[OSL | CCIE_Voice] LAB 5A - failover

2009-07-14 Thread Voice Guy
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:

9.1[2-9]xx[2-9]xx  - pt-hq-ld  :

rl-hq-br1-ld :

rg-hq -- hq-sip,rg-br1 mgcp
-

91[2-9]xx[2-9]xx - pt-xlate-internal

Translation Pattern  - rl-local-gw --- standard local route group

It appears that it is failing over to the MGCP gateway as intended but it is
not working.

No dial peers on MGCP.



Any help appreciated.

Thanks.

Here are the 2 debugs, DNA and then a trace file.

?xml version=1.0 standalone=yes ?
 - # DigitAnalysisResult
 - # ResultsSummary
   CallingPartyInfo CallingParty=*15002* Partition=** DeviceCSS=**LineCSS
=*CSS-XLATE-INTERNAL-LOC-LD* AARGroup=** AARCSS=** /
   DialedDigits916178632683/DialedDigits
   MatchResultRouteThisPattern/MatchResult
   MatchedPatternInfo Pattern=*9.1[2-9]XX[2-9]XX* Partition=*
pt-hq-ld* TimeSchedule=** /
   CalledPartyNumber916178632683/CalledPartyNumber
   TimeZoneGreenwich Standard Time/TimeZone
   EndDeviceRL-HQ-BR1-LD/EndDevice
   DeviceLocationOffNet/DeviceLocation
   InterDigitTimeoutNO/InterDigitTimeout
   AllowDeviceOverrideDisabled/AllowDeviceOverride
   OutsideDialToneNO/OutsideDialTone
  /ResultsSummary
 - # CallFlow
 - # TranslationPattern Pattern=*91[2-9]XX[2-9]XX*
   PositionalMatchList9:16178632683/PositionalMatchList
   CallingPartyNumber15002/CallingPartyNumber
   PreTransformCallingPartyNumber5002/PreTransformCallingPartyNumber
   PreTransformCalledPartyNumber916178632683/
PreTransformCalledPartyNumber
 - # CallingPartyTransformations
   ExternalPhoneNumberMaskYES/ExternalPhoneNumberMask
   CallingPartyMask /
   Prefix1/Prefix
   CallingLineIdPresentationDefault/CallingLineIdPresentation
   CallingNamePresentationDefault/CallingNamePresentation
   CallingPartyNumber15002/CallingPartyNumber
  /CallingPartyTransformations
 - # ConnectedPartyTransformations
   ConnectedLineIdPresentationDefault/ConnectedLineIdPresentation
   ConnectedNamePresentationDefault/ConnectedNamePresentation
  /ConnectedPartyTransformations
 - # CalledPartyTransformations
   CalledPartyMask /
   DigitDiscardingInstructionNone/DigitDiscardingInstruction
   Prefix /
   CalledNumber916178632683/CalledNumber
  /CalledPartyTransformations
  /TranslationPattern
 - # RoutePattern Pattern=*9.1[2-9]XX[2-9]XX*
   PositionalMatchList9:16178632683/PositionalMatchList
   DialPlan /
 - # RouteFilter
   FilterName /
   FilterClause /
  /RouteFilter
   RequireForcedAuthorizationCodeNo/RequireForcedAuthorizationCode
   AuthorizationLevel0/AuthorizationLevel
   RequireClientMatterCodeNo/RequireClientMatterCode
   networkLocation /
   PreTransformCallingPartyNumber15002/PreTransformCallingPartyNumber
   PreTransformCalledPartyNumber916178632683/
PreTransformCalledPartyNumber
 - # CallingPartyTransformations
   ExternalPhoneNumberMaskNO/ExternalPhoneNumberMask
   CallingPartyMask /
   Prefix /
   CallingLineIdPresentationDefault/CallingLineIdPresentation
   CallingNamePresentationDefault/CallingNamePresentation
   CallingPartyNumber15002/CallingPartyNumber
  /CallingPartyTransformations
 - # ConnectedPartyTransformations
   ConnectedLineIdPresentationDefault/ConnectedLineIdPresentation
   ConnectedNamePresentationDefault/ConnectedNamePresentation
  /ConnectedPartyTransformations
 - # CalledPartyTransformations
   CalledPartyMask /
   DigitDiscardingInstructionNone/DigitDiscardingInstruction
   Prefix /
   CalledNumber916178632683/CalledNumber
  /CalledPartyTransformations
  /RoutePattern
 - # RouteList RouteListName=*RL-HQ-BR1-LD*
 - # RouteGroup RouteGroupName=*rg-hq*
   PreTransformCallingPartyNumber15002/PreTransformCallingPartyNumber
   PreTransformCalledPartyNumber916178632683/
PreTransformCalledPartyNumber
 - # CallingPartyTransformations
   ExternalPhoneNumberMaskDefault/ExternalPhoneNumberMask
   CallingPartyMask /
   Prefix /
   CallingPartyNumber15002/CallingPartyNumber
  /CallingPartyTransformations
 - # CalledPartyTransformations
   CalledPartyMask /
   DigitDiscardingInstructions /
   Prefix /
   CalledNumber916178632683/CalledNumber
  /CalledPartyTransformations
 - # Device Type=*SIPTrunk*
   EndDeviceNamehq-sip/EndDeviceName
   PortNumber0/PortNumber
   DeviceStatusUnKnown/DeviceStatus
   AARGroup /
   AARCallingSearchSpace /
   AARPrefixDigits /
   DeviceDestinationUse System Default/DeviceDestination
   CallingPartySelectionOriginator/CallingPartySelection
   CallingLineIdPresentationDefault/CallingLineIdPresentation
   CallerIDDn /
  /Device
  /RouteGroup
 - # RouteGroup RouteGroupName=*rg-br1*
   PreTransformCallingPartyNumber15002/PreTransformCallingPartyNumber
   PreTransformCalledPartyNumber916178632683/
PreTransformCalledPartyNumber
 - # CallingPartyTransformations
   

Re: [OSL | CCIE_Voice] Heads Spinning? Question about comment at Techtorial

2009-07-14 Thread Nara Shikamaru
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 meant to get someone’s attention like when
 you see a good looking girl walking down the street.  As I said, if they
 make it too hard, they are doing themselves a disservice.



 *From:* Jason Granat [mailto:j...@slash128.com]
 *Sent:* Friday, July 10, 2009 3:18 PM
 *To:* Michael Ciarfello; ccie_voice@onlinestudylist.com

 *Subject:* RE: [OSL | CCIE_Voice] Heads Spinning? Question about comment
 at Techtorial



 Thanks, I appreciate the perspective. I was concerned that this meant they
 were really increasing the difficulty level. Here’s the tweet:



 The new lab is gonna make candidates heads spin...that coming from the
 guy who writes the lab:-(

 8:07 AM Jun 28th http://twitter.com/IPexpertVoice/status/2372249598 from
 web



 And the link:



 http://twitter.com/IPexpertVoice



 *From:* Michael Ciarfello [mailto:mciarfe...@iplogic.com]
 *Sent:* Friday, July 10, 2009 10:53 AM
 *To:* Jason Granat; ccie_voice@onlinestudylist.com
 *Subject:* RE: [OSL | CCIE_Voice] Heads Spinning? Question about comment
 at Techtorial



 The end of the cheating / easy-way out, braindumping, etc (for a while at
 least.)



