I am new to the list and just wanted to say hey!
Got a question about standards people have see on the lab for mapping DSCP
to COS. The old school way was marking call setup and tear down as AF31
(DSCP 26), but since cisco has changed to conforming to RFC to using COS3
(DSCP 24) which way is the
From the QOS SRND it looks like 2 different standard they are saying for the
port based COS to Queue mappings
for a 1P3Q1T port use:
wrr-queue bandwidth 5 70 25 1
wrr-queue cos-map 1 1
wrr-queue cos-map 2 0
wrr-queue cos-map 3 2 3 4 6 7
wrr-queue cos-map 4 5
Though Auto QoS creates:
wrr-queue
Quick question about low bandwidth circuits. If you have a low speed link,
what do you set the priority queue. Looks like the SRND states to always
use 33% for the priority queue though auto qos puts out 70% (on a 384
speed).
Any thoughts?
Any idea how much codec G711 and G729 uses per call when using cRTP. 4.1(3)
SRND shows that G729 uses 10kbps but that is without layer 2 FRF .12.
Don't really see a calculation on how they came up with that also.
Can anyone tell me either Cisco states how much a cRTP G729 call takes up
(including
I am fairly new to gatekeepers. Just trying to play around with my own
scenario.
I have 1 gatekeeper with 2 local zones, 1 is CCIE for Call Manager, 1 is for
CCIERemote for CME router. Why is the routers matching on the CCIE zone
when it should look at the zone prefix match send the call to the
few questions... first what version of CUE is on the lab now?
Also attached is the config for my CME/CUE router. I get a fast busy when i
try to dial into CUE. Any reason from my local CME phones.
I also get the following error that comes across in the CME router:
Oct 21 03:14:46.144:
Nevermind Monday detail. I wasn't sending my SIP dial-peer to the right IP
address.
On Mon, Oct 20, 2008 at 10:15 PM, Ryan Trauernicht [EMAIL PROTECTED]
wrote:
few questions... first what version of CUE is on the lab now?
Also attached is the config for my CME/CUE router. I get a fast busy
CMGKCME Phone then to CUE on no answer. Call just drops when trying to
goto CUe. I know CUE supports G711 only, so I have a transcoder on the CME
router and shows registered show sdspfarm units.
Any reason on CUE that would happen from the following version:
Installed Packages:
- Installer
Of *Ricardo Arevalo
*Sent:* Monday, October 20, 2008 10:08 PM
*To:* Ryan Trauernicht
*Cc:* ccie_voice@onlinestudylist.com
*Subject:* Re: [OSL | CCIE_Voice] Same Config CM Phone drops to CUE
Ryan,
In telephony-service add the commando:
call-forward pattern .T
//r.a.
On Mon, Oct
Do you have a ephone-dn assigned to a park slot?
On Thu, Nov 6, 2008 at 3:07 PM, Olson, Pete [EMAIL PROTECTED] wrote:
Yes, it is in the connected state when the park is grayed out.
*Pete Olson*
[EMAIL PROTECTED]
425-965-2577
--
*From:* Vik Malhi
Anyone get this bug in the lab for RDNIS while in SRST.
*CSCsb11565*
I can not seem to get RDNIS to work when a PSTN call comes in. IP phone to
IP Phone while in SRST works great, but not from the PSTN.
I have phone A at the headquarters that has a MRG1 of Sub (multicast) and
MRG2 of Pub (Unicast). Then in my MRGL I have MRG1 then MRG2.
If I shut down the Subscriber I lose all MOH. Am I missing something to
allow a Multicast MOH server failover to a Unicast MOH server?
Do you have the config?
On Thu, Nov 13, 2008 at 4:45 PM, James Key [EMAIL PROTECTED] wrote:
Are you using the MGCP bind commands? If so, try removing these and
re-add.
James
*From:* [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED] *On Behalf Of *Erwan Erwan
*Sent:* Thursday, November
When you are in SRST MOH will not stream IP to IP internal to that SRST
site. It will however stream to the PSTN, so the PSTN users will still hear
MOH.
In normal mode (MGCP or H323) IP to IP phone work fine and PSTN work fine.
On Wed, Nov 19, 2008 at 4:53 AM, Kumar, Narinder [EMAIL PROTECTED]
Do you have the poll status boxed checked also on the 6608?
