33 AM
To: Mouhammad Nasser
Cc: ipexpert
Subject: Re: [OSL | CCIE_Voice] Gatekeeper CUBE
Hello All:
I am having troubles getting a SIP phone registered to the BR2 gateway.
The phone will boot up in SIP code to the point where it says Phone
Unprovisioned on the phone itself. I can get past this by adding this line
to the gateway:
tftp-server flash:SIP000C851CE88B.cnf
Although I do
g711ulaw
On Sat, Jul 10, 2010 at 12:31 PM, Adam Thompson phoe...@fatturtle.comwrote:
Do you have this phone configured in your CME? SIP phones need a DN
configured in order to register.
-Adam
On Sat, Jul 10, 2010 at 1:41 PM, CCIE Voice cc...@corb.net wrote:
Hello All:
I am having troubles
at 2:49 PM, CCIE Voice cc...@corb.net wrote:
Hi Adam:
Thanks and yes I do have the phone along with a DN build for it. My phone
is setup as voice register pool 2. Config follows:
voice register global
mode cme
source-address 10.10.202.1 port 5060
max-dn 2
max-pool 2
load 7960
Hi all,
I am trying to figure out how to make UCCX say, You are caller number X in
the queue. I believe this is done with the Create TTS Prompt function
within UCCX. I then tried to play it with the Play Prompt function
referencing the TTS variable. The intended text shows up, however, the
letters (P[], S[], etc), let me know, I'm interested in what you find out.
Thanks again!
-Original Message-
From: Randall Saborio [mailto:ill2...@gmail.com]
Sent: Monday, July 12, 2010 2:40 PM
To: CCIE VOICE
Cc: ipexpert
Subject: Re: [OSL | CCIE_Voice] UCCX TTS Configuration
I believe
]
Sent: Monday, July 12, 2010 3:01 PM
To: CCIE Voice
Cc: ipexpert
Subject: Re: [OSL | CCIE_Voice] UCCX TTS Configuration
If you are just passing one number on the variable, it should be fine.
What do you have on that var?
I found the other options. If you go to the expression editor, and go
at
www.ipexpert.com/communities http://www.ipexpert.com/communities and our
public website at www.ipexpert.com http://www.ipexpert.com/
From: CCIE Voice ccievoiced...@gmail.com
Date: Mon, 12 Jul 2010 19:26:37 -0400
To: 'Tanner Ezell' tanner.ez...@gmail.com, 'Randall Saborio'
ill2...@gmail.com
Cc
Jeff is correct...whenever I have to make a change to CME SRST config,
especially the ephones, I copy all of my related config to notepad and issue
the no telephony-service command. I then paste everything back in and it
works fine. Don't know if this is a bug or what but it happens every single
If you have put a newer image on the cucm server make sure you have restarted
the tftp service. The server will not recognize new files until the service has
restarted.
--
On Jul 28, 2010, at 4:58, Shady Hasan shady@gmail.com wrote:
I know you may made this before, but:
1. make a
OK folks, I REALLY do not understand the following command. Cisco's
explanation states that it configures the drop thresholds for queue 2 to 40
and 60 percent of the allocated memory, guarantees 100% of the allocated
memory, and configures 200% as the maximum memory that this queue can have
I have run into a strange problem that I can not figure out. Dialing digits
on phone at BR2 (with what I can tell are correct CSS/partitions, gateway
assignments) disconnect immediately after completing the dialing. e.g.
Dialing 912123942123 call disconnects the moment that last digit is dialed.
Tried with 1 sip phone and 1 sccp phone. Route pattern was set to route. Thanks
for the ideas though.
--
On Aug 22, 2010, at 14:13, bkvalent...@gmail.com bkvalent...@gmail.com
wrote:
Was the phone using SIP?
- Reply message -
From: CCIE Voice cc...@corb.net
Date: Sun, Aug 22
registered.
Sent from my iPhone
On Aug 22, 2010, at 5:30 PM, Pavan pav.c...@gmail.com wrote:
In such cases grabbing detailed SDL SDI traces would immensley help.
