Congratulations
De: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] En nombre de adam compton
Enviado el: jueves, 30 de junio de 2011 18:52
Para: gogli...@gmail.com; ccie_voice@onlinestudylist.com
Asunto: Re: [OSL | CCIE_Voice] Congrats
George,
You
Hi Randall,
In your cme you need this
Ip http server
Ip http path flash: (path for GUI files for CME)
Telephony-service
web admin system name administrator password cisco
De: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] En nombre de Randall Crumm
Hi,
If change the mwi mechanism in CUE to Subscribe-Notify
Then CME
sip-ua
mwi-server ipv4:142.102.66.253 expires 3600 port 5060 transport udp
you will need mwi sip command
De: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] En nombre de Greg
Enviado
Hi all,
If in the exam in Qos WAN exercise were asked to configure MLF or FR12 between
HQ and SB, supposing there is only one physical interface in HQ, the traffic
shaping applied will affect the BR2 subintertace appliying a default mincir
28000 which is a problem for more than one call
If BR1 works as teho GW for HQ calls the inbound voip dial-peer should select
g729.
For br1 phone local calls - g711
De: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] En nombre de Greg
Enviado el: miércoles, 15 de junio de 2011 1:15
Para:
Hi all
In CME If nothing is specified regarding maximum number of calls or busy
trigger. What do you configure for primary extensions of SITEC, dual or octo?
Thanks!
___
For more information regarding industry leading CCIE Lab training, please visit
Hi Greg,
Busy-trigger-per-button only applies for octo-lines. Is it the case?
Regards,
Francesc
De: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] En nombre de Greg
Enviado el: domingo, 12 de junio de 2011 21:17
Para: ccie_voice@onlinestudylist.com
Hi Chris,
I have read several times but I haven’t been able to understand the scenario.
Could you clarify it a little bit?
Thanks!
De: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] En nombre de Chris Green
Enviado el: sábado, 11 de junio de 2011 21:18
Hi Michael,
From my point of view
1 ok.
2 Why are they needed? isdn send-alerting and isdn sending-complete. I
haven't never configured them. I would configure
configuration MGCP HQ
interface Serial0/0/0:23
no ip address
encapsulation hdlc
isdn switch-type primary-ni
isdn
Congratulations Kobel!
De: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] En nombre de Miron Kobelski
Enviado el: miércoles, 08 de junio de 2011 7:07
Para: ccie_voice@onlinestudylist.com
Asunto: [OSL | CCIE_Voice] #29164
I received my number yesterday
Hi,
For US you need configure
HQ
Ntp server xx.xx.xx.xx
Clock timezone pst -8
Clock summertime pst recurring
For Honkong
Ntp server xx.xx.xx.xx
Clock timezone hkg 8
Do no use recurring, China does not have savelight
Verify the time the phones get checking the real time and the timezones
Hi,
Working with srst once mode dn or all in site configured as H323 GW for
Callamanger once the ephones register in SRST mode we have
BRANCH1#sh dial-peer voice summary
ADPRE PASSOUT
TAGTYPE MIN OPER PREFIXDEST-PATTERN
Hi all,
As the current exam has more troubleshooting requirement I was reviewing Cisco
IOS Voice Troubleshooting and Monitoring Guide, Release 12.4T
There is a very feature Cisco VoIP Internal Error Codes that gives great
information about why a gateway or gatekeeper release or refuse a call
: sábado, 07 de mayo de 2011 13:25
Para: Roig Borrell, Francesc Xavier
CC: ccie_voice@onlinestudylist.com
Asunto: Re: [OSL | CCIE_Voice] More troubleshooting in the exam - How to use
Internal Error Codes with CLI
Hi,
Internal Error Codes provide very useful troubleshooting tool for IOS gateways.
IEC
troubleshooting.
You can then use those for your purposes.
