I believe that the bug is related to the intra/inter region issues - don't
have the bug ID handy but it has been discussed before in the forum(Hence,
it is suggested to hard-code the region relationships and set the intra
region to G.729).
You can set the BRQ enabled Service Parameter to TRUE and
interface Serial0/3/0.1 point-to-point
frame-relay interface-dlci 201
auto qos voip
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Shared line support between SCCP and SIP phones on CUCM is available. Not
available on CME.
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On Sun, Nov 20, 2011 at 12:16 AM, Kshitij Singhi
martinian.ksin...@gmail.com wrote:
Shared line support between SCCP and SIP phones on CUCM is available. Not
available on CME.
On Sat, Nov 19, 2011 at 10:30 PM,
ccie_voice-requ...@onlinestudylist.comwrote:
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this in lab scenario.
Regards,
Brajesh.
On Mon, Nov 14, 2011 at 6:06 PM, Kshitij Singhi
martinian.ksin...@gmail.com wrote:
Hi Sonu,
So for a call on the Branch site, I can think of a few scenarios:
1. Call coming in for an IP Phone at the branch site from the PSTN.
2. Call being
Hi Sonu,
So for a call on the Branch site, I can think of a few scenarios:
1. Call coming in for an IP Phone at the branch site from the PSTN.
2. Call being made from a phone at Branch Site 1 to another phone at Branch
Site 1.
3. Call being made from a phone at Branch Site X to a phone at Branch
Hi Gregg,
Yes, you are missing the dots after the MWI ON/OFF number. The reason why
CUE is not seeing the MWI on/off numbers is because an MWI outcall is
made by CUE in the following format:
MWI_ON_NUMBEREXTENSION_NUMBER - - - - - - - - For switching MWI on
MWI_OFF_NUMBEREXTENSION_NUMBER - - -
Hi Mike,
In the lab we get:
CUCM version 7.0.1-something
CME version: 7.0.x (IOS 12.4(20)Tx
CUE version: 7.0.x
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Hi Edgar,
I've always used the Phone button template. Never used the modify phone
buttons option.
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: Contents of CCIE_Voice digest...
Today's Topics:
1. Re: Modify button items vs Phone button template (Kshitij Singhi)
2. Re: UCCX G711 G729 (eng_firasoq...@yahoo.com)
3. Re: can not save script in script repository (CCIEVoiceKP)
4. Question about CUBE Gatekeepers (ccielabrat
Just wondering whether you got this:
Maybe this will help:
*Station and Backup Server KeepAlive Interval:* [image: Required
Field] This parameter designates the interval between KeepAlive
messages that are
sent from Cisco IP Phones to the secondary Cisco CallManager. This is a
required field.
Maybe this will help:
*Station and Backup Server KeepAlive Interval:* [image: Required
Field] This parameter designates the interval between KeepAlive
messages that are
sent from Cisco IP Phones to the secondary Cisco CallManager. This is a
required field. Default: 60 Minimum: 10
The reason why an incoming dial-peer is created is because PID 0 (or dial
peer 0) has certain characteristics which cannot be modified and don't work
too well when matched. For a VoIP dial peer 0:
- Supports any codec
- No DTMF relay
- IP precedence 0
- VAD-enabled
- No RSVP
Hi Randall,
Are you trying to go through the Initialization Wizard on CUE? Just
wondering where exactly are you being asked for authentication - is it
during the CME GUI login (via http://IP_OF_CME/ccme.html) or after you
have logged in to the CUE GUI and are going through the initialization
Hi Ken,
From a lab perspective, I didn't find any difference between SJ and
Brussels with respect to:
1. Speed
2. Ease of access
3. Proctors
The only difference was that the phones in SJ are mounted on wooden boards,
4 on one side and 4 on the other so you can simply reach out and test.
Phones
, 2011 at 5:29 AM, Kshitij Singhi
martinian.ksin...@gmail.com wrote:
Hi Ken,
From a lab perspective, I didn't find any difference between SJ and
Brussels with respect to:
1. Speed
2. Ease of access
3. Proctors
The only difference was that the phones in SJ are mounted on wooden
boards, 4
Hi Donny,
What I would suggest is:
0. Pray
1. Check replication
2. If broken, reload Pub, wait for about 5 minutes and then reload sub.
3. Start reading the lab during the wait for about 5 minutes period.
4. Once the sub reload has been initiated, start configuring the GWs
again and again.
