Hello All,
On the fastethernet port connecting to the core switch, i have the duplex
setting as auto
i have this message popping up on the console the HQ router
CDP-4-DUPLEX_MISMATCH: duplex mismatch discovered on fastethernet0/1 (not full
duplex), with SW1 ..
i issue no c
Duy,
OK I forgot that it is going to be an ephone dialing this number so the
ITS is processing each digit in real time and will never let you hit the
5th digit once its made a match on the first 4.
Chris
Duy Nguyen wrote:
Chris,
It actually keeps hitting the "Dial-peer voice 2300 voip" bec
I've tested in on CCM and CME but over the PSTN.
On CCM side:
Create a Voice-Mail profile "to_vm", select the Voice-mail pilot and put the
"external phone number mask" as .
Create a Route Point with DN 22xxx, do call-forward all to VM, select the VM
profile "to_vm".
To make it work over th
Was trying to reconfigure my PSTN router last evening for the V3 labs.
The tables .pdf file for the V3 blueprint does not include a table 12
for Site Numbering Plan.
Are we to assume that the same site numbering plan is being used from
the V2 labs?
CDP isn't a duplex issue. If the port is connecting to a PC, sometimes it's
best to hardcode it to 100Mb Full.
On Tue, Mar 31, 2009 at 2:43 AM, adefila seun wrote:
> Hello All,
>
> On the fastethernet port connecting to the core switch, i have the duplex
> setting as auto
>
> i have this message
Jin,
Remove the FXS, configure Transcoding for 2 sessions, see if it works then
shutdown and plug the fxs back in. Always insert the VIC's from right to
left.
On Sun, Mar 29, 2009 at 10:32 PM, CCIE OSL wrote:
>
> I have purchased a 1760 router with PVDM-12 and CME.and FXS
> Just so that I can
Hi
i have a basic question :) if i need to call from cm site to crs pilot number
via gatekeeper, do i need to add a transcoder, in case i need this, where
should i apply this transcoder?
Thx
On the HQ side.
On Tue, Mar 31, 2009 at 9:57 AM, ccie Me wrote:
> Hi
>
> i have a basic question :) if i need to call from cm site to crs pilot
> number via gatekeeper, do i need to add a transcoder, in case i need this,
> where should i apply this transcoder?
>
> Thx
>
>
Thanks Cliff,
I was planning to put this together yesterday,
But I could not reserver the rack, since it was all taken.
I guess proctorlabs is only allowing a few v2.0 labs. !!!
In any case, Thanks for you input,
It actually looks good,
one think it that the requirement was to use single zone.
I
That's good point,
/Jin Jung..
Wesley Lim wrote:
Interesting.
How I would tackle this.
Lets assume we have 2 dialpeers as of the following:
1. 2...
2. 22...
No matter what, both dialpeers are overlapping. Hence for the dialpeer
with 2... , i would put a timeout at the end of the destination
Well, I used a single zone, but yes, you can use multiple prefixes within a
zone (as I did).
I could think up about 3 other ways to do this that would work, but the
devil is in the details. Without being able to see the exact requirements,
all I could do is figure out things that will work to
Hi
We’ve come across a typical requirement wherein we need to enable
FAC(Forced Auth Code) on an Internal DN.
I was thinking of creating a RP & hair pinning it back from a GW to
the required DN.This DN could be only in this GW's CSS.
Is this possible with MGCP or I need to have only H.323 GW? Wh
MGCP can make it work only if it's going to be hairpinned via the PSTN.
If you use H323, you can run it in and out on VOIP dial-peers.
H323 would likely be a better way to go if you must do this. That said,
this is a pretty strange requirement. I would ask some hard questions about
exactly WH
Hello,
Do you know how to recreate the audio stream, when you're sniffing a voice
network using wireshark?
thanks
This bugged me. I kept thinking about it.
Went back and started googling for some additional gatekeeper documentation.
Found what I was looking for. To summarize the problem as I understand it:
1. Trunks coming in from CCM to GK. Trunk to sub is priority for incoming
calls from GK.
2.
What are you attempting to do? Capture it and convert it back so you can
listen to the RTP stream?
- Original Message -
From: Cristobal Priego
To: ccie_voice@onlinestudylist.com
Sent: Tuesday, March 31, 2009 1:53 PM
Subject: [OSL | CCIE_Voice] Recreate Audio Stream
Hello,
Yes, that is correct
I'm still dealing with the one way audio issue. I have a few sniffers from
both ends, and I'd like to recreate the audio and keep moving until I find
which hop is dropping the packets
2009/3/31 Cliff McGlamry
> What are you attempting to do? Capture it and convert it back
You don't need to "hear" it. The stream won't be there at all at the point
when you hit the point where it's gone one way.
- Original Message -
From: Cristobal Priego
To: Cliff McGlamry
Cc: ccie_voice@onlinestudylist.com
Sent: Tuesday, March 31, 2009 2:29 PM
Subject: Re
I believe they've changed. Also, I think IPexpert is going to have
different numbering plans in different labs so that their users don't get
used to just memorizing answers (I think I remember reading it as one of the
Proctorlabs' new features.)
