The only way I've gotten it to work is when I use ntpdate.exe x.x.x.x (NTP
Server) after I restart the NTP service. It's takes about 5-10 minutes before
the time synchs.
Chad Stachowicz <[EMAIL PROTECTED]> wrote:
Me either.
:<
On Sat, Apr 19, 2008 at 8:02 AM, Gregory Jost (grjost)
Hello all,
I'm trying to make something that I thought was very simple work. However, I
was wrong, and I'm ripping my hair out.
I'm calling from CME, through a GK into a IPCC trigger. IPCC = G711 and CME -
GK = G729. Trigger works fine from CCM.
I'm using default-prefix on the GK, henc
Ok, so I'm wondering how the dspfarm transcoder maximum sessions command is
different from the sdspfarm transcoder sessions under telephony-services? I
know that using the dspfarm command, you set up the actual number of sessions
you want configured, but why the number of sessions different unde
it's controlled by using the description command under the ephone-dn.
Chad Stachowicz <[EMAIL PROTECTED]> wrote:you know on call manager how the
external phone number mask will populate the top right portion of a phone
display, in the black bar at the top. Is there a particular field that
Just to verify, in order to configure pass through on the ATA186,
set audiomode to "0x00140014" and connectmode to "0x0400".
I keep seeing different values, and I assume these values can be set
differently and still have it work?
Thanks!
---
Someone asked this good question some time ago.
"I have a couple of questions about IPMA and redundancy. Assuming I have a
CallManager Publisher and Subscriber, with the Subscriber being the first
choice for call processing and the Publisher backing it up, in a cluster and
the IPMA servic
I'm using B-ACD on CME with drop-through mode. Calls are queuing just fine, but
I thought that I should be able to see queuing statistics on the phone display,
ie. "1 call in queue" for example. Or did I just have a bad dream?
Thanks!
-
Looking for l
-matt
>
> ccievoice1 wrote:
>> 24K/80K are for voice codec, aren't they? I think Earnieball78 was
>> wondering on the video codec...
>>
>> On Fri, Feb 15, 2008 at 11:32 AM, Matthew Saskin >> >> wrote:
>>
>> It was my understanding that CallM
tthew Saskin <[EMAIL PROTECTED]> wrote:
Don't quote me, but I believe audio is fixed as 24k for G.729 and 80K
for G.711
not sure on video...
-matt
Earnieball78 wrote:
> Hello all,
> I'm am trying to find out where one would fine the table that lists
> the audio and
Hello all,
I'm am trying to find out where one would fine the table that lists the audio
and video codec rates information required to configure locations based CAC in
CCM?
Thanks!
-
Never miss a thing. Make Yahoo your homepage.
George,
Do a debug voice dialpeer and see what dial-peer it matches.
Are you sure you are only receiving 4 incoming digits?
George Cassels <[EMAIL PROTECTED]> wrote:
I setup B-ACD and hunt list per lab 22.21 as follows:
dial-peer voice 3000 pots
service aa
incoming called-number 3
ter it's NM-HDV or
NM-HDV-2 ?
-Anil
--- Earnieball78 wrote:
> Hi Anil,
> Not sure what you are asking. The way I interpret
> the output is that you already have a NM module in
> the 2811, which is the 16port etherswitch module.
> The 2811 only has one available NM slot...
>
Hi Anil,
Not sure what you are asking. The way I interpret the output is that you
already have a NM module in the 2811, which is the 16port etherswitch module.
The 2811 only has one available NM slot...
Remember, the 2811 does not have the same architecture as the old 2600
series, where you
Hello all,
I have question regarding MoH. In a scenario where you provide both multicast
and unicast streams, is it possible for the MRG/MRGL to automatically switch
between multicast and unicast depending on the capabilities of the network?
Let's say that I have two MRGs, one with multica
I am wondering what the relevance of the AAR configuration is on the gateways
(H323,MGCP and Trunks). In what scenario would this be important.
Basic site to site AAR works just fine without it.
Thanks!
Christian
-
Be a better friend, newshound,
for your response!
Christian
Scott Monasmith <[EMAIL PROTECTED]> wrote:
For Scenario 1: Transcoding codecs between inbound and outbound SIP calls
is not supported.
For Scenarios 2 & 3: What inbound and outbound DTMF types are you using?
On Feb 1, 2008 3:30 PM, Earnieball
Hello,
I am refering to task 4.9 in the workbook. I have a few issues I'd like some
help straighting out.
Scenario 1: CCM --> H323 G711 to HQGW --> SIP G729 to CME
Calling from CCM, calls works fine, can pick up CME phone and talk. I see
transcoder on HQRTR being invoked so all is good.
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