For #2, just filter “sip” in your captured session to see just the sip message
flow. To see a ladder diagram, click Telephony on the top menu and select VoIP
Calls. Wireshark will then show a pop up window with all captured voice calls.
Double click the call to see a ladder diagram.
This is a
From a Cisco partner business perspective I don’t see how it would make sense
to have your current CCIE’s retake the lab when they could just as easily
migrate to Collaboration after taking their written renewal. Time is money, and
consuming your top resources with another lab certification
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...@gmail.com wrote:
Hello Somphol/Justin,
I have
/products_configuration_example09186a00808f9666.shtml
He would see the issue in the debugs
On Thu, Jan 30, 2014 at 6:51 AM, Mark Holloway m...@markholloway.com wrote:
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
Regards,
Moataz Tolba
On Thursday, 30 January 2014, 15:17, Mark Holloway m...@markholloway.com
wrote:
I understand how DTMF works on SIP Trunks, what I’m not clear on is how “no
supp services” would have an impact on his DTMF issue. I’m trying to
understand the logic of something
What is it with RTP and fish? The day I took my lab there they served chicken
la cordon bleu, which is just as bad when your stomach is in knots. In San Jose
we went to the cafeteria and I had salad. Much better, and I passed.
On Dec 16, 2013, at 12:37 PM, Bill Lake whl...@gmail.com wrote:
3 months? You have until 2016 to convert from CCIE Voice to CCIE
Collaboration.
--snip--
Pass the CCIE Collaboration Written Exam and then permanently convert your CCIE
Voice certification to a CCIE Collaboration certification between November 21,
2013 and February 13, 2016.
On Sep 18,
This is a valid question. I hope the answer is yes.
On Jun 2, 2013, at 10:28 PM, Karen Johnson karen.johnson...@yahoo.ca wrote:
My question is :
If we only pass ccie voice lab then get ccie number and we keep renewing it .
When cisco retired it in Feb 2014. Does my ccie number count for
from multiple different voice study
group to have a migration track to Collaboration please share your thoughts
guys
On 28 May 2013 18:56, Mark Holloway m...@markholloway.com wrote:
Bummer, I was really hoping CCIE Voice candidates would transition to
Collaboration without any
Bummer, I was really hoping CCIE Voice candidates would transition to
Collaboration without any additional lab exams.
On May 28, 2013, at 7:08 PM, Vik Malhi vma...@ipexpert.com wrote:
For my initial reaction read here:
http://bit.ly/12MNK5t
Vik Malhi – CCIE #13890
Managing Partner -
You should not concern yourself with the blue print change at this point. Focus
on passing the current blue print and you will make it happen. If the blue
print changes, you have 6 months to continue studying, but even when Cisco
starts testing the new blue print it's going to take some time
If memory serves me correctly, if they reach a point in the lab grading where
you've already reached the failing mark, they don't proceed to grade the rest
of the exam and you just get Fail for remaining sections. I cannot verify that
though and it has been two years since I passed.
I took my
Congratulations. It's hard to believe there are 10,000 more CCIE's in just two
years. I earned mine on October 28, 2010. I just took my written again a couple
of days ago.
Mark
CCIE #27384 (Voice)
On Oct 24, 2012, at 11:51 AM, Bruno Nonogaki wrote:
Hello guys,
I have just received my
CUCM 7.01 has a bug. The VM Profile is a work around. Every time I would
reset my CUCM VM's sometimes I would get the expected display as shown below,
other times I wouldn't. A few other folks confirmed this as well. It's a
display issue with 7.01 and VMware
On Mar 31, 2011, at 12:19 PM,
Usually this is because you are using Single Button Cbarge on CUCM which breaks
Cbarge in SRST. It's a known issue. Use standard Cbarge in CUCM and it should
work in SRST.
On Mar 29, 2011, at 9:08 AM, adam compton wrote:
Do you see the CBarge softkey but the conference gives you a busy
I thought I was the only one who picked up on that. :-)
Although, it's 81 degrees in Phoenix with sunny skies. So technically San Jose
is in a crisis situation compared to Phoenix.
