We had the same problem. Unfortunately it's a user training issue. We decided
to take out the 911 pattern and put stickers on the phones which specified to
dial 9911 for emergency. False 911 calls dramatically reduced after that.
HTH
> On May 26, 2014, at 8:31 AM, Ben John wrote:
>
> Guys,
>
First of all, mind yr manners on the forum before you say anything outlandish
on different races. Secondly using the website at the first place to get lab
dumps shows yr character on how you approach CCIE.
> On Mar 17, 2014, at 10:20 AM, kit yee wrote:
>
> Yes, ccielabdumps some indian cheate
> importantly I don't revisit config pahes just to setup aar.
>
> Hope this helps...
>
> -Justin
>
> On Nov 4, 2013 6:57 AM, "CISCO CCIE VOICE" wrote:
>
>> Hi Guys,
>>
>> i am trying to configure AAR between 2 sites, when i assign AAR-GROUP a
Hi Guys,
i am trying to configure AAR between 2 sites, when i assign AAR-GROUP and
AAR-CSS on device pool it does not take effect rather i have to apply it
each phone device and GW inorder for it to work.Is there any thing i am
missing
Thanks
___
For mo
CDR/CAR should be able to provide breakdown by PRI since it's MGCP.
On Sep 4, 2013, at 5:34 PM, Edgar Feliz wrote:
> TELCO can provide a usage report for each PRI, who is the SP?
>
> Edgar
>
>
> On Tue, Sep 3, 2013 at 2:23 PM, Hesham Abdelkereem
> wrote:
>> Dear Experts,
>>
>> I have 12
great, but they have their place. I don't think telecom will ever
> be a total cloud based solution.
>
> From: Bill Lake
> To: Drake J
> Cc: "ccie_voice@onlinestudylist.com"
> Sent: Thursday, August 29, 2013 8:12 AM
> Subject: Re: [OSL | CCIE_Voice] Is
Hi Experts,
I am not able to restrict max-call-per-button 2 and busy-trigger-per-button
1 using ephone-template
ephone-dn-template 1
call-forward busy 2220
call-forward noan 2220 timeout 20
huntstop channel 1
ephone-template 1
softkeys remote-in-use Newcall CBarge
max-calls-per-button 2
n...@onlinestudylist.com] On Behalf Of CCIE Voice
> Aspirant
> Sent: Thursday, July 25, 2013 12:45 PM
> To: ccie_voice@onlinestudylist.com
> Subject: [OSL | CCIE_Voice] HWIC-4ESW
>
> Hello list
>
> I am looking for 2 HWIC-4ESW cards
Sounds good, thanks Hesham!
On Jul 25, 2013, at 1:29 PM, Hesham Abdelkereem
wrote:
> Just get that model is good enough and cheap 3550-PWR-24
>
>
> On 25 July 2013 11:28, Hesham Abdelkereem wrote:
>> Yes sure just make DOT1Q on port 24
>> switchport mode trunk
>> switchport trunk encapsula
gt;
> Thanks,
> Hesham
>
>
> On 25 July 2013 09:45, CCIE Voice Aspirant
> wrote:
>> Hello list
>>
>> I am looking for 2 HWIC-4ESW cards for my lab, does anyone have spares I can
>> buy?
>>
>> Thanks
>> _
Hello
Does anyone have spare HWIC-4ESW and 3560 I can buy?
Thanks
___
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Hello list
I am looking for 2 HWIC-4ESW cards for my lab, does anyone have spares I can
buy?
Thanks
___
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Are you a CCNP or CCIE and looking for a job? Ch
hannel selection order is not bound to your telco
> configuration.
> You can set top-down or bottom-up with no issue at all.
