Hi,
I am working on Lab 5A task 5.8 - HQ phone calling PSTN , failover from HQ
to BR1 MGCP gateway.
After shutting the HQ voice port, calls cannot fail over to the BR1-MGCP
gateway - 'your call cannot be completed...'
As per the Proctor guide solution ,here is the route plan:
Yes, I agree. Don't focus too much on spinning heads, focus on learning the
material and becoming a good engineer.
On Fri, Jul 10, 2009 at 12:45 PM, Michael Ciarfello
mciarfe...@iplogic.comwrote:
Maybe the heads spin didn’t mean you will come out dizzy and
disoriented. Maybe heads spin
Hi,
From workbook question 9.2 . BACD is doesn't work when called from any BR2(
HQ,BR1) IP phone. The relevent configuration is
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
dial-peer voice
The au file is already in G711 format (normally). To confirm, debug
ephone moh right BEFORE putting the moh music-on-hold.au cmd under
call-man-fallback section - this will produce output that will tell
you the codec that the moh file was created as.
HTH,
--
Mark Snow
CCIE #14073
I'm thinking you posted to the wrong list :) I believe regions determine
codec the same for SIP as it does SCCP.
On Sun, Jul 12, 2009 at 11:32 AM, Mike Thompson mthompson...@gmail.comwrote:
I’m going back over some of the earlier labs again and I think I’m just
overanalyzing something.
Only document I found on this is here:
http://www.ciscotaccc.com/kaidara-advisor/voice/showcase?case=K20653428
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Michael Ciarfello
Sent: Friday, July 10,
Regarding these two Catalyst commands:
auto qos voip cisco-phone
auto qos voip cisco-softphone
Does the Catalyst use CDP to detect softphones (IP Communicator or Personal
Communicator) or is CDP only used on hardphones? Second, how should a
switchport be configured if you need autoqos for a
Okay this one really has me stumped. I am about half way through lab 5 and
any time I make an LD or international call form HQ Phone 2, I get a reorder
tone after 3 digits. When you run the DNA though, it shows the call being
routed, and routed properly. Anyone have any ideas?
Hello,
anyone from this list attended Networkers CCIE Tech sessions?. Is
troubleshooting going to be towards the end of the lab session ?
Or will this be just part of the whole day ?.
Thanks,Padhu
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In module 3, I'm setting up a dialplan for my 7960 SIP phone in BR2 but
despite configuring one, creating the profile, and resetting the number only
gets sent out after ITT. A couple of curiosities;
1) On BR2 under *voice register dialplan 1* I need to type *type
7940-7960-others* before I can
Did you put Gatekeeper trunk in a G729-only region?
You only configured CAC mechanism until now.
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Scott ODonnell
Sent: Wednesday, July 08, 2009 11:04 PM
To:
test
If I can be of further assistance, please let me know.
--
Drew LePla Comp TIA A+, CCNA
Technical Support Engineer - IPexpert, Inc.
Telephone: +1.810.326.1444 x204
Fax: +1.810.454.0130
Mailto: dle...@ipexpert.com
--
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Any idea about this issue guys?
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Aamir
Panjwani
Sent: Saturday, 11 July 2009 12:17 AM
To: OSL Group
Subject: [OSL | CCIE_Voice] Voicemail and calling party transformation
When I
And that's one of the reasons you set the H-323 setup timer to 3 seconds.
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Michael
Ciarfello
Sent: Friday, July 10, 2009 3:48 PM
To: Nara Shikamaru
Cc: OSL Group; GRAFL Philipp
Has there been any confirmation of what documentation is available during the
lab? I've heard mixed comments about it only being the config guides and
SRNDs. But, then others have mentioned that the technotes and configuration
examples documents are available as well. I'm taking it in San
I have not received any new posts to the list since 7/10...wondering if the
list is down or something on my account?
MDD
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After logging into BR2 Ph3, the is not showing for any line on forst button but
showing speed dial on second. There is no dial tone on the phone and calls
can't be completed.
I am not able to find how to troubleshoot. Any thoughts...
Vineet
Hello,
I am trying to install Cisco UC in VMWare for my home/work lab but am unable
to log into the application interface after the system finishes installing.
I've been trying different install scenarios for over a week now with little
success. I've tried on ESXi 3.5 update 4 I've tried in
Also check the route plan and route plan report, for the DN you are trying
to add for the CTI route point.