On Feb 16, 2011, at 7:46 AM, Rrcrumm wrote:
Lol
the weather has the city on lockdown
Sent from my
I have a few pieces of gear that I'm willing to part with if anyone is
interested. Primarily 2811's and 2821 fully equipped, which works great with
the IPX labs. It's best to take this discussion off the list, but I'd much
rather see my equipment find a home with someone who is working towards
I only had to show my driver's license. I took my first attempt in RTP and the
second in San Jose.
On Jan 26, 2011, at 10:22 AM, Matteo B. wrote:
Hello People...
next Friday i'm going to sit for the lab...( cross your finger!! )
On the confirmation email there is a note that say i've to
I recommend that you know both ways. The lab might tell you what you cannot do
which means there is only one other option remaining in order to get the
question right.
On Jan 18, 2011, at 11:18 PM, bruno wrote:
Dear all
in vol1 network infrastrure,
why we need to configure trunk mode on
If you configure Single Button cBarge in CUCM then it won't work in SRST.
On Jan 18, 2011, at 11:29 AM, Amit Batra wrote:
Hello guys
May be I am wrong. But I kind of remember this issue. People talked a
lot about it. And is a known bug. If this is the same issue which we are
Like Matt said, perform a sip debug. You will most likely see that CUCM is
responding with something. It may be something like 404 not found. If that is
the case look at the Called number in the SIP Invite and make sure it matches
your dial plan in CUCM.
On Dec 31, 2010, at 9:00 AM, Matthew
I want to say thank you to everyone on the OSL who has participated in any of
my discussions or helped resolve issues that I encountered. I went to San Jose
for my second attempt on Friday and received the news yesterday that I passed.
CCIE #27384.
Thanks,
Mark
I want to create a qos policy to police sip traffic on my HQ 3750 and remark
excess to dscp 0. Can someone explain the difference in a policy-map between
'set dscp 24' and 'set ip dscp 24'? Also, is it accurate to set the burst to
8000 or should it be a minimum of 16000 burst, or is it
I ran into this same issue. Supposedly it's a problem with CUCM 7.0 and VMWare.
If you go to the Device Phone and click on the Site B phones Line and
specifically assign the Voicemail Profile to the Line it might work. I had
success a couple of times doing this, but then after resetting my
-called number from to the e164 that you are after.
Cheers
On Mon, Oct 18, 2010 at 11:36 AM, Mark Holloway m...@markholloway.com wrote:
I ran into this same issue. Supposedly it's a problem with CUCM 7.0 and
VMWare. If you go to the Device Phone and click on the Site B phones
Line
I had to upgrade my CUE from 2.1.2 to 7.01.
http://www.markholloway.com/blog/?p=595
On Oct 12, 2010, at 3:33 AM, Amr Sherif wrote:
Hello Experts,
I have CUE version 3.2.3 and i want to upgrade to 7.0 which is the exam
version . The license for my CUE is CME mode and it's embedded
Has anyone ever experienced an issue where you assign DnD to a softkey template
in CUCM, assign that template to a phone (which also has voicemail in Unity
Connection), but when a call comes into that phone with DnD, it still rings
instead of going to voicemail even if DnD is Active? Even if I
Hmm, I actually had to go to the Device Phone HQPH1 and scroll down to Do
Not Disturb and hard-set it to Call Reject (instead of Default Profile
Behavior). I thought the default behavior was in fact Reject.
On Oct 12, 2010, at 11:46 AM, Mark Holloway wrote:
Has anyone ever experienced
I always shut down the Serial interface of the Frame Relay WAN link.
On Oct 10, 2010, at 9:44 PM, Pithog Oil wrote:
Hi experts,
What is the best and quickest way to invoke SRST in the labs, for me the only
way i have tested as at now is the ip expert proctorlabs way, of creating sub
) with
training locations throughout the United States, Europe, South Asia and
Australia. 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: Mark Holloway m
(I
experience different behavior each time I reset my rack too), so as odd as it
is I think it's important to know the Voicemail profile assignment is a valid
fix.