I believe realtime is only supported with syslog
On Sat, May 7, 2011 at 5:36 AM, Roig Borrell, Francesc Xavier
francesc.ro...@tecnocom.esmailto:francesc.ro...@tecnocom.es wrote:
Hi all,
As the current exam has more troubleshooting
[mailto:linuxbos...@gmail.com]
Enviado el: sábado, 23 de abril de 2011 2:09
Para: Roig Borrell, Francesc Xavier; 'Peter Farkas';
ccie_voice@onlinestudylist.com
Asunto: RE: [OSL | CCIE_Voice] CME busy-trigger-button Problems
Roig,
This will do , lets assume you made 5 calls to the dn.
ephone-dn 10 octo
Hi Peter,
You are right. Testing it the busy trigger applies in the shared line
So how could we achieve this?
shared line
-1) Maximum 5 incoming calls into the DN
-2) 1st phone should not receive not more than 4 incoming call and 2nd phone
Should not receive more than two
I have seen
nombre de Peter Farkas
Enviado el: jueves, 21 de abril de 2011 22:19
Para: Roig Borrell, Francesc Xavier; ccie_voice@onlinestudylist.com
Asunto: Re: [OSL | CCIE_Voice] CME busy-trigger-button Problems
In CME busy-trigger is the same common value for the shared line. Let it be
only one
Hi all!!
I am having a behavior that It does not make any sense with this easy scenario
ephone-dn 11 octo-line
number 4500
ephone 1
mac-address 0024.97AA.1B49
busy-trigger-per-button 1
type 7945
button 1:11
ephone 1
mac-address 0024.14B3.765E
busy-trigger-per-button 1
type 7965
Hi all,
Working with MVA with hairpinin I have found and issue that I don't know how to
workaround it
Hqph2 2002
Remotedestination, 6178632683
When I call to hqph1 from remote the identification is OK, 2002
The problem appears with DISA service. I does not recognize the remote
destination
Yes! It was right under my nose…. And It was a very easy workaround ☺
Thank you very Claude!
Regards,
Francesc
De: Friderich Claude [mailto:cfrider...@netcore.lu]
Enviado el: lunes, 28 de febrero de 2011 21:45
Para: Roig Borrell, Francesc Xavier; ccie_voice@onlinestudylist.com
Asunto: RE: [OSL
Hi all!
I'm Working with Vol2 lab2 5.2 and I have found a problem that I do not
understand. When SIP hqph2 calls using the BR1 mgcp gateway the PSTN rings one
and the call fails. Looking at CCM traces a I can find the error:
SIP/2.0 503 Service Unavailable
Reason: Q.850;cause=47
Date: Mon, 14
-interface Loopback0
mgcp bind media source-interface Loopback0
mgcp behavior g729-variants static-pt
!
Thanks you very much!
Francesc
De: Miron Kobelski [mailto:findko...@gmail.com]
Enviado el: lunes, 14 de febrero de 2011 22:05
Para: Roig Borrell, Francesc Xavier; ccie_voice
You don't need 3 ntp refrences in CCM, in fact a you do not need to configure
any ntp reference unless you have SIP phones. In this case one NTP reference
pointing to HQ (ntp server)
For SCCP phone you need to synchronize the PUB with HQ router, different date
time groups will define the offset
Hi Daniel,
This is not a limitations of your phones. The reason why B or C can't add
parties is because this is an ad-hoc conference started by A, so it is the only
phone who can add parties
Regards,
Francesc
-Mensaje original-
De: ccie_voice-boun...@onlinestudylist.com
Hi all,
I am trying synchronize CCX with NTP. In SystemCisco Unified Callmanger, NTP
(PUB 10.10.210.10).
I think that the expected behavior should be that CCX gets the UTC from NTP and
the timezone in W2003 acts as the offset.
However, if I run a simple script with
Set time=T[Now],
In order
is the reason for NTP server entry
in the UCCX menu: System Cisco Unified CM Configuration NTP?
Francesc
De: Shrini [mailto:linuxbos...@gmail.com]
Enviado el: miércoles, 02 de febrero de 2011 21:29
Para: Roig Borrell, Francesc Xavier
CC: ccie_voice@onlinestudylist.com
Asunto: Re: [OSL | CCIE_Voice] CCX
Congratulations!
Francesc
De: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] En nombre de akash patel
Enviado el: jueves, 20 de enero de 2011 18:46
Para: ccie_voice@onlinestudylist.com
Asunto: [OSL | CCIE_Voice] I passed my Voice CCIE
I took my exam
Hi all,
Testing em with cme I have found a problem with the Clear Call History.
