hi Guys,
I am looking fornbsp; learningnbsp; resources on the following .nbsp; Could
someone guide me to some videos andnbsp; books for this ...
1)nbsp; Sippnbsp; (Open Source test tool )
2) Asterisk PBXnbsp;
Regards,
Vir
___
Free CCIE RS,
hi All,
In CRCX message in mgcpnbsp; if wenbsp; seenbsp;
S0/SU1/DS1-0/2...@r.comnbsp;nbsp;nbsp; then does it mean that that
thisnbsp;nbsp; is a T1 controllernbsp; ( 0/1/0)nbsp; ornbsp; E1 controller
( 0/1/0)nbsp; ?
- Vir
___
Free CCIE RS,
Hello Guys,
Could anyone help me with the followingnbsp; :-
1) I am looking at convertingnbsp; a sip and H323nbsp; pcap ( wireshark file
)nbsp; into a plain textnbsp; filenbsp; ( .txt format)nbsp; . Are there any
easy options to do this ?nbsp;nbsp; I sawnbsp; some tcpdump options however
not
Hello Guys,
I have collected a Packet capture ( wireshark) from a cisco CME .nbsp; I need
to knownbsp; following :-
1) The filters used on wireshark to see all traffic that passed between the ip
phonenbsp; and the CME.
2) Would also like tonbsp; how to filter wireshark captures based on ip
hello,
nbsp;I have my pstn called number coming into my gatewaynbsp; as 8 digits (
2404)nbsp; and if I am strippingnbsp; the number to the last 4
digitsnbsp;nbsp; at the voice port .
Now my question is if I am required to call from the pstn to numbernbsp;nbsp;
24044000 thennbsp;nbsp; can
hi Guys,
I was practicing the lab exercises today . When I was trying to configure BACD
I went to the cisco product link for the doc on this location:
http://www.cisco.com/cisco/web/psa/default.html?mode=prod
And found thatnbsp; CME docnbsp; unders Voice amp; unified communication
gt;gt; IP
To: virajith lt;vir...@rediffmail.comgt;
Cc: ccie_voice@onlinestudylist.com lt;ccie_voice@onlinestudylist.comgt;
Subject: Re: [OSL | CCIE_Voice] QOS WAN
At first glance those steps look ok. nbsp;What do you see from show traffic
and show frame-r pvc # on all three site routers?
For odd audio issues I would
Hello All,
Any update? I am still waiting for a reply.
Please assist guys.
From: virajith lt;vir...@rediffmail.comgt;
Sent: Sat, 21 Sep 2013 07:18:00
To: ccie_voice@onlinestudylist.comlt;ccie_voice@onlinestudylist.comgt;
Subject: QOS WAN
hi All,
I am trying to connect HQ and SB
hi All,
I am trying to connect HQ and SB with a 384 k frame relay PVC.Enable FRF.12
link fragmentation and interleave on the Frame Relay connections to
fragment large data packets and interleave voice packets to minimize delays.
Max delay between fragments should be
set at 10 ms . Also
hi All,
I am trying to connect HQ and SB with a 384 k frame relay
PVC.Enable FRF.12 link fragmentation and interleave on the Frame Relay
connections to
fragment large data packets and interleave voice packets to minimize delays.
Max delay between fragments should be
set at 10 ms . Also
hi Guys,
What if we are required to use Embedded Call-Queue and AA Tcl Scriptsnbsp; in
the lab. Currently I am practicing by using tcl scripts in the flash
(3.0.0.2.tcl)
Questions:
=nbsp;
1) Do all 12.4 IOS versions support Embedded scripts?
2) How do I check if my IOS
HI All,
I am trying to write a uccx script for the following requirement and not able
understanding the logic here could someone help? I there a simple script that
can be written for this?
I have two IPCC extensions 1 on HQ Phone 2 ( ext: 2102) and 1 on SiteB Phone 2
(ext:
3102).