>
> hth
> lorenz
>
>
> On Wed, Jul 3, 2013 at 10:58 AM, CISCO CCIE VOICE
> wrote:
>
>> Hi Guys,
>>
>> I am trying to understand the
Hi Guys,
I am trying to understand the "Channel selection order", i know that if
TELCo is using bottom Up then i have to use opposite Direction i.e Top
Down .My Telco is uses last channel for IN-bound calls and set Top down on
my side for OUT-bound calls but still i can see that its taking bottom
38404
>
> “Designing and Implementing Cisco Unified Communications on Unified
> Computing Systems”
>
> ** **
>
> *From:* CISCO CCIE VOICE [mailto:ccievoic...@gmail.com]
> *Sent:* Monday, July 01, 2013 12:58 PM
> *To:* Sears, Michael (msears)
> *Cc:* ccie_voice@
Thank You Michael, Question asked about L1 should me NETWORK SIDE and L2
USER SIDE in that case
controller t1 0/0/0
clock source line -L1 Network Side (This is by default
enable no need to add it)
and for L2: USER SIDE do we need to add any additional commands under
serial int
Hi Michal,
As i can see that Clock source line command which is by default enable
further can u give an example for both "User Side" and "Network Side"
means can u past the commands that indicates both user and Network Side
Thanks
On Mon, Jul 1, 2013 at 7:00 PM, wrote:
> Send CCIE_Voice mai
Hi Amit,
Remove the TCL application and add it again i will work
Thanks
On Sun, Jun 30, 2013 at 9:00 AM, wrote:
> Send CCIE_Voice mailing list submissions to
> ccie_voice@onlinestudylist.com
>
> To subscribe or unsubscribe via the World Wide Web, visit
> http://onlinestudylist
Hi Todd,
Please add the below commands and let me know the result...
voice service voip
allow-connections h323 to h323
h323
emptycapability
h225 id-passthru
h225 connect-passthru
Thanks
On Wed, Jun 26, 2013 at 5:38 AM, wrote:
> Send CCIE_Voice mailing list submissions to
> cc
ning: parameter number-of-hunt-grps has not been registered under
> app-b-acd namespace
> Branch2(config-app-param)#do show run | begin application
> associate application SCCP
> !
> dspfarm profile 5 conference
> codec g711ulaw
> codec g711alaw
> codec g729ar8
> codec g7
t; !
>>
>> service app-b-acd-aa <-- you can't change the name of the embedded
>> BACD AA script
>>. (detail remove for brevity).
>>param service-name app-b-acd <-- refer to the embedded BACD Queue
>> script
>>param handoff-
ueue script
>param handoff-string app-b-acd-aa
>. (detail remove for brevity).
> !
>
> dial-peer voice 222 voip
> service app-b-acd-aa <-- refer to the name of the embedded BACD AA
> script
>. (detail remove for brevity)...
> !
>
>
at, Jun 22, 2013 at 1:00 AM, CISCO CCIE VOICE
> wrote:
>
>> Hi,
>>
>> I am trying to Remove B-ACD configuration but still showing in the
>> running configuration i have restarted the router but no look any guess?
>>
>>
>> thanks
>>
>>
>>
Hi,
I am trying to Remove B-ACD configuration but still showing in the running
configuration i have restarted the router but no look any guess?
thanks
___
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HI experts,
I am trying to remove B-ACD cnofigs from the router but still show in
running config,i have reloaded the router but still no luck any guess ?
Thks
___
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an stay in a call queue and
> retry to connect with a hunt group before the call is sent to an alternate
> destination number.
>
> •*seconds*—Maximum period of time, in seconds. The range is from 30 to
> 3600. The default is 600.
>
>
> On Mon, Jun 17, 2013 at 5:11 PM, CISCO
hi experts,
I am trying to understand timer for "*param* *max-time-call-retry" can
anyone share there knowledge about how does it effect the bacd script and
the purpose of this field*
*
*
*Thnks*
* *
___
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fairly quickly. Good luck.
>
>
> Sent from an Apple iOS device with very tiny touchscreen input keys.
> Please excude my typtos.
>
> On Jun 16, 2013, at 10:06 AM, "CISCO CCIE VOICE"
> wrote:
>
> I have Branch 2 site which contains Transcoder moreover all my
ctions. Otherwise you would get a busy signal if
> region said g729 and CCX was set for g711.
>
>
>
>
>
>
> Sent from an Apple iOS device with very tiny touchscreen input keys.
> Please excude my typtos.
>
> On Jun 16, 2013, at 9:41 AM, "CISCO CCIE VOICE"
> w
ebook<http://www.facebook.com/pages/Alexander-Open-Systems-AOS/109484829074064>
> | LinkedIn <http://www.linkedin.com/company/aos>
>
>
>
>
> Sent from an Apple iOS device with very tiny touchscreen input keys.
> Please excude my typtos.
>
> On Jun 16,
t;
> I think the easiest way is to check UCCX Service Parameters under System
> menu.