 Open-ended questions sound like they are going to generate some head
 spinning.  They are said to be easy if you have legitimately done the work.
 There are a LOT of complaints over on Netpro from a discussion on them from
 the RS.  See the certifications forum.  Sounds like it’s working.  IPExpert
 has a small quizzer that will give a sample of what those questions will be
 like so as we are studying, we have somewhat of an idea what we really
 should be paying attention to (more theory for those questions).  If you can
 configure, say gatekeeper, but can’t answer a simple question, given the
 diagram, what router can you put the gatekeeper on, then you have NO
 understanding of Gatekeeper.  I guess the theory is if you know how to build
 a car engine, but don’t know what kind of gas you have to put in
 it—something is wrong.



 I don’t think they will make the lab so hard as to not produce any CCIE’s.
 It’s in their best interest to generate CCIEs-Can’t configure equipment,
 can’t sell it.  They just don’t want to generate CCIE’s that took a
 shortcut, etc. that devalue the certification.



 Do you have the link to that?



 *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
 ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Jason Granat
 *Sent:* Friday, July 10, 2009 1:11 PM
 *To:* ccie_voice@onlinestudylist.com
 *Subject:* [OSL | CCIE_Voice] Heads Spinning? Question about comment at
 Techtorial



 I saw a twitter update that said something to the effect that “the new lab
 is going to make candidates heads spin”.



 Any comments/thoughts/etc on this?


  --



 http://slash128.com


  --



 http://slash128.com

 ___
 For more information regarding industry leading CCIE Lab training, please
 visit www.ipexpert.com




-- 
-Shikamaru
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[OSL | CCIE_Voice] BACD VOIP calls ver3

2009-07-14 Thread vineet sanghi

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 222 voip
 service aa
 destination-pattern 3500
 session target ipv4:10.10.29.1
 incoming called-number 3500
 dtmf-relay h245-alphanumeric
 codec g711ulaw
 no vad

Calling from PSTN phone directly to BACD ext(3500) works like a charm. 

Checked debug voice dialpeer and it is matching the incoming dialpeer.

(First matching the outgoing dialpeer) 
ep  7 21:15:08.615: //-1//DPM/dpMatchPeersMoreArg:
   Result=SUCCESS(0) 
   List of Matched Outgoing Dial-peer(s): 
 1: Dial-peer Tag=222
(Then incoming)
ep  7 21:15:08.627: //-1//DPM/dpAssociateIncomingPeerCore:
   Calling Number=3500, Called Number=, Voice-Interface=0x0,
   Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
   Peer Info Type=DIALPEER_INFO_SPEECH
Sep  7 21:15:08.627: //-1//DPM/dpAssociateIncomingPeerCore:
   Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=222


The call is not going to GK and GK doesn't have any BW statements. Also tried 
with shutting down gatekeepeer dialpeer on CME.

Thanks,
vineet


  
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Re: [OSL | CCIE_Voice] MOH - AU to G711 Conversion

2009-07-14 Thread Mark Snow
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 (Voice, Security)

Senior Technical Instructor - IPexpert, Inc.

Telephone: +1.810.326.1444
Fax: +1.309.413.4097
Mailto: ms...@ipexpert.com
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Join our free online support and peer group communities: 
http://www.IPexpert.com/communities
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IPexpert - 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  
CCIE Storage Lab Certifications.
--


On Jul 10, 2009, at 1:59 PM, Mike Brooks 2xcci...@gmail.com wrote:

 When spoofing MOH what converts the .AU file to G711 on the router ?

 I wouldnt think DSPs are used 

 Thx,

 Mike Brooks
 CCIE#16027 (RS)
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Re: [OSL | CCIE_Voice] [cisco-voip] Lab2A Q2.4

2009-07-14 Thread Tanner Ezell
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.



 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 manipulate the Regions so that
 the inter region default isn’t used?



 Thanks for your input,

 MT





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 cisco-v...@puck.nether.net
 https://puck.nether.net/mailman/listinfo/cisco-voip


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Re: [OSL | CCIE_Voice] Question about route lists and mixing MGCP +H323 route groups

2009-07-14 Thread Cristi Radescu
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, 2009 10:48 PM
To: Nara Shikamaru
Cc: OSL Group; GRAFL Philipp
Subject: Re: [OSL | CCIE_Voice] Question about route lists and mixing MGCP 
+H323 route groups

Did you wait enough time for the h323 call to fail before it kicked over to the 
MGCP gateway entry?  I remember it taking a while.  I can try it later tonight. 
 Also, what do your trace files indicate is happening?

From: Nara Shikamaru [mailto:shikam...@kagadis.com]
Sent: Friday, July 10, 2009 2:27 PM
To: Michael Ciarfello
Cc: GRAFL Philipp; OSL Group
Subject: Re: [OSL | CCIE_Voice] Question about route lists and mixing MGCP 
+H323 route groups

The detail for this setting states that it's for ICT calls, so I don't know if 
it will work for this particular issue.  I've worked through the issue by 
adding voip dial peer pointed at an alternate gateway with lower preferences 
that are used in case the pots dial peers fail.  Of course, the side effect is 
that I have a ton of dial peers now.  Maybe Vik or Mark have an opinion?

It seems to me that this would be a problem for both H323 and SIP gateways.
On Fri, Jul 10, 2009 at 10:57 AM, Michael Ciarfello 
mciarfe...@iplogic.commailto:mciarfe...@iplogic.com wrote:

You bring up an interesting question that would be good to get down solid.  I 
haven't researched it FULLY yet, but try this as an answer.  I'm sure someone 
has a more exacting answer or knows where the answer is.



I think because the call could not go out the H323 connection (for whatever 
reason), the error returned will be unallocated number and cause us to go to 
the route list member.  You should be able to look at a trace file to see what 
happens as your H323 call fails and the error that returns.  You should also 
see it try the next member in the route list (MGCP this time.)



From: 
ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com
 
[mailto:ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com]
 On Behalf Of Nara Shikamaru
Sent: Friday, July 10, 2009 12:56 PM
To: GRAFL Philipp
Cc: OSL Group
Subject: Re: [OSL | CCIE_Voice] Question about route lists and mixing MGCP 
+H323 route groups



Sorry, I have no idea how this flag is related to my original question.  Can 
you explain?





