to see the expected/normal debug
output for the dtmf on this working scenario.
Hope this helps...
-Justin
(Sent from my phone, please excuse and/or laugh at any typos.)
On Jan 29, 2014 5:40 PM, Somphol Boonjing somp...@gmail.com wrote:
On Thu, Jan 30, 2014 at 8:48 AM, Vignesh Sethuraman
at 6:51 AM, Mark Holloway m...@markholloway.comwrote:
Something doesn't seem to add up in my head. Supp Services shouldn't
effect DTMF. Did you change anything related to the SIP Trunk on CUCM? Or
anything DTMF related on a dial-peer?
On Jan 30, 2014, at 6:22 AM, Vignesh Sethuraman sethuvign
Hello All,
I am facing an issue with dtmf-relay. PhoneA registered to CUCM (SiteA)
calls PhoneD registered to CUCME (Site C). Between Regions G729 codec is
negotiated. PhoneD call-forward no answer to Voicemail. CUE integrated with
CME. After leaving the Voicemail from PhoneA to PhoneD, when I
Dear All,
I have a basic question, when a site is in SRST and when I dial a PSTN
number or try to reach other site over PSTN, how come the GW at the SRST
site knows the called/calling party numbering plan and type. I do not have
any configuration related to that on the dial-peer or on the
supporting SIP.
Thanks,
Viki
On Sun, Jan 12, 2014 at 2:08 PM, Vignesh Sethuraman
sethuvign...@gmail.comwrote:
Dear All,
During the IPMA configuration, when I tried to do the Assistant
configuration for a user ID, the Device Name, Intercom Line Primary Line
are not listed.
But the manager
Hello All,
I could not get this question solved as per the solution mentioned in the
Proctor guide.
The SRST fallback incoming call at extn 1003 from PSTN hit the VM but I
don't get the Subscriber greeting but get a default message From a touch
tone telephone dial any extn.
I followed the OSL
Hello Mark,
yes, I do have *mgcp dtmf-relay voip codec all mode out-of-band.*
Thanks,
Viki
On Tue, Jan 21, 2014 at 8:57 PM, Mark Thrash (marthras)
marth...@cisco.comwrote:
Do you have the command
Mgcp dtmf codec all out
In your mgcp config
From: Vignesh Sethuraman sethuvign
Hello All,
Unity Connection not recognizing the password (no DTMF) when the call
is routed as following during a high availability situation.
SiteB PH2/PH3 --- MGCP T1 Port of SiteB GW My PSTN GW (use to switch
call between all sites via pots dialpeers) - SiteA H323 GW - CUCM
SUB
Hello All,
Is there a possibility to change the sampling rate on CUCM. If so, please
let me know where can I find it.
Thanks,
Viki
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Hello All,
I have configured the following in the CUCME. On the Skinny phone, the
Login Softkey is greyed out so that I could not able to login to test the
EM feature.
I am not sure why the login softkey is greyed out and let me know how to
activate it.
telephony-service
no auto-reg-ephone
ip http server is also configured.
On Mon, Jan 13, 2014 at 8:51 PM, Vignesh Sethuraman
sethuvign...@gmail.comwrote:
Hello All,
I have configured the following in the CUCME. On the Skinny phone, the
Login Softkey is greyed out so that I could not able to login to test the
EM feature.
I am
Hello All,
I am trying to find out the IP Phone services URL for IPMA from the DocCD
http://www.cisco.com/cisco/web/psa/default.html
Could you please someone point me to exact navigation.
Thanks,
Viki
___
Free CCIE RS, Collaboration, Data Center,
and Services Guide,
Release 7.0(1) do a search for MAservice or find chapter called Cisco
Unified Communications Manager Assistant With Proxy Line Support and do a
search for http, this one is a bit easier to remember.
Regards,
Attila
2014.01.12. 11:55 ezt írta (Vignesh Sethuraman sethuvign
of us have had, don't be surprised if the timers don't react the way that
they are configured. Please see other posts on the subject.
Josh
On Jan 10, 2014 3:26 PM, Vignesh Sethuraman sethuvign...@gmail.com
wrote:
Hi,
I am getting a fastbusy tone and unallocated number messager when I tried
Dear All,
During the IPMA configuration, when I tried to do the Assistant
configuration for a user ID, the Device Name, Intercom Line Primary Line
are not listed.
But the manager configuration is showing the relevant details in the
details in the drop down.
For the Assistant configuration, I
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Hi,
I am getting a fastbusy tone and unallocated number messager when I tried
to call the BACD pilot number from the PSTN and also from the CME
registered phones. Here is my config.
application
service aa flash:app-b-acd-aa-3.0.0.2.tcl
param number-of-hunt-grps 2
paramspace english index 1
Hello All,
I have one PVDM3-16 on my BR1 Router. I can use it for IOS transcoder and
also for IOS CFB. In the ipexpert VoD, I heard Vik Mahli saying the DSP
resources cannot be shared between transcoder and Conference bridge but
when I tried in my Lab it is been shared. I hope PVDM3-16 has got
Hello All,
I have 2 gateways one H323 GW (HQ) and one MGCP GW (BR1).
For 911 calls, I made the calling party transformation pattern as use
Device pool calling party transformation pattern CSS where I masked the
calling number as 7 Digits.
For Local calls, I created 2 RP one for BR1 and other
Hello,
I am trying to find out the document Configuring Conferencing and
Transcoding for Voice Gateway Routers' on the IOS 15MT using the
products/Technology page but could not see it.
I am not able to find it out on any of the config guides in the below URL.
Dear All,
Is this document available for the Voice candidates in the Lab.
