Congratulations!!!
On Nov 22, 2011 10:43 PM, Mark Reed marklr...@gmail.com wrote:
I took my second attempt at RTP yesterday and passed. Thank you to
IPExpert for the great study materials. Going through their materials 60
hours a week made this possible. Vik, thank you for the great
This is wrong:
tftp-server flash:flash:/Desktops/320x212x12/voicesmall.png
tftp-server flash:flash:/Desktops/320x212x12/TN-voiceslarge.png
Change to:
tftp-server flash:/Desktops/320x212x12/voicesmall.png
tftp-server flash:/Desktops/320x212x12/TN-voiceslarge.png
On 22 September 2011 18:40,
The best way is to test on your rack to see what happens without and then
with those commands during srst and in normal operation having enabled your
voice and voip dialpeer debugs. Just my two cents towards being an expert.
Rogers
CCIE #28970 (Voice)
On 29 June 2011 09:01, donny f
You actualy don't need pim dense mode when route in below command
*multicast moh** multicast-address* *port* *port* [*route* *ip-address-list*]
route ip-address-list—(Optional) Declares the IP address or addresses from
which the flash MOH packets can be transmitted.
Rogers
On 27 June 2011
Congratulations!
On 8 June 2011 08:06, Miron Kobelski findko...@gmail.com wrote:
I received my number yesterday :)
This list has been very helpful during my preparations, so thank you all
and I wish you good luck with your exams!
best regards
kobel
On HQ side do trust if you've done LAN and all other sites dont trust since
you dont have LAN QoS
On Jun 6, 2011 4:03 AM, Michael Luo hout...@gmail.com wrote:
When configure auto QoS for WAN (on router), shall we use trust keyword
or
not? What's the best practice?
Thanks!
Michael
Ever seen this?
http://www.2bccie.com/2011/06/03/conflicting-ccie-numbers-29052.html
I've used the verification tool to confirm it.
___
For more information regarding industry leading CCIE Lab training, please visit
www.ipexpert.com
Are you a CCNP or
Your bet won :)
Rogers CCIE #28970
On May 25, 2011 4:44 AM, George Goglidze gogli...@gmail.com wrote:
If you have to change it, my bet is they will provide the FTP server...
Sent from my iPad
On 24 May 2011, at 19:26, Cristobal Priego cristobalpri...@gmail.com
wrote:
hello all
i was
Next business day. A friend took his test on Thursday in Brussels and got
the results on Friday. Check at CCIE Portal.
Rogers - CCIE #28970 (Voice)
On 28 May 2011 22:52, hmmm de...@wp.pl wrote:
Hi,
I was CCIE Voice lab In Brussels on Tuesday now is Saturday and I haven't
score report and
Then you surely know to raise the matter with cert support? :)
Rogers CCIE #28970 (Voice)
2011/5/29 hmmm de...@wp.pl
Hi,
I know that normally results are next business day, that wasn’t my first
attempt :-)
But on CCIE Portal i „Test passed” and „score” are still blank…
Congratulations Fatai, Enjoy!
Rogers - CCIE #28970 (Voice)
On 27 May 2011 20:57, Fatai Adekunle fatai_adeku...@yahoo.com wrote:
Hot and fresh. I passed my lab yesterday in brussels. Now i will take out
time to rest and relax. It was a sressful but worthy journey.
Fatai Adekunle CCIE
Freshly minted! I took my exam yesterday in Bangalore and the good news is
here!!!
Thanks to my study partners Michael, Fatai and Rahul!
CCIE#28970 - Voice
Rogers Ochieng
___
For more information regarding industry leading CCIE Lab training, please
--
Message: 2
Date: Fri, 20 May 2011 10:11:47 -0500
From: Michael Luo hout...@gmail.com
To: Rogers Ochieng rogersochi...@gmail.com
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Passed CCIE#28970!!!
Message-ID: BANLkTi=bdsnur2pjax0l5ahxlkfca-1
I'd say use the most efficient method that you can master. I prefered a
simple recording script so once i enter the UCCX section i don't leave the
UCCX Server until am done with it, need to hop on to Unity Connection to
record then import to UCCX.
CCIE# 28970 - Voice
Rogers
On 21 May 2011 05:19,
In which Device Pool and Region are your cue cti port and cti route point?