>
> --Somphol.
>
>
> --Somphol
>
>
> On Sun, Jun 16, 2013 at 7:36 PM, CISCO CCIE VOICE
> wrote:
>
>> Hi,
>>
>> How ca
Sent from Mailbox <https://www.dropbox.com/mailbox> for iPad
>
>
> On Sun, Jun 16, 2013 at 3:37 AM, CISCO CCIE VOICE
> wrote:
>
>> Hi,
>>
>> How can i verify that UCCX is using G729 codec native
>>
>> Thnajs
>>
>
>
___
Hi,
How can i verify that UCCX is using G729 codec native
Thanks
___
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Hi,
How can i verify that UCCX is using G729 codec native
Thnajs
___
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It will be a better option to cancel or reschedule CCIE voice lab exam and
take it after Feb 2014 ,its wast of money,time and effort on CCIE Voice
On Wed, May 29, 2013 at 10:27 AM, wrote:
> Send CCIE_Voice mailing list submissions to
> ccie_voice@onlinestudylist.com
>
> To
Hi experts,
i am also searching for that solution thanks i will also try that and let
u know, which command i need to run on SB gateway to get the output,do i
need call start fast under voice service voip in SB gateway ?
Thanks
On Tue, May 21, 2013 at 11:41 AM, wrote:
> Send CCIE_Voice mail
, 2013 at 10:21 PM, CISCO CCIE VOICE
> wrote:
>
>> HI experts,
>>
>> When the Phones are in CME SRST its does not shows Extension on Phone
>> display this happens when i change the name +86223033001 to S
HI experts,
When the Phones are in CME SRST its does not shows Extension on Phone
display this happens when i change the name +86223033001 to SiteC Phone 1
under Ephone-dn's
Thnks
___
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Hi Experts,
When i dial UCCX Trigger i am not able to play custom welcome prompt rather
its playing MOH.
thanks
___
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Are you a CCNP or CCIE and looking fo
Hi Experts,
How can i send Calling name to PSTN over MGCP and H323 Gateway?
thanks
___
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I am facing the same problem when is use default icd.aef script
___
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ly
> validated
> 4-Then go to UCCX Admin > Script Management -> Upload the Script
> --> Overwrite the Script --> Then Refresh The Script
> 5-Refresh the Application Script
>
> Then you should be good to go
>
> Thanks,
> Hesham
>
>
> On 29 Ap
Hi Experts,
I am getting an error "i am sorry we r currently facing system problm and r
unable t process ur call" during CSQ
Thanks
___
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Are you a CCNP or
OK, The difference is the variation
> between the INVITE and the ACK.
>
> This link has further description.
> http://www.iplogos.fr/English-Resources/Focus/sip-early-media-early-offer-en.html
>
>
>
> On Sun, Apr 28, 2013 at 2:51 AM, CISCO CCIE VOICE
> wrote:
>
>> Thank you
Thank you suresh i will check ths on my pod...
On Sun, Apr 28, 2013 at 11:26 AM, Suresh Bhandari wrote:
> Content Length 0 means, there will be an OK followed with SDP So
> basically both are same.
>
>
> On Sun, Apr 28, 2013 at 1:56 PM, CISCO CCIE VOICE
> wrote:
>
>
, Suresh Bhandari wrote:
> If you are busying out the channels in CUCM using automatic mgcp
> configuration, it will not be Fractional PRI... Experts, please correct me
> if I am wrong!
>
>
> On Sun, Apr 28, 2013 at 2:01 PM, CISCO CCIE VOICE
> wrote:
>
>> HI Thanks Again,
>
> Notice the boldface text that represents your gateway.
>
> Further, you have to check Enable status Poll in your gateway page.
>
> Hope this helps.
>
>
> On Sun, Apr 28, 2013 at 12:45 PM, CISCO CCIE VOICE
> wrote:
>
>> HI Thanks,
>>
>> but if suppo
including SDP message and so its an early offer.
>
> HTH
>
>
> On Sun, Apr 28, 2013 at 1:15 PM, CISCO CCIE VOICE
> wrote:
>
>> HI Experts,
>>
>> Which SIP Message contains Delay Offer and Early Offer message ? i i am
>> bit confuse as some document
HI Experts,
Which SIP Message contains Delay Offer and Early Offer message ? i i am bit
confuse as some documents says SIP INVITE message contains both Delay and
Early Offer and some documents say its SIP 200 OK message
Thanks
___
For more information
ster for you.