Regards
Amit
Sent from my iPad
On 4/11/2011, at 7:00 AM, Kshitij Singhi martinian.ksin...@gmail.com
wrote:
Hi Donny,
What I would suggest is:
0. Pray
1. Check replication
2. If broken, reload Pub, wait for about 5 minutes and then reload sub.
3. Start reading
Hi Abbas,
Here is the data sheet for the ATA 187.
http://www.andovercg.com/datasheets/cisco-ata-187-analog-telephone-adaptors.pdf
Only supports T.38 faxing and fax pass-through.
Not sure about the franking machine being used but if does only modem
passthrough, then we are running into
with Franking machines (Kshitij Singhi)
--
Message: 1
Date: Tue, 1 Nov 2011 19:51:59 -0700
From: William Affeldt william.affe...@yahoo.com
To: Jonathan Preston jonathan_preston2...@yahoo.com
Cc: ccie_voice@onlinestudylist.com
Hi Baktha,
Voiceview auth in SRST isn't supported.
As far as the COR question goes, according to me, 4 digit dialing between
sites over the WAN would not count as international calls since they won't
be using the PSTN. However, in the event of WAN failure, if the call
attempts to go through the
Let me know if you would like to get on a Remote Support session and we can
see if this can be sorted out.
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Please send across the following:
1. Latest show run from the Site router and from the Telco.
2. debug voip dialpeer, debug voice ccapi inout, debug isdn q931 for a
call where you don't have the forward digits 7. In this case, make sure that
the destination-pattern is 9[2-9].. Please take
.
On Tue, Oct 18, 2011 at 1:25 AM, Kshitij Singhi
martinian.ksin...@gmail.com wrote:
Hi Ray,
For configuring B-ACD, you can browse to:
PSA webpage (this is available when you open IE. It may also be in the
bookmark but looks a lot like the download pages. Remember that search
I think bland was a typo for vlans :)
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not too familiar with US telco requirement so
this is good to know.
Sent from my iPad
On 15 Oct 2011, at 20:39, Kshitij Singhi martinian.ksin...@gmail.com
wrote:
Hi Brian,
That's quite a common requirement for Telcos. (not so much for 11 digit
calls but definitely
/ dnis
(Kshitij Singhi)
--
Message: 1
Date: Mon, 17 Oct 2011 00:23:53 -0700 (PDT)
From: mgscip gpsvoiceexpe...@yahoo.com
To: ccie ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] SRST Behaviour
Message-ID
Hi Ray,
For configuring B-ACD, you can browse to:
PSA webpage (this is available when you open IE. It may also be in the
bookmark but looks a lot like the download pages. Remember that search is
disabled on this, as are a few other options that are not specific to the
exam):
Here is what I had followed:
ALWAYS send a plan of ISDN.
If it has been stated that the Telco expects a specific plan/type, then
ensure that the plan/type is sent in the ani/dnis. Also, for an outgoing
call ensure that you take care of the digits seen on the calling phone ONLY
if specified in
for the response Kshitij
just curious - but what is the thinking behind sending type as unknown when
not presenting leading digits in the US?
Brian
On Sat, Oct 15, 2011 at 8:27 AM, Kshitij Singhi
martinian.ksin...@gmail.com wrote:
Here is what I had followed:
ALWAYS send a plan of ISDN
Hi Alejandra,
Can you try the following:
telephony-service
cnf-file perphone
url director http://IP OF CME/localdirectory
create cnf
Reset the phone(s) in question
and then test.
On Fri, Oct 14, 2011 at 3:00 AM, ccie_voice-requ...@onlinestudylist.comwrote:
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MGCP Gateway Endpoint
Configuration page of CUCM GUI as well.
Thanks,
Ken
On Fri, Oct 7, 2011 at 12:36 PM, Kshitij Singhi
martinian.ksin...@gmail.com wrote:
OK. Let's dance.
Given below is my configuration (the pertinent section):
show run | sec controller
controller T1 0/0/0
Hi Jason,
Check the CFWD CSS - make sure that it has access to the partition in
question.
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/mgcp
Should be good to go.
On Mon, Oct 3, 2011 at 11:17 AM, Kshitij Singhi martinian.ksin...@gmail.com
wrote:
Fractional MGCP controlled PRIs are not supported by TAC. It's not possible
to configure a fractional PRI by downloading the config from CUCM via the
following commands:
ccm-manager
Hi NoWork_onlyfun,
The debug that you need to run in order to figure this out will be debug
cch323 h225 and debug cch323 h245.