On 17 May 2011 00:49, Stephen Manuel srman...@bellsouth.net wrote:
In my home lab, I have the following
2811 router w/NM-CUE module w/7.0.1 software and CCM license.
VM Ware Call Manager 7.0.1 software
Router
OSL is fast loosing reputation as a list where CCIE NDA is openely violated.
On 20 May 2011 04:33, Peter Slow peter.s...@gmail.com wrote:
This thread shows blatant disregard for the CCIE NDA. You really
should not be posting your lab questions to anyone, let alone an
entire user group.
Remove the bind commands and let it register then you can put them back in
On May 17, 2011 1:03 AM, Hough, Earl earl.ho...@pcmallservices.com
wrote:
Randall,
Dumb question, but is your Loopback0 have an IP address and not shutdown?
Also, if you source a ping from the loopback0, is it able to
I see if you have single DN mapped to an ephone then busy-trigger is OK but
when you have two DN's on the ephone and want to forward on busy for one DN
then huntstop channel
2011/5/12 voice boy voice...@hotmail.com
can any one confirm on this
I know huntstop channel is used for shared lines,
After configuring E1 then maximum available will be the remaining resources.
if asked about maximum session 8 then its that 8. Don't use maximum
available because how else will you put in the 8 as in the requirement?
2011/5/8 voice boy voice...@hotmail.com
Hi,
I'm confused about the needed
-ignore*
* snmp trap link-status*
* frame-relay interface-dlci 501*
* ip rsvp bandwidth 40*
Is there any step by step guide to configure or troubleshoot it?
Warm Regards,
Vinay Kumar
From:
Vinay Kumar6/India/IBM@IBMIN
To:
Rogers Ochieng rogersochi...@gmail.com
Cc: ccie_voice
Here is the link with troubleshooting guid
http://www.cisco.com/en/US/docs/ios/voice/cminterop/configuration/guide/vc_rsvp_agent.pdf
On 29 April 2011 10:32, Rogers Ochieng rogersochi...@gmail.com wrote:
1. show sccp
2. show sccp connections
3. show sccp connections details
4. show sccp
do you have this under your IOS dspfarm profile Mtp configuration
rsvp
codec pass-through
On 29 April 2011 07:24, Vinay Kumar6 vinayjaisw...@in.ibm.com wrote:
Hi,
Trying to configure Location based CAC using RSVP, have done the
configuration buut it always says *not enough bandwidth*
That is expected as it will look for rsvp agent and find non so the call
will fail since its Mandatory. If you think about it they'll not use the
word Mandatory if the call can be allowed without rsvp agent, it would just
be bad English :)
On 27 April 2011 06:07, donny f f.faraday...@gmail.com
You need ip ospf network point-to-point on the int service-engine 0/0 too
On 18 April 2011 18:11, romain mullier romain.mull...@gmail.com wrote:
Hi guys,
So instead of using a vlan interface to bind the CUE module on BR2, I tried
to use a loopback and for some reason I cannot ping the
I'm on proctorlabs rack now and decided to have a look and my config below
works. I can ping the CUE IP
interface Service-Engine0/0
ip unnumbered Loopback1
ip ospf network point-to-point
service-module ip address 10.10.222.2 255.255.255.252
service-module ip default-gateway 10.10.222.1
I'm on proctorlabs rack now and decided to have a look and my config below
works. I can ping the CUE IP
interface Service-Engine0/0
ip unnumbered Loopback1
ip ospf network point-to-point
service-module ip address 10.10.222.2 255.255.255.252
service-module ip default-gateway 10.10.222.1
-Engine0/0 from what I saw. Is the
Design guide incorrect?
*From:* ccie_voice-boun...@onlinestudylist.com [mailto:
ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *romain mullier
*Sent:* Monday, April 18, 2011 9:58 AM
*To:* Rogers Ochieng
*Cc:* ccie_voice@onlinestudylist.com
*Subject:* Re
Please see config example in the support pages here:
http://www.cisco.com/en/US/products/sw/voicesw/ps5520/products_configuration_example09186a0080289ef0.shtml
Works for me
On 18 April 2011 18:46, Randall Crumm randall.cr...@flextronics.com wrote:
HI,
I was able to get my CTI ports to
.