On Oct 9, 2010, at 9:39 AM, Mark Holloway wrote:
Ok, the secret to getting it to work every time is going to Device Phone
Line
If I want limit BR2Ph1 to 3 incoming calls and BR2Ph2 to 6 incoming calls, how
can I control the total number of incoming calls to each phone if there is more
than one ephone-dn assign to the phone? For example, if 6001 is an octo line
assigned to Ph1, 6002 is an octo line assigned to Ph2, and
Hmm, PSTN to BR1 and IP to IP (inter and intra site) play multicast MoH piano
music from route flash just fine, but for some reason when calling from BR1 to
the PSTN and pressing HOLD on the BR1 phone it plays beep beep beep.
Usually the issue is PSTN to IP because you need a voice class
Does anyone know if/what UCCX wav file says Please try again later
Thanks,
Mark
___
For more information regarding industry leading CCIE Lab training, please visit
www.ipexpert.com
Has anyone ever seen this before?
I login to Unity Connection then click on my BR1PH1 user so I can record a
custom greeting.
inline: PastedGraphic-2.png
When I press the Record button I get the following error.
inline: PastedGraphic-3.png___
I'm trying to get my CFUR to work so it shows the External Mask in the For and
By part of the call presentation but instead I am only getting it to show the 4
digit extension. For example, lets say HQ 5001 calls BR1 3001 (3001 is
unregistered and has CFUR set in CUCM to dial out the PSTN
to external mask everytime the 5xxx makes
a call out of that gateway.
HTH
Prashant
On Fri, Oct 8, 2010 at 6:39 PM, Mark Holloway m...@markholloway.com wrote:
I'm trying to get my CFUR to work so it shows the External Mask in the For
and By part of the call presentation but instead I am only
I have had it working before, but it's odd because sometimes when I reset the
lab rack I can get it work and other times it does not work the way I want.
I'm trying to figure out if I keep overlooking something.
On Oct 8, 2010, at 4:08 PM, Mark Holloway wrote:
I do not want to modify 5XXX
I'm trying to create a policy map that matches the skinny signaling protocol
that will police it and rewrite the exceeded packets from dscp 24 to 0. I am
pretty sure I have the policy map created correctly but when I do 'show
policy-map interface interface' I am not seeing the counters
managing the list at
ccie_voice-ow...@onlinestudylist.com
When replying, please edit your Subject line so it is more specific
than Re: Contents of CCIE_Voice digest...
Today's Topics:
1. MoH SRST (Stream from Flash)` (Mark Holloway)
2. Re: MoH SRST (Stream from Flash)` (Prashant
is) and prefix the 555. Other sites will still show
the full E.164 number.
Graham
On 1 Oct 2010, at 18:00, Mark Holloway wrote:
The crazy thing is I tried this but I couldn't get it to work.
PT_HQ_CALLING_OUT assigned to CSS_HQ_CALLING_OUT assigned to CALLING Number
Transform
the results to get a
good understanding of where patterns are matched and which transformations
overwrite other ones.
Dial Number Analyzer is also pretty handy here if something doesn't match and
you think it should
Graham
On 4 Oct 2010, at 18:41, Mark Holloway wrote:
Fantastic
then there is probably a codec mismatch and this increments the counter.
The Device Pool assigned to the MOH server needs to have a region that does
g711 with all HQ or BR1 or BR2 regions.
HTH
Prashant
On Sun, Oct 3, 2010 at 9:08 PM, Mark Holloway m...@markholloway.com wrote:
Sorry
.
On Oct 4, 2010, at 10:16 PM, Mark Holloway wrote:
It's the strangest thing.
I couldn't get Multicast MoH to work on my BR1 H323 router. I wiped out my
call-manager-fallback configuration, re-entered everything, put my router in
SRST mode (to practice other things) and just for the hell
:16 PM, Mark Holloway m...@markholloway.com wrote:
It's the strangest thing.