Check in perfmon to see if the 6608 is in maintenance mode.
On Wed, Nov 19, 2008 at 12:23 PM, anil batra [EMAIL PROTECTED] wrote:
can you confimr your chanel selection is - top-down
--- On Wed, 11/19/08, Studychris [EMAIL PROTECTED]
you need to put a
call-forward pattern .T
in your telephony-service
On Thu, Nov 20, 2008 at 6:57 AM, [EMAIL PROTECTED] wrote:
why do you need to bind sip to loopback interface?
i will put allow sip to h323, s to s and h323 to h323
sara
*Pardeep Singh (pardsing) [EMAIL PROTECTED]* wrote:
Anyway to get MWI working on DNs in CME for a GDM box.
For example...
I have GDM 5999
CME DN's that are subscribers for GDM at 5000 and 5001.
When a message is left in the GDM I was MWI for 5999.
Thanks!
Ryan Trauernicht
So what if you get a question saying use to use annunciator to play a
message to pstn users. Just skip it or use unity then, since that is your
only way to play a message to PSTN users.
On Thu, Nov 20, 2008 at 10:44 AM, Alex Arseniev [EMAIL PROTECTED]wrote:
Correct and please treat as such.
button.
Rgds
Alex
- Original Message -
*From:* Ryan Trauernicht [EMAIL PROTECTED]
*To:* ccie_voice@onlinestudylist.com
*Sent:* Friday, November 21, 2008 1:30 AM
*Subject:* [OSL | CCIE_Voice] GDM for CUE
Anyway to get MWI working on DNs in CME for a GDM box.
For example...
I
Can someone shed some light on Wait for Far End H.245 Terminal Capability
Set. Should that always be unchecked on a GK trunk, or what are the
situations that should be checked and when shouldnt it be checked?
Thanks!
On Sat, Nov 22, 2008 at 6:53 AM, anil batra [EMAIL PROTECTED] wrote:
Hello
if you are using a H323 gateway did you goto the service parameters and
enable stop routing on user busy and existed route list? Set those to
false.
On Sat, Nov 22, 2008 at 8:33 AM, Shadab Abbasi (moabbasi)
[EMAIL PROTECTED] wrote:
Hello All,
I am facing some strange issue here.
I
you have an association problem. Did you create a user with the user id of
ac all lowercase and password 12345. This needs to be done. You also
need to associate the phone with this user.
You are failing on call control. You might want to reset the Telephony
dispatch service as well.
On Sun,
phone to 4111? What happens then?
- Original Message -
From: Ryan Trauernicht
To: [EMAIL PROTECTED]
Cc: Pardeep Singh \(pardsing\) , ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Can't call to CUE pilot number from HQ
Date: Thu, 20 Nov 2008 19:20:43 -0600
you need
that is a preference. You need to understand how to control codec's on
dial-peers both ways. My preference is hard coding it. I don't want the
router doing any decisions for me.
Thanks,
Ryan Trauernicht
On Sun, Dec 7, 2008 at 3:05 PM, Mike Brooks [EMAIL PROTECTED] wrote:
Hi everyone
. Once I change them to the same
file name it works fine. I am guessing this is a caveat for MOH from the
flash, I was just looking for a sanity check.
Thanks,
Ryan Trauernicht
and
that is why it is not working. I would double check on that.
You can test by calling is 2003 (that isnt a DN on a phone) to alias that to
2002 with cfw to the DID.
See if that works.
Thanks,
Ryan Trauernicht
On Thu, Dec 4, 2008 at 9:55 PM, anil batra [EMAIL PROTECTED] wrote:
Hello Experts
Correct Scott, but for codec choice, it really is up to the preference you
have, but make sure you understand how to do both of them.
On Sun, Dec 7, 2008 at 7:43 PM, Hardesty, Scott [EMAIL PROTECTED]wrote:
You should use voice class to set tcp timeout. By default q931 will
timeout before
G711 hardcoded on the dial-peer to the
GK.
Thanks,
Ryan Trauernicht
On Thu, Dec 4, 2008 at 11:30 AM, Khaled Al-Amoodi [EMAIL PROTECTED]wrote:
Dears
I 'd like to set my GK bandwidth to allow only 4 g729 calls between CCM
CME from both sites
I set up my devices as follwoing
In CCM
Does you user show up in IPCC as a resource?