Stop Routing on Unallocated Number Flag: Error! Filename not specified.


This parameter determines routing behavior for intercluster trunk calls to an 
unallocated number. An unallocated number represents a dialed directory number 
that does not exist in a Cisco cluster. Valid values specify True or False. 
When the parameter is set to True and a call that is being routed to a remote 
Cisco cluster through a route list is released by a remote Cisco CallManager 
because of the unallocated number, a local Cisco CallManager will stop routing 
the call to a next device in the route list. When the parameter is set to 
False, the local Cisco CallManager will route the call to the next device.




On Fri, Jul 10, 2009 at 1:10 AM, GRAFL Philipp 
philipp.gr...@nextiraone.atmailto:philipp.gr...@nextiraone.at wrote:

Try in CCM-Service parameter: Stop routing on unallocated number flag = false.







Von: 
ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com
 
[mailto:ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com]
 Im Auftrag von Nara Shikamaru
Gesendet: Donnerstag, 09. Juli 2009 18:08
An: OSL Group
Betreff: [OSL | CCIE_Voice] Question about route lists and mixing MGCP +H323 
route groups



I'm working on a situation whereby a route list has two route groups, each 
contain end points with different call controls.  Primary is H323, secondary is 
MGCP.  I've run into an unfortunate shortcoming that I hope I'm wrong abount.  
In order to get calls routed to the second route group in the list, it LOOKS 
like the H323 end point has to be completely inaccessible.  In other words, 
it's not enough that the PRI is down; the whole device needs to be unreachable. 
 I suspect this is due to the fact that on an H323 gateway the PRI is not being 
backhauled to CUCM so the cluster has no way of knowing that the circuit is 
down, so it continues to be engaged.  I would LIKE to set up a situation 
whereby if the PRI is not functional, the second route group in the list is 
used.  Does this mean that I would have to configure the first gateway as MGCP 
instead?

--
-Shikamaru



--
-Shikamaru



--
-Shikamaru
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[OSL | CCIE_Voice] cisco-phone vs. cisco-softphone QoS

2009-07-14 Thread David Simes
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 hardphone or a softphone
on the same port depending on what gets plugged in?
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[OSL | CCIE_Voice] Lab 5 issue

2009-07-14 Thread Bill Hatcher
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?
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[OSL | CCIE_Voice] anyone from this list attended Networkers CCIE Tech sessions?

2009-07-14 Thread Padmanabhan, Padhu
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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[OSL | CCIE_Voice] SIP dialplan not working

2009-07-14 Thread Nara Shikamaru
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 enter the pattern.  The PG makes no note of
this.
2) Despite specifying a timeout of 0 in *pattern 2 999 timeout 0*, the
running config makes no mention of the timeout setting (although it does if
I set it to anythng else.)  I'm assuming it sets the timeout to 0 by default
and therefore doesn't state it explicity in the config?

-- 
-Shikamaru
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Re: [OSL | CCIE_Voice] CM to GK BRQ behavior

2009-07-14 Thread Cristi Radescu
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: OSL Group
Subject: [OSL | CCIE_Voice] CM to GK BRQ behavior

I'm seeing something strange in making calls from CM to CME via GK.

I've enabled the BRQ service parameter in CM.
I've included bandwidth total default 16 in my gk config and did a shut/no 
shut

When I make calls from CM to CME the deb h225 asn1 shows (I think) that 128k is 
being requested.

Am I missing something obivous here ?
Currently all my calls get rejected from the GK and go via the HQ GW.
If I remove the bandwidth command from the Gatekeeper config, the call works 
using g729.


- Scott


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[OSL | CCIE_Voice] test

2009-07-14 Thread Drew LePla
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 communities:
http://www.IPexpert.com/communities -- IPexpert - 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 CCIE Storage Lab Certifications. 

--

 

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Re: [OSL | CCIE_Voice] Voicemail and calling party transformation

2009-07-14 Thread Aamir Panjwani
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 press message button on HQ or BR1 phones it plays general CUC
greeting instead of prompting for user password. 

 

Calling internally between HQ and BR1 phones I see 4 digit ANI on called
phone. But when I press message button on the phone and look at unity
connection port monitor in RTMT I am seeing 7 digit ANI (HQ Phones) and
10 digit ANI (BR1 phones) being sent to unity connection (which means
calling party transformation pattern is kicking in). Off course I can
fixed this problem my adding 7 or 10 digit alternate extension in CUC
but that is just a workaround.

 

Use device pool calling party transformation CSS is definitely not
checked on the phone page itself yet somehow when I press message button
calling party transformation is kicking in! I have looked everywhere but
still can't figure out why dialing voicemail pilot would invoke calling
party transformation pattern??

 

HQ phones: 5001 and 5002

BR1 phones: 1001 and 1002

Calling party transformation 5XXX

Voicemail Pilot: 5050

 

Any help would be highly appreciated

 

Thanks

 


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Re: [OSL | CCIE_Voice] Question about route lists and mixing MGCP +H323 route groups

2009-07-14 Thread Mike Thompson
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
Subject: Re: [OSL | CCIE_Voice] Question about route lists and mixing MGCP
+H323 route groups

 

Did you wait enough time for the h323 call to fail before it kicked over to
the MGCP gateway entry?  I remember it taking a while.  I can try it later
tonight.  Also, what do your trace files indicate is happening?

 

From: Nara Shikamaru [mailto:shikam...@kagadis.com] 
Sent: Friday, July 10, 2009 2:27 PM
To: Michael Ciarfello
Cc: GRAFL Philipp; OSL Group
Subject: Re: [OSL | CCIE_Voice] Question about route lists and mixing MGCP
+H323 route groups

 

The detail for this setting states that it's for ICT calls, so I don't know
if it will work for this particular issue.  I've worked through the issue by
adding voip dial peer pointed at an alternate gateway with lower preferences
that are used in case the pots dial peers fail.  Of course, the side effect
is that I have a ton of dial peers now.  Maybe Vik or Mark have an opinion?

 

It seems to me that this would be a problem for both H323 and SIP gateways.

On Fri, Jul 10, 2009 at 10:57 AM, Michael Ciarfello mciarfe...@iplogic.com
wrote:

You bring up an interesting question that would be good to get down solid.
I haven't researched it FULLY yet, but try this as an answer.  I'm sure
someone has a more exacting answer or knows where the answer is.

 

I think because the call could not go out the H323 connection (for whatever
reason), the error returned will be unallocated number and cause us to go to
the route list member.  You should be able to look at a trace file to see
what happens as your H323 call fails and the error that returns.  You should
also see it try the next member in the route list (MGCP this time.)

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Nara Shikamaru
Sent: Friday, July 10, 2009 12:56 PM
To: GRAFL Philipp
Cc: OSL Group
Subject: Re: [OSL | CCIE_Voice] Question about route lists and mixing MGCP
+H323 route groups

 

Sorry, I have no idea how this flag is related to my original question.  Can
you explain?