>
> --
> Date: Sun, 28 Apr 2013 09:10:30 +0300
> From: ccievoic...@gmail.com
> To: ccie_voice@onlinestudylist.com
> Subject: [OSL | CCIE_Voice] MGCP Auto vs Manual configuration
>
>
> Hi Experts
>
> Which method to be use in CCIE Voice lab exam
Hi Experts
Which method to be use in CCIE Voice lab exam in order to configure MGCP
Auto or Manually.
Thanks
___
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Are you a CCNP or CCIE and looking for
HI experts,
When Branch-2 MGCP router is in CME SRST mode ,i am unable to see Alerting
message on inbound call from PSTN,moreover when i call from PSTN my call
is directly connected and there is no ring back tone ,Can any one help to
fix the issue.
ISDN Se0/0/0:15 Q931: RX <- SETUP pd = 8
Hi Guys,
Can i group H323 GW and SLRG in one Route List ? it that the best practice
?can any one provide me with the reference for best practice to which
protocol in can group together like h323, mgcp,sip trunk,GK Trunk ect in
to Route List.
Thanks
___
Hi Guys,
Regarding E1 Framing if the question states the below
FOR E1 use the following:
switch type:net5
framing :CRC
linecode :HDB3
then do i need to use the NO-CRC or CRC ?
___
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ISPLAY IE in outgoing ISDN messages is allowed
>>
>> and
>> *isdn outgoing ie ?
>> *
>>...
>>displayDISPLAY IE in outgoing ISDN messages is allowed
>>...
>>
>> so this may be otherwise option for the same display-ie.
>>
>>
Hi guys,
what is the difference between IE display and display IE command
interface Serial0/1/0:15
no ip address
encapsulation hdlc
isdn switch-type primary-net5
isdn incoming-voice voice
isdn bind-l3 ccm-manager
isdn outgoing *display-ie VS ie dislay*
isdn outgoing ie redirecting-number
Hi Experts,
Can any one share there knowledge and experience on how to troubleshoot
one-way audio when the call is answer from PSTN phone which messages do i
need to look at on RTMT and which traces do i need to enable on CUCM to
check the One way audio problem ..
Thanks
_
Hi experts,
i have been trying to test the Call forward unregister whn the call arrive
on destination phone its show on the display as FORWARD,FOR and BY on the
phone screen so is there a way to display "FROM" on the destination phone
screen
thanks
___
Hi experts,
i am facing a problem with CUCM DHCP ,my phones are taking an ip address as
first IP from starting range and so on actually it has to take last ip
first and so on
eg: CUCM DHCP range :177.1.11.10 -177.1.11.30
MY phones getting an ip as 177.1.11.10
thank
iguration/guide/b_cflash.html#wp23144
>
>
> ** **
>
> ** **
>
> *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
> ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *CISCO CCIE VOICE
> *Sent:* Tuesday, March 12, 2013 10:51 AM
> *To:* ccie_voice@onlin
Hi Experts,
can any one help me I want to create directory folder on router without
formatting flash when i use mkdir command its saying that invalid input
thnks
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HI experts,
If the question ask to configure H323 Trunk to backbone router then in that
do i need to configure it as H323 Gateway or H323 Non -GK controlled trunk?
Thnks
___
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Hi Experts,
Can anyone explain me how to collect h323 trunk debugs from RTMT ...
Thanks
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HI Guys,
Can any one Share H323 and SIP Trunk configuration that need to be done on
PSTN Router in order for it to work properly ...
Thanks
___
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Are you a
HI All
When the Branch-1 is in SRST Proper Mode (call-manager-fallback) When i am
trying t press voice mail button its playing system greeting not the user
greeting ,its there any thing i need do on CUCM in order to play user
greeting when the phones are in SRST..