On Tue, Dec 2, 2008 at 8:55 AM, Sergio Polizer [EMAIL PROTECTED]wrote:
Hi,
My CAD can not logon. The system says The ID you entered was not found
I have checked at CM the ID, password, CTI check box and the rmjtapi
associtation to the device.
You need to use DNS I believe. I have never read a doc or heard of anyone
doing it via IP address.
Thanks,
Ryan Trauernicht
On Mon, Nov 24, 2008 at 4:00 PM, James Key [EMAIL PROTECTED] wrote:
Hi Rob,
I think you can only do ip addresses when networking CUE to CUE. CUE to
Unity requires
Reason why this is a much talked about topic is from examples people are
throwing around is: to use the annuciator to please the following message
with DID XYZ is dialed. No where in those wordings does it say to use
Unity. It said to use the annunciator, which I am pretty sure this is not
even
Do you have your CM to GK setup as a ICT H225 trunk?
You might want to try this in your GK
voice service voip
h323
call start interwork
Also make sure you have your h245 capabilities box unchecked on the CM
trunk.
Ryan Trauernicht
On Wed, Dec 10, 2008 at 6:22 PM, [EMAIL PROTECTED] wrote
Any way to play a IPCC prompt when an agent becomes ready while the call is
in queue?
You need to prefix on the RP, or RL.
On Sat, Dec 13, 2008 at 10:31 PM, Norma Exel normae...@writeme.com wrote:
I know I could probably find out the answer if I remember to remind myself
the next time I have a vRack session but would anyone know that, if a h225
trunk is configured to register
I have heard some conflicting discussions on what people would put for QOS
KB.
If the lab asks you to use 5% for BW of a 384 pipe but don't use the
percent.
384 * .05 = 19.2kb
Would you put 19 or do you always round up?
Do the proctors give you some leeway on how what numbers they are looking
to G729 only call, it bombs out right way with a fast
busy.
Is this not an option to have 729 across the wan to a SIP trunk that needs
to be 711.
Thanks,
Ryan Trauernicht
that is why your DN is only showing
up.
Thanks,
Ryan Trauernicht
On Wed, Dec 17, 2008 at 7:59 AM, Kumar, Narinder narinder.ku...@uxcg.com.au
wrote:
I want to receive full E164 number CLID when SiteC PH call PSTN.
Hq(MGCP GW)--CCM-(SiteB MGCP GW
That would prob work, but wouldnt it be mls qos trust dscp since only
801.1q is cos values?
On Wed, Dec 17, 2008 at 1:28 PM, Sergio Polizer spoli...@hotmail.comwrote:
Hi,
The Q.22.39 asks for mark, at Cat3550, DSCP PHB label as EF for RTP traffic
from the port connected to IP Phones.
This is not possible without RSVP.
On Thu, Dec 18, 2008 at 10:55 AM, jeremy co jeremy.coo...@gmail.com wrote:
Hi,
If I want to maintain 4 digit dialing from HQ to BR1 what should I do?
I'm confused since I donnow what to put in Route list. I know HQ GW is
backup to dial BR1 which is in
That is a solution in 4.2 and 5 and above. There is a field for
unregistered which you can point to the DID of the phone.
Thanks,
Ryan Trauernicht
On Thu, Dec 18, 2008 at 3:58 PM, Kumar, Narinder narinder.ku...@uxcg.com.au
wrote:
It is not possible with CCM 4.1(3)
*From:* ccie_voice
You also need to take into account how you are using MOH from CM on IPCC.
If you use the HoldDelayUnhold that is User hold then when the call
is presented to an agent that is network hold.
Both are streamed via CTI Ports.
Thanks,
Ryan Trauernicht
On Fri, Dec 19, 2008 at 1:52 PM, Kevin
on a IOS hardware Xcoder. You must add that codec in.
Let me know if that helps.
Thanks,
Ryan Trauernicht
On Sun, Dec 21, 2008 at 12:58 PM, jeremy co jeremy.coo...@gmail.com wrote:
By the way this the trace output:
Cisco CallManagerStationD: (017) (1,100,132,243) CallInfo
You are right Christian I guess I was assuming to much. I went on the
assumption that you are putting the CTI Route Points and the CTI Ports in
the same Device Pools.