 

 


Stop Routing on Unallocated Number Flag: Error! Filename not specified.

This parameter determines routing behavior for intercluster trunk calls to
an unallocated number. An unallocated number represents a dialed directory
number that does not exist in a Cisco cluster. Valid values specify True or
False. When the parameter is set to True and a call that is being routed to
a remote Cisco cluster through a route list is released by a remote Cisco
CallManager because of the unallocated number, a local Cisco CallManager
will stop routing the call to a next device in the route list. When the
parameter is set to False, the local Cisco CallManager will route the call
to the next device. 

 

On Fri, Jul 10, 2009 at 1:10 AM, GRAFL Philipp philipp.gr...@nextiraone.at
wrote:

Try in CCM-Service parameter: Stop routing on unallocated number flag =
false.

 

 

 

Von: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] Im Auftrag von Nara
Shikamaru
Gesendet: Donnerstag, 09. Juli 2009 18:08
An: OSL Group
Betreff: [OSL | CCIE_Voice] Question about route lists and mixing MGCP +H323
route groups

 

I'm working on a situation whereby a route list has two route groups, each
contain end points with different call controls.  Primary is H323, secondary
is MGCP.  I've run into an unfortunate shortcoming that I hope I'm wrong
abount.  In order to get calls routed to the second route group in the list,
it LOOKS like the H323 end point has to be completely inaccessible.  In
other words, it's not enough that the PRI is down; the whole device needs to
be unreachable.  I suspect this is due to the fact that on an H323 gateway
the PRI is not being backhauled to CUCM so the cluster has no way of knowing
that the circuit is down, so it continues to be engaged.  I would LIKE to
set up a situation whereby if the PRI is not functional, the second route
group in the list is used.  Does this mean that I would have to configure
the first gateway as MGCP instead?  



-- 
-Shikamaru






-- 
-Shikamaru




-- 
-Shikamaru

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[OSL | CCIE_Voice] Documentation at voice lab exam

2009-07-14 Thread Kevin Damisch
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 Jose.

Thanks,
Kevin


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[OSL | CCIE_Voice] Test: ccie_voice list down?

2009-07-14 Thread Michael Dietrich
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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[OSL | CCIE_Voice] CME EM Task 9.5

2009-07-14 Thread vineet sanghi



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


  
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[OSL | CCIE_Voice] Cisco UC in VMWare

2009-07-14 Thread Jason L. Hawkins
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 Workstation 6.  I've
tried different disk emulation modes and sizes from 72GB to 120GB.  I've
tried different amounts of memory but everything I've tried doesn't seem to
allow me to log into the application web interface.

I have used a CUCM 6.1.2 DVD with UC 2.1 to install UC with also tried using
a CUCM 6.0.1 DVD with UC 2.0.1.  The install completes successfully when
installed in as a VMWare Workstation 6 image.  After the install I can log
into the CLI, the OS administration website, and the Disaster Recovery
System.  But when I try to log into any of the application websites (Cisco
Unified Serviceability, Cisco Unity Connection Administration, Cisco Unity
Connection Serviceability) it always returns with a login failure.

From with in the CLI logged in as the admin I've tried using utils
reset_ui_administrator_name and utils reset_ui_administrator_password to
set a new application admin user name and password.

The utils reset_ui_administrator_name runs successfully but when I try
using utils reset_ui_administrator_password I get this:

admin:utils reset_ui_administrator_password
New password:
Confirm new password:
  utils reset_ui_administrator_password

Reset user interface password
appadmin
07/11/2009 18:39:48.050 | Update FAILED: code = 1, AUTHORIZATION FAILURE
admin:

What am I doing wrong here?

The other main scenario is trying to load UC on ESXi.  The install has
failed each time so far.  The install gets about an hour in then one of the
commands times out.  Then install program kicks me out.

The ESXi system is a 7825 H2.
The VM Workstation system is a 64bit - quad core with 4GB RAM.

I know I must be missing something small but this is really driving me
crazy.  Please let me know what I'm doing wrong.

and FYI the CUCM install from both versions works no problem.

Jason
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Re: [OSL | CCIE_Voice] UCCX 5.X Trigger Configuration

2009-07-14 Thread Sumit Ahuja
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 get the following error

 * Failed to create the CTI Route Point and corresponding Line on Unified
 CM.

 what's wrong ?

 thank you

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[OSL | CCIE_Voice] Test to see if mail server is back up properly ...

2009-07-14 Thread Mark Snow

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
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CCIE Storage Lab Certifications.

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[OSL | CCIE_Voice] clone vs snapshot vs any better method??

2009-07-14 Thread Joel Jose
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 my vmwares into another backupdir.. so i copied the pub clone and thot i
was done... first of all.. the db wipes clean.. yeah..yeah..obviously.. but
now i cant even find the sub as an available node for my cucm-group..
however all sub configurations can be changed and is reflected on the pub
gui itself... yes i have cross checked the serviceablity for both nodes and
thats fine
my backup strat is pretty boring i know.. i still havent figured what the
snapshots do... if anyone can suggest a better way to backup my servers it
would be great.. my head is spinning on the thoughts of a cucm reinstall
from dvd... even a quadcore q9550 with 8gb ram has its limits on sluggish
algorithms(redhat-cisco-installation-scripts) u see ;( .

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[OSL | CCIE_Voice] SIP/2.0 405 Method Not Allowed , Lab 4A

2009-07-14 Thread Erwan Erwan
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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[OSL | CCIE_Voice] Lab2A Q2.4

2009-07-14 Thread Michael Thompson

  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 manipulate the Regions so that
 the inter region default isn’t used?

 Thanks for your input,

 MT

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[OSL | CCIE_Voice] test

2009-07-14 Thread Drew LePla
test

 

If I can be of further assistance, please let me know. 

 

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Telephone: +1.810.326.1444 x204 

Fax: +1.810.454.0130 

Mailto: dle...@ipexpert.com 

 

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[OSL | CCIE_Voice] LAB 5A - HQ FAILOVER TO BR1 MGCP

2009-07-14 Thread Voice Guy
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:

9.1[2-9]xx[2-9]xx  - pt-hq-ld  :

rl-hq-br1-ld :

rg-hq -- hq-sip,rg-br1 mgcp
-

91[2-9]xx[2-9]xx - pt-xlate-internal

Translation Pattern  - rl-local-gw --- standard local route group

It appears that it is failing over to the MGCP gateway as intended but it is
not working.

MGCP is working

I tried removing rg-hq from the route list - rl-hq-br-ld , leaving only
rg-br1 to see if the call goes through.

I did a isdn q931 debug on br1 and I can see the call will go through.  The
output is identical to HQ's debug when the call goes through to the PSTN .
However, I still get the message 'your call cannot be completed..'