On Fri, Jul 10, 2009 at 11:55 AM, Cristobal Priego
cristobalpri...@gmail.com wrote:
Experts,
When I go to Application Management and if I try to add a trigger to my
application .
I
Pls reply if you get this.
Cheers,
--
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:
i am so sick and tired with this... the main problem for which i thot vmware
had a solution is still stinking me... how do i do a proper backup and
restore?? i mena i have pub and sub... my pub had a db connectivity
error(due to an abrupt power failure)...sub was fine... i had already cloned
all
hi,
i am testing the Lab 4A, calling to 3...@ipxcme,.com , and called failed ,
find from debug
SIP/2.0 405 Method Not Allowed
any suggestion, what is wrong ?
tks
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I’m going back over some of the earlier labs again and I think I’m just
overanalyzing something.
The question states to use Codec G711u for all calls. It’s under the SIP
Endpoint section and I’m weak on SIP. Is there something in SIP that needs
to be done or is this simply a statement to
test
If I can be of further assistance, please let me know.
--
Drew LePla Comp TIA A+, CCNA
Technical Support Engineer - IPexpert, Inc.
Telephone: +1.810.326.1444 x204
Fax: +1.810.454.0130
Mailto: dle...@ipexpert.com
--
Join our free online support and peer group
Hi,
I am working on Lab 5A task 5.8 - HQ phone calling PSTN , failover from HQ
to BR1 MGCP gateway.
After shutting the HQ voice port, calls cannot fail over to the BR1-MGCP
gateway - 'your call cannot be completed...'
As per the Proctor guide solution ,here is the route plan:
Hullo
Best regards
Mick
E: m...@pobox.net.uk
From: Mark Snow ms...@ipexpert.com
Date: Mon, 13 Jul 2009 11:02:13 -0400
To: CCIE Voice Maillist ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Test to see if mail server is back up properly
...
Pls reply if you get this.
Cheers,
Roger.
On Mon, Jul 13, 2009 at 8:02 AM, Mark Snow ms...@ipexpert.com wrote:
Pls reply if you get this.
Cheers,
--
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
I got it
2009/7/13 Mark Snow ms...@ipexpert.com
Pls reply if you get this.
Cheers,
--
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
I just got a 7961 sip phone
installed on cme 7.1
it keeps failing to upgrade
config etc
BR2#sh run | b voice reg
voice register global
mode cme
source-address 10.11.203.1 port 5060
max-dn 14
max-pool 3
load 7961 SIP41.8-4-2S
load 7960-7940 P0S3-08-6-00
time-format 24
date-format
BlankFirst my apologies, as this is not the right avenue for this
I am currently looking for two strong Resources, with a strong exposure to IPCC
Contact Center environment with
High Availability Reliability. This position is available in the North East
(New Jersey/New York City area).
i triedn changing DN number.. but in one of the phones(cipc) its not
changing... i then deleted the entry from pub..and still the phone does not
unregister or even change DN.. i checked the device-phone and there is no
phone with that mac...but still the phone is there perfectly registered...
and
Looks good. Wondering where everyone went. I even put on deodorant today.
From: ccie_voice-boun...@onlinestudylist.com
[ccie_voice-boun...@onlinestudylist.com] On Behalf Of Mark Snow
[ms...@ipexpert.com]
Sent: Monday, July 13, 2009 11:02 AM
To: CCIE Voice
Hii All,
Please anyone help me to find out the root cause?
I cannot accessing the web interface of MCU 3315.
I've already configure the ip add for both interface.
I can ping one ip address and the other is cannot.
and i cannot accessing the web interface
http://10.68.10.115/board/admin
Please
As stated is proctor lab guide 5c, task 5.4, pdf page 204 and what
actually happens for ucm 7.0(1), is that the calling party
transformation pattern matching is based on the calling number at the
point the route pattern match takes place,
However, the ucm's srnd states that calling party
Hello.
Thanks and appreciated to yhis wonderful group.
Thks
See the Web#39;s breaking stories, chosen by people like you. Check out
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For more information regarding industry leading CCIE Lab training,
Not sure if this hit the mail server so I'll try again
Regarding these two Catalyst commands:
auto qos voip cisco-phone
auto qos voip cisco-softphone
Does the switch use CDP to detect softphones (IP Communicator or Personal
Communicator) or is CDP only used on hardphones? Second, how
OSL Voice, RS, Security, and SP constituents,
As many of you now know, some changes were made Friday night (without
being tested) that caused mail service to be interrupted over the
weekend.