Incoming
hinbsp; Guys,
My cme meet me conference does not work. I dial 4321 ...it says Cannot
Complete Conference
Here is my config ...
voice-card 0
nbsp;dspfarm
nbsp;dsp services dspfarm
!
ephone-dnnbsp; 7nbsp; octo-line
nbsp;number 4321 no-reg primary
nbsp;conference meetme
!
telephony-service
Trnsfer
From: Cory Gray lt;corygray22...@hotmail.comgt;
Sent: Wed, 13 Feb 2013 18:46:23
To: 'virajith ' lt;vir...@rediffmail.comgt;,
lt;ccie_voice@onlinestudylist.comgt;
Subject: Re: [OSL | CCIE_Voice] CME Meet me conference not working
Did you add “meet me” to the seized layout?
nbsp;1.nbsp
TrnsfVM Trnsfer
From: Cory Gray lt;corygray22...@hotmail.comgt;
Sent: Wed, 13 Feb 2013 18:46:23
To: 'virajith ' lt;vir...@rediffmail.comgt;,
lt;ccie_voice@onlinestudylist.comgt;
Subject: Re: [OSL | CCIE_Voice] CME Meet me conference not workingDid you add
“meet me” to the seized layout? nbsp;1
hi Guys,
How do I write anbsp; UCCX script fornbsp; the following...
HQ Phone 2 is ready to take a call, callershould hear “ Thank you for waiting,
we will connect you now to X. Where X is the user name. On thesame token if
SiteB Phone 2 is ready to take a call, it should play SBPH2. The
hi Jamie,
Thanks for your response.
Could you elaborate the exact steps to be done?
Thanks,
Vir
From: Jamie Parr (jamparr) lt;jamp...@cisco.comgt;
Sent: Mon, 04 Feb 2013 17:26:50
To: virajith lt;vir...@rediffmail.comgt;
Cc: ccie_voice@onlinestudylist.com lt;ccie_voice@onlinestudylist.comgt
hi Guys,
I am just wondering if we are required to download the TCL scripts for the BACD
config or it will be present in the flash and we can therefore just go ahead
with the configuration.
Also if it is not present in the flash . Are we to assume that BACD
configuration needs to be done
Hi Guys,
I am just wondering in the ip addressing scheme UCCX is 142.100.64.14 , CUPS is
142.100.64.15 and CUPC is 142.100.64.16.
1) I am just trying to understand how the CUPCnbsp; client is 142.100.64.16
given that it is on the UCCX machinenbsp; ( windows) which has an ip address
of
hi All,
Any feedback on the below email?
-Vir
From: virajith lt;vir...@rediffmail.comgt;
Sent: Sun, 30 Dec 2012 21:31:01
To: ccie_voice@onlinestudylist.comlt;ccie_voice@onlinestudylist.comgt;
Subject: LAN QOS - Need clarification
Hi All,
I need to map cos 5 to EF 46. The gig port for CUPC
Hi All,
I need to map cos 5 to EF 46. The gig port for CUPC client needs to be marked
for dscp3 and
guaranteed BW of 32k for CUPC signalling
Excess traffic should be marked to dscp 8 and then transmitted.
My configuration on the Headquater switch is as follows...
mls qos
mls qos map cos-dscp
hi All,
I am noticing that for my cisco unity connection integratednbsp; with
CUCMnbsp; - the VM pilot and profile that I have made as default VM pilot and
profile is not getting automatically applied on my phones and therefore any
call forward busy or no answer gets a busy tone.
I have to
hi Guys,
I tried out everything to get the deskphone mode working on my CUPC but still
no good. The following was done...
-verified the end points are registered
-user is associated with the line
-rebooted the CUPS server
-redid the entire integration twice
-still no good .
I get the
Hi Zahar,
Coolnbsp; :)
Thanks for the procedure brother. I will try it out.
-Vir
From: Zahwar Hussain lt;911c...@gmail.comgt;
Sent: Sun, 23 Dec 2012 06:10:26
To: virajith lt;vir...@rediffmail.comgt;
Cc: ccie_voice@onlinestudylist.com lt;ccie_voice@onlinestudylist.comgt;
Subject: Re: [OSL
HI Guys,
I have configured cme-srst on my router tonbsp; support phonesnbsp; amp;
gateway functionnbsp; when the wan is down and when my mgcp gateway unreg from
callmanger. Now the requirement is that voicemail forwarding using CUE should
work between ip phones as well as PSTN calls and that
hi All,
Do anyone know to configure unity connection such that when a phone user
presses the Meet me
conf button ..it announces the participant name and when somebody dials the
meet me number
it asks who may I say is calling
Please elaborate the configuration steps used if possible?