Thanks,
Ryan Trauernicht
On Mon, Dec 22, 2008 at 2:38 AM, Christian Hennrich
christian.hennr...@intact-is.com wrote:
What
I dont see codec G729br8 in there.
That is the codec Call Manager uses for G729.
You did get the call-forward pattern .T which is good, but the codec is
missing.
Thanks,
Ryan Trauernicht
On Mon, Dec 22, 2008 at 6:33 PM, Chris Parker cpar...@cparker.us wrote:
What do you see when you type
.
Thanks,
Ryan Trauernicht
On Mon, Dec 22, 2008 at 1:04 PM, anil batra anil...@yahoo.com wrote:
Hi All,
Does anyone already configured and tested on cisco ipcc express where the
caller will hear Ring Back Tone instead of MOH when an agent transfers a
call to someone.
Thanks for your help
More then likely is a codec issue. You must go G711ulaw to BACD. If you
want G729 across the WAN/GK you will need a Xcoder at the SiteC RTR to hear
audio from a HQ phone.
Thanks,
Ryan Trauernicht
On Fri, Dec 26, 2008 at 9:57 PM, kamal yousaf lovingprin...@gmail.comwrote:
Hi,
I have setup
it is not the cable b/c that cable works with SiteB RTR to PSTN router.
I did a show port 3/1 to get the MAC Address of the 6608 port and plugged
that into MGCP gateway of 6000 T1 VOIP GATEWAY.
any ideas?
Thanks,
Ryan Trauernicht
Can you post your config? What does debug voice dialpeer on CME router
look like?
Thanks,
Ryan Trauernicht
On Sat, Dec 27, 2008 at 10:59 PM, anil batra anil...@yahoo.com wrote:
On GK when I give bandwidth total default 64 while there is only one
zone confgiured under GK. The GK trunk
to geto your CME
router.
Thanks,
Ryan Trauernicht
On Sun, Dec 28, 2008 at 11:41 AM, Jose Gregorio Linero (jlinero)
jlin...@cisco.com wrote:
Hi:
I am testing MOH and using Subscriber as a multicast source and publisher
as a unicast source, everything is working good, however when I am testing
. Not sure if
this is possible or not.
Thanks,
Ryan Trauernicht
What SR version for CM is on the lab I just looked at mine can I have
assisstant console plugin under install plugins.
I am running 4.13SR7
On Mon, Dec 29, 2008 at 8:40 AM, James Key j...@jackhenry.com wrote:
Hi Asif,
The IPMA assistant console isn't located on the plugins page. Here
I can't paste in my screen shot but I have Cisco Unified CallManager
Assistant Console I also have Attendant Console plugin as well.
4.13SR7
Thanks,
Ryan Trauernicht
On Mon, Dec 29, 2008 at 2:03 PM, James Key j...@jackhenry.com wrote:
Hi Ryan,
Are you referring to Attendant Console or IPMA
in
the lab on whether it is there or not. If it is, bonus! Just in case it
may not be, probably still a good idea to know where to find the link in
documentation fast. My BLS solutions shows using the Doc CD to install
assistant console.
James
*From:* Ryan Trauernicht [mailto:ryanstudyvo
Anyway to allow call blocking on a particular DN if you have an H323 Gateway
in regular and SRST mode.
For example... 2001 should be allowed to accept calls during normal mode but
call reject in failover mode?
thanks.
Ryan Trauernicht
1.1.1.1 192.168.187.1
time-zone 8
cor outgoing 3002 4 3002
any ideas?
Thanks,
Ryan Trauernicht
On Fri, Jan 2, 2009 at 3:26 PM, Alex alex.arsen...@gmail.com wrote:
COR?
!
dial-peer cor custom
name Block
name PSTN
!
dial-peer cor list Block
member Block
!
dial-peer cor list PSTN
member
Didnt really think it was needed. I will give it a shot.
On Fri, Jan 2, 2009 at 3:47 PM, graeme watson gree...@googlemail.comwrote:
Hi Ryan - you do not appear to have any cor on the incoming dialpeer 1?