On Mon, Apr 18, 2011 at 11:49 AM, Rogers Ochieng rogersochi...@gmail.com
wrote:
You need ip ospf network point-to-point on the int service-engine 0/0 too
On 18 April 2011 18:11, romain mullier romain.mull...@gmail.comwrote:
Hi guys,
So instead of using a vlan interface to bind the CUE
H323 is peer to peer, no registration to CUCM. So it can communicate to
unlimited clusters in this sense
On Apr 17, 2011 9:59 AM, Mann Chaddha mann.chad...@gmail.com wrote:
Hi Experts
I have these 2 questions related to H323 behavior on Cisco VGs:
1. Can I trombone a call to the same gateway
have to
configure for ACD AA application.
--
*From:* Rogers Ochieng rogersochi...@gmail.com
*To:* mgscip gpsvoiceexpe...@yahoo.com
*Cc:* ccie ccie_voice@onlinestudylist.com
*Sent:* Wed, April 13, 2011 8:30:20 PM
*Subject:* Re: [OSL | CCIE_Voice] B-ACD Not working
Used find the route pattern to hit to enable the phone ring the cellphone
and also when you press the mobility softkey to send the call to the
cellphone.
On 15 April 2011 07:33, Erwan Erwan e_er...@yahoo.com wrote:
hi guys,
wondering do we need Rerouting CSS in RDP Profile for SNR ?
-boun...@onlinestudylist.com [mailto:
ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Rogers Ochieng
*Sent:* Wednesday, April 13, 2011 8:00 AM
*To:* mgscip
*Cc:* ccie
*Subject:* Re: [OSL | CCIE_Voice] B-ACD Not working
param voice-mail is mandatory even if you are not sending the call
Have you followed the example on CUCM help pages? There's the h323 example
and the hairpining example.
On 14 April 2011 07:38, Shrini linuxbos...@gmail.com wrote:
Hi Experts,
I am able to configure MVA in h323 site but in MGCP I create similar voip
dial-peers and application but it is not
not possible with call-manager-fallback. You still have your 3 party
conference just that you can't configure hardware conferencing. Use
telephony-service SRST
On 13 April 2011 06:00, Erwan Erwan e_er...@yahoo.com wrote:
hi all,
why i can not register my IOS conf with call-manager-fallback ?
param voice-mail is mandatory even if you are not sending the call to voice
mail, you can configure a hunt pilot number or dn number
On 13 April 2011 16:35, mgscip gpsvoiceexpe...@yahoo.com wrote:
Hi ,
We tested with B-ACD in CME . whenever we dial the pilot number call
disconnect.
The simplest way is to create the phones in CUCM, choose SCCP as the
protocol, have your dhcp tftp settings set as the CUCM IP. Go to settings
and type **# and erase the configs to force the phone to acquire new
settings.
On 14 April 2011 07:24, Mann Chaddha mann.chad...@gmail.com wrote:
Hi All
them to be
directed to vm directly also right?
Thanks
Shingei
On Mon, Apr 11, 2011 at 1:30 PM, Rogers Ochieng
rogersochi...@gmail.comwrote:
Off the top of my head am thinking do call forward all and apply outgoing
voice translation rule under the ephone-dn to change your redirecting
That's why we have the have the option ccm-manager fallback-mgcp for MGCP
gateways. So configure that and either CME SRST or call-manager-fallback.
On 12 April 2011 00:51, Randall Crumm randall.cr...@flextronics.com wrote:
HI
In question 6 BACD, It says BR2 is in WAN outage and needs PSTN IB
CME 7 Admin Guide Talks abiut such restriction and mentions prebuilding
configuration to provide service similar to that during normal operation, on
page 1234
On 10 April 2011 15:28, Stern, Larry larry.st...@nuvt.com wrote:
I have ran into the same thing, see the note below from one of my
Looks similar to requirement of IPExpert Workbook 1 Lab 12A - 12.2
On 10 April 2011 15:40, Naoufal Kerboute naou...@mhdinfotech.com wrote:
The + symbol is a string so it can be match. My script is working if I
set the condition like If Calling Number == “+3434141891” then
redirect call
CME 7 Admin Guide Talks about such restriction and mentions prebuilding
configuration to provide service similar to that during normal operation, on
page 1234.