- Vir
hi All,
Had a few questions ...
1) is switch type and plan required to be configured in the srst mode for Mgcp
and h323 gateways?
2) Does any have have working configurations for callmanger fallback and
cme-srst for the branch sites ? if yes can u email me the configs.
Thanks Guys,
Vir
hi Gazzaz,
Thanks for the links.
1) If the digit strip is done on the callmangernbsp; then if the topology is...
ip phonecucm--h323 gateway--pstn
Then are you using seperate set of dial peers for the calls made in normal mode
and the ones made in srst mode?
Thanks,
Vir
hi Guys,
I have a question regarding if we need to carry any identity documents when we
go to thenbsp; exam center such as
- driver's license or passport photocopy .
- Also do we need to carry lunch and water or isnbsp; it provided?
- any other detailsnbsp; or documents we need to carry with us
hi Guys,
In the client I get the option of deskphone mode in CUPC but when I select
itnbsp; the mode does not get applied.
In show server health I see the Desk phone mode says Partially
Connected(Primary) - Cannot Connect to Phone.
I have checked the cti server / gateway profile and it is
Yes it is
From: Chrysostomos Christofi lt;ch.christ...@logicom.netgt;
Sent: Wed, 19 Dec 2012 12:29:23
To: virajith lt;vir...@rediffmail.comgt;, ccie_voice@onlinestudylist.com
lt;ccie_voice@onlinestudylist.comgt;
Subject: Re: [OSL | CCIE_Voice] CUPS- Presence - deskphone mode not active
hi Guys,
My CUE module is rebooting erratically...
SiteC#service-module integrated-Service-Engine 1/0 session
Trying 142.161.120.98, 2066 ... Open
Waiting for IOS to register IP address.
nbsp;- waited 30 seconds...
Waiting for IOS to register IP address.
nbsp;- waited 40 seconds...
Waiting for
hi Guys,
I want to Modify ICD script to play pre-recorded prompts before call is
delivered to
the agent. For example, if HQ Phone 2 is ready to take a call, caller should
hear “ Thank you for waiting, we will connect you now to X. Where X is the
user name. On the same token if branch 1 Phone 2
hi Guys,
on UCCX my 1 button login works , also my cti route points are
registered,nbsp; I can also call the AAnbsp; but when I call the ICD it gives
a message we are currently experiencing system problems and are unable to
connect your call
Observation:
-One button login works but Inbsp;
hi guys,
I am unable to clear / remove the mls and qos commands from my switch...
Switch#sh running-config | b mls
mls qos map cos-dscp 0 8 16 24 32 46 48 56
mls qos srr-queue input bandwidth 90 10
mls qos srr-queue input threshold 1 8 16
mls qos srr-queue input threshold 2 34 66
mls qos
then this is going to eat into the allocated exam time?
-Vir
From: Steffen Bruening lt;stbruen...@gmail.comgt;
Sent: Sat, 01 Dec 2012 12:24:35
To: virajith lt;vir...@rediffmail.comgt;
Cc: ccie_voice@onlinestudylist.com lt;ccie_voice@onlinestudylist.comgt;
Subject: Re: [OSL | CCIE_Voice
Hi All,
I am wondering which would be the best interface to bind for transcoders ,
gateways , cfbs and for sccp configurations I know loopback is the best for
binding but with like to clarify with the example below...
For example...
R3#sh ip int brief
Interfacenbsp;nbsp; IP-Addressnbsp;nbsp;
hi All,
Is there any default script which I can modify on UCCX to this something
similar to get Thank you for calling if you dialed this by mistake press 1 to
contact operation else someone else will be with you shortly . Press 1 --gt;
should go to 9001
Thanks,
Vir
nbsp;
hi Chrysostomos,
Thanks for your reply.
