What configuration should we use for the TFTP servers if it is not
spelled out explicitly in our lab? A single TFTP on the PUB? Or on both
SUB and PUB? I guess the safest bet is just to have it on on both?
I'm focusing on the SIP portion of the 3rd module (for v3 lab) and am having
problems upgrading my SCCP 7941 to SIP. After applying all of the syntax
needed to set up a SIP phone in CME and configuring the tftp-server syntax
to point to the SIP firmware, it seems that when the phone attempts to
do
Thanks, Vik.
On Mon, Feb 23, 2009 at 5:04 PM, Vik Malhi wrote:
> Copying all the “tftp-server” commands is OK- just a question of
> readability and managing the size of your config.
> --
> Vik Malhi – CCIE #13890, CCSI #31584
> Senior Technical Instructor - IPexpert, Inc.
>
> Telephone: +1.810.3
Copying all the ³tftp-server² commands is OK- just a question of readability
and managing the size of your config.
--
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 onlin
In module 3 of the v3 lab material, a section describes the need to add
tftp-server syntax in telephony-service mode fo both SCCP and SIP phones.
The material explains that it's easier to copy the syntax from
CME-7-0-full-readme-v.1.0.txt (by invoking the "more" command).
The tftp-server syntax is
Hi Alex:
I could have a MWI in a envelope way in each IP Phone with an additional button
doing some voice translations to the number coming from the CUE as a MWI to the
pilot number of the hunt group, as I told you I am testing things to practice
BACD, GDM and the MWI for the GDM, and I am t
:)
--
Mark Snow
CCIE #14073 (Voice, Security)
Senior Technical Instructor - IPexpert, Inc.
Telephone: +1.810.326.1444
Fax: +1.309.413.4097
Mailto: ms...@ipexpert.com
--
Join our free online support and peer group communities:
http://www.IPexpert.com/communities
--
IPexpert - The Global Lead
NM, I read Mark's email; sorry. Read first then type later...
On Mon, Feb 23, 2009 at 2:33 PM, Jeremy Combs wrote:
> Curious, what is the requirement for FastStart being enabled on an ICT
> Gatekeeper Controlled Trunk again?
>
>
> On Mon, Feb 23, 2009 at 2:03 PM, Kevin Hogan (kevhogan) <
> kevho
Curious, what is the requirement for FastStart being enabled on an ICT
Gatekeeper Controlled Trunk again?
On Mon, Feb 23, 2009 at 2:03 PM, Kevin Hogan (kevhogan)
wrote:
> Yep...Here it is...
>
>
> --
> *From:* Mark Snow [mailto:ms...@ipexpert.com]
> *Sent:* Monday, F
Thanks will fix the screenshot.
But call does indeed work fine now?
--
Mark Snow
CCIE #14073 (Voice, Security)
Senior Technical Instructor - IPexpert, Inc.
Telephone: +1.810.326.1444
Fax: +1.309.413.4097
Mailto: ms...@ipexpert.com
--
Join our free online support and peer group communities:
That makes sense. All I have been using for the past year or so was
12.4(20)T in lab stuff, so it would have worked without it. Page 93 of
lab 1 volume 3 is the screen shot.
From: Mark Snow [mailto:ms...@ipexpert.com]
Sent: Monday, February 23, 2009 1:53 PM
To:
So it works now?
It will not work without it (if it shows unchecked in the answer key
then it is a typo - what page and I will get it corrected right away).
Reason is this: H323 DO (delayed offer) to SIP EM (early media) has
only begun to be supported in the latest 12.4(xx)T train (20T I
The one thing I missed...Fast Start with G.729 on the CCM Trunk.
Thought I tried it, but apparently not! Curious though, in lab 1 of
volume 3 this scenario does not have FastStart checked on the GK Trunk.
Should it work without that?
From: ccie_voice-boun...@onli
No luck...Here is the NM that I am using for DSP Farm (using different
NM for PRI just to make sure it wasn't a conflict or something):
NAME: "High Density Voice", DESCR: "High Density Voice"
PID: NM-HDV= , VID: 1.0, SN: JAB034004DA
NAME: "PVDM 3-C549 Simm", DESCR: "PVDM 3-C549 Simm"
I see that the DSP is engaging from a previous post and that you have
changed your DSPfarm too so that removes the possibility of HW problems. One
more thing to try if you haven¹t already
I don¹t know how many DSP¹s you have available. Bring down the max sessions
down to ³2² (in both places) and
Sorry, forgot to paste show dspfarm units
BR2#show sdspfarm units
mtp-1 Device:MTP001ae2a4c080 TCP socket:[3] REGISTERED
actual_stream:20 max_stream 20 IP:10.1.202.1 26763 MTP YOKO keepalive
28
Supported codec: G711Ulaw
G711Alaw
G729
G729a
Just erased the config, re configured it with same result. I have also
tried (and currently using) non-GK controlled ICT to take GK out of the
mix (just incase). Calls go through fine with g711 forced end to end,
but with 729 to CME they fail. Attached is the config as it stands
right now.