If i want names as in CUCM then I use mode autoprovision none and prebuild
my ephone dn's with the names as needed
On 9 April 2011
send your CME configs
On 11 April 2011 02:10, 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 forwarded. So it plays
...@onlinestudylist.com] *On Behalf Of *Rogers Ochieng
*Sent:* Tuesday, April 05, 2011 6:55 AM
*To:* ccie_voice@onlinestudylist.com
*Subject:* [OSL | CCIE_Voice] Debug Gatekeeper trunk call - codec mismatch
Which debug output will show any codec mismatch? I know that i need to
check for codec as one problem
...@onlinestudylist.com] *On Behalf Of *Naoufal Kerboute
*Sent:* Tuesday, April 05, 2011 12:14 AM
*To:* Rogers Ochieng; ccie_voice@onlinestudylist.com
*Subject:* Re: [OSL | CCIE_Voice] Debug Gatekeeper trunk call - codec
mismatch
You can run: debug gatekeeper call 10 or debug voice ccapi inout
NaoufaL
Also add isdn outgoing ie redirecting-number
On 4 April 2011 08:35, Hannes Dippenaar hanne...@hotmail.com wrote:
Hi,
Please see below:
interface Serial0/0/0:23
description # To Hannes HQ Router #
no ip address
encapsulation hdlc
isdn switch-type primary-net5
isdn
0 is the PSTN access code, and one 0 is LD code then 00 international code
So calling LD i dial 0 then my LD number starting with 0 0.0[1-9]XXX
International 0.00!
On 4 April 2011 21:12, Randall Crumm randall.cr...@flextronics.com wrote:
HI I was just looking over V2 L4 and have a
Which debug output will show any codec mismatch? I know that i need to check
for codec as one problem if a call between two endpoints get drop on answer.
I need an expert level debug i can send to an ITSP and tell them, here's the
mismatch.
___
For more
It depends on what you are asked to do.
On 2 April 2011 16:12, Michael Luo hout...@gmail.com wrote:
On LAN QoS, I've seen a configuration like below:
g0/1/0
srr-queue bandwidth share 1 30 40 30
srr-queue bandwidth shape 4 0 0 0
priority-queue out
From my understanding, priority-queue
have it i don't get a for and by field at all. To my knowledge, the
redirect ie is turned on under the serial interface by default.
A heads up, right now I have the following display
forwarded 5001
for: 1001
by: 1001
Thanks for all your help!
On Thu, Mar 31, 2011 at 1:51 PM, Rogers
Check to see that association there on the directory with the user, Line
association on the RD and RDP.
On 1 April 2011 06:08, Baktha Muralidharan muralic...@gmail.com wrote:
Hello folks
I am having trouble getting Mobile connect to work and would appreciate any
tips on the same.
- I have
I've only been able to get below:
From hq ph 1
(5001)
For: +16178631001 ( 1... )
by : +16178631001 ( 1... )
if i restrict calling name i get
From
(5001)
For: +16178631...
by : +16178631001 ( 1... )
Using the VM profile solution i've managed below without (1...) or +
From
(5001)
For:
at 12:36 PM, Rogers Ochieng
rogersochi...@gmail.com wrote:
Add
ephone-template 1
softkeys remote-in-use Cbarge NewCall
And also no huntstop on the conference ephone-dn
On 28 March 2011 19:01, Rahul Kapor rahul.kapo...@gmail.com wrote:
Hi all ,
Cbarge in SRST not working
here
Your sccp and telephony service config will help
On 28 March 2011 18:57, Rahul Kapor rahul.kapo...@gmail.com wrote:
Hi all ,
I am trying to register my hardware conference to cucm/cme but it is
stuck in TCP_CONN_ERROR error state.
sh sccp
Conferencing Oper State: ACTIVE_IN_PROGRESS -
Add
ephone-template 1
softkeys remote-in-use Cbarge NewCall
And also no huntstop on the conference ephone-dn
On 28 March 2011 19:01, Rahul Kapor rahul.kapo...@gmail.com wrote:
Hi all ,
Cbarge in SRST not working
here is my config
ephone-dn-template 1
call-forward busy 914082026002
are your directory numbers in a partion and do you have a css assigned to
the phone that can see the phone and pstn partitions?