It looks like the it is not even failing  over to BR1 because when I have
the 2 route groups in place and i shut the voice-port on HQ, there are
totally no debug outputs on BR1.

It looks like a RP problem but I am not sure.

Any help appreciated.

Thanks.

Here are the 2 debugs, DNA and then a trace file.

?xml version=1.0 standalone=yes ?
 
-http://?ui=2view=jsname=jsver=xWWCalDiPoM.en.am=%211cqNLSaoA8C5he3iwbQ2woaWCD5IGcpXuAaDtVE1#12274b5dfe452ce2_12274acf2e86fd9d_12273906eb36d252_
DigitAnalysisResult
 
-http://?ui=2view=jsname=jsver=xWWCalDiPoM.en.am=%211cqNLSaoA8C5he3iwbQ2woaWCD5IGcpXuAaDtVE1#12274b5dfe452ce2_12274acf2e86fd9d_12273906eb36d252_
ResultsSummary
   CallingPartyInfo CallingParty=*15002* Partition= DeviceCSS=LineCSS
=*CSS-XLATE-INTERNAL-LOC-LD* AARGroup= AARCSS= /
   DialedDigits916178632683/DialedDigits
   MatchResultRouteThisPattern/MatchResult
   MatchedPatternInfo Pattern=*9.1[2-9]XX[2-9]XX* Partition=*
pt-hq-ld* TimeSchedule= /
   CalledPartyNumber916178632683/CalledPartyNumber
   TimeZoneGreenwich Standard Time/TimeZone
   EndDeviceRL-HQ-BR1-LD/EndDevice
   DeviceLocationOffNet/DeviceLocation
   InterDigitTimeoutNO/InterDigitTimeout
   AllowDeviceOverrideDisabled/AllowDeviceOverride
   OutsideDialToneNO/OutsideDialTone
   /ResultsSummary
 
-http://?ui=2view=jsname=jsver=xWWCalDiPoM.en.am=%211cqNLSaoA8C5he3iwbQ2woaWCD5IGcpXuAaDtVE1#12274b5dfe452ce2_12274acf2e86fd9d_12273906eb36d252_
CallFlow
 
-http://?ui=2view=jsname=jsver=xWWCalDiPoM.en.am=%211cqNLSaoA8C5he3iwbQ2woaWCD5IGcpXuAaDtVE1#12274b5dfe452ce2_12274acf2e86fd9d_12273906eb36d252_
TranslationPattern Pattern=*91[2-9]XX[2-9]XX*
   PositionalMatchList9:16178632683/PositionalMatchList
   CallingPartyNumber15002/CallingPartyNumber
   PreTransformCallingPartyNumber5002/PreTransformCallingPartyNumber
   PreTransformCalledPartyNumber916178632683/
PreTransformCalledPartyNumber
 
-http://?ui=2view=jsname=jsver=xWWCalDiPoM.en.am=%211cqNLSaoA8C5he3iwbQ2woaWCD5IGcpXuAaDtVE1#12274b5dfe452ce2_12274acf2e86fd9d_12273906eb36d252_
CallingPartyTransformations
   ExternalPhoneNumberMaskYES/ExternalPhoneNumberMask
   CallingPartyMask /
   Prefix1/Prefix
   CallingLineIdPresentationDefault/CallingLineIdPresentation
   CallingNamePresentationDefault/CallingNamePresentation
   CallingPartyNumber15002/CallingPartyNumber
   /CallingPartyTransformations
 
-http://?ui=2view=jsname=jsver=xWWCalDiPoM.en.am=%211cqNLSaoA8C5he3iwbQ2woaWCD5IGcpXuAaDtVE1#12274b5dfe452ce2_12274acf2e86fd9d_12273906eb36d252_
ConnectedPartyTransformations
   ConnectedLineIdPresentationDefault/ConnectedLineIdPresentation
   ConnectedNamePresentationDefault/ConnectedNamePresentation
   /ConnectedPartyTransformations
 
-http://?ui=2view=jsname=jsver=xWWCalDiPoM.en.am=%211cqNLSaoA8C5he3iwbQ2woaWCD5IGcpXuAaDtVE1#12274b5dfe452ce2_12274acf2e86fd9d_12273906eb36d252_
CalledPartyTransformations
   CalledPartyMask /
   DigitDiscardingInstructionNone/DigitDiscardingInstruction
   Prefix /
   CalledNumber916178632683/CalledNumber
  /CalledPartyTransformations
  /TranslationPattern
 
-http://?ui=2view=jsname=jsver=xWWCalDiPoM.en.am=%211cqNLSaoA8C5he3iwbQ2woaWCD5IGcpXuAaDtVE1#12274b5dfe452ce2_12274acf2e86fd9d_12273906eb36d252_
RoutePattern Pattern=*9.1[2-9]XX[2-9]XX*
   PositionalMatchList9:16178632683/PositionalMatchList
   DialPlan /
 
-http://?ui=2view=jsname=jsver=xWWCalDiPoM.en.am=%211cqNLSaoA8C5he3iwbQ2woaWCD5IGcpXuAaDtVE1#12274b5dfe452ce2_12274acf2e86fd9d_12273906eb36d252_
RouteFilter
   FilterName /
   FilterClause /
  /RouteFilter
   RequireForcedAuthorizationCodeNo/RequireForcedAuthorizationCode
   AuthorizationLevel0/AuthorizationLevel
   RequireClientMatterCodeNo/RequireClientMatterCode
   networkLocation /
   PreTransformCallingPartyNumber15002/PreTransformCallingPartyNumber
   PreTransformCalledPartyNumber916178632683/
PreTransformCalledPartyNumber
 
-http://?ui=2view=jsname=jsver=xWWCalDiPoM.en.am=%211cqNLSaoA8C5he3iwbQ2woaWCD5IGcpXuAaDtVE1#12274b5dfe452ce2_12274acf2e86fd9d_12273906eb36d252_

Re: [OSL | CCIE_Voice] Test to see if mail server is back up properly ...

2009-07-14 Thread Mick Vaites
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,
 

-- 
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
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Re: [OSL | CCIE_Voice] Test to see if mail server is back up properly ...

2009-07-14 Thread Tanner Ezell
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
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 RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and
 CCIE Storage Lab Certifications.
 --





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Re: [OSL | CCIE_Voice] Test to see if mail server is back up properly ...

2009-07-14 Thread Cristobal Priego
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 peer group communities:
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 RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and
 CCIE Storage Lab Certifications.
 --





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[OSL | CCIE_Voice] sip 7961 upgrade failure

2009-07-14 Thread c george

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 D/M/Y
 tftp-path flash:
 create profile sync 58112668185A
!
voice register dn  1
 number 5501
 name CarolineSpain
!
voice register dn  2
 number 5502
 name CutesSpain
!
voice register dn  3
 number 5503
 name CharlesSpain
!
voice register template  1
 dialplan 1
!
voice register dialplan  1
 type 7940-7960-others
 pattern 1 3...
 pattern 2 4...
 pattern 3 5...
 pattern 4 911
 pattern 5 9911
!