I couldn't get Multicast MoH to work on my BR1 H323 router. I wiped out my
call-manager-fallback configuration, re-entered everything, put my router in
SRST mode (to practice other things) and just for the hell
, 2010, at 4:30 AM, Roger Källberg wrote:
Turn off privacy
Roger Källberg
CCIE # 26199 (Voice)
Unified Communication Consultant
Cygate AB
From: Mark Holloway [mailto:m...@markholloway.com]
Sent: den 2 oktober 2010 06:24
To: Graham Hopkins
Cc: CCIE Voice Maillist
Subject: Re: [OSL
Never mind, I believe I have done it correctly. When the PSTN phone answers an
SNR call, then only way I can get HQPH3 to show In Use Remote is to actually
press the Line 1 button on HQPH3 and then it displays In Use Remote on the
phone.
On Oct 3, 2010, at 10:18 AM, Mark Holloway wrote
I thought I had this figured out but I'm slipping up somewhere. Could use some
help. :)
I'm configuring multicast moh at BR1 using G.711 and streaming from BR1 router
flash. BR1 is an H323 gateway.
call-manager-fallback
max-dn 24
max-ephones 2
ip source address 10.20.30.254 this is the
[ccie_voice-boun...@onlinestudylist.com] On Behalf Of Mark Holloway
[...@markholloway.com]
Sent: Sunday, October 03, 2010 7:17 PM
To: CCIE Voice Maillist
Subject: [OSL | CCIE_Voice] MoH SRST (Stream from Flash)`
I thought I had this figured out but I'm slipping up somewhere. Could use
I'm having a hard time when an internal extension calls another internal
extension that uses SNR, the From phone number on the PSTN phone is 4 digits
instead of 7. For example, extension 2001 calls 2003, and 2003 simultaneously
rings a PSTN phone number. The display on the PSTN phone says
Graham, same thing here.
This is a summary of what I've done to get it working correctly. I eliminated
using Translation Profiles as I didn't find them necessary for this.
Create PT_SNR which is assigned to CSS_SNR
Create a Remote Destination Profile and assign CSS_SNR to both Calling Search
, at 8:56 AM, Mark Holloway wrote:
Graham, same thing here.
This is a summary of what I've done to get it working correctly. I eliminated
using Translation Profiles as I didn't find them necessary for this.
Create PT_SNR which is assigned to CSS_SNR
Create a Remote Destination Profile
local to any gateway - at
least not here in the UK so would be a national call from anywhere.
Graham
On 1 Oct 2010, at 17:10, Mark Holloway wrote:
Sorry, I meant Translation Patterns, not Profiles. Still working on the
From number presentation. I'm assuming that if HQ1 calls HQ3
without prefix \
On Fri, Oct 1, 2010 at 7:00 PM, Mark Holloway m...@markholloway.com wrote:
The crazy thing is I tried this but I couldn't get it to work.
PT_HQ_CALLING_OUT assigned to CSS_HQ_CALLING_OUT assigned to CALLING Number
Transform on the Outbound portion of the HQ gateway
was referring to RP/RL transformations...
i tested it and i got the same in my lab
so i guess, as you already mentioned before, the way to do it is to actually
put Calling Party Transform Mask to be XXX on the RL (for RG member).
On Fri, Oct 1, 2010 at 7:52 PM, Mark Holloway m
! with
XXX works, but as Mark says this doesn't do what he requires
Graham
On 1 Oct 2010, at 19:23, Mark Holloway wrote:
The only issue with this is you don't know if the calling party is
Subscriber, National, or International, so you can't use XXX because if
BR2 or BR1
FAIL. Based on my score report it was not by much. I will go back in 30 days to
try again. It was my first Voice attempt. Next time I will go to San Jose
instead of RTP just because it's closer. Although, the proctor in RTP was very
nice and helpful. I wish she would go to San Jose.
Is there an option so instead of the VM greeting being Sorry, Jon Doe is not
available it says Sorry, extension 5001 is not available. Basically, I want
it to say the extension instead of the name.
Thanks,
Mark
___
For more information regarding
Thanks. :)
On Sep 24, 2010, at 3:01 PM, bkvalent...@gmail.com wrote:
Remove the display name.