Those responsible (not me - I just got the privilege of having to FIX
it :) have been drug into a
Testing OSL
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www.ipexpert.com
HI All,
I've got line group TestHuntGroup with distribution broadcast with member
1001 and 1002
I've got hunt list TestHuntList with the selected group is TestHuntGroup
I've got the hunt pilot 1000 with Hunt list TestHuntList
Whenever ext 1005 dial 1000. so ip phone ext 1001 and 1002 both
got it !
2009/7/14 ccie_voice-requ...@onlinestudylist.com
Send CCIE_Voice mailing list submissions to
ccie_voice@onlinestudylist.com
To subscribe or unsubscribe via the World Wide Web, visit
http://onlinestudylist.com/mailman/listinfo/ccie_voice
or, via email, send a message
got it
From: ccie_voice-requ...@onlinestudylist.com
ccie_voice-requ...@onlinestudylist.com
To: ccie_voice@onlinestudylist.com
Sent: Tuesday, July 14, 2009 11:19:16 PM
Subject: CCIE_Voice Digest, Vol 41, Issue 67
Send CCIE_Voice mailing list submissions to
Got it.
Thanks
Larry Hadrava
CCIE #12203 CCNP
Sr. Support Engineer – IPexpert, Inc.
URL: http://www.IPexpert.com
On Tue, Jul 14, 2009 at 11:04 AM, Mark Snow ms...@ipexpert.com wrote:
OSL Voice, RS, Security, and SP constituents,
As many of you now know, some changes were made Friday night
Michael -
Deodorant and it's not even Saturday:-)
Thanks
Larry Hadrava
CCIE #12203 CCNP
Sr. Support Engineer – IPexpert, Inc.
URL: http://www.IPexpert.com
On Mon, Jul 13, 2009 at 9:20 PM, Michael Ciarfello
mciarfe...@iplogic.comwrote:
Looks good. Wondering where everyone went. I even put on
Good to go
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Mark Snow
Sent: Tuesday, July 14, 2009 11:04 AM
To: CCIE Voice Maillist; ccie...@onlinestudylist.com;
ccie...@onlinestudylist.com; CCIE Security Maillist
Subject: [OSL |
Got it.
Sent via BlackBerry from T-Mobile
-Original Message-
From: Mark Snow ms...@ipexpert.com
Date: Mon, 13 Jul 2009 14:58:11
To: CCIE Voice Maillistccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Test to see if mail server is back up properly
...
Got it.
Sent via BlackBerry from T-Mobile
-Original Message-
From: Mark Snow ms...@ipexpert.com
Date: Tue, 14 Jul 2009 11:04:08
To: CCIE Voice Maillistccie_voice@onlinestudylist.com;
ccie...@onlinestudylist.com; ccie...@onlinestudylist.com; CCIE Security
Got it.
Thx,
Jarrod
On Jul 14, 2009, at 8:04 AM, Mark Snow ms...@ipexpert.com wrote:
OSL Voice, RS, Security, and SP constituents,
As many of you now know, some changes were made Friday night
(without being tested) that caused mail service to be interrupted
over the weekend.
Those
Jose:
Congratulations on the achievement
Thanks
Larry Hadrava
CCIE #12203 CCNP
Sr. Support Engineer – IPexpert, Inc.
URL: http://www.IPexpert.com
On Tue, Jul 14, 2009 at 12:04 AM, Jose Gregorio Linero (jlinero)
jlin...@cisco.com wrote:
Hi all,
Finally I passed the lab exam, thanks
After logging into BR2 Ph3, the is not showing for any line on forst button but
showing speed dial on second. There is no dial tone on the phone and calls
can't be completed.
I am not able to find how to troubleshoot. Any thoughts...
Vineet
Got it.
Best regards,
--
Neil Apolzan
Senior Training Advisor
IPexpert, Inc.
--
Mobile: +1.810.305.4848
Skype: Neil.Apolzan
IPexpert Blog: www.IPexpert.ccieblog.com
--
Email sent via mobile phone...
On Jul 14, 2009, at 11:04 AM, Mark Snow ms...@ipexpert.com wrote:
OSL Voice, RS, Security,
I was not paying attention. I di not add all the tftp-server commands for the
7961 sip code.
All is well now
Respectfully
Charles George
From: cisco...@hotmail.com
To: ccie_voice@onlinestudylist.com; ccie_voice-boun...@onlinestudylist.com
Date: Mon, 13 Jul 2009 22:23:37 +
Subject: [OSL
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