Without them it is difficult to guess
Sent from my phone
On Aug 22, 2010, at 3:46 PM, CCIE Voice cc...@corb.net wrote
All:
When a G.729 call from an HQ phone to the BR2 CME phone CFNA's to CUE Pilot
- the CUE prompt doesn't play (call appears to be connected bu no audio) and
when looking at the tx/rx stats of the HQ calling phone - you can see it
transmitting all day long and never receiving.This isn't a
, Miron Kobelski findko...@gmail.com wrote:
Hi,
do you have the default gateway configured for CUE module?
Check if you have IP connectivity from CUE CLI to calling phone's subnet.
regards
kobel
On Sat, Sep 18, 2010 at 02:55, CCIE Voice cc...@corb.net wrote:
All:
When a G.729 call from
Make sure you do not have called transformation applied on the dev pool or the
phone. This could be localizing your display back to 7 digits.
--
On Nov 11, 2010, at 0:55, Mujaddid Ahmed mujaddi...@yahoo.com wrote:
Hi,
Please confirm what can be the reason of globalization not working
Make sure you add the subscriber to the publisher list of call manager servers.
--
On Nov 25, 2010, at 14:25, Wael Agina waelag...@gmail.com wrote:
Dear All,
I am implemnting new UC project at one customer location.
I installed the first node - publisher - succefully.
Now whenevr i
I think it is 7.01
--
On Jan 1, 2011, at 11:41, George Goglidze gogli...@gmail.com wrote:
Hi all,
I was wondering if anyone knew the exact version in a lab.
7.0 ? 7.1 ?
I know the blueprint says that any major version could be on a lab, but I was
wondering if anyone knew any better.
Hi Joli:
Make sure your trunk has been assigned a device pool that has a call manager
group with both servers listed. If it is using the Default DP it would only
be registering to the PUB.
hth...scd
On Sun, Jan 9, 2011 at 2:39 PM, Joli-coeur Wouter jwou...@gmail.com wrote:
Hello,
I am
. Be sure to visit our online communities at
www.ipexpert.com/communities *http://www.ipexpert.com/communities* and
our public website at www.ipexpert.com *http://www.ipexpert.com/*
From: CCIE Voice cc...@corb.net
Date: Wed, 19 Jan 2011 09:18:08 -0700
To: OSL Group ccie_voice
Hi All,
I am studying QoS and I am facing some problems to understand some concepts.
Anybody can help me to understand the following terms: ( I am now studying 3750
QoS):
1- Stack ring.
2- Queue-set.
I have other terms but I will add them later,
Regards,
of these template ?
From: William McCoy beci...@gmail.com
To: Ccie Voice v.c...@yahoo.com
Sent: Sun, January 23, 2011 7:22:06 AM
Subject: Re: [OSL | CCIE_Voice] QoS terms
This links provides a good description of the stack ring process:
http://www.cisco.com/en/US/prod
Hi all,
any body can help me to understand the following command:
Switch(config)#mls qos queue-set output 1 threshold 1 ?
1-3200 enter drop threshold1 1-3200
Switch(config)#mls qos queue-set output 1 threshold 1 2000 ?
1-3200 enter drop threshold2 1-3200
Switch(config)#mls qos
Hi all,
I did not enter CCIE LAB before and I am now preparing to the lab
I need to know how I can access the documents in the LAB and how I can practice
on this now.
Please if you attend the exam, share your experience with us
___
For more
Hi all,
any body try to configure:
QoS for CUPC ?
Is it detectable with CDP ?
Regards,
___
For more information regarding industry leading CCIE Lab training, please visit
www.ipexpert.com
Hi Michael,
it seems translation rule problem.
What is the called number from outside?
are you expecting 3500 or 53500 ?
From: Michael Luo hout...@gmail.com
To: ccie_voice@onlinestudylist.com
Sent: Tue, January 25, 2011 6:50:45 PM
Subject: [OSL |
It does not require CDP to use:
auto qos voip cisco-softphone
Thanks in advanced.