!
voice register pool  2
 id mac 0018.18D8.8251
 type 7961
 number 1 dn 2
 template 1
 cor incoming user1 default
 dtmf-relay sip-notify
 username 5502 password cisco
 description 7314685502
 codec g711ulaw
!
 
tftp-server syncinfo.xml url flash:/syncinfo.xml
tftp-server SIPDefault.cnf url flash:/SIPDefault.cnf
tftp-server softkeyDefault_kpml.xml url flash:/softkeyDefault_kpml.xml
tftp-server softkeyDefault.xml url flash:/softkeyDefault.xml
tftp-server softkey1_kpml.xml url flash:/softkey1_kpml.xml
tftp-server softkey1.xml url flash:/softkey1.xml
tftp-server cme_dialplan_1.xml url flash:/cme_dialplan_1.xml
tftp-server SIP000BBEF9E5A6.cnf url flash:/SIP000BBEF9E5A6.cnf
tftp-server SEP001818D88251.cnf.xml url flash:/SEP001818D88251.cnf.xml
 
JuJul 13 22:20:29.343: TFTP: Looking for SIP41.8-4-2S.loads
Jul 13 22:20:29.347: TFTP: Opened flash:SIP41.8-4-2S.loads, fd 7, size 638 for p
rocess 123
Jul 13 22:20:29.351: TFTP: Finished flash:SIP41.8-4-2S.loads, time 00:00:00 for
process 123
Jul 13 22:20:30.099: TFTP: Looking for jar41sip.8-4-1-23.sbn

BR2#sh flash | in jar
243 502269 Jul 13 2009 22:07:40 jar41sip.8-4-1-23.sbn
***
7960 has no problems upgrading to sip
 
it is requesting file jar41sip.8-4-1-23.sbn. The file is in flash but it will 
not pull it for some reason.
 
phone just sits with a error on it
 



Respectfully Charles George


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[OSL | CCIE_Voice] Looking For a Strong Cisco IPCC Resource

2009-07-14 Thread Tech Guy
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).

If you fit the bill or you know someone that does; please send me a private 
email; (NO BROADCAST EMAILS PLEASE!) 
and I will send you the detail job description.

Thanks.

Godswill Oletu
CCIE #16464 (RS)
CCNP,CCDP,CCSP,INFOSEC,MCP, et al.



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[OSL | CCIE_Voice] autoregistered phone... acting like zombie!!

2009-07-14 Thread Joel Jose
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 able to call..etc.. i have had this issue once..and i dont remmeber how
i solved it... may be i resinstalled from scratch.. i dont remember... why
doesnt cisco have a clean SQL db implementation??... uhhh... i am from
India...if this is the kinda quality our outsourcing companies are giving...
then i have good news for my american brothers!! ;).. LOL... somewhere
something is broken... or may be i am just dreaming...
joel.

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Re: [OSL | CCIE_Voice] Test to see if mail server is back up properly ...

2009-07-14 Thread Michael Ciarfello
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 Maillist
Subject: [OSL | CCIE_Voice] Test to see if mail server is back up properly ...

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.commailto:ms...@ipexpert.com
--
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[OSL | CCIE_Voice] how to access web interface mcu 3315

2009-07-14 Thread A Tommy
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 anybody help me.

Thanks..
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[OSL | CCIE_Voice] Calling party transformation patterns, where do they take place?

2009-07-14 Thread Otto Sanchez
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 transformation pattern
match occurs after the rp/rl calling party transformations/manipulations
are performed,

 

From srnd dial plan chapter, route patterns section:

 The calling and called party numbers resulting from the digit
transformations configured in the route pattern and/or route lists are
then processed by any Transformation Patterns configured for the devices
contained in the chosen Route Group.

 

How do we contrast these two statements?, and where should the calling
party transformations patterns be taking place for gateway's outgoing
calls?

 

Thanks!,

 

Otto

 

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[OSL | CCIE_Voice] Hi Guys

2009-07-14 Thread Mikal Jackson
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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[OSL | CCIE_Voice] cisco-phone vs. cisco-softphone QoS

2009-07-14 Thread David Simes
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 should a
switchport be configured if you need autoqos for a hardphone or a softphone
on the same port depending on what gets plugged in?  The administrative
overhead of having to manage port settings (autoqos hardphone vs. softphone)
would be a real pain in dynamic environment where people move around a lot.







-- 


Regards,

David Simes
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[OSL | CCIE_Voice] Test to all OSL Mailing Lists

2009-07-14 Thread Mark Snow

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 back alley and promptly dispatched of.


Anyway - big test message to see if mail is processing correctly. If  
you do get this message, REPLY TO ALL - that way all of the messages  
float through the OSL mail server and we can confirm in a larger way  
that mail is back up properly.



Thanks!

--
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: 
http://www.IPexpert.com/communities
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[OSL | CCIE_Voice] Test

2009-07-14 Thread Mike Elston
Testing OSL
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[OSL | CCIE_Voice] hunt pilot need help

2009-07-14 Thread A Tommy
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 ringing.
When 1001 pick up the phone, 1002 stop ringing.
There's a message that 1 new missed call on the 1002 ip phone.

Because the phone call already answer by the 1001 phone, how to configure
that 1002 don't display 1 new missed call?
Is it possible?

Thanks for your help

Regards
Cok.
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[OSL | CCIE_Voice] Tre: est to see if mail server is back up properly

2009-07-14 Thread Kamran Ahsanullah
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 with subject or body 'help' to
ccie_voice-requ...@onlinestudylist.com

 You can reach the person managing the list at
ccie_voice-ow...@onlinestudylist.com

 When replying, please edit your Subject line so it is more specific
 than Re: Contents of CCIE_Voice digest...