- Reply message -
From: Mark Holloway m...@markholloway.com
Date: Fri, Sep 24, 2010 5:53 pm
Subject: [OSL | CCIE_Voice] Unity Connection Greeting
To: osl osl ccie_voice
When BR2 is part of CUCM and CUE is integrated with CUCM through JTAPI I could
never get MWI to work if BR2 is in SRST. Has anyone been able to get this
working? Cisco docs say with older version of CUE it doesn't work. I think
they are referring to 2.x or 3.x, not 7.x.
Thanks,
Mark
] On Behalf Of Mark Holloway
Sent: Friday, September 24, 2010 3:53 PM
To: osl osl
Subject: [OSL | CCIE_Voice] Anyone get MWI to work with CUE in SRST?
When BR2 is part of CUCM and CUE is integrated with CUCM through JTAPI I
could never get MWI to work if BR2 is in SRST. Has anyone been able to get
Parabens!
On Sep 24, 2010, at 8:11 PM, Marcelo Alexandria wrote:
Hello Guys ,
Since last year i m trying to pass in the lab , so in last Friday I was make
my 2nd attempt in version 3.
On Monday I got my results and I cant believe ..i passed!!!
So , I never post nothing here in
If I send a call from the HQ through the PSTN to BR2 and the calling number
from HQ is in E.164 format, the BR2 phone's main display (ie. large fonts)
doesn't show the + in front of 1XX but on the bottom part of the 7965's
screen it does show +1XX in smaller fonts. Is this
Från: Mark Holloway [...@markholloway.com]
Skickat: den 11 september 2010 07:47
Till: osl osl
Ämne: [OSL | CCIE_Voice] CUCM Corporate Directory (BLF Presence Status)
I have HqPh1 and HqPh3 both assigned to the same SUBSCRIBE CSS and HqPh3 has
BLF Speed Dial
Yes, each End User's Extension Mobility section is populated with the same
Presence Group and SUBSCRIBE Calling Search Space.
On Sep 11, 2010, at 11:57 AM, Brian Mulgrew wrote:
Hi Mark - are your end users set for a Presence Group?
Thk
Brian
On 11/09/2010, Mark Holloway m
Thanks Tam. I thought the Primary Extension was the only required association
on the End User page. Of course the Device Phone also had the End Use
associated. Since the Phone Number entry on the End User page is not a
required field I didn't even think to populate it. :)
On Sep 11,
I have HqPh1 and HqPh3 both assigned to the same SUBSCRIBE CSS and HqPh3 has
BLF Speed Dial assigned to watch HqPh1's primary extension and everything works
great on HqPh3's line key that watches HqPh1. However, I am trying to access
the corporate directory on HqPh3 and expect to see presence
If I want a priority queue to have 25% of the port bandwidth, I have configured
shape 4. I want queues 2, 3, and 4 to share 40%, 40%, and 20% of the remaining
bandwidth. All the examples I have seen for shape/share show a value of 1 for
priority queue in share regardless of the fact shape is
, 2010 at 6:16 PM, Mark Holloway m...@markholloway.com wrote:
If I want a priority queue to have 25% of the port bandwidth, I have
configured shape 4. I want queues 2, 3, and 4 to share 40%, 40%, and 20% of
the remaining bandwidth. All the examples I have seen for shape/share show a
value of 1
In the real Cisco lab can you assign your own Key Mappings in Secure CRT (for
copy/paste functionality)? For example, I am running Secure CRT 4 at home and
I can assign Page-Up to copy and Page-Down to paste. I hope CTRL-C and CTRL-V
work in Notepad. :/
In UCM how do you determine whether you are assigning single button cBarge or
normal cBarge?