From: ccieid1ot ccieid...@gmail.com
To: Ccie Voice v.c...@yahoo.com
Cc: CCIE Study ccie_voice@onlinestudylist.com
Sent: Tue, January 25, 2011 9:10:46 PM
Subject: Re: [OSL
...@gmail.com
To: Ccie Voice v.c...@yahoo.com
Cc: CCIE Study ccie_voice@onlinestudylist.com
Sent: Tue, January 25, 2011 9:28:43 PM
Subject: Re: [OSL | CCIE_Voice] CUPC and 3750 switch
You ask for QOS. CDP should be detected for cisco hardware or software.
On Tue, Jan 25, 2011 at 12:25 PM, Ccie
thank you too much :)
From: Friderich Claude cfrider...@netcore.lu
To: Ccie Voice v.c...@yahoo.com; CCIE Study ccie_voice@onlinestudylist.com
Sent: Tue, January 25, 2011 11:29:24 PM
Subject: RE: [OSL | CCIE_Voice] CUPC and 3750 switch
Hi,
Look at this url
Check the below table:
DTMF Relay Type at Origination Side
DTMF Relay Type at Termination Side
H323 - H323
H245 alphanumeric H245 alphanumeric
H245 signal H245 signal
RFC 2833 RFC 2833
H245 alphanumeric H245 signal
RFC 2833 H245 signal
RFC 2833 H245 alphanumeric
H323 - SIP
H245
Hi All,
I am sorry for this silly question but I really need your help to understand
when I should click these check boxes and how to test the functionality:
Display IE Delivery
Redirecting Number IE Delivery-Outbound
Regards,
___
For more
,
From: Shrini linuxbos...@gmail.com
To: Ccie Voice v.c...@yahoo.com; CCIE Study ccie_voice@onlinestudylist.com
Sent: Fri, January 28, 2011 6:51:29 AM
Subject: RE: [OSL | CCIE_Voice] IE Delivery
Nothing is silly here :-)
Display ie - means when a user calls PSTN line
:05.283: ISDN Se0/0/0:23 Q931: RX - CONNECT_ACK pd = 8 callref =
0x0087
Regards,
From: Michael Luo hout...@gmail.com
To: Ccie Voice v.c...@yahoo.com
Cc: CCIE Study ccie_voice@onlinestudylist.com
Sent: Fri, January 28, 2011 6:57:58 AM
Subject: Re: [OSL
==
if I make call form PSTN to IP phone:
PSTN : to XX.2001
IP Phone: fromEmergency Services
Is it OK like this??? this is just with MGCP for H.323 still I have the
previous
problem :)
Regards,
From: Michael Luo hout...@gmail.com
To: Ccie Voice v.c...@yahoo.com
Why I need redirect number in this case??
From: Friderich Claude cfrider...@netcore.lu
To: Ccie Voice v.c...@yahoo.com; CCIE Study ccie_voice@onlinestudylist.com
Sent: Fri, January 28, 2011 6:07:39 PM
Subject: RE: [OSL | CCIE_Voice] IE Delivery
Hi
change the type of call so maybe this instruct the
router to deal with it in different way, I am just saying maybe :)
Regards,
From: Erik Goppel egop...@gmail.com
To: Ccie Voice v.c...@yahoo.com
Sent: Fri, January 28, 2011 8:10:13 PM
Subject: Re: [OSL | CCIE_Voice
Hi all,
anybody can help me to understand what is the exact meaning of the following
requirements:
Take clocking for Layer 1 from Network side?
Your PRI circuit Layer 2 should be user side?
Actually I am not able to understand what is Layer 1 and Layer 2
configuration is
the default.
Layer1 and Layer2 are referencing to the Layer1 and Layer 2 of
the
ISDN Protocol.
I hope this helps a little bit.