On 29 March 2011 10:01, Rashid Khan me_rashid...@yahoo.com wrote:
Hi friends,
I have a cisco 7941 IP Phone, I am able to receive calls, but not able to
place any
Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30
Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization
Period: 30
reloaded the router again but no help.
thx,
Rahul
On Tue, Mar 29, 2011 at 12:28 PM, Rogers Ochieng
rogersochi...@gmail.comwrote
I think you'll have to configure 3 place holder ephone-dn's with no number
say
ephone-dn 5 dual-line
ephone-dn 6 dual-line
ephone-dn 7 dual-line
then
button 2:5 3:6 4:7
this will force the blf speed dial to be pushed to button 5
On 28 March 2011 07:31, Michael Luo hout...@gmail.com wrote:
If i have a requirement to only use local gateway for a route pattern, is it
advisable to use a specific route list pointing to the route group with the
gateway or a route list pointing to a SLRG will do?
Rogers
___
For more information regarding
With CoS mapping to a queue and doing share/shape on the trunk port to
router
On 27 March 2011 18:43, a...@ipcomconsult.com wrote:
Hey Experts
Anybody can clarify on this topic?
How to GUARANTEE bandwidth for incoming traffic on 3750?
Thanks
- Forwarded message from
March 2011 19:24, Roger Carpio roger.car...@gmail.com wrote:
How many sites are supposed to use this RP? If only one site is meant to
use it; SLRG will have an extra step (assigning the RG to the DP).
Regards,
Roger Carpio.
On Sun, Mar 27, 2011 at 9:22 AM, Rogers Ochieng
rogersochi
remove mgcp bindings if any and after it registers put them back, you can
also try shut, no shut the controller
On 22 March 2011 09:31, Shrini linuxbos...@gmail.com wrote:
On CCM it is registered to right interface and right CCM.
On router sh ccm-manager returns Primary call manager
http://www.cisco.com/en/US/partner/products/sw/voicesw/ps5520/products_configuration_example09186a0080289ef0.shtml
On 21 March 2011 07:38, Michael Luo hout...@gmail.com wrote:
Can anyone send me a link to CUE/CME MWI configuration example?
Whenever I left a message, CUE VM pilot will call the
Use this
http://www.cisco.com/en/US/partner/products/sw/voicesw/ps5520/products_configuration_example09186a008037f2a9.shtml
On 21 March 2011 07:38, Michael Luo hout...@gmail.com wrote:
Can anyone send me a link to CUE/CME MWI configuration example?
Whenever I left a message, CUE VM pilot
As expected you are not supposed to have control of PSTN so you can't go
configure MOH over there to play MOH for PSTN held calls. But in our case on
proctorlabs to satisfy your need, configure MoH on the PSTN router.
On 20 March 2011 17:34, Michael Luo hout...@gmail.com wrote:
I was trying to
If the Lab ask for it or in real world if your network operator ask for it.
On 19 March 2011 00:24, Michael Luo hout...@gmail.com wrote:
We usually set the called party Type (Subscriber, National, International),
so provider and do appropriate routing.
What's the purpose of setting it on the
Lets be polite guys! People ask OT questions all the time, if i have a quick
answer i'll give it or politely mention a useful link or forum. I saw
someone throwing a tirade at Ash sometime back for OT, i think he didn't
know that the guy is a CCIE and has a great blog,the kind of contribution
the
Did you configure the no-reg after the gateway to GK registration? if yes
then you need to restart the gateway
On 13 March 2011 13:01, Adil Shaikh adil.sha...@gmail.com wrote:
Hi All,
I have configured 'no-reg both' on the BR2-RTR's ephone-dn and
telephony-service, still ephone-dn get
Was wondering why my session only had 7 hours left from my usual lab
starting time.Did a quick google of EST to GMT and saw my lost one hour :)
..No daylight saving in my region so was caught unawares today of this
change until 6th November.
___
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I'm getting the same error, i thought it was my browser. I've tried in IE
and Firefox
On 13 March 2011 16:33, Kalyan iyer kparam2...@gmail.com wrote:
Hi guys,
I have the rack rental session from 8AM to 4 PM. I am trying to connect to
proctorlabs rack and load initial configs. I keep getting
I've gotten the same issues devices going off, i put them on and they get
back off after a short time. I've already wasted close to 3 hours, so
frustrating.