By default is enabled, and optionally can be disallowed
Telephone service
Em keep-history
So without em keep-history the behavior that should be expected would be
After logout the call history was empty but it
Hi,
Thanks for the inputs. I will test with 14.4.22T and see how it goes
Regards,
Francesc
De: Rogers Ochieng [mailto:rogersochi...@gmail.com]
Enviado el: sábado, 15 de enero de 2011 19:07
Para: Roig Borrell, Francesc Xavier
CC: ccie_voice@onlinestudylist.com
Asunto: Re: [OSL | CCIE_Voice] Auto
Hi all,
I am experiencing a very strange issue applying (Auto qos voip fr-atm/ Auto qos
voip trust fr-atm) command in HR-RTR. Sometimes it does not create the virtual
template and creates all the config as auto qos voip / auto qos voip trust was
executed. I haven't been able to find a
Hi Prashant,
Yes I always apply the bandwidth command before configuring autoqos .
Thanks!!
Francesc
De: Prashant Patel [mailto:prashantpatel...@gmail.com]
Enviado el: sábado, 15 de enero de 2011 17:14
Para: Roig Borrell, Francesc Xavier
CC: ccie_voice@onlinestudylist.com
Asunto: Re: [OSL
Hi Justin,
After set up the clock timezone you have to execute create cnf under telephony
service. Then reset the phone
Regards,
Francesc
De: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] En nombre de Justin Brady
Enviado el: sábado, 15 de enero de
Hi Satoshi,
Sometimes the PGs uses L2 header 9, 11 or 13. It depends the Cisco
documentation used. This is the problem...
Here you have a link with a very interesting discussion of this issue and Mark
Snow's conclusions
http://www.mail-archive.com/ccie_voice@onlinestudylist.com/msg06644.html
: Miron Kobelski [mailto:findko...@gmail.com]
Enviado el: martes, 11 de enero de 2011 21:27
Para: Roig Borrell, Francesc Xavier
CC: ccie_voice@onlinestudylist.com
Asunto: Re: [OSL | CCIE_Voice] Inter- Presence CCM-CME
Hi,
on thing you might be missing is Presence Group configuration on IP phone
Hi Joli,
From my experience after complete all the JTAPI integration I have the same
problem and I always need a extra CUE reload in order to register JTAPI
subsystem to Callmanager
Regards,
Francesc
De: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com]
Hi All,
I am trying to configure blfs between CME and CCM. I have not been able to find
good documentation for this integration.
CME offers a way to configure subsribe prsence to external phones using the
allow subscribe all command and a server ip address under presence. The
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Roig Borrell,
Francesc Xavier
Sent: Tuesday, January 11, 2011 7:11 AM
To: CCIE_Voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Inter- Presence CCM-CME
Hi All,
I am trying to configure blfs between CME and CCM. I have not been
Hi Shrini,
This is a normal behavior for CUCME.
Regards,
Francesc
De: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] En nombre de Shrini
Enviado el: lunes, 10 de enero de 2011 7:12
Para: ccie_voice@onlinestudylist.com
Asunto: [OSL | CCIE_Voice] + display
Hi all,
I am going crazing with Phone View configuration with Unity Connection. I have
configured following
http://blog.ipexpert.com/2010/11/17/setting-up-phone-view/
I have reviewed the autentication URL
(http://10.10.210.10:8080/ccmcip/authenticate.jsp), I think that UC needs to
Hi Shingei,
Thanks for the input but I am using 7965s.
De: ShinGei Yong [mailto:shingei.y...@gmail.com]
Enviado el: lunes, 10 de enero de 2011 14:50
Para: Roig Borrell, Francesc Xavier; ccie_voice@onlinestudylist.com
Asunto: Re: [OSL | CCIE_Voice] Phone View UC
Hi Francesc,
What is the phone
Hi everyone!
I am trying to understand the right way to calculate the priority value in LLQ
with a RSVP configuration.
I have not been able to find documentation clarifying this.
So supposing HQ-BR1 4 calls g729
ip rsvp bandwitdh = 24*3 + 40 = 112
No problem with the rsvp bandwith, 3 calls
a= Worst case = (L3+PL)*8*100 = (40+10)*8*100 = 40k
b= One G729 = ((L2+L3)+PL)*50*8 = ((40)+20)*8*50 = 24 L2+L3 =
IP+RTP+UDP = 40
De: Shrini [mailto:linuxbos...@gmail.com]
Enviado el: miércoles, 05 de enero de 2011 18:33
Para: Roig Borrell, Francesc Xavier
CC: ccie_voice
Hi Jeff,
Great! Then we agree with the solution for this requirement. :)
Thank you very much!!