/Robert
Von:ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] Im Auftrag
von
Ccie
Hi,
I do not have too much experience in CUE, So I just need to know if there is
some features must be done through CLI and I am not able to do it through CUI
and Is there any restrictions in the CCIE Lab to use GUI or CLI?
Regards,
___
For
Hi all,
How I can know if the registered endpoints it is from local zone or remote
zone.
just by using the command show gatekeeper endpoints
and one more thing what is the difference between VoIP-GW and H323-GW in the
type column?
Regards,
Hi Michael,
Did you solve your problem.
I faced some of this problem and I solved and I did not solved it in the same
time :)
By chance I found that when I used Attendant Console software downloaded from
CUCM 6 it works with CUCM 7 fine but when I am using the Client downloaded from
the
But what about FTP???
CUE Require FTP not TFTP?? so how I can deal with it
Regards,
From: Ki Wi kiwi.vo...@gmail.com
To: ccieid1ot ccieid...@gmail.com
Cc: OSL Questions ccie_voice@onlinestudylist.com; Roger Källberg
roger.kallb...@cygate.se; voice boy
Hi All,
I am a little bit confused about how to set the value for Calling and Called
Party Number Plan.
let us say HQ Phone 1 Calls local Call in this case I think I have to set:
Calling Party Number Type to: Subscriber.
Called Party Number Type to: Subscriber.
What about Long Distance:
To: Ccie Voice v.c...@yahoo.com; CCIE Study ccie_voice@onlinestudylist.com
Sent: Thu, March 3, 2011 6:41:12 PM
Subject: SV: [OSL | CCIE_Voice] Calling and Called Party Number Type
Hi,
You need to look at this from the originating endpoint and the outgoing
gateway. For a more detailed explanation
confirm what Call Manager is sending to the PSTN.
On Thu, Mar 3, 2011 at 1:52 PM, Ccie Voice v.c...@yahoo.com wrote:
Thank you all for your reply,
I just need to know if the PSTN router in the LAB will accept
the call or no if it is not set to the proper
Thank you all for your clarification
From: Steve Denney (stdenney) stden...@cisco.com
To: Roger Källberg roger.kallb...@cygate.se; Ccie Voice v.c...@yahoo.com;
CCIE Study ccie_voice@onlinestudylist.com
Sent: Fri, March 4, 2011 5:49:48 PM
Subject: RE: [OSL
Hi All,
I did the following scenario:
HQ and BR1
I enabled MoH g711ual Multi-cast and I configured multicast under SRST
configuration.
if BR1-Ph1 called BR1-PH2 and press hold I can heat MoH from the flash of BR1
Router.
The problem is when I am calling form PSTN. I have FXS port if called
:10.30.30.254
gig0/0: 10.30.20.254
From: ccieid1ot ccieid...@gmail.com
To: Ccie Voice v.c...@yahoo.com
Cc: CCIE Study ccie_voice@onlinestudylist.com
Sent: Sat, March 5, 2011 12:17:33 AM
Subject: Re: [OSL | CCIE_Voice] Multicast MoH for Remote branches
I just need to add one note:
I stopped the CUCM service, so the phone registered with SRST router at that
point, if I called from analog phone to IP phone and press hold I can hear MoH
form the flash.
From: ccieid1ot ccieid...@gmail.com
To: Ccie Voice v.c
Did you activate AXL service in CUCM?
From: adam compton com...@gmail.com
To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
Sent: Sat, March 5, 2011 12:08:46 AM
Subject: Re: [OSL | CCIE_Voice] CUE to CUCM integration problems
I had this setup
Goglidze gogli...@gmail.com
To: Ccie Voice v.c...@yahoo.com
Cc: ccieid1ot ccieid...@gmail.com; CCIE Study ccie_voice@onlinestudylist.com
Sent: Sat, March 5, 2011 2:28:00 AM
Subject: Re: [OSL | CCIE_Voice] Multicast MoH for Remote branches
do you have ccm-manager music-on-hold ??? this is needed
Hi All,
Anybody can tell me how the MWI works with CUE integrated with CUCM?