On Aug 7, 2010, at 9:35 AM, cisco voip wrote:
That bug is for srst mode auto provision none.. for provision all, it should
work
The problem you are facing of having cbarge for split second is
I had the same problem. When the phones go into SRST mode and then I call from
the PSTN to Ph1's shared line and put the call on hold, I go to Ph2 and press
Ph2's shared line and see the cBarge softkey for a split second then it changes
to a ghosted Redial softkey. I could never get it to
Sorry, I didn't mean to say put the call on hold, what I meant was I let the
call sit idle while going to Ph2 and pressing the shared line. I would see the
cBarge softkey for a split second then it changes to Redial.
On Aug 4, 2010, at 8:39 AM, Mark Holloway wrote:
I had the same problem
Graham,
Are you configuring this in your own lab or using Proctor Labs? I am using my
own lab (2800's, 12.4.24T3, 7965 phones) and I couldn't get cBarge to work in
SRST with auto provision none. Others using Proctor Labs said they could get
it to work. Perhaps it's a difference between IOS
:28 AM
To: Berry, Matthew J.
Cc: Mark Holloway; osl osl
Subject: Re: [OSL | CCIE_Voice] CBarge in SRST ~(Again)
Sorry to jump in on the topic. Matt, just curious were you successful with
this configuration? It does not work for me with auto-provision none and an
ephone-template under
Angel - What kind of phones did you test srst cBarge with? I can't get this to
work with my 7965 phones. I needed to add more details under the ephone
configuration in order for ephone-template 1 to be applied to the phone which
should make the cBarge softkey available during srst.
softkeys appear
for a split second, then they disappear and the normal softkeys (CallFwd,
etc) appear.
On Jul 20, 2010, at 2:18 PM, Mark Holloway wrote:
Angel - What kind of phones did you test srst cBarge with? I can't get this
to work with my 7965 phones. I needed to add more details
to an MRGL that
is assigned to DP_BR1. When I configured my dspfarm profile on BR1 the max
sessions I could configure is 2. I believe that should still be adequate.
On Jul 14, 2010, at 11:12 PM, ccieid1ot wrote:
Set privacy to off.
On Thu, Jul 15, 2010 at 12:30 AM, Mark Holloway m
Man, I forgot I was working with Locations and AAR last night and I forgot to
remove the low bandwidth I set for BR1 to test AAR.
cBarge is working now based on setting Privacy to Off. Thank you to everyone
who helped. It is greatly appreciated. :)
On Jul 14, 2010, at 11:28 PM, Mark
So, I have two phones Br1Ph1 and Br1Ph2 sharing the same DN. Both have the
same softkey template I created called Standard User-CBarge which includes
Remote in Use - CBarge. Both phones are in Device Pool BR1 which includes the
BR1_MRGL which has MRG_BR1_HW_CONF assigned. This MRG contains
of no
supplementary-service h225-notify cid-update - or any other way of
preventing the 9 appearing on the phone display.
Regards
Graham
On 13 Jul 2010, at 03:53, Mark Holloway wrote:
Ok, so this is how set my H.323 gateway to operate. For example, a single
POTS dial peer
Thanks. In the GUI it's under Voicemail VM Configuration Play Caller ID for
External Callers = YES
On Jul 11, 2010, at 11:52 PM, Graham Hopkins wrote:
Add the line
voicemail callerid
not sure where it is in the GUI - must check
Graham
On 12 Jul 2010, at 06:42, Mark Holloway m
, but this 9 will still show up on the phone unless you use
voice service voip
no supplementary-service h225-notify cid-update
Regards
Graham Hopkins
On 9 Jul 2010, at 19:21, Mark Holloway wrote:
Sounds like you have the PSTN to CUCM part working ok.
This is what I have been
voice service voip
no supplementary-service h225-notify cid-update
Regards
Graham Hopkins
On 9 Jul 2010, at 19:21, Mark Holloway wrote:
Sounds like you have the PSTN to CUCM part working ok.
This is what I have been doing.
On the H323 router create the following dial
I'm not quite sure what's causing this issue, but when any PSTN number calls
Br2Ph1 or Br2Ph2 I can see the Calling party information fine in the ISDN setup
and on the display of the phones, but if I let it go to voicemail and then
check messages from the phones after MWI lights up, CUE always
You can have a data T1 and a PRI T1, or a data E1 and a PRI E1, but you can't
split T1/E1 across the same card.
