You have no dialplan-pattern under telephony service or voice
translation-rules?
From: jeremy co [mailto:[EMAIL PROTECTED]
Sent: Saturday, November 15, 2008 12:24 AM
To: Hardesty, Scott
Cc: Greg Miglucci (gmiglucc); CCIE Voice Maillist
Subject: Re: [OSL |
Hi Greg,
As u can see my configs, no I don't have.
when I press message button, I can connect to pilot number , but I cannot
hear any thing, even I can see call connects to pilot number form sh call
active voice
Jeremy
On Sat, Nov 15, 2008 at 7:18 PM, Greg Miglucci (gmiglucc)
[EMAIL
Is your question about overhead calculation for voice calls or something else?
My explanation for calculating overhead for voice calls.
Example:
CIR 512,
Allow priority b/w for 5 g.729 calls with FRF.12 (can be MLPPP, FR:
Allow 10% overhead.
I would calculate 5x 27.2kbps = 136kbps
Add 10%
Possible your transcoder is not getting invoked.
You've HQ region set to use G.711 within itself and G.729 with others.
I believe CTI RP and ports would be in HQ region and IPCC Express is configured
for G.711 codec. So you get fast-busy. When you change the region settings to
use G.711 you
On route-pattern you may set CLID Name/Number to restricted/allowed.
James Key [EMAIL PROTECTED] wrote: Block calling nameWhat is the
best way to block calling name on certain route patterns, while still allowing
it on others? Example: hq local send calling name + number, hq
translation-profile incoming on ephone-dn and translate the voicemail number to
CUE or Unity Pilot. Leave other phone without translation.
Balamurugan Singaram [EMAIL PROTECTED] wrote: Hi,
We have two cme phones in BR2 two different unity systems:
1st phone press messages button and go
Hi,
Here is the config:
hostname se-200-0-0-100
ip domain-name cue.voice.com
ip name-server 114.0.0.1
ntp server 200.0.0.254
groupname Administrators create
groupname Broadcasters create
groupname SupportQ create
groupname second create
username admin create
username ph1 create
username
Hi,
I think I should enable something to enable trace configuration.
when I go to trace configs, every thing is disabled and I cannot change any
parameter.
Jeremy
hi,
I configured AAR and it kicks in.(I get the AAR message that call is
rerouted) but ccm wouls not forward call to H323 Gateway.
There is no location on GW configured adn no partition or CSS configured.
When I call directly the external number ,ccm forwards call but when AAR
kicks in ,it
1. check route pattern at cm, do you have a route pattern for external
phone-mask of br1. is it in the ccs of the gw?
2. check debug isdn q931 at pstn see any calls coming in?
jeremy co [EMAIL PROTECTED] wrote: hi,
I configured AAR and it kicks in.(I get the AAR message that call is
Hi,
When I can call to external mask correctly means route pattern matched and
pstn is working.
As I said be4 there is no parition or CSS in my ccm.
Jeremy
On Sun, Nov 16, 2008 at 1:57 AM, [EMAIL PROTECTED] wrote:
1. check route pattern at cm, do you have a route pattern for external
hi,
I try to config IOS Conf Bridge and Transcode in 2811 , DSP type C5110
- Config for Transcode succeed
- But config for Conference failed, it said max session 0, i think I still hv
enough DSP
Hello Everyone,
I believe this issue has started to become mor familiar with all CCIE Voice
candidates. The situation is that if I place a call over the GK to B-ACD
located at Site C, on the show gatekeeper call there are 2 call legs showing
up, one as G.729 and the other one as G.711:
PVDM 16 - Single DSP supporting 6 high complexity xcoder or 2 * 8 party
conference.
It is my understanding a single 5510 DSP can not support both conference and
trancoder. Check the following reference
http://www.cisco.com/en/US/docs/ios/12_3/vvf_c/interop/intcnf2.html#wp100328
7
I
You can also try to turn up your conference dspfarm profile first, and then
the transcoder. I have run into the same issue when trying to do the
transcoder first, so just shut on dspfarm profile 2, then no shut on
dspfarm profile 1, and finally no shut on dspfarm profile 2. Let me know
if that
Deniz,
Did you create your JTAPI user and RMJTAPI users before attempting to create
the JTAPI triggers? That is an odd error I have never seen it before, let
me know about the users.
On Sat, Nov 15, 2008 at 11:59 AM, M.Deniz KIZILCABOLUK [EMAIL PROTECTED]
wrote:
Hi,
When I try to add a new
You must have a free DSP to configure any sessions for conferencing.
With a PVDM2-16 you will be able to configure 2 conferencing sessions as
long as nothing else has grabbed any of the channels. Transcoding will
share DSP's with Voice Port terminations, so I agree that it is best
to configure
Kevin, Are you sure about the xcoder with this type of secondary DSP? It is
the old style of DSP on the NM-HDV, I believe.
The 2 conferences will take the 5510. If it is a T1 and all Bchans are
allocated both the additional PVDM 12 will be used in the NM-HDV.
I find with the older DSP it
Thks Kevin and Jacobs,
I have not tried it yet, but I believe must work w/ your workaround,cause I
remember it worked before w/ the same amount of DSP, and Kevin's explanation
really make sense why it failed
--- On Sun, 11/16/08, Kevin Porter [EMAIL PROTECTED] wrote:
From: Kevin Porter
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