Regards
___
For more information regarding industry leading CCIE Lab training, please visit
www.ipexpert.com
Hi all,
I configured CUE to be integrated with CUCM and everything is OK then I tested
it with CME-SRST and it was also OK. But the problem is: CTI Route Points is
not
registered after SRTS all phones in BR2 registered now with CUCM but CTI Route
Point and CTI ports are not registerd, I
lapb-ta mop udptn v120 ssh
line vty 0 4
password cisco
login
!
scheduler allocate 2 1000
ntp source GigabitEthernet0/0
ntp server 10.10.10.254
end
BR02-CME#
BR02-CME#
BR02-CME#
From: amit batra batraji...@yahoo.com
To: Ccie Voice v.c...@yahoo.com
Sent
,
From: George Goglidze gogli...@gmail.com
To: Ccie Voice v.c...@yahoo.com
Cc: CCIE Study ccie_voice@onlinestudylist.com
Sent: Sat, March 5, 2011 4:13:22 AM
Subject: Re: [OSL | CCIE_Voice] CUE Integrated with CUCM and MWI
Hi,
CTI Manager takes care of MWI ON/OFF, and just
I am having the exact same problem. virtual-access1, virtual-access2, and
virtual-template200 are all in a down/down or up/down state. Not sure how
to rectify it. Anyone else experienced this and figure out what was wrong?
On Tue, Nov 23, 2010 at 12:33 PM, Romain Mullier
Hi All,
During the reading of Cisco 3750 QoS Configuration Example I found the
following:
If you use the default policed-DSCP values, it does not make sense to
use
policing. For example, you have configured to police the traffic at the
rate
of 10 Mbps. The incoming packet
Thank you for your reply,Same what I am thinking about, I used the Policed-dscp map table. show mls qos map policed-dscp.From: Friderich Claude cfrider...@netcore.luTo: Friderich Claude cfrider...@netcore.lu; Ccie Voice v.c...@yahoo.com; CCIE Study ccie_voice@onlinestudylist.comSent: Mon, March 7
Hi all,
could you please tell me what is the most important things that I should adjust
if I need to use Auto QoS for MLP.
i know it depends on the requirements but there is some headlines like:
auto qos voip trust fr-atm
1- set the priority BW for voip to 33
2- change the mincir in
Sounds like your pstn router is not configured for moh.
--
On Mar 20, 2011, at 8:34, Michael Luo hout...@gmail.com wrote:
I was trying to configure a CME router to stream music on hold to its
e-phones. The CME router is also the voice gateway to PSTN.
Here are the results:
Scenario
You might need to type the ephones too.
--
On Mar 22, 2011, at 0:07, Rahul Kapor rahul.kapo...@gmail.com wrote:
Hi All,
earlier BLF call list in CME was working but not sure why its not working
now.
here is my config
sip-ua
presence enable
presence
presence call-list
Debug mgcp packet.
--
On Mar 25, 2011, at 18:20, adam compton com...@gmail.com wrote:
I'm not sure if the call could stay active unless it is the call manager that
fails. If you lost network connectivity or the T1 went down the call would
fail. I know you need the following command for
Hi Shaun:
I have 4 2811s in my lab. The minimum DSP resource I have on each is 2
PVDM2-16s. I have a voice T1/E1 from each site to the PSTN and a data T1/E1
to the PSTN. The OSPF is handled over the data connection via frame-relay.
On Sat, Mar 26, 2011 at 4:52 PM, Shaun P
Rahul:
You may need a 'no huntstop' under your ephone-dn 10
Example:
ephone-dn 5 octo-linec
number 4300 no-reg primary
conference ad-hoc
no huntstop
If that does not work do a 'sh sccp' and post that output.
On Mon, Mar 28, 2011 at 10:01 AM, Rahul Kapor rahul.kapo...@gmail.comwrote:
Hi
Your ccm group is configured to only register with 1 call mgr...
sccp ccm 14.160.110.22 identifier 3 version 7.0
reset the resources in call manager and do a 'no sccp/sccp' on the gateway.