MGCP and Related Protocols Configuration Guide
http://www.cisco.com/en/US/docs/ios-xml/ios/voice/mgcp/configuration/12-4t/vm-12-4t-book.html
If not, please let me know the on the SRND, on which topic the MGCP GW
Hello All,
I have integrated VG202 with CME using H323. The integration is through IP
connectivity.
I can make calls from IPphone registered to CME to the Analog phone
connected to VG202.
I could not hear any dial-tone when the Analog phone goes off-hook nor I
can dial any Cisco IP phone
Authentication Proxy Rights
• Standard Tab Sync User
Phone User
• Standard CTI Enabled
• Standard CTI Allow Control of Phones supporting Connected Xfer and conf
Or do I need to do all the above mentioned setup during my session?
Thanks,
Viki
On Thu, Oct 31, 2013 at 2:44 PM, Vignesh Sethuraman
Hello Experts,
I am trying to use Phone view (lab edition) software to control my Lab
phones. I can see registered phones on the Phone view but when dial any
extension from any of the registered phones, for example 5001, I see
message at the bottom on the activity log stating Command
Dear All,
In the CCIE Voice, IPX volume 1 task 11.3, I am unable to understand what
would be testing result if I press 3 as the caller input.
For caller input 3, the question says, option 3 should allow callers to
modify and enable any greeting for the call handler (including Alternate
will be prompted to enter
your user ID and password, example 5002 and a vm password of 12345. Once
you have been authenticated it will ask you to enter the number of the call
handler you wish to change followed by #. After that just follow the
prompts.
On Mon, Oct 21, 2013 at 12:21 PM, Vignesh
Hi Samson,
Have you hard coded the isdn channel to ascending or descending. If so try
to remove that and check.
Did you try isdn bchan-negotiate?
Did you see any errors on the output of show controllers t1/e1, and show
isdn status
Thanks,
Viki
On Monday, October 21, 2013, Samson Kareem wrote:
I meant the owner of HQph2 and BR1ph2 as the call handler owner.
On Monday, October 21, 2013, Vignesh Sethuraman wrote:
Hello Martin and Bill,
I have already assigned HQph2 and BR1ph2 as call handler owners, is this
you mean as assigning the role or something else?
Thanks,
Viki
Dear All,
I am facing issue in opening the PDF workbook and accessing my account in
ipexpert today.
I tried to send e-mail to supp...@ipexpert.com but still waiting for the
answer.
Just wanted to ensure if this problem exists only for me or to every
ipexpert customer.
Thanks,
Viki
Hello Experts,
I have registered my hardware 9971 SIP phone to CME. I would like to know
how to change the softkey template of 9971 SIP Phone to have the ad-hoc
conference facility. Moreover, do I need to do anything specific to make
9971 as dual-line as like it is did in Skinny Phones.
I tried
Hello,
I am trying to setup the Extension Mobility on CME, but when I press the
Mobility key, it shows key is not active
here is my config
*telephony-service*
* no auto-reg-ephone*
* authentication credential username password*
* em keep-history*
* max-ephones 1*
* max-dn 2 no-reg both*
Hi,
I am working on IPX Vol1 Lab 5C Task 5.8, I am not able to get the calls
working. When I checked the output of debug gatekeeper main 10, I could see
the following logs on the PSTN router (Remote Gatekeeper).
PSTNRouter#
Apr 27 09:55:21.071: ////GK/gk_rassrv_lrq:
Hi,
I am working on IPX Vol1 Lab 5C Task 5.8, I am not able to get the calls
working. When I checked the output of debug gatekeeper main 10, I could see
the following logs on the PSTN router (Remote Gatekeeper).
PSTNRouter#
Apr 27 09:55:21.071: ////GK/gk_rassrv_lrq:
Hello Experts,
I am working on Task 5.7 from Vol1. Question is to block the 91900? numbers. I
have configured a Route pattern to block this number but this Route pattern is
overridden by a another Route pattern 9.1[2-9]XX[2-9]XX which I have
created for Task 5.6.I understand the longest
Hello Experts,
I am not sure if this the right forum to post my question but giving a try.
I am in the initial stage of a project for Migrating Nortel to Cisco UC. I need
a Generic Design document and the implementation plan which I can use as a
reference to start with the project and will
Hi Farooq,
Did you see calls coming into the mgcp gw, use debug isdn q931 to check.
Try no mgcp mgcp on the Hq gw.
Thanks,
Viki
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Hi,
I am working on CCIE Voice IPX Vol1 task 5.1, as mentioned in the question, I
removed the SIP and tried to configure the HQ Router as H.323 GW. The issue is
it is affecting the task that I did in Lab 4A (4.6 and 4.7).
Basic question,can I have a router acting as H.323 GW and also as the
Hello All,
I was listening to Vol1 workbook video solution, in Task 4.2, question was to
add BR2 as the H323 GW but the video solution is about adding HQ RTR as MGCP GW
and configuring Route Groups etc.
Am I missing something or my understanding of Task is wrong?
Thanks,
gateway that should correct the problem. Its caused because you are
communicating between a SIP trunk and an H323 trunk.
On Tuesday, February 5, 2013, vignesh sethuraman wrote:
Hello,
In the task 4.7 of IPX Vol1 Lab 4A, am able to call and answer the calls to
BR2 from HQ and BR1
From: vignesh sethuraman sethuvign...@yahoo.co.in
To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
Sent: Wednesday, 6 February 2013 10:51 AM
Subject: Re: [OSL | CCIE_Voice] CCIE Voice - IPX Vol1 - Lab 4A - Task 4.7
Hello All,
The solution for my issue was both
Hello,
In the task 4.7 of IPX Vol1 Lab 4A, am able to call and answer the calls to BR2
from HQ and BR1.
But when I dial HQ phones or BR1 phones from BR2, am getting a ring back tone
but I could not answer the calls. Even after picking the handset, I could hear
the ringback on the BR2 phones.
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