Re: [OSL | CCIE_Voice] Is the list actually down?

2014-06-26 Thread Ramcharan Arya
I believe there is apps name tap a talk  for this new forum. If you logon
into new forum on right  under Ipexpert Chatter  it is mentioned  If you'd
like to be able to access our Member's Community from your iPhone or
Android, download this PDF http://community.ipexpert.com/TapaTalk.pdf for
instructions.

Regards,
Ramcharan Arya
CCIE # 28926 (Voice/RS)




On Thu, Jun 26, 2014 at 7:06 PM, Robert McGhee bobwmcg...@verizon.net
wrote:

 I think there is a smartphone app for the new forum...

 Sent from my iPhone

 On Jun 26, 2014, at 6:44 PM, Josh Petro josh.pe...@gmail.com wrote:

 Wayne,
 If you're still reading off this email list,  I really do like the new
 IPExpert site, but since I'm not always able to login to it, especially
 when I'm on my phone,  its not as appealing as this study list. I may have
 missed where you can sign up for the forum and have the topics and
 discussions of the day emailed to you, but if not, I probably won't use it
 as much. The email list is just too convenient.
 Is there a way to only email people with active accounts or something?
 Maybe that would help weed out all of the SPAMers and jerks trying to
 cheat. Just a thought (and I'm sure you've already looked at it as an
 option). Anything to keep things coming to an email would be sweet. Or, a
 smartphone app would be even better, but I'm sure that's overkill for what
 you're trying to do.
 Josh
 On Jun 26, 2014 1:23 PM, Edgar Mauricio Diaz Orellana 
 eorellan1...@gmail.com wrote:

 i get the email!

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Re: [OSL | CCIE_Voice] RIP CCIE-Voice :-)

2014-02-15 Thread Ramcharan Arya
Those who are choosing option # 3 - for those candidate IPX should provide
a different training program just to cover only specific features in new
blueprint. This might be beneficial. Assumption that candidates who have
CCIE voice certification  expert level understanding on infrastructure,
gateway protocols ( H.323, MGCP, SIP etc..), QOS and other topics from
previous blueprint.


I would think about option # 3 after Data Center lab planned this year.

Thanks,
Ramcharan Arya
CCIE # 28926 (Voice/RS)





On Sat, Feb 15, 2014 at 3:05 PM, Mark Holloway m...@markholloway.com wrote:

 From a Cisco partner business perspective I don't see how it would make
 sense to have your current CCIE's retake the lab when they could just as
 easily migrate to Collaboration after taking their written renewal. Time is
 money, and consuming your top resources with another lab certification with
 no foreseeable gain seems like a waste. I imagine most folks working for
 partners are already deep into the same platforms on the Collaboration
 blueprint. At minimum, they should at least have the UC NFR software setup
 in their own lab already.

 On Feb 15, 2014, at 12:10 AM, m george m.george00...@gmail.com wrote:

 Will anyone here who already passed voice lab preparing to undertake
 collaboration lab for 2nd CCIE title ? I have talked to many folks  me 
 my colleagues we plan to convert our titles to Collab IE with written
 rather than going for another hectic lab. What's your guys take on this ?
 What will you do  ?


 On Sat, Feb 15, 2014 at 3:54 AM, Abel ... midga...@gmail.com wrote:

 Upgrading my home lab already, kind of expensive with new 29xx. But just
 for the knowledge sake.


 On Sat, Feb 15, 2014 at 7:15 AM, wilson.sam...@bt.com wrote:

  Aha Nicolas, you have a point sir.

 Anyway, I just wanted to make the passage of the track / version
 somewhat memorable that's all.

 No need to get serious on this now (note to myself as well)

 Lets get the Colloboration done..

 Btw, who is attempting it on tihs forum and how you have prepared for
 it? Lab Gear??

 Regards

  --
 *From:* Mergenthal, Chase [chase.mergent...@bestbuy.com]
 *Sent:* Friday, February 14, 2014 3:01 PM
 *To:* Nicolas MICHEL; Samuel,W,Wilson,JKH3 R
 *Cc:* Online Study
 *Subject:* RE: [OSL | CCIE_Voice] RIP CCIE-Voice :-)

   It's funny you mention that, on my second or so attempt; at the end
 of the exam UCCX wasn't working at all... I got 100% on UCCX...


 --

 Chase Mergenthal


 *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
 ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Nicolas MICHEL
 *Sent:* Friday, February 14, 2014 1:41 PM
 *To:* wilson.sam...@bt.com
 *Cc:* Online Study
 *Subject:* Re: [OSL | CCIE_Voice] RIP CCIE-Voice :-)


 Wilson, I am already a CCIE in RS so I know what to expect when I am
 taking a CCIE exam.