Make sure your cfb in call mgr is defined correctly with identical name:
BR1-CONF
On Tue, Mar 29, 2011 at
Not sure if there is a CLI method to reset just the PVDM...would be good to
know. Power cycling the router would do it for sure though. :) (instead of
just a reload)
Sounds like a bugger of a problem...
On Tue, Mar 29, 2011 at 9:32 AM, Rahul Kapor rahul.kapo...@gmail.comwrote:
Hi All,
I
To replicate the lab scenario you should have 3 phones at hq, 2 at sites
br1/br2
and 1 pstn phone. Pstn phone can be 7960. Others should be 7965 or 7962 for
most accuracy.
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On Apr 2, 2011, at 2:36, Mohamed Gazzaz mgaz...@hotmail.com wrote:
Hello,
Ap per Ipexpert's voice
me just because they are CCIE and I
have more experience and knowledge.
Regards,
From: Alex a...@ipcomconsult.com
To: Justin Brady jbr...@tsginc.biz; ccie_voice-boun...@onlinestudylist.com;
George Goglidze gogli...@gmail.com; Ccie Voice v.c...@yahoo.com
Cc
Try leaving the controller in a shutdown state until everything is configured.
Then do a no shut on it. This seems to work for me every time.
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On Apr 3, 2011, at 21:56, Naoufal Kerboute naou...@mhdinfotech.com wrote:
Hi gents,
Please I need your help for the below.
I’m always
Hi all,
I need to run more than one auto attendant on the same router, depending on the
called number.
I usually do it by adjusting Tcl script, I need to know if you have better than
this solution
Regards,
___
For more information regarding
Hi,
For LAN you need to understand this document:
Cisco Catalyst 3750 QoS Configuration
Exampleshttp://www.cisco.com/en/US/products/hw/switches/ps5023/products_tech_note09186a0080883f9e.shtml
For WAN you have to use Auto-QoS
Regards,
From: Erwan Erwan
make sure that if you apply: show clock on the router it is showing the correct
time.
if the time OK under telephony-services issue: create cnf
Regards,
From: Rahul Kapor rahul.kapo...@gmail.com
To: ccie_voice@onlinestudylist.com
Sent: Thu, April 7, 2011
have to
adjust
auto-qos configuration.
Regards,
From: Erwan Erwan e_er...@yahoo.com
To: ccie_voice@onlinestudylist.com; Ccie Voice v.c...@yahoo.com
Sent: Thu, April 7, 2011 6:50:32 PM
Subject: Re: [OSL | CCIE_Voice] QoS example --quick reference
If I do
Enable sip diversion header.
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On Apr 10, 2011, at 17:10, Divin Mathew John divinj...@gmail.com wrote:
Call Flow
###
HQ- Phone CUCM -- SIP CME Br3-Phone-1 CFA
- CUE
Now in this call flow, the problem is that, CUE has no idea that the
Call was
It is user extension, I did this before in my lab and I imported users from AD
before and after the extension and I found that unity added some parameters
inside users information.
This what I know, maybe other guys here can give you more details.
Regards,
You need to run the sdspfarm commands and you need to do that in telephony
service. Don't think you can use CMF.
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On Apr 12, 2011, at 21:00, Erwan Erwan e_er...@yahoo.com wrote:
hi all,
why i can not register my IOS conf with call-manager-fallback ?
it show TCP_CONN_ERROR
I have a vouchers for sale that do not expire until after June if anyone is
interested. $10 each... pay via paypal.
Email if interested.
steve
___
For more information regarding industry leading CCIE Lab training, please visit
www.ipexpert.com
For h323 pstn gateway you only need the h323 VoIP bind command. The other h323
VoIP commands are for gatekeeper reg.
--
On May 1, 2011, at 17:17, Chevy chevy.man...@gmail.com wrote:
You were right I figured it out just now. A really stupid mistake.
On Sun, May 1, 2011 at 6:17 PM, Shrini
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