 Today's Topics:

   1. Test to see if mail server is back up properly... (Mark Snow)
   2. Re: silly doubt regarding lab2A (Michael Thompson)
   3. LAB 5A - failover (Voice Guy)


 --

 Message: 1
 Date: Mon, 13 Jul 2009 11:02:13 -0400
 From: Mark Snow ms...@ipexpert.com
 Subject: [OSL | CCIE_Voice] Test to see if mail server is back up
properly...
 To: CCIE Voice Maillist ccie_voice@onlinestudylist.com
 Message-ID: 07302333-5bba-4c35-a262-a2d773b94...@ipexpert.com
 Content-Type: text/plain; charset=us-ascii

 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:
 http://www.IPexpert.com/communities
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 --

 Message: 2
 Date: Sun, 12 Jul 2009 14:57:02 -0400
 From: Michael Thompson mthompson...@gmail.com
 Subject: Re: [OSL | CCIE_Voice] silly doubt regarding lab2A
 To: Joel Jose joeljose...@gmail.com
 Cc: Mark Snow ms...@ipexpert.com, ccie_voice@onlinestudylist.com,
Vik Malhi vma...@ipexpert.com
 Message-ID:
363f21ee0907121157pa554eabiff3becebbb854...@mail.gmail.com
 Content-Type: text/plain; charset=iso-8859-1

 Joel,
a few key points:
1)Only 1 UCM Group can have auto registration enabled.  if you have 12
 groups in the UCM cluster, still only 1 can have auto reg.  NOTE that this
 means that no matter WHAT device pool you WANT a phone to end up in, it's
 going to take the auto-register information from the UCM config and apply
 it
 to that phone.
2) you can have all 3 servers in your UCM group (if you had as many)
 with auto reg turned on.  style points whether each of those servers have
 different DN ranges or they pull from the same 'pool' of numbers...I
 personally try to break them up (helps you tell which server assigned the
 DN).  which leads me to my next point...
3) I don't know if this is a technical edict or just experience
 speaking.  The first UCM server in the UCM group will exhaust its entire
 pool of numbers before it goes to the second server to look for a DN
 (keeping in mind those pesky 'unassigned numbers in the Route Plan Report).


 The PFM as it relates to getting a DN of 1000, probably a snapshot
 sequencing issue.  the only downside I've found with how quickly these guys
 busted their asses to get these products out as fast as they did.  All in
 all, I'm good with a few typos :)

 On Sat, Jul 4, 2009 at 11:05 AM, Joel Jose joeljose...@gmail.com wrote:

  inorder to have HQ phones and B1phones autoregister.. how do we go about
  it?? i mean they have diff DN ranges right...so got a little confusing...
 in
  the PG the screenshots solving this are causing me to be confused... both
  pub and sub will have the same autoregister DN range?? and can more than
 1
  server have autoregistration enabled in the same cluster? how does in
  the
  subsequent screenshot the BR1 phone gets DN of 1000 range automagically??
  thanks for your time,
  joel.
 
  --
  it's not the years in your life that count. It's the life in your
 years.
  Abraham Lincoln
 
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 --

 Message: 3
 Date: Mon, 13 Jul 2009 18:02:41 +0800
 From: Voice Guy learn.vo...@gmail.com
 Subject: [OSL | CCIE_Voice] LAB 5A - failover
 To: OSL Group ccie_voice@onlinestudylist.com
 Cc: Vik Malhi vma...@ipexpert.com
 Message-ID:
fa495cf90907130302y6a5f7093r37a6374b36532...@mail.gmail.com
 Content-Type: text/plain; charset=iso-8859-1

 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 

Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 41, Issue 67

2009-07-14 Thread Suresh Solomon
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
    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 with subject or body 'help' to
    ccie_voice-requ...@onlinestudylist.com

You can reach the person managing the list at
    ccie_voice-ow...@onlinestudylist.com

When replying, please edit your Subject line so it is more specific
than Re: Contents of CCIE_Voice digest...


Today's Topics:

  1. Test to see if mail server is back up properly    ... (Mark Snow)
  2. Re: silly doubt regarding lab2A (Michael Thompson)
  3. LAB 5A - failover (Voice Guy)


--

Message: 1
Date: Mon, 13 Jul 2009 11:02:13 -0400
From: Mark Snow ms...@ipexpert.com
Subject: [OSL | CCIE_Voice] Test to see if mail server is back up
    properly    ...
To: CCIE Voice Maillist ccie_voice@onlinestudylist.com
Message-ID: 07302333-5bba-4c35-a262-a2d773b94...@ipexpert.com
Content-Type: text/plain; charset=us-ascii

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
--
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http://www.IPexpert.com/communities
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Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and  
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Message: 2
Date: Sun, 12 Jul 2009 14:57:02 -0400
From: Michael Thompson mthompson...@gmail.com
Subject: Re: [OSL | CCIE_Voice] silly doubt regarding lab2A
To: Joel Jose joeljose...@gmail.com
Cc: Mark Snow ms...@ipexpert.com, ccie_voice@onlinestudylist.com,
    Vik Malhi vma...@ipexpert.com
Message-ID:
    363f21ee0907121157pa554eabiff3becebbb854...@mail.gmail.com
Content-Type: text/plain; charset=iso-8859-1

Joel,
    a few key points:
    1)Only 1 UCM Group can have auto registration enabled.  if you have 12
groups in the UCM cluster, still only 1 can have auto reg.  NOTE that this
means that no matter WHAT device pool you WANT a phone to end up in, it's
going to take the auto-register information from the UCM config and apply it
to that phone.
    2) you can have all 3 servers in your UCM group (if you had as many)
with auto reg turned on.  style points whether each of those servers have
different DN ranges or they pull from the same 'pool' of numbers...I
personally try to break them up (helps you tell which server assigned the
DN).  which leads me to my next point...
    3) I don't know if this is a technical edict or just experience
speaking.  The first UCM server in the UCM group will exhaust its entire
pool of numbers before it goes to the second server to look for a DN
(keeping in mind those pesky 'unassigned numbers in the Route Plan Report).


The PFM as it relates to getting a DN of 1000, probably a snapshot
sequencing issue.  the only downside I've found with how quickly these guys
busted their asses to get these products out as fast as they did.  All in
all, I'm good with a few typos :)

On Sat, Jul 4, 2009 at 11:05 AM, Joel Jose joeljose...@gmail.com wrote:

 inorder to have HQ phones and B1phones autoregister.. how do we go about
 it?? i mean they have diff DN ranges right...so got a little confusing... in
 the PG the screenshots solving this are causing me to be confused... both
 pub and sub will have the same autoregister DN range?? and can more than 1
 server have autoregistration enabled in the same cluster? how does in  the
 subsequent screenshot the BR1 phone gets DN of 1000 range automagically??
 thanks for your time,
 joel.

 --
 it's not the years in your life that count. It's the life in your years.
 Abraham Lincoln

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--

Message: 3
Date: Mon, 13 Jul 2009 18:02:41 +0800
From: Voice Guy learn.vo...@gmail.com
Subject: [OSL | CCIE_Voice] LAB 5A - failover
To: OSL Group ccie_voice@onlinestudylist.com
Cc: Vik Malhi vma...@ipexpert.com
Message-ID:
    fa495cf90907130302y6a5f7093r37a6374b36532...@mail.gmail.com
Content-Type: text/plain; charset=iso-8859-1

Hi,

I am working on Lab 5A task 5.8  - HQ phone calling PSTN 

Re: [OSL | CCIE_Voice] [OSL | CCIE_Security] Test to all OSL Mailing Lists

2009-07-14 Thread Larry Hadrava
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 (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 back alley and promptly dispatched of.