In the debugs, you should see:
Remote phone picking up and sending the connect to the CUBE.
CUBE will not forward the connect to the originator since it will wait for
Hi Stuart,
Here are the things that I would check:
1. Make sure that the GW CSS has access to the Route Pattern
2. Make sure that the Signifcant digits set on the GW are equal to the MVA
DN.
3. Make sure that the two service parameters in CUCM for MVA are enabled.
(one is for enabling MVA and
Multicast MOH is not working (Kshitij Singhi)
2. Huge Discount for ProctorLabs Vouchers (Marko Milivojevic)
3. NTP (Ray)
4. ntp (Ray)
--
Message: 1
Date: Sun, 18 Sep 2011 01:29:38 +0530
From: Kshitij Singhi
Hi Darsh,
1. Is the MOH Server in a Device Pool which has a region relationship of
g711ulaw to ALL other regions?
2. Have you checked the multicast option on the MRG?
3. Do you have the following commands on the Router:
call-manager-fallback
moh flash:file_name
multicast moh 239.1.1.1 port 16384
, Kshitij Singhi martinian.ksin...@gmail.com
wrote:
Hi Stuart,
Can you follow the steps given below in order and let us know if it works:
1. voice register pool 2
2. dtmf-relay rtp-nte
3. voice register global
4. create profile
5. voice register pool 2
6. reset
7. dial-peer voice xxx voip
8
Hi Stuart,
Can you follow the steps given below in order and let us know if it works:
1. voice register pool 2
2. dtmf-relay rtp-nte
3. voice register global
4. create profile
5. voice register pool 2
6. reset
7. dial-peer voice xxx voip
8. dtmf-relay rtp-nte
9. voice-class sip dtmf-relay force
Hi Emerson,
Can you send across the following please:
show run
show ephon phone-load
debug ephone detail mac-address MAC ADDRESS OF THE PHONE
debug tftp event
Once you enable the debugs, reset the phone in question. (The phone for
which you have enabled the ephone detail debug stated above)
simply
branding them as cheaters (maybe they just have more experience).
Also, your faith in TAC is great to see and I sincerely hope that we
continue to live up to your expectations.
*Thanks and regards,*
*Kshitij Singhi,*
*CCIE Voice # 29705*
*Cisco TAC - Unified Communications (Tech Lead
update-calendar
ntp server 192.168.13.57
ntp server 192.168.11.58
interface Ethernet 0/0
ntp broadcast
vines time use-system
On Fri, Sep 9, 2011 at 4:28 PM, Kshitij Singhi
martinian.ksin...@gmail.com wrote:
Here is what needs to be done:
Use clock
it and we will appreciate this , but leave all the other share them thoughts
whether you like them or not :)
Ash
On Fri, Sep 9, 2011 at 4:43 PM, Kshitij Singhi
martinian.ksin...@gmail.com wrote:
Hi Ken,
Whatever I had sent across in my email is verified. The theory is vast
with respect
Hi Ray,
47 is used since:
We add 7 bytes for LFI
2 for cRTP
20 for G.729 with a ptime of 20ms.
This equals to 29 bytes * 50 * 8 = 11.6 (this is the BW being used for 1
G.729 call)
4 G729 calls = 4 * 11.6 = 46.4
Hence, we put 47.
I have not read the question completely but I think this is what
attachment was scrubbed...
URL:
/archives/ccie_voice/attachments/20110910/22fda6ba/attachment-0001.html
--
Message: 3
Date: Sat, 10 Sep 2011 18:47:06 +0300
From: Ashraf Ayyash ash.ayy...@gmail.com
To: Kshitij Singhi martinian.ksin...@gmail.com
Cc: ccie_voice
Here is what needs to be done:
Use clock summer-time TIMEZONE offset from GMT
Hence, as an example, for PST/CST/EST it will be:
PST - - - clock time-zone PST -8
CST - - - clock time-zone CST -6
EST - - - clock time-zone EST -5
For ALL US sites, set the Daylight Savings to recurring. Hence,
Oops.. sorry... the second line should be:
Use clock time-zone TIMEZONE offset from GMT
On Fri, Sep 9, 2011 at 4:28 PM, Kshitij Singhi
martinian.ksin...@gmail.comwrote:
Here is what needs to be done:
Use clock summer-time TIMEZONE offset from GMT
Hence, as an example, for PST/CST/EST
know about exact timezone then how to set it?
as if ask for uk then what we have to set?
if ask for france or india what we have to set in router?