I have tried to modify the default ICD script many times . However it does not
work for me.
Please assist..
-Vir
From: Chrysostomos Christofi lt;ch.christ...@logicom.netgt;
Sent: Wed, 28 Nov 2012 15:51:03
To: virajith lt;vir...@rediffmail.comgt
hi Guys,
Have a short query ..
How do we ensure that the cos vaule 5 is mapped to DSCP EF . On the interface
all incoming
traffic from the CUPC should be marked CS3 and a bandwidth of 32 k. If there be
anything
in excess should be first marked down to dscp value of 8 before transmitting.
I
hi guys,
I am trying to check the media capabilities( of the gateway) and ringing events
( callmanger) in mgcp debugs .
Questions:
1) In the 200 ok I don't see any media cap . I only see it in MDCX (this is a
callmanger message). No where else is the media cap see . Am I checking the
right
hi guys,
I am trying to check the media capabilities( of the gateway) and ringing events
( callmanger) in mgcp debugs .
Questions:
1)
In the 200 ok I don't see any media cap . I only see it in MDCX (this
is a callmanger message). No where else is the media cap see . Am I
checking the right
hi All,
Just wondering if you can help me with the following..
1)Is there an easy onenbsp; translation rule one can use to translate any
number of digits that come from pstn to a 4 digit number on phones registered
to cme or in srst?
2) Can num -exp be used to achieve the same as in
hi Guys,
I am just wondering if one can practice the labs without a ESW module .
I have a setup in with I am using a switch (3750) for the vlan config.
Is an ESW module necessary for the lab practice?
How is the above config on 3750 different from using an ESW module?
-Vir
Hi Guys,
If we need to configure RSVP between 2 sites and There can be 4 concurrent
calls. G711 CODEC to be used for
multi-directional audio.
this mean that we set things up as this : -
dspfarm profile 1 mtp
codec g711u
codec pass‐through
rsvp
maximum sessions software 4
associate
allocation .
Hence why is there a need to specify Mandatory in locations on thenbsp;
callmanager?
-Vir
From: Cory Gray lt;corygray22...@hotmail.comgt;
Sent: Mon, 22 Oct 2012 18:49:20
To: 'virajith ' lt;vir...@rediffmail.comgt;,
lt;ccie_voice@onlinestudylist.comgt;
Subject: Re: [OSL
hi Guys,
If a phonenbsp; is ready to take a call, caller should hear “ Thank you for
waiting, we will connect you now to Y. Where Y is the user name. On the same
token if another phone is ready to take a call, it
should play the phone's display name. The initial prompt “Thank you for
calling”
hi Guys,
If a phonenbsp; is ready to take a call, caller should
hear “ Thank you for waiting, we will connect you now to Y. Where Y is
the user name. On the same token if another phone is ready to take a
call, it
should play the phone's display name. The initial prompt “Thank you for
hi Guys,
If we want only 1 phone to display No services Configured when pressing
corporate directory button.
What is thenbsp; easiest way to achieve this config? Could someone guide me
with the steps?
-Vir
From: ccie_voice-requ...@onlinestudylist.com
Sent: Wed, 10 Oct 2012 21:31:21
To:
hinbsp; ALL,
I want to know how to setup frame relay and point to point config for a lab
setup on with I am practicing voice lab exercises.
Currently I have the entire setup connected via fastethernet cables . However I
am told that serial cables are required for wan configurations.
Donbsp;
It is 7960 phone...
From: Kevin Spicer lt;ke...@kevinspicer.co.ukgt;
Sent: Sat, 22 Sep 2012 14:25:43
To: virajith lt;vir...@rediffmail.comgt;
Cc: ccie_voice@onlinestudylist.com lt;ccie_voice@onlinestudylist.comgt;,
Abel ... lt;midga...@gmail.comgt;
Subject: Re: [OSL | CCIE_Voice] need
hi guys,
I need to understand or have a simple way of understanding + dialing and number
localization. Can anyone help me wiht this?