 When you skip the UCCX task because you ran out of time and when you
 score report says : UCCX = 100%, to me it means complete nonsense :)



 Lots of people have different opinion and point of view :)



 Cheers



 Nicolas


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Re: [OSL | CCIE_Voice] wireshark capture analysis

2014-01-18 Thread Ramcharan Arya
To access the VoIP calls analysis use the menu entry Telephony-VoIP
Calls...

http://wiki.wireshark.org/VoIP_calls

Thanks,
Ramcharan Arya
CCIE # 28926 (Voice/RS)



On Sat, Jan 18, 2014 at 8:17 PM, virajith vir...@rediffmail.com wrote:

 Hello Guys,

 I have collected a Packet capture ( wireshark) from a cisco CME .  I need
 to know  following :-


 1) The filters used on wireshark to see all traffic that passed between
 the ip phone  and the CME.

 2) Would also like to  how to filter wireshark captures based on ip
 address  and  mac address of the phone .


 -Vir





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 Get your own *FREE* website, *FREE* domain  *FREE* mobile app with
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Re: [OSL | CCIE_Voice] NM-CUE integration with CUCM

2013-12-10 Thread Ramcharan Arya
Check you WAN QoS policy. Fragmentation may cause this issue.

Thanks
Ramcharan Arya
CCIE #28926(voice/RS)

Sent from my iPhone

 On Dec 9, 2013, at 6:02 PM, Rahul Tapiawala dean0...@gmail.com wrote:
 
  
 Hello,
  
 I am trying to integrate NM-CUE (7.0.6) with CUCM (7.0.2) without success.
 Ports wont register as well as Jtapi user cant login. I have tried everything 
 suggested online
 as well as Cisco website. CUE can ping both PUB/SUB and vice versa.
  
 Has anyone encountered similar issue?
  
 Please advise. Thanks.
 ___
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Re: [OSL | CCIE_Voice] Our FREE vLectures are back!

2013-11-14 Thread Ramcharan Arya
Hi Wayne,

One of the Data Center vLecture is schedule on 28th Nov Thanksgiving day.

Data Center: MDS FC Security Features (November 28th, 3 pm EST)

*(REGISTER) http://www.ipexpert.com/Company/vlecture-registration*
Is this still going to be happen on Thanksgiving day.? Is there any please
reschedule for another day.

Please let me know.

Thanks  Regards,
Ramcharan Arya
CCIE # 28926 (Voice/Routing  Switching)



On Fri, Oct 25, 2013 at 9:25 AM, Wayne Lawson
waynelawson-...@ipexpert.comwrote:

 *Free CCIE Training! (Our vLectures Are 
 Back!)*http://blog.ipexpert.com/2013/10/23/free-ccie-training-our-vlectures-are-back/?utm_source=rssutm_medium=rssutm_campaign=free-ccie-training-our-vlectures-are-back

 Back by popular demand – our vLectures (free 1 to 2 hour training
 lectures) are now on the schedule, and free to anyone and everyone! A few
 things to note. First, these vLectures are done via our state of the art –
 Online-HD-ILT training solution. Secondly, if you’re unable to attend – we
 will be posting all vLectures on our YouTube channel – and linking to them
 from our website! *To REGISTER, please visit this link.
 http://www.ipexpert.com/Company/vlecture-registration*

 Here’s our vLecture schedule for the remainder of the year. We will be
 posting 2014 Q1 within the next few weeks.

- Data Center: UCS Disjoint Layer 2 (October 28th, 3 pm EST) *(REGISTER)
http://www.ipexpert.com/Company/vlecture-registration*
- RS: MPLS 101 (Nov 6th, 10 am EST)*(REGISTER)
http://www.ipexpert.com/Company/vlecture-registration*
- Collaboration / Voice. H.323 FastStart (November 8th, 4 pm 
 EST)*(REGISTER)
http://www.ipexpert.com/Company/vlecture-registration*
- Security: Access Point 802.1x (Nov 13th, 10am EST) *(REGISTER)
http://www.ipexpert.com/Company/vlecture-registration*
- Wireless: CAPWAP AP controller discovery process  (November 18th, 1
pm EST) *(REGISTER)
http://www.ipexpert.com/Company/vlecture-registration*
- Data Center: EvPC and FCoE (November 18th, 3 pm EST)*(REGISTER)
http://www.ipexpert.com/Company/vlecture-registration*
- Data Center: MDS FC Security Features (November 28th, 3 pm 
 EST)*(REGISTER)
http://www.ipexpert.com/Company/vlecture-registration*
- Security: IKEv2 L2L VPN (Dec 4th, 10am EST) *(REGISTER)
http://www.ipexpert.com/Company/vlecture-registration*
- RS: Multicast basics (Dec 6th, 10 am EST) *(REGISTER)
http://www.ipexpert.com/Company/vlecture-registration*
- Collaboration / Voice: 3750 Quality of Service (December 6th, 4 pm
EST)*(REGISTER)
http://www.ipexpert.com/Company/vlecture-registration*
- Wireless: Troubleshooting the CAPWAP AP Join process (December 9th,
1 pm EST) *(REGISTER)
http://www.ipexpert.com/Company/vlecture-registration*