Graeme
2009/1/2 Ryan Trauernicht ryanstudyvo...@gmail.com
Thanks alex... just tried
good catch... what is the reason behind needing it on the inbound. I don't
really see the cor list for incoming on the inbound dial-peer doing anything
really.
On Fri, Jan 2, 2009 at 3:51 PM, Ryan Trauernicht
ryanstudyvo...@gmail.comwrote:
Didnt really think it was needed. I will give
to validate this off of:
http://www.cisco.com/en/US/tech/tk1077/technologies_white_paper09186a00800c5f67.shtml
Thanks,
Ryan Trauernicht
On Fri, Jan 2, 2009 at 7:02 PM, anil batra anil...@yahoo.com wrote:
Hi Vik,
I am again stuck on same issue. I restarted CCM, HQ and BR2 router even
,
Ryan Trauernicht
Just an update on this that old link on the new 4.13 SR does not work
anymore. I just plugged it in and goes to a unavailable page now. You need
to download it from the plugins page.
Thanks,
Ryan Trauernicht
On Mon, Dec 29, 2008 at 4:15 PM, anil batra anil...@yahoo.com wrote:
Thanks James
Sergio thank you very much... not sure how I missed that.
Thanks,
Ryan Trauernicht
On Sun, Jan 4, 2009 at 3:09 PM, Sergio Polizer spoli...@hotmail.com wrote:
Ryan
You can use show frame pvc dlci
Cheers
Sergio
- Mensagem Original -
De: Ryan Trauernicht ryanstudyvo...@gmail.com
ntp master 3
I think I got it figured out. I had to add in a souce command which I can't
think of off the top of my head since everything was going to the loopback
of my NTP server. I will post the command when I get home.
Thanks,
Ryan Trauernicht
On Sun, Jan 4, 2009 at 6:41 PM, Cliff
Technically no you will need to enable multicast routing on the router
(assuming the 6500 dosnt have a mfsc).
thanks
Ryan Trauernicht
On Mon, Jan 5, 2009 at 8:03 AM, marwa marwa_ah...@seegypt.com wrote:
hi,
as i don't have 6500 to test this can anyone tell me how we enable the
music
Sergio, I thought I saw the output for the policy-map when I did a show
frame-relay pvc 101 but now all i see if DLCI info. I dont see the stats
under it anymore.. I made sure my service-policy is applied to the
map-class and still nothing.
Thanks,
Ryan Trauernicht
On Sun, Jan 4, 2009 at 3:17
Nevermind I forgot the traffic-shaping command on the interface.
You must have that on there for the stats to show up.
On Tue, Jan 6, 2009 at 10:50 AM, Ryan Trauernicht
ryanstudyvo...@gmail.comwrote:
Sergio, I thought I saw the output for the policy-map when I did a show
frame-relay pvc 101
It would be post compression.
The priority queue is based on output assuming you are applying it on the
output of the interface.
Thanks,
Ryan Trauernicht
On Tue, Jan 6, 2009 at 10:36 PM, wafers44 wafer...@gmail.com wrote:
If we apply cRTP on the FR interface using frame-relay ip rtp
header
Not sure what your voip config looks like b/c if it is the voip dial-peer
below it will not work.
First, BACD needs not to be SIP. Also, the only way I could get voip to
work is by having the destination-pattern and incoming called-number pattern
be the same DN.
On Wed, Jan 7, 2009 at 9:30 AM,
aa
destination-pattern 3500
session target ipv4:177.3.254.1
incoming called-number 3500
dtmf-relay h245-alphanumeric
codec g711ulaw
ip qos dscp cs3 signaling
no vad
!
Regards
Ryan Trauernicht schrieb:
Not sure what your voip config looks like b/c if it is the voip dial-peer
below
No real work around. Kind of a limitation that I can find. Unless anyone
else can add to this topic and find a workaround for non-ras voip dial-peers
with GK BW limiations.
Thanks,
Ryan Trauernicht
On Wed, Jan 7, 2009 at 10:59 AM, Christian Hennrich
christian.hennr...@intact-is.com wrote
Why would you need both directions. If you mark them on the ingress of the
fast ethernet on each location you dont need the other direction.