On 13 March 2011 17:02, Erwan Erwan e_er...@yahoo.com wrote:
I got same issue lost 2h now, and they said their web has issue
And
Also check that the interface of the ip address you used to add the gateway
in CUCM has the h323-gateway voip bind srcaddr to be sure that h323 is
sourced from that IP otherwise CUCM will reject the call unless connection
from unknow source is allowed in service parameters. Please send the other
If you are using CUBE then you shouldn't have dial peer to remote gateway,
the call should be sent back to the gatekeeper, hope you've registered the
HQ gateway to the CUBE zone, you only need the outvia as you are only
sending calls to the remote zone so the dial peer you need would be like
on
the gateway which is g729 and so your phone will try to listen to the g729
ip/port. If you handle your codec requirement well then this is a problem
that doesn't come to mind
On 8 March 2011 06:48, Rogers Ochieng rogersochi...@gmail.com wrote:
*The below two lines are not needed as we are routing
AS you've stated you are using CUE which in normal operations you've
integrated using jtapi CUE integration, i assuem the CUE module is on the
BR2 router. So for SRST create a voip dial-peer using sip protocol and codec
g711ulaw, dtmf sip-notify, to route calls to CUE and set CFB and CFNA, setup
8, 2011 at 10:26 PM, Rogers Ochieng
rogersochi...@gmail.comwrote:
AS you've stated you are using CUE which in normal operations you've
integrated using jtapi CUE integration, i assuem the CUE module is on the
BR2 router. So for SRST create a voip dial-peer using sip protocol and codec
Why would you need all those regions? ICD is in HQ Device Pool so traffic
within the same region will be G711, CUE will be on the Br2 device pool so
g7711 with thin that region. Since interegion codec is g729 calls from BR1
will cause transcoder to be invoked
On 8 March 2011 22:49, Rahul Kapor
you try in your lab and let me know about the result ?
thx,
Rahul
On Mon, Mar 7, 2011 at 10:46 AM, ccieid1ot ccieid...@gmail.com wrote:
Thanks Roger! Hopefully Rahul agrees as well. :P
duy
ccie #27737 voice
tmobile g2
On Mar 6, 2011 10:53 AM, Rogers Ochieng rogersochi...@gmail.com
Whats your number? I see most people are always excited enough to flash
their number :)
On 6 March 2011 11:38, voicemail voicemail voicemail20...@gmail.com wrote:
YAH
I am so happy to pass.. CCIE VOICE:))
So here is the updates
If there is no VoIP activity in the MGCP gateway, after the defined
receive-rtcp timer, the gateway tries to disconnect the call
The following command should sort it out
no mgcp timer receive-rtcp
Rogers
On 6 March 2011 18:18, Mritunjay Kumar mjs...@gmail.com wrote:
Hi All,
After stopping
phone , not by BR1 phone . becos still Br1 phone will try
listen the MOH on base port. so voice class is mandatory.
what u think ?
thx,
Rahul
On Wed, Mar 2, 2011 at 7:56 PM, Rogers Ochieng
rogersochi...@gmail.comwrote:
If you are sourcing MOH from Flash for a CUCM Branch phone
will be
negotiated as gateway and phone are in same region and voice class codec is
configured with g711
i would like to have your feedback.
thx,
Rahul
On Wed, Mar 2, 2011 at 9:11 AM, Rogers Ochieng rogersochi...@gmail.comwrote:
When there is a requirement for HQ Phones calls to BR1 area
I don't get what you mean by Somehow when SRST,the softkeys cbargedoesn't
appear on the phone when the
shared line was answered by another ip phone
Do you mean when you press the shared line on the phone or you want it to
appear in idle state?
On 25 February 2011 17:42, ShinGei Yong
What is at SiteB and SiteC? CUCM, CUCME? some configs might help
On 23 February 2011 10:03, Ki Wi kiwi.vo...@gmail.com wrote:
I have enabled MMOH for both sites.
If Site B or Site C calls PSTN and put it on hold, PSTN can receive the
multicast MOH music. However it i use either site B to
If you are using H323 gateway then you will face this challenge, best to use
srst mode auto provision none so that the ephone-dn configs are not availab
On 23 February 2011 04:43, want ccie wantc...@gmail.com wrote:
Hi All,
I setup my H323 GW w/ SRST and calls are working fine. I'm trying to
Have you checked the phone configuration to ensure it has picked the SRST
reference? Try restarting tftp service and reset the phones to pick
configuration
On 25 January 2011 16:29, ShinGei Yong shingei.y...@gmail.com wrote:
Hi,
I've couple of 7961G ipphone unable to register to SRST
during
As Ash said, either way and as Roger put it learn both ways. The crazy Lab
might just ask you to use a specific one
2011/1/19 bruno bruno.juni...@gmail.com
thanks for discuss. anyone can give me some confirm?