Thanks
___
>3. Re: Phone NTP Reference Vs NTP Reference (Cory Gray)
>4. Re: Phone NTP Reference Vs NTP Reference (CCIEing)
>5. AAR HELP (CISCO CCIE VOICE)
>6. Re: AAR HELP (Jamie Parr (jamparr))
>7. Re: AAR HELP (CISCO CCIE VOICE)
>
>
> --
*
> Engineer - IT
> jamp...@cisco.com
> Phone: *+44 20 8824 2641*
> Mobile: *+44 7590622049*
>
> ** **
>
> *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
> ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *CISCO CCIE VOICE
> *Sent:* 05 March 2013 08:13
&g
Hi experts,
Can any one help me to understand the below Question
Call from HQ to Branch-2 should invoke AAR base on RSVP bandwidth,If
HQPH1 calls BR2PH1 then it should display E.164 number and Network
congestion Re-routing Message on Phone
So ,my doubt is that do i need to use AAR and l
n error, please contact the sender and delete the material from any
> computer.
>
> __
>
> ** **
>
> From: CISCO CCIE VOICE
> Date: Tuesday, February 26, 2013 2:44 PM
> To: "ccie_voice@onlinestudylist.com&q
Hi,
Is it safe to use BAT in real lab exam?
thanks
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, 2013, at 6:45 AM, "CISCO CCIE VOICE"
> wrote:
>
> Hi
>
> Can any one help me with Dial Plan consideration when calling from HQ Site
> to Branch 1 Site,following what i have configure.But the problem is that on
> B1PH1 screen its showing as below
>
&g
Hi
Can any one help me with Dial Plan consideration when calling from HQ Site
to Branch 1 Site,following what i have configure.But the problem is that on
B1PH1 screen its showing as below
*From +14082021001
Forward by:2001*
*HQ SITE:*
*Extension Range*:1XXX
*Partition:*Branch_1_SRST_PT
*CSS
HI,
Does MWI works on SRST Proper mode (call-manager-fallback).?
Thanks
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HI
When in SRST either SRST Proper or CME as SRST ,when i call from HQ to
branch office do i need to assign unregistered DN in order to have ext to
ext calling working case of WAN fails
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Hi,
I have my personal rack for ccie voice lab practice, if you want we can
share and it will be cheapest, you can contact me on
ccievoicedub...@gmail.com or 00973-33457553
Thanks
Mick
___
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Hi,
I have the required devices to study CCIE LAB v.3, it does not match the
requirements exactly but it is very good. To save my time I need to upload the
initial configuration for each device at the beginning of each lab.
I need to build an automated way to do this rather than do everything
>
>> Best of Luck
>> Ash
>>
>> On Thu, Nov 24, 2011 at 9:16 PM, Ccie Voice wrote:
>> > Hi all,
>> > I added my credit card to pay for Cisco but they did not proceed the
>> > payment. I did not take care about it, because someone told me that
>
Hi all,
I added my credit card to pay for Cisco but they did not proceed the payment. I
did not take care about it, because someone told me that Cisco now proceeding
the payment after the lab, I went for CCIE lab and I did my lab but till now I
did not receive my result and the payment still p
I am not sure also.
but can we use auto qos voip trust?
From: datucha123 datucha123
To: Ccie Voice
Cc: CCIE Study
Sent: Monday, November 21, 2011 1:19 PM
Subject: Re: [OSL | CCIE_Voice] WAN QoS
As I know, we have to calculate the FRF.12 size based on
Hi all,
Could you please help me to solve this:
There is 384 frame-rely PVC between HQ and BR1.
Enable FRF.12 on this
circuit , 10 ms as sampling rate.
16k for signaling traffic and 4 g.729 calls.
Assume all RTP traffic is marked with EF and signaling with CS3
Header Compression should be enabl
added it under
frame-relay interface-dlci 202
let me know if it is OK?
From: Raees Shaikh
To: "ccie_voice@onlinestudylist.com"
Sent: Sunday, November 20, 2011 9:18 AM
Subject: [OSL | CCIE_Voice] auto qos command does not show up
From: Raees Shaik
thank you for your reply,
but I did not understand exactly what I should configure?
Could you please explain to me how huntstop channel can solve this?
and what I should use huntstop cahnnel 1, 2 .. ?
Regards,
From: Ashraf Ayyash
To: Ccie Voice
Cc: CCIE
: Ccie Voice
Cc: CCIE Study
Sent: Sunday, November 20, 2011 1:20 AM
Subject: Re: [OSL | CCIE_Voice] SRST Advanced Config
you can use huntstop channel command under the call manager fall back
which will limit both in and out number of the calls on the dual/octo
lines
Ash
On Sat, Nov 19, 2011 at 9
Hi all,
is it possible to configure:
1- Maximum Number of Calls
2- Busy Trigger
In SRST Call-Manger-fallback NOT CME SRST?