Thanks,
Vir
From: ccie_voice-requ...@onlinestudylist.com
Sent: Thu, 20 Sep 2012 21:29:38
To: ccie_voice@onlinestudylist.com
Subject: CCIE_Voice Digest, Vol
...@gmail.comgt;
Sent: Fri, 21 Sep 2012 17:45:34
To: 'virajith ' lt;vir...@rediffmail.comgt;,
lt;ccie_voice@onlinestudylist.comgt;
Subject: Re: [OSL | CCIE_Voice] need to understand + dialing and number
localization
Hinbsp;Watch IP Expert + dialing lecture from Vik.nbsp;Best way to
understand itnbsp
as +14081515111nbsp; it gets stored as just 515.
Why does this happen is my config wrong? Please help.
Regards,
Vir
From: Nicolas MICHEL lt;mcl.nico...@gmail.comgt;
Sent: Fri, 21 Sep 2012 17:45:34
To: 'virajith ' lt;vir...@rediffmail.comgt;,
lt;ccie_voice@onlinestudylist.comgt;
Subject: Re
To: 'virajith ' lt;vir...@rediffmail.comgt;,
lt;ccie_voice@onlinestudylist.comgt;
Subject: Re: [OSL | CCIE_Voice] need to understand + dialing and number
localizationHinbsp;Watch IP Expert + dialing lecture from Vik.nbsp;Best way
to understand itnbsp;nbsp;nbsp;From: ccie_voice-boun...@onlinestudylist.com
of
the number getting stored as +14081515111nbsp; it gets stored as just 515.
Why does this happen is my config wrong? Please help.
Regards,
Vir
From: Nicolas MICHEL lt;mcl.nico...@gmail.comgt;
Sent: Fri, 21 Sep 2012 17:45:34
To: 'virajith ' lt;vir...@rediffmail.comgt;,
lt;ccie_voice
Hi guys,
I am trying to achieve the following with UCCX :
1)users should hear a script saying Thank u for calling and All our reps are
busy at this time please stay on the line someone will be with u shortly
2) The script should also be capable of playing to the caller his position in
hi folks,
I am noticing while usingnbsp; 9011.!nbsp;nbsp; asnbsp; the route pattern
for International numbersnbsp; only the the first digit after the the predot
strip is being sent to PSTN.
For eg. if I dial 901187222023005nbsp;nbsp; only the the 8 is sent in the
called party number and not
hi guys,
what is the procedure to get a phone subscribed to corporate directory.
Currently under the phone gt; subscribe/unsubscribenbsp; I only see Intercom
calls. But not Corporate directory.
-Vir
From: ccie_voice-requ...@onlinestudylist.com
Sent: Fri, 31 Aug 2012 00:28:31
To:
hi All,
When usingnbsp; international Route pattern ofnbsp;nbsp;
9011.!nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp; with predot strip .nbsp; I
see that the international number I dial is shows called party number of 8 and
not the other digits I dial after that.nbsp; If I make the the international
hi Guys,
A phone user should be able to divert an incoming ringing call to VM without
answering . Also he should be able to transfer active calls on his phone to
voicemail by dialing a specific number (mapped to VM). How should I achieve
this requirement using Unity
connection?
Thanks,
Vir
...@ucguerrilla.comgt;
Sent: Tue, 21 Aug 2012 03:00:46
To: Dan Quinlan (daquinla) lt;daqui...@cisco.comgt;
Cc: Online Study lt;ccie_voice@onlinestudylist.comgt;, virajith
lt;vir...@rediffmail.comgt;
Subject: Re: [OSL | CCIE_Voice] UNITY CONNECTION ISSUE! (virajith )
Woops. That would be 0 points! Thanks
hi Guys,
How to write a script which will transfer calls to available agent based on
longest idle time. The phone user has requested some time after ending a
call. During this time, phone user should be marked NOT READY.
User has to change status to READY manually before call can be directed.
HI Guys,
This is first email to onlinestudy list :) I amnbsp; replying on top of an
existing chain as my first email bounced from this alias as I sent is as a
separate email to onlinestudy list. Below is my question...
I am trying to config the following for Switch QOS:
COS 5 should be in
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