 Anyway - big test message to see if mail is processing correctly. If you do
 get this message, REPLY TO ALL - that way all of the messages float through
 the OSL mail server and we can confirm in a larger way that mail is back up
 properly.


 Thanks!

 --
 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:
 http://www.IPexpert.com/communities http://www.ipexpert.com/communities
 --
 IPexpert - 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
 CCIE Storage Lab Certifications.
 --





 ___
 For more information regarding industry leading CCIE Lab training, please
 visit www.ipexpert.com


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Re: [OSL | CCIE_Voice] Test to see if mail server is back up properly ...

2009-07-14 Thread Larry Hadrava
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 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 Maillist
 *Subject:* [OSL | CCIE_Voice] Test to see if mail server is back up
 properly ...

  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:
 http://www.IPexpert.com/communities http://www.ipexpert.com/communities
 --
 IPexpert - 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
 CCIE Storage Lab Certifications.
 --





 ___
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 visit www.ipexpert.com


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Re: [OSL | CCIE_Voice] Test to all OSL Mailing Lists

2009-07-14 Thread Steve Sarrick
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 | CCIE_Voice] Test to all OSL Mailing Lists

 

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 back alley and promptly dispatched of.

 

Anyway - big test message to see if mail is processing correctly. If you
do get this message, REPLY TO ALL - that way all of the messages float
through the OSL mail server and we can confirm in a larger way that mail
is back up properly.

 

 

Thanks!


-- 

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:
http://www.IPexpert.com/communities

--

IPexpert - 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 CCIE Storage Lab Certifications.

--

 

 

 

 

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Re: [OSL | CCIE_Voice] Test to see if mail server is back up properly...

2009-07-14 Thread bkvalentine
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
...


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Re: [OSL | CCIE_Voice] Test to all OSL Mailing Lists

2009-07-14 Thread bkvalentine
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 
Maillistccie_secur...@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Test to all OSL Mailing Lists


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Re: [OSL | CCIE_Voice] Test to all OSL Mailing Lists

2009-07-14 Thread Jarrod Phelps

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 responsible (not me - I just got the privilege of having to  
FIX it :) have been drug into a back alley and promptly dispatched of.


Anyway - big test message to see if mail is processing correctly. If  
you do get this message, REPLY TO ALL - that way all of the messages  
float through the OSL mail server and we can confirm in a larger way  
that mail is back up properly.



Thanks!

--
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: 
http://www.IPexpert.com/communities
--
IPexpert - 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 CCIE Storage Lab Certifications.

--




___
For more information regarding industry leading CCIE Lab training,  
please visit www.ipexpert.com
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Re: [OSL | CCIE_Voice] CCIE#24857

2009-07-14 Thread Larry Hadrava
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 to Vik for an excellent bootcamp, I
 learnt a lot, and thanks to everyone who participate in this group, it was
 my lucky 5th attempt, and I can tell you that all the effort, all the
 nights and weekends studyng worth this moment. Keep focus and with
 perseverance you can have your number.



 Regards,



 [image:
 http://www.cisco.com/global/EMEA/brand/signature/default/spacer.gif]

 *Jose Gregorio Linero Welcker*
 *Systems Engineer*
 *Sales / Channels**
 *
 jlin...@cisco.com
 Phone: *5713256052*
 Mobile: *573102634216*
 Fax: *5713256090*

 *Cisco Systems, Inc.*
 Cra 7 # 71-21 Torre A Piso 17
 Colombia
 Cisco.com http://www.cisco.com/global/CO/





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 http://www.cisco.com/global/EMEA/brand/signature/default/footerHead.gif]



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[OSL | CCIE_Voice] CME extension mobility

2009-07-14 Thread vineet sanghi

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



  
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Re: [OSL | CCIE_Voice] [OSL | CCIE_Security] Test to all OSL Mailing Lists

2009-07-14 Thread Neil Apolzan

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
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Email sent via mobile phone...

On 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  
(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 back alley and promptly dispatched of.


Anyway - big test message to see if mail is processing correctly. If  
you do get this message, REPLY TO ALL - that way all of the messages  
float through the OSL mail server and we can confirm in a larger way  
that mail is back up properly.



Thanks!

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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
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Re: [OSL | CCIE_Voice] sip 7961 upgrade failure FIXED

2009-07-14 Thread c george

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 | CCIE_Voice] sip 7961 upgrade failure








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 D/M/Y
 tftp-path flash:
 create profile sync 58112668185A
!
voice register dn  1
 number 5501
 name CarolineSpain
!
voice register dn  2
 number 5502
 name CutesSpain
!
voice register dn  3
 number 5503
 name CharlesSpain
!
voice register template  1
 dialplan 1
!
voice register dialplan  1
 type 7940-7960-others
 pattern 1 3...
 pattern 2 4...
 pattern 3 5...
 pattern 4 911
 pattern 5 9911
!

!
voice register pool  2
 id mac 0018.18D8.8251
 type 7961
 number 1 dn 2
 template 1
 cor incoming user1 default
 dtmf-relay sip-notify
 username 5502 password cisco
 description 7314685502
 codec g711ulaw
!
 
tftp-server syncinfo.xml url flash:/syncinfo.xml
tftp-server SIPDefault.cnf url flash:/SIPDefault.cnf
tftp-server softkeyDefault_kpml.xml url flash:/softkeyDefault_kpml.xml
tftp-server softkeyDefault.xml url flash:/softkeyDefault.xml
tftp-server softkey1_kpml.xml url flash:/softkey1_kpml.xml
tftp-server softkey1.xml url flash:/softkey1.xml
tftp-server cme_dialplan_1.xml url flash:/cme_dialplan_1.xml
tftp-server SIP000BBEF9E5A6.cnf url flash:/SIP000BBEF9E5A6.cnf
tftp-server SEP001818D88251.cnf.xml url flash:/SEP001818D88251.cnf.xml
 
JuJul 13 22:20:29.343: TFTP: Looking for SIP41.8-4-2S.loads
Jul 13 22:20:29.347: TFTP: Opened flash:SIP41.8-4-2S.loads, fd 7, size 638 for p
rocess 123
Jul 13 22:20:29.351: TFTP: Finished flash:SIP41.8-4-2S.loads, time 00:00:00 for
process 123
Jul 13 22:20:30.099: TFTP: Looking for jar41sip.8-4-1-23.sbn

BR2#sh flash | in jar
243 502269 Jul 13 2009 22:07:40 jar41sip.8-4-1-23.sbn
***
7960 has no problems upgrading to sip
 
it is requesting file jar41sip.8-4-1-23.sbn. The file is in flash but it will 
not pull it for some reason.
 
phone just sits with a error on it
 



Respectfully Charles George


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