I see, I have make sure my RDP CSS include the 4 digit ext. but still
failed.
On Sun, Apr 7, 2013 at 2:44 PM, William Bell b...@ucguerrilla.com wrote:
Not quite. The RDP CSS is used by the MVA process in CUCM to make the
final call routing decision.
--
William Bell
blog:
hi all,
does anyone know the quickest trick on provision phone and DN in exam?
i ever read ,there is super copy.
tks
d
On Sat, Apr 6, 2013 at 12:50 AM, Peter Simmons pe...@grayrigg.com wrote:
Bill,
Your feedback has been awesome - much appreciated by all here, I'm certain
- it has
Hi Hugo,
If I remember correctly, This is an issue with IPexpert setup. Since Both SA
and SB users are imported to CUC, MWI will work but SC user is not imported to
CUC and hence MWI will not work. It sounds weird but to confirm this, delete SA
user from CUC and try to leave a VM for SB user.
I
Hello Experts!
I have HQ as the GK and CME, CUCM publisher and subscriber are registered
to GK in single zone. Further, GK Trunk is in separate region/DP with
hard-coded G.729 codec with other regions/itself as well.
When I call from HQ to CME side, and check sh gatek call it shows that
the call
Hi Suresh,
I think you are hitting a known bug . Please go to service parameters -- call
manager and change the following to G729
Intraregion Audio Codec Default: G729
Regards,Mohamed Gazzaz
Date: Mon, 8 Apr 2013 19:02:14 +0545
From: bring...@gmail.com
To: ccie_voice@onlinestudylist.com
Will check and let you know.
BTW you mentioned a known bug can you please post a link of it?
Thanks.
On Mon, Apr 8, 2013 at 7:41 PM, Mohamed Gazzaz mgaz...@hotmail.com wrote:
Hi Suresh,
I think you are hitting a known bug . Please go to service parameters --
call manager and change the
I don't have a the link to it but I read about it in this OSL.
Date: Mon, 8 Apr 2013 19:47:35 +0545
Subject: Re: [OSL | CCIE_Voice] CME - GK - UCM bandwidth
From: bring...@gmail.com
To: mgaz...@hotmail.com
CC: ccie_voice@onlinestudylist.com
Will check and let you know.
BTW you mentioned a known
Hi Donny,
Partial/Complete match is NOT related to 4 digit dial. I thought you were
asking about separate issue with matching remote destination.
Let me ask you this, you're not trying to call the extension on the phone
for which you have Remote Destination/Profile configured, are you?
Sergey
Here is bug id
CSCsl74701 Bug Details ARQ requests 1280 when no regions are defined to use
g711
Regards,
Ramcharan Arya
CCIE # 28926 (RS)
On Mon, Apr 8, 2013 at 9:02 AM, Suresh Bhandari bring...@gmail.com wrote:
Will check and let you know.
BTW you mentioned a known bug can you please
Sergey, and all,
I had hard-coded the G729 codec from UCM side, and when I did the same for
voip dial peer pointing to RAS, it didn't show up, as it is the default.
Tried with voice-class codec as well, but no luck.
Will check the Bug as well. Thank Ramcharan for the bug id.
On Mon, Apr 8,
The workaround is to change the service parameter for default intra-region
codec to g729. You will then obviously need to update your regions for
site a b c to use g711 rather than 'default' which is now g729.
This bug mentioned above is where a gk send a call to cucm and it doesn't
look at the
The brq parameter does does apply here (ie, won't fix the issue) since a
brq is sent after a call is already connected and is requesting a *change*
in bandwidth.
The initial call setup is done with an arq that contains the initial
bandwidth request. If you debug this issue end to end, you will
...looks like I didn't read you email correctly the first time and you
already tried both methods :-)
My recommendation (may or may not be the way the lab is graded) is that you
should use intra region param. Reason is the grading script might not
connect the call or it will only look at debugs
Justin,
I hope you read my earlier mail.
I had my results when I used BRQ and when I used Intraregion codec.
Like they say, my (and hopefully others' including you too) answer
depends on what is asked.
Thanks.
On Mon, Apr 8, 2013 at 10:41 PM, Justin Carney justin.s.car...@gmail.comwrote:
And this last mail I received when I just pressed send. Anyways, thanks for
sharing your views.
On Mon, Apr 8, 2013 at 10:47 PM, Justin Carney justin.s.car...@gmail.comwrote:
...looks like I didn't read you email correctly the first time and you
already tried both methods :-)
My
Bill,
Good to know I was concerned it might have been GK or something else. I will
try this next time I lab.
Regards,
Hugo
From: William Bell [mailto:b...@ucguerrilla.com]
Sent: Sunday, April 07, 2013 11:59 PM
To: Barrera, Hugo
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice]
Hi All,when they asked to make sure VM pilot (+1408200) can be called
directly from PSTN.do i have to make a call from PSTN and dial +1408200? ,
but pstn phone not support to dial '+'.is that mean dial 1408200 from pstn
phone or some tricks behind here?
thanks advance-ikizoo
When you have the strip digits at gateway level set to 4, and as long as
2220 is a DID then from wherever it is called, it will go to the VM.
Regarding plus dialing of VM DID from PSTN, it will be/is registered to CME
(h.323), so I don't expect it will send a plus.
My two cents.
On Tue, Apr 9,
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