De: givemeccievoice2...@gmail.com [mailto:givemeccievoice2...@gmail.com]
Enviado el: miércoles, 05 de enero de 2011 20:53
Para: Roig Borrell, Francesc Xavier; 'Shrini'
CC: ccie_voice
...@gmail.com [mailto:givemeccievoice2...@gmail.com]
Enviado el: miércoles, 05 de enero de 2011 22:42
Para: 'Miron Kobelski'
CC: Roig Borrell, Francesc Xavier; 'Shrini'; ccie_voice@onlinestudylist.com
Asunto: RE: [OSL | CCIE_Voice] RSVP LLQ priority value calculation
After I just agreed with you
Hi all!
I have been testing with Barge in Callmanager. The features and services guide
says:
The original call requires G.711 codec. If G.711 is not available, use cBarge
instead
However, with my tests I am able to barge a g722 established call (hqph1-hqph2)
from br1ph1 using ilbc or g729.
Hi Divin,
Great! I have tested, it works
Thanks!
De: Divin Mathew John [mailto:divinj...@gmail.com]
Enviado el: sábado, 13 de noviembre de 2010 10:38
Para: Roig Borrell, Francesc Xavier
CC: givemeccievoice2...@gmail.com; Pavan; ccie_voice@onlinestudylist.com
Asunto: Re: [OSL | CCIE_Voice] BLF
Hi All!
I have my own lab with the same topology of vRacks (all the infrastructure at
the office an VPN connection home to office with all the phones). Could anyone
tell me how do you manage to test QoS in this way? Or are there local phones in
the vRacks and you can control them in order to
[mailto:pav.c...@gmail.com]
Enviado el: jueves, 11 de noviembre de 2010 0:30
Para: Roig Borrell, Francesc Xavier
CC: Prashant Patel; ccie_voice@onlinestudylist.com
Asunto: Re: [OSL | CCIE_Voice] BLF call list - Presence group issue
To rule out css, can you put the problem phones in the null partition
Hi All!!
Working with lab 13 and blf call list I've found and issue that is driving me
crazy!!
I have two Presence groups
Standard Presence Group , Employees Presence group.
The group Relationship: Use System Default (Dissallow Subscriptions)
hqph1, hqpph2 (Employees)
Xavier
CC: ccie_voice@onlinestudylist.com
Asunto: Re: [OSL | CCIE_Voice] BLF call list - Presence group issue
Hi Francesc,
On the phones check the subscribe css. It should be the one with the phone
partitions in it.
HTH
Prashant
On Wed, Nov 10, 2010 at 5:43 PM, Roig Borrell, Francesc Xavier
Hi All!
I am trying to understand strict-match option in ip source command. But,
there's something doesn't make sense for me.
Vik explains in the lectures that if ip option 150 doesn't match source
address with strict-option the phone will not register. But in test scenario
the phone
-nte RFC 2833
sip-notify sip-notify
sub-notify subscribe notify
http://www.mail-archive.com/ccie_voice@onlinestudylist.com/msg16088.html
Thanks. Regards,
XAvi
De: Miron Kobelski [mailto:findko...@gmail.com]
Enviado el: sábado, 16 de octubre de 2010 10:03
Para: Roig Borrell
Xavier
CC: ccie_voice@onlinestudylist.com
Asunto: Re: [OSL | CCIE_Voice] Dtmf-Relay Negotiation
Roig,
On Sat, Oct 16, 2010 at 11:14, Roig Borrell, Francesc Xavier
francesc.ro...@tecnocom.esmailto:francesc.ro...@tecnocom.es wrote:
Hi,
Yes, I know H.245 alphanumeric is not supported with SIP
Hi All,
I am doing some tests with dtmf-relay and I have found an scenario I do not
understand:
Call SCCPphone CME to CUE
dial-peer voice 4600 voip
destination-pattern 4[126]00
session protocol sipv2
session target ipv4:10.10.202.2
incoming called-number 409[89]
dtmf-relay
Hi all,
I have this scenario
Hardware CFB configured in HQ
Ad-hoc conferene started by hq phone1 with br1 phone1 and br1 phone2. No
problem with this
HQ#SH sccp connections
sess_idconn_idstype mode codec ripaddr rport sport
33556436 33554594 conf sendrecv g729b
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