On Tue, Mar 31, 2009 at 6:59 AM, Girard, Jeffrey CO
Hey, Cliff,
Man!
First, Thanks for all you time on this.
I see your method, and it's wonderful that this works..
One question,
Let's say, if you are allow to use bandwidth command on Gk, would you
use same method??
bandwidth interzone 16
/Jin Jung...
Cliff McGlamry wrote:
This bugged me. I
Cliff, your statement bring up my attention.
When I tested it out in my lab, it is not what you said
> G729 is the preferred codec for
> H323/CME/SRST.? It will go to that if it can.? It won't negotiate up
> if the other end supports G729.? It WILL negotiate down if it attempts G711
> and the
Hi All,
Can somebody please provide the steps to configure the following:
For instace:- Tech Prefix 1#, Zone Prefix 1*, 2*.
First call should go
through Subscriber, if one call in progress, second call should go through
Publisher.
What else do I need other than following:
a. Make Pref
Cliff,
This looks great I will try it out tonight. Been thinking about this
all day too but my pesky job has been keeping me busy ... :-)
Good work
Chris
Cliff McGlamry wrote:
This bugged me. I kept thinking about it.
Went back and started googling for some additional gatekeeper
documen
I hope, you are not dreaming about it!
/Jin Jung...
Chris Parker wrote:
Cliff,
This looks great I will try it out tonight. Been thinking about this
all day too but my pesky job has been keeping me busy ... :-)
Good work
Chris
Cliff McGlamry wrote:
This bugged me. I kept thinking
this is the topology
7200(PRI)--3560(192.168.x.x)--3560(192.168.x.x)--3560(192.168.x.x)--
7960(192.168.x.x)
|
| DS3
3745(10.10.x.x)--3750(10.10.x.x)--CCM servers(10.10.x.x)
I just capture a call with one way audio and what I see on the sniffer and
the trace is that the negotiation of ip
It's not needed to accomplish this. But if told to use it to accomplish
some other requirementmaybe.
It would just depend.
- Original Message -
From: "CCIE OSL"
To: "Cliff McGlamry"
Cc:
Sent: Tuesday, March 31, 2009 3:23 PM
Subject: Re: [OSL | CCIE_Voice] gatekeeper question - f
Appropriate prepending and "max-conn 1" under outbound dialpeers on your
originating H.323 GW?
You need prepending to prevent calls to the same number landing on one GK trunk
on CCM side.
That is assuming you are allowed only 2 calls from H.323 GW to CCM at any time.
If you need to load-balance r
Wouldn't be the first time :-)
CCIE OSL wrote:
I hope, you are not dreaming about it!
/Jin Jung...
Chris Parker wrote:
Cliff,
This looks great I will try it out tonight. Been thinking about this
all day too but my pesky job has been keeping me busy ... :-)
Good work
Chris
Cliff
Can someone from IPExpert please address my question?
---
Jeffrey T. Girard ("Jeff")
COL, 53
Future Forces Integration Directorate (FFID), Deputy - Networks
office: (915)568-1240 DSN 978
Mobile: (915)727-4222
reply to: jeffrey.gir...@us.a
Guys:
I'll get you an answer on this.
Larry Hadrava
CCIE #12203 CCNP CCNA
Sr. Support Engineer – IPexpert, Inc.
URL: http://www.IPexpert.com
On Tue, Mar 31, 2009 at 9:59 AM, Girard, Jeffrey COL MIL USA <
jeffrey.gir...@us.army.mil> wrote:
> Was trying to reconfigure my PSTN router last evening
Apologies for not getting back to this earlier.
The PSTN phone has the following DN's configured:
ephone-dn 1
number 911
label Emergency 911/999
description PSTN Phone
name Emergency Services
!
!
ephone-dn 2
number 2123942123 no-reg primary
label HQ-NYC +1-212-394-2123
!
!
epho
We have run into a bug- can you make sure that you are running the following
IOS or later:
12.4.21 Mainline IOS code.
--
Vik Malhi CCIE #13890, CCSI #31584
Senior Technical Instructor - IPexpert, Inc.
Telephone: +1.810.326.1444
Fax: +1.810.454.0130
Mailto: vma...@ipexpert.com
Join our free o
Hi,
I am not a regular here but I have a question that I can not find answer to.
Anybody knows how to change default message duration for IPCCX prompts?
Thank you in advance,
Onur.
The 1711 is currently running 12.4.23 (c1700-advipservicesk9-mz.124-23.bin).
PT
From: Vik Malhi [mailto:vma...@ipexpert.com]
Sent: Tue 3/31/2009 7:40 PM
To: Paul Thomas; OSL Group
Subject: Re: [OSL | CCIE_Voice] Trouble using EZVPN on 1711 to Pod 32
We have ru
Thanks Cliff
Basically we are interfacing a Public Announceement System(PAS) with
an analog port on a VG248. The PAS doesn't have any builtin
authentication & so I need to enforce FAC on the DN itself.
Mann.
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