On Fri, Jan 9, 2009 at 12:42 PM, Jose Gregorio Linero (jlinero)
jlin...@cisco.com wrote:
Hi Majdi:
Actually you have to do it in both directions for
--
*From:* Ryan Trauernicht [mailto:ryanstudyvo...@gmail.com]
*Sent:* Viernes, Enero 09, 2009 3:40 PM
*To:* Jose Gregorio Linero (jlinero)
*Cc:* Majdi Harb; ccie_voice@onlinestudylist.com
*Subject:* Re: [OSL | CCIE_Voice] marking on routers
Why would you need both directions. If you mark
fine. Any ideas? Thanks, Ryan Trauernicht
just bring it to the proctors attention? thanks, Ryan Trauernicht
12.4.5b
It doesnt happen all the time but it has happened a few times.
On Fri, Jan 9, 2009 at 6:18 PM, anil batra anil...@yahoo.com wrote:
Wt's the version of IOS on your GW? I have seen this happening with older
IOS.
--- On *Sat, 1/10/09, Ryan Trauernicht ryanstudyvo...@gmail.com
Should I be checking the MTP box on the GK (225 controlled) trunk? I didnt
think I was suppose to have that checked.
On Fri, Jan 9, 2009 at 6:21 PM, anil batra anil...@yahoo.com wrote:
wt's ur MTP setting on GK. Try enabing it.
--- On *Sat, 1/10/09, Ryan Trauernicht ryanstudyvo...@gmail.com
...@onlinestudylist.com [mailto:
ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Ryan Trauernicht
*Sent:* Saturday, 10 January 2009 11:08 AM
*To:* ccie_voice@onlinestudylist.com
*Subject:* [OSL | CCIE_Voice] isdn bind-l3 ccm-manager command dissappears
Anyone run into that face that if you
The TCS was unchecked already. MTP required was unchecked. So no good.
Any other ideas?
Thanks,
Ryan Trauernicht
On Fri, Jan 9, 2009 at 9:14 PM, Vik Malhi vma...@ipexpert.com wrote:
Wait for H245 TCS on the trunk page within CCMAdmin should be unchecked.
Let us know how you get
Prob a stupid question is there any way to set a preference command for
the hopoff command on a gatekeeper?
so I dont want CM to register with a tech-prefix but I want to use the
hopoff. I don't see a priority command like do you with a normal zone
prefix command.
thanks,
Ryan Trauernicht
)
--
Message: 1
Date: Fri, 09 Jan 2009 19:14:57 -0800
From: Vik Malhi vma...@ipexpert.com
Subject: Re: [OSL | CCIE_Voice] CM call to CME Phone and CME puts CM
on park... call disconnects
To: Ryan Trauernicht ryanstudyvo...@gmail.com
I agree with Ryan H. Only way that would work is if you checked the VM box.
Verify again that your IPMA CTI Route Point has the CSS for Managers and
not Everyone.
Call-forward no coverage is the one you need would need to be set to
managers CSS.
Thanks,
Ryan Trauernicht
2009/1/10 Ryan Hicks
reply.
Thanks,
Ryan Trauernicht
On Sat, Jan 10, 2009 at 9:54 AM, Ryan Hicks ryanhicks0...@yahoo.com wrote:
Erwan,
If this is the 6608. For fractional PRI you have to set the status poll
check box in the gateway. Then under services, call manager, click the
advanced button, then search for b
What type of hardware MTP are you using. You need to remember that some
older MTP hardware only does Xcoding and not MTP. I believe PVDM1's do not
do MTP... only Transcoding (even though it is called MTP).
6608 port can do hardward MTP.
thanks,
Ryan Trauernicht
On Sat, Jan 10, 2009 at 7:09 PM
Yeah I can prob do that... just looking to see what the flexibility is.
thanks,
Ryan Trauernicht
On Sun, Jan 11, 2009 at 4:18 PM, wafers44 wafer...@gmail.com wrote:
AFAIK, the GK will load balance (random select) b/w the endpoints
registered to the zone that you are trying to hopoff to. I
What is the work around for IPCC express when trying to add in a JTAPI
controlled group with the default DP?
It just sits there and does nothing.
Only way I found around it is to set the JTAPI integration only to the
publisher and configure everything... then add in the sub later.