On Fri, Sep 9, 2011 at 1:58 PM, Kshitij Singhi
martinian.ksin...@gmail.com wrote:
Here is what needs to be done:
Use clock summer-time
, 2011 at 4:28 PM, Kshitij Singhi
martinian.ksin...@gmail.com wrote:
Here is what needs to be done:
Use clock summer-time TIMEZONE offset from GMT
Hence, as an example, for PST/CST/EST it will be:
PST - - - clock time-zone PST -8
CST - - - clock time-zone CST -6
EST - - - clock time-zone EST
Hi Ray,
Do the following:
1. Ensure the Site is NOT in SRST.
2. Shut down the serial interface
3. Remove ISDN L3 binding
4. Shut down the voice port
5. Shut down the controller
6. Do a no pri-g time 1-12,15-16 ser mg
7. Add pri-g time 1-12 ser mg
8. No shut the controller
9. Add L3 binding on
Hi Chevy,
Have never drawn a topology for the lab. Frankly speaking, for the Voice lab
it's not required. What I would suggest is ensuring that you have your own
flow or way of doing things. For example, I always created my dial peers
in the following way:
dial-peer 1 was for incoming calls
Hi Dew,
Most of the replies are inaccurate with respect to your question. A DSP is
invoked in ALL scenarios, except when there is a POTS - - - to - - - POTS
flow involved. In order to invoke the DSP when a POTS-to-POTS call flow is
involved, we use the command no local-bypass under voice-card.
= == === == == = === === == == === ==
C5510 23.8.0 g711ulaw 001 01 03 3/0/0:23 003 001 03 03 0 0
0 274/312
END OF FLEX VOICE CARD 3
Thanks Again
Ash
On Tue, Aug 23, 2011 at 9:00 PM, Kshitij Singhi
Hi Mike,
Have you stripped off the tech prefix on CUCM?
If not, add a Translation Pattern in the none partition:
1#.[23]XXX with a DDI of Pre-dot (assuming your extensions at HQ and BR1 are
2XXX and 3XXX respectively - modify as set up in your lab).
On Sun, Aug 21, 2011 at 11:59 PM,
Hi Alpesh,
Here is what I could find:
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note09186a00801786cb.shtml#sol3
Apparently, this seems to take care of the issue in most cases.
Randall - the sarcasm is unwanted and uncalled for.
On Thu, Aug 18, 2011 at 4:09 PM,
Hi,
Make the following change and test:
sip-ua
mwi-server ipv4:255.255.255.255 transp udp unsolicited
On Wed, Aug 10, 2011 at 8:18 PM, ccie_voice-requ...@onlinestudylist.comwrote:
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Forgot to add:
Also make sure that the gateway address is correct in ccn subsystem sip on
the CUE.
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Hi Mike,
It appears as if the PSU on the switch is faulty. Have you tried connecting
a non-PoE device to see if that works fine? Or perhaps give power to the
phone via a power brick and see if it registers as expected.
If you don't have any device connected to the switch, then there should
Hi Jason,
Add the type command under the voice register pool, do a create profile
under voice register global and then test.
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To
If it is a 7965 phone then that is the default. The + doesn't show up on the
main screen, but does show up in the bottom left hand corner when there is
an incoming call.
Options to work around this:
1. Use the incoming prefix settings on the H.323 GW page based on ISDN
plan/type
2. Use calling
, 27 Jul 2011 15:49:50 -0700 (PDT)
From: Vega Wong vega2...@yahoo.com.au
To: Kshitij Singhi martinian.ksin...@gmail.com,
ccie_voice@onlinestudylist.com, CCIE for Me cciefo...@hotmail.com
Subject: Re: [OSL | CCIE_Voice] Plus sign on display of phone -
assistance
Message-ID
for this ?
BR
Deepak
*From:* Kshitij Singhi martinian.ksin...@gmail.com
*To:* ccie_voice@onlinestudylist.com
*Sent:* Saturday, 2 July 2011 11:36 PM
*Subject:* Re: [OSL | CCIE_Voice] CUE MWI
Hi Deepak,
This is expected behavior. The MWI off event is sent only once the message
is deleted/saved
Hi Deepak,
This is expected behavior. The MWI off event is sent only once the message
is deleted/saved. Reading the message and leaving it in the inbox does not
trigger an MWI off.
On Sat, Jul 2, 2011 at 9:30 PM, ccie_voice-requ...@onlinestudylist.comwrote:
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