Hi all,
Quick question, having the connection between CME and CCM trough an IPIPGW
where the CME leg is SIP and the CCM leg is H323, and the call flow is from CME
to CCM do we need always the DTMF relay:
dtmf-relay rtp-nte digit-drop
or can we use just the command:
dtmf-relay rtp-nte
T
Yes, I do.
On Mon, Feb 25, 2008 at 7:43 PM, boonchin .ng <[EMAIL PROTECTED]> wrote:
> What about the CallManager services parameter *Automated Alternate Routing
> Enable* *set to TRUE?
>
>
> On 2/26/08, jason sung <[EMAIL PROTECTED]> wrote:
> >
> > Hello,
> >
> > i am having issue getting AAR t
What about the CallManager services parameter *Automated Alternate Routing
Enable* *set to TRUE?
On 2/26/08, jason sung <[EMAIL PROTECTED]> wrote:
>
> Hello,
>
> i am having issue getting AAR to work (so simple but still...)
>
> when I call from a phone in site A to site B, the Phone is site A dis
Hello,
i am having issue getting AAR to work (so simple but still...)
when I call from a phone in site A to site B, the Phone is site A displays
network congestion but does not actually dial out instead gives a busy.
I have groups configured and assigned it to phones,
I have proper locations co
Please disregard the mail below - I found the missing config - it was a
user problem, as usual: the addressing options on the primary locations
page do the trick of course :-D
This makes my day ;)
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ju
Hi,
I have a problem getting VPIM to work in the Unity => CUE direction. The
opposite direction works just fine: leaving messages on the Unity
server, by dialing the locationID + HQ/BR1 phone number, leaves a
message in the corresponding Unity subscriber's mailbox.
But when I try the same in the
Hi Mark,
thanks a lot for the comforting feedback. In fact, where it all went
'wrong' - I realize from your last response - is that I jumped over to
ex 15&16 after doing Unity (lab9), quickly checking the prerequisites
for these exercises.
Thing is that sometimes some config is being used from
Another thing to throw at you guys on the subject of call forwarding from
phones whilst in SRST mode.
You don't need the alias command to forward calls phones registered in SRST
mode to unique telephone numbers.
>From the example below, I wanted to fwd 2001 to 1661, 2002 to 1662, 2003 to
1663, e
Hey Juan,
So I will take this as a perfect example of when, in the actual CCIE
Lab, you would definitely want to try and seek clarification from the
Proctor to make sure you understand the requirements of the task
properly. The more questions the better. In the case that you
configured mo
Hi Devildoc,
thanks for the reply on this - really. However, the solution you provide
is exactly the same as what I described below - apart from the fact that
I used CUE's AA script from lab 15 to dial into the B-ACD pilot number,
instead of directly calling the AA pilot (3200 in your solution -
I just have it in my magic box of goodies... as every ccm guy
should... that and translaterX, the triple combo, and a crowbar...
Jonathan
On Mon, Feb 25, 2008 at 9:01 AM, Matthew Saskin <[EMAIL PROTECTED]> wrote:
> No dblhelper in the lab as it's not installed by default on the
> callmanager.
Juan,
The question asks you to create a GDM for the Support Queue so that people can
leave a voice message in case there is no agent available to take the call.
To do this you'll need to do the followings:
1. On CUE, create a group called SupportQ and give it an extension number that
is t
No dblhelper in the lab as it's not installed by default on the
callmanager. You typically have to grab it from another CM that has it,
or get it from TAC.
-matt
Jonathan Charles wrote:
Jumping in late, just subscribed... do they not have DBLHelper on the
Publisher? Is it there on the CCIE l
Thanks Mark for the info. That helps a lot explaining one issue that i had
with the aa dialpeer. I had a combined dialpeer for both inbound and outbound
for the aa application. As a test, i took out the incoming called-number to
the aa pilot and that broke the working of the application. It
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