Thanks,
Ryan
ip dscp cs3 sign already on there so i do not believe it
is the mgcp sending the packet. I also have no voip dial-peers .
Any help would be much appreciated.
Thanks,
Ryan Trauernicht
my looking into the telephony-service on the CME.
I missed
Transfer-pattern .T
Transfer-system full-consult
Sorry for the spam!
Thanks,
Ryan Trauernicht
On Sun, Jan 11, 2009 at 4:20 PM, wafers44 wafer...@gmail.com wrote:
From the CME router, can you collect the following debugs
This is a known bug. of the show policy-map int command. Is it strictly
cosmic. What version of code do you have? 12.2.25see?
Thanks,
Ryan Trauernicht
On Sun, Jan 11, 2009 at 10:49 PM, jeremy co jeremy.coo...@gmail.com wrote:
Hi,
Interestingly no traffic is matched by class map, anybody
Very true. Yeah I have verified the packet it was. It was the 6608
keepalive messages. Anyway to remark them on the application layer (aka in
CM). I know I can create an access-list to remark them, but I wanted to see
if there was alternatives.
thanks,
Ryan Trauernicht
On Sun, Jan 11, 2009
Jeremy, The would tell you in the lab what to do. If they say to configure
3 Unity ports and don't give you any more specific details... assume that
all 3 ports can do all rolls. answer calls, TRAP, MWI notification and
notification dialout.
thanks,
Ryan Trauernicht
On Mon, Jan 12, 2009
and
getting supplementary features to work?
thanks,
Ryan Trauernicht
.
--
*From: *Ryan Trauernicht ryanstudyvo...@gmail.com
*Date: *Sun, 11 Jan 2009 16:46:39 -0600
*To: *ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
*Subject: *[OSL | CCIE_Voice] IPCC JTAPI Control Group Hanging
What is the work around for IPCC express when trying to add
.
*From: *Ryan Trauernicht ryanstudyvo...@gmail.com
*Date: *Mon, 12 Jan 2009 13:30:19 -0600
*To: *Vik Malhi vma...@ipexpert.com
*Cc: *ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
*Subject: *Re: [OSL | CCIE_Voice] IPCC JTAPI Control
Correct.
In IPCC for the JTAPI Provider put Sub,Pub (192.168.187.12,192.168.187.11)
Then try to add in a group. It doesnt matter what DP I pick.
If I change the Provider to only the Pub and you are good to go.
Also restarting the node manager.
Thanks,
Ryan Trauernicht
On Mon, Jan 12, 2009
- The Global Leader in Self-Study, Classroom-Based,
Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE
RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and
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--
*From: *Ryan Trauernicht
What prompt do you choose to play when configuring the timeout to 30?
IF you dont pick a valid prompt doesnt it goto unsuccessful?
On Sun, Jan 11, 2009 at 4:09 PM, Ryan Trauernicht
ryanstudyvo...@gmail.comwrote:
That options seems like it would save alot of time. I have always just
pulled
policy-map way. You must do it the old school
map-class way for (cir, mincir, bc, be).
Vik can you comment on that or anyone else who knows for sure.
Thanks,
Ryan Trauernicht
Another question is.
If you use the traditional way for low speed links... (cir, mincir, bc, be)
it is required to do the frame-relay traffic-shaping on the physical isnt
it?
On Mon, Jan 12, 2009 at 4:49 PM, Ryan Trauernicht
ryanstudyvo...@gmail.comwrote:
That is what I thought.
do you
:
ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Ryan Trauernicht
*Sent:* Tuesday, 13 January 2009 10:05 AM
*To:* Vik Malhi
*Cc:* ccie_voice@onlinestudylist.com
*Subject:* Re: [OSL | CCIE_Voice] QOS MLP 1 link and FRF Shaping the other
verification
Another question is.
If you use
way
for FRF.12 (aka nested policy-maps)
Thanks,
Ryan Trauernicht
On Mon, Jan 12, 2009 at 5:49 PM, anil batra anil...@yahoo.com wrote:
1. Let's say BR1 to HQ we are to use MLP with LFI but for BR2 to HQ we are
to use FRF.12 Fragmentation. What I understand is we will use MLP with LFI
between HQ
Do you need fast start enabled if you have an H323 Gateway that calls into
CUE?
Thanks,
ryan Trauernicht
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