On Jul 9, 2010, at 7:05 AM, Kevin Damisch wrote:
I’ve seen this question before but can’t find it. On a VWIC2-2MFT-T1/E1
card, can you configure or are there issues with having
Sounds like you have the PSTN to CUCM part working ok.
This is what I have been doing.
On the H323 router create the following dial-peer
dial-peer voice 10 pots
destination-pattern [2-9]..$
port 0/0/0:23
On CUCM have a Route Pattern that handles \+1414.[2-9]XX for calls
originated
I'm attempting to police VoIP signaling on Fast1/0/1 of a 3750 switch that is
configured as a trunk port connecting to the HQ router. I can't apply the
service-policy in the output direction. Am I thinking about this the wrong way
because I can apply it in the inbound direction.
# show run
If a router (for example, HQ) is configured with the ntp server x.x.x.x
command to sync time from another source, but I want another device (such as
PUB) to get its time from the HQ router, do I also need to configure the HQ
router with ntp server stratum X or can UCM simply get the time sync
Yikes, I meant ntp master stratum X not ntp server stratum X
On Jul 8, 2010, at 3:57 PM, Mark Holloway wrote:
If a router (for example, HQ) is configured with the ntp server x.x.x.x
command to sync time from another source, but I want another device (such as
PUB) to get its time from the HQ
Cool, thanks Graham and Randall.
On Jul 8, 2010, at 4:11 PM, Graham Hopkins wrote:
Default stratum is 8 so a simple ntp master will work
Graham
On 8 Jul 2010, at 23:59, Mark Holloway m...@markholloway.com wrote:
Yikes, I meant ntp master stratum X not ntp server stratum X
On Jul 8
...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Mark Holloway
Sent: Wednesday, July 07, 2010 1:40 PM
To: OSL osl
Subject: [OSL | CCIE_Voice] isdn plan
When tasked with setting the call type to unknown, subscriber, national, or
international, are you guys also setting
When tasked with setting the call type to unknown, subscriber, national, or
international, are you guys also setting the plan to isdn or are you just
specifying the type and leaving the plan as unknown even though all the pstn
access is isdn?
___
.
Matthew Berry, CCVP, Sr. Unified Communications Engineer
mjbe...@kroll.com
-Original Message-
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Mark Holloway
Sent: Wednesday, July 07, 2010 1:40 PM
To: OSL osl
Subject: [OSL
If calls should complete using G.729 from HQ/BR1 to CUE on BR2 which is G.711u,
can the transcoding be configured on the BR2 router locally or does it need to
happen via the originating party's transcoding resources in UCM?
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For more information
Codec : g729abr8, Maximum Packetization Period : 60
On Jul 6, 2010, at 10:23 AM, Graham Hopkins wrote:
You'll need to do it at BR2 - if you do it at HQ/BR1 it will be G.711 across
the WAN.
Graham
On 6 Jul 2010, at 17:41, Mark Holloway wrote:
If calls should complete using
the default which is g729.
Matthew Berry, CCVP, Sr. Unified Communications Engineer
mjbe...@kroll.com
-Original Message-
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Mark Holloway
Sent: Tuesday, July 06, 2010 12:44
fully
I can see the dspfarm profile has registered so just taking a guess really.
However did this myself this afternoon - had some dtmf-relay issues but
transcoder was ok - post the whole config if you like.
Graham
On 6 Jul 2010, at 18:43, Mark Holloway m...@markholloway.com wrote
Is your Site B router MGCP or H323? With an H323 gateway I could get the
router to stream the local piano music while the MoH server is set to one hop
in UCM. With an MGCP gateway I couldn't get this to work and it always streams
from UCM unless the router is in SRST mode then it plays piano
Anyone have any luck getting MWI to work when CUE is integrated with CCM?
According to Cisco's documentation you do not need MWI numbers when JTAPI is
used. Voicemail is working as expected but I cannot get MWI to light up on the
phones using that are using the UCM Voicemail Profile I have
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