-- Original --
*From: * Rogers
Mobile Voice Access is accessed by calling a system-configured DID number
that is answered and handled by an H.323 or SIP VoiceXML (VXML) gateway. The
VoiceXML gateway plays interactive voice response (IVR) prompts to the
Mobile Voice Access user, requesting user authentication and input of a
It's in the CUCM 7x SRND which is also available in the lab.
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/7x/uc7_0.html
On 19 January 2011 19:58, mihal caro mihalc...@gmail.com wrote:
Hi Miron
thanks for the reply
I've try ed Sip-Notify same issue. I can get it work with h323.
Hi Michael,
I see you are mixing technologies here by looking at the peer to peer h323
configuration and the Gatekeeper controlled h323 configuration. I like the
way it's been intuitively named Gatekeeper-controlled Trunk. peer to peer
h323 configuration will work as you are thinking but when the
if you apply bandwidth higher than 768 then it will not create virtual
template
On 15 January 2011 17:38, Roig Borrell, Francesc Xavier
francesc.ro...@tecnocom.es wrote:
Hi all,
I am experiencing a very strange issue applying (Auto qos voip fr-atm/ Auto
qos voip trust fr-atm) command in
CCM service paramters MVA number is used when integreting with Mobile
Communicator which is not in the blueprint. I can swear Vik does mention
this in Volume 1 Walkthrough
On 13 January 2011 19:53, Randall Saborio ill2...@gmail.com wrote:
I see what you mean. As a matter of chance I was doing
Congratulations man.
Any new topics introduced? I'm taking my 3rd attempt on 11th
Rogers
Kenya
On 25 December 2010 14:33, Sriharshaa Prabhakar
sriharshaa.prabha...@mannai.com.qa wrote:
Hi All,
I have cleared my CCIE Voice Certification at Brussels and my number is
27816, this is an
You don't need UPC client in CUCM to have Deskphone control. You have to
configure Application Cisco Unified Personal Communicator and Application
Deskphone. If i need Softphone only i add the client in CUCM and remove
CTI gateway from the CUPC settings.
On 28 December 2010 20:24, Casey Lee
graceful is differ from immediate
with respect to call preserve ?
thx,
Rahul
On Thu, Dec 9, 2010 at 12:06 AM, Rogers Ochieng
rogersochi...@gmail.comwrote:
Calls are supposed to be preserved during switchover and switchback
events. Think about it, otherwise mgcp could not have been
Jason he's talking about CME not CUE. I get through
http://myrouterip/telephony-service.html.
On 8 December 2010 20:45, Jason Aarons (US) jason.aar...@us.didata.comwrote:
Don’t you have to session into the module to run initial setup, before the
Web GUI will be active.
In the session you
I use this link http://myrouterip/telephony_service.html
This link is helpful https://supportforums.cisco.com/docs/DOC-8172
On 8 December 2010 20:19, Baktha Muralidharan muralic...@gmail.com wrote:
Hi folks
As part of Lab 3A, I configured for GUI access into the CME, as follows
ip http
Calls are supposed to be preserved during switchover and switchback events.
Think about it, otherwise mgcp could not have been recommended for
redundancy and we would prefer h323 with call-preserve feature.
On 8 December 2010 20:57, Rahul Kapor rahul.kapo...@gmail.com wrote:
Hi Mate,
on
In Phone configuration there is Do Not Disturb Select DND Option*
Call Reject
On 25 November 2010 14:56, Rahul Kapor rahul.kapo...@gmail.com wrote:
Hello Mate ,
how to divert incoming call to voice mail in DND mode or by just pressing
DND button/softkey ?
Regards,
Rahul
On the GK Trunk in CUCM, uncheck Wait for Far End H.245 Terminal Capability
Set
On 18 November 2010 11:44, Shrini linuxbos...@gmail.com wrote:
Hi Experts:
When I redial a HQ missed call (+1408XXX) on Br2 Ph1 (International) ,
HQ Phone is recieving the call, I am able to pickup but in
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