 The following 2 vLectures are available now – for viewing (delivered last
 week):

 *Security: ISE Profiling – Device Sensor
 http://www.youtube.com/watch?v=h571K91GWCAfeature=youtu.be*

 *RS:  http://www.youtube.com/watch?v=mGblyTKAGIkPrivate VLANs –
 Configuration, Operation and Troubleshooting
 http://www.youtube.com/watch?v=mGblyTKAGIk*


 Regards,



 Wayne A. Lawson II - CCIE #5244

 Founder  CEO - IPexpert, Inc.


 :: Message sent from iPhone

 On Oct 25, 2013, at 6:02 AM, Bill Hatcher wchatc...@gmail.com wrote:

 Stefano,

 When you go to that URL with your browser, using Internet Explorer go to
 view and source.  This will allow you to see the xml format on the page.
 You should be able to edit from that.

 Bill Hatcher


 On Thu, Oct 24, 2013 at 8:08 AM, StefanoS stefan...@gmail.com wrote:

 Hey Somphol thank you for your reply.

 You mean this?:
 http://PUB-IP:8080/ccmcip/xmldirectory.jsp
 There must be something that we have to add in this URL I believe because
 as it is doesn't show anything. I remember something similar but I can't
 figure it out right now. I'm exercising on something else. So if you can
 explain further it would be perfect :)

 I added the IP Phone Customization links for things like XML writing,
 Background changes for the 7965 etc

 Stefanos


 On Thu, Oct 24, 2013 at 3:53 PM, Somphol Boonjing somp...@gmail.comwrote:

 Hi StefanoS,

 Just a tiny addition on the good collection of the link you've gathered,
 with the long navigation to Cisco IP Phone customization, you may want to
 get the the template from

 The url directories under enterprise parameters

 There you will see a URL that you can use to retrieve a template.

 If I am not mistaken, I think this tip is from one of William Bell's
 email or one of his blog posts.

 Regards,
 --Somphol


 On Thu, Oct 24, 2013 at 11:04 PM, StefanoS stefan...@gmail.com wrote:

 Oh never mind I found it! It looks like I'm too tired after all.

 So it's under Products  Unified Communications  Call Control  Mid
 Market Call Control  Cisco Unified Communication Manager Express

 Anyway since I've started this maybe we could make a new list with the
 new paths we might need

Re: [OSL | CCIE_Voice] Voice Lab dates all gone?

2013-10-30 Thread Ramcharan Arya
Hi Bill,

You can  open a support case with Cisco Certification support team
www.cisco.com/go/certsupport. they might be able to assist you if any dates
available.

I had similar issue in 2009 when RS blueprint 4.0 change announced.

Thanks,
Ramcharan Arya
CCIE # 28926 (Voice/Routing  Switching)



On Wed, Oct 30, 2013 at 4:21 PM, Bill Tolentino btolent...@hotmail.comwrote:

 Hello everyone,
 Kind of off topic for most as I'm sure none of you waited until today to
 schedule/pay for your voice lab, unlike myself.  I just wanted to confirm
 from someone else that has access to the online scheduling site.
 Two days ago there were 8 seats left in San Jose.  Last night there was
 5.  This morning at 7am Pacific (Wed Oct 30th, 2013) there were no seats
 left in San Jose.  Fine, I looked up other sites and RTP had about 5-6
 seats (within the next 10 days).  Tokyo had about 8 or so.  Bangalore had
 many open seats (20+) over the next three months.  I think Vancouver had
 one seat.  Then at approximately 7:10am they were all gone.  I have been
 re-checking all day and still no dates for any sites.?  Technically someone
 can be dropped from their 90+ day reservation date over the next 2 weeks
 and a slot may open, assuming they have not yet paid.  Then there's a slim
 shot at sneaking in...  Can someone confirm all slots have been filled?