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Are you a CCNP or CCI
Hi everybody,
I am going to do my 3rd attempt very soon, and I do not need to search about
study partners but I need something else.
Anyone he attempted, already knows that there is some tricks and it is not easy
to pass without them. Or after receiving the report from Cisco - you failed-
you
that I am sending the exact digits and
I am hitting the exact dial-peer but no call routing.
and I checked the disconnect cause, it was 1 = unassigned number.
I will try to rebuild the lab soon and check again :)
From: Chris Martin
To: Ccie Voice
Cc: CCIE Study
Thank you but I do not think I am correct :)
From: Robert Schuknecht
To: 'Ccie Voice' ; 'CCIE Study'
Sent: Friday, October 21, 2011 4:50 PM
Subject: AW: [OSL | CCIE_Voice] Voice Gateway and Signaling
I think you are correct.
/Robert
Shaikh
To: Ccie Voice ; CCIE Study
Sent: Friday, October 21, 2011 6:21 PM
Subject: Re: [OSL | CCIE_Voice] Call Routing
Hi
What digits are going when you dont add forward digits 7?
Regards,
Raees
From: Ccie Voice
To: CCIE Study
Sent: Friday, October 21, 2011
Hi all,
I have very strange problem, and I need someone to help me to understand why?
I am trying to study call routing, local calls
I have the following setup
SCCP Phone >> RP >> Local RL>> H.323 GW>> PSTN
in the GW I added the following dial-peer:
dial-peer voice 15 pots
translation-profi
Hi everybody,
If a question asked you to configure the voice gateways as the following:
use T1 Framing: ESF, line coding :B8ZS
isdn sw: primary-ni
take clocking for layer 1 from network side
your pri circuit layer 2 should be user side
for me the answer will be the following configuration
But I am using router flash memory Not CUCM___
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Hi everybody,
I am trying to configure Multicast MOH in Branch 1 router, I added the file to
router flash and I configure CUCM to use multicast.
when I am calling from one phone to another in BR1 I am able to hear the file
that I uploaded to router flash memory. but if I tried to use the followi
I did this before, you can use
debug cch323 h225
debug cch323 h245
after Ringing you will find the party under waiting h245 capabilities, I
recommend to read SRND it has some good information about this.
Regards,
From: DeShon Crayton
To: 'Nowork_onlyfun' ; cc
but need to configure
each router in the path & register each as a MTP in CUCM
On Wed, Sep 28, 2011 at 10:30 AM, Ccie Voice wrote:
Hi everybody,
>
>
>When I have to use CAC location based and when to use CAC RSVP?
>- Multi-link, are there any other reasons or scenarios?
>
Hi everybody,
When I have to use CAC location based and when to use CAC
RSVP?
- Multi-link, are there any other reasons or scenarios?
What is the meaning for the following?
"When using locations in hub-and-spoke topology,
devices at the hub site should assigned to the Location"
Hi all,
I need to add a router for my home lab and I need your advise about which one I
should choose.
I know that the best one is 2801 but it is very expensive if I need to compare
with other old routers.
what I know, I can use one of the following routers:
3725
2620MX
1760
3640 ( I thin
Hi Vega,
1. What can we bring in to the lab? Water bottle? No thing :)
2. How long is the lunch break? Is it allocated at a specific time? it is 30
mins the proctor should tell you about it.
3. Do we have to do the patching of cables? No
4. What do they provide apart from the Cisco PDF? Any wri
Hi All,
I did my first attempt, I did not pass but the report was very good and most of
my marks 100%. The report encourages me to do my second attempt. I prepared for
the second attempt and I did my second attempt.
Second attempt was much better than the first one. I studied very well and
tr
Dear Friends,
I have created a group on Skype where we can communicate online with each
other to discuss our matters on urgent basis, you guys can add (
ccie.voice.group ) where you can be added to the group. Just copy paste this
request while adding "ccie.voice.group" ( I am a Cisco Voice User, a
Dear Friends,
I have created a group on Skype where we can communicate online with each
other to discuss our matters on urgent basis, you guys can add (
ccie.voice.group ) where you can be added to the group. Just copy paste this
request while adding "ccie.voice.group" ( I am a Cisco Voice User, a
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