Check the output of ³show sdspfarm units² on the CME- how does it look?
Also can you send your full CME router config.
--
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
Can ping the 6608 Xcoder IP from the CUE and other way. G711 works
fine. H.
From: Vik Malhi [mailto:vma...@ipexpert.com]
Sent: Monday, February 23, 2009 11:21 AM
To: Kevin Hogan (kevhogan); OSL Group
Subject: Re: [OSL | CCIE_Voice] CCM-->CUE call fail
So
Sounds like a routing issue somewhere along the media path.
- Check the default gw for the CUE.
- If the call is passing through an MTP on the CCM side check the xcoder
default gw (6608). A problem here would also cause dead air on calls to CME
phones
- if the two things above are fine then remove
Yes, as well as from PSTN. I can also force the calls to g711 from CCM
and it works. Only get this with 729. CME router shows transcoders
getting activated and I can see the call coming into CUE with a trace
voicemail all. I though it might just a dsp issue and that is why I
tried upgrading cod
Do you get audio when you dial locally to CUE (from a CME phone).
--
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 online support and peer group communities:
http://www.
Already have it in there (forgot to put that in last email) and the call
is hitting it. Still dead air when CUE answers. Here is the incoming
DP it is hitting:
dial-peer voice 1001 voip
translation-profile incoming strip-prefix ---strips tech prefix.
incoming called-number 2#3... ---tried w
I had it setup that way, then changed it just the sub interface wasn't
causing issues. Didn't work either way.
From: Alex [mailto:alex.arsen...@gmail.com]
Sent: Monday, February 23, 2009 3:42 AM
To: Kevin Hogan (kevhogan); ccie_voice@onlinestudylist.com
Subject:
I think you need an inbound VOIP dial-peer in order to fix the codec to g729
and thereby intiating the transcoder.
Try this:
Dial-p v xx voip
incoming called-number .
codec g729r8 !! DEFAULT COMMAND- FYI only
--
Vik Malhi CCIE #13890, CCSI #31584
Senior Technical Instructor - IPexpert, Inc
Exact, thats the better answer. I forgot the softkeys
/Robert
>>> Alex schrieb am Montag, 23. Februar 2009 um 14:04
>>> in
Nachricht a435bb691314cb0821748dbc8d2de766:
> Create an ephone-dn template and remove Confrn softkey from "connected"
> phone state. Then assign template to your ep
Create an ephone-dn template and remove Confrn softkey from "connected"
phone state. Then assign template to your ephone and reset.
Rgds
Alex
- Original Message -
From: "Robert Schuknecht"
To: ;
Sent: Monday, February 23, 2009 12:00 PM
Subject: [OSL | CCIE_Voice] Antw: CME
I would
I tried the SIP-TRUNK another time, and it is still not working. Now i was
calling from CME to CCM, and i am using only g711 and the software mtp. When
the call is established (both phone off-hook) i am able to invoke supplementary
services (hold/unhold) but from CCM to CME the DMTF is not worki
I would only bind one single-line ephone-dn to the phone.
/Robert
>>> omar itani schrieb am Montag, 23. Februar 2009 um 08:31 in
Nachricht 3e39d9c0a60c649e169eff7cad816c12:
> WHAT IS THE COMMAND FOR BR2 PHONE 3 UNABLE INITIATE A CONFERENCE ?
>
> _
is subscriber shoul be the primary call process agent and the publisher should
be as the backup
we must logon to the seubscriber ccm server and into call manager create
another server for publisher ?
and choose priority ccm group for sub ahead the pub or we just log in and
config
Kevin,
Are you sure about this:
sccp local FastEthernet0/1 (shouldn't it be Fas0/1.?)
Check where is your IP@ 10.1.202.1 assigned, to which interface and make sure
SCCP bound to this interface.
This way it always worked for me.
Rgds
Alex
From: Kevin Hogan (kevhogan)
To: ccie_voice@onlinestud
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