 Thanks,

 Bill Tolentino


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

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Re: [OSL | CCIE_Voice] Outbound PSTN call drops

2013-10-21 Thread Ramcharan Arya
Hi,

What is the settings in your gateway CUCM configuration. Have you checked
early offer option.

Please send debug of following command.


debug voip ccapi inout

debug h225 q931

debug h245 asn1

Thanks,
Ramcharan Arya CCIE # 28926 (Voice/RS)




On Mon, Oct 21, 2013 at 12:32 PM, Samson Kareem
samson_kar...@hotmail.comwrote:

 Hi All,

 I am hitting an issue where outbound PSTN calls are failing after the call
 initially seems to setup ok.
 Then the H323 gateway sends a disconnect to the PSTN and the call drops as
 per below.


 Oct 21 16:42:59.876: ISDN Se0/3/0:15 Q931: TX - SETUP pd = 8  callref =
 0x008C
 Bearer Capability i = 0x8090A3
 Standard = CCITT
 Transfer Capability = Speech
 Transfer Mode = Circuit
 Transfer Rate = 64 kbit/s
 Channel ID i = 0xA98383
 Exclusive, Channel 3
 Display i = 'Tobias Funke'
 Calling Party Number i = 0x1181, '17707022001'
 Plan:ISDN, Type:International
 Called Party Number i = 0x91, '97148037333'
 Plan:ISDN, Type:International
 Oct 21 16:42:59.908: ISDN Se0/3/0:15 Q931: RX - CALL_PROC pd = 8  callref
 = 0x808C
 Channel ID i = 0xA98383
 Exclusive, Channel 3
 Oct 21 16:42:59.916: ISDN Se0/3/0:15 Q931: RX - ALERTING pd = 8  callref
 = 0x808C
 Progress Ind i = 0x8188 - In-band info or appropriate now available
 Oct 21 16:42:59.944: ISDN Se0/3/0:15 Q931: TX - DISCONNECT pd = 8
 callref = 0x008C ==
 Cause i = 0x80AF - Resource unavailable, unspecified
 Oct 21 16:42:59.952: ISDN Se0/3/0:15 Q931: RX - RELEASE pd = 8  callref =
 0x808C
 Oct 21 16:42:59.956: ISDN Se0/3/0:15 Q931: TX - RELEASE_COMP pd = 8
 callref = 0x008C


 I first thought it was a DSP resource issue as the disconnect cause is
 Cause i = 0x80AF - Resource unavailable, unspecified
 but I checked the status of DSPs using #show voice dsp group all and no
 problems there.

 I found an old post from OSL Nov 2011 where someone suggested a codec
 mismatch with the H245 negotiation which causes
 the call to drop immediately after the alerting message.

 The thing is I have a voice class codec configured and in any case, the
 gateway was originally an MGCP gateway and calls were failing with the same
 ISDN disconnect cause (I only changed it to a H323 g/w for troubleshooting
 purposes).

 I tried using debug cch323 h245 and have pasted the output received after
 the ISDN alerting message below. I believe this is when codec negotiation
 takes place on the voip leg of the call between CUCM and the g/w.

 Oct 21 16:25:16.653:
 //51/8043215D0100/H323/cch323_h245_channel_established_ind: Using fd=4 to
 send msgs
 Oct 21 16:25:16.653:
 //51/8043215D0100/H323/cch323_send_event_to_h245_connection_sm: Changing to
 new event H245_ESTABLISHED_EVENT
 Oct 21 16:25:16.653: //51/8043215D0100/H323/cch323_h245_connection_sm:
 H245_LISTEN: Received event H245_ESTABLISHED_EVENT while at H245_WAITING
 state
 Oct 21 16:25:16.653: //51/8043215D0100/H323/cch323_h245_set_new_state:
 Changing from H245_WAITING state to H245_CONNECTED state
 Oct 21 16:25:16.653: //51/8043215D0100/H323/run_h245_iwf_sm: received
 IWF_EV_H245_CONNECTED while at state IWF_IDLE
 Oct 21 16:25:16.653: //51/8043215D0100/H323/h245_iwf_set_new_state:
 changing from IWF_IDLE state to IWF_AWAIT_CAP_MSD_RESP state
 Oct 21 16:25:16.657: //51/8043215D0100/H323/cch323_run_h245_cap_out_sm:
 Received H245_EVENT_CAP_REQ while at state IDLE
 Oct 21 16:25:16.657: //51/8043215D0100/H323/h245_cap_out_set_new_state:
 changing from IDLE state to AWAITING_RESPONSE state
 Oct 21 16:25:16.657: //51/8043215D0100/H323/cch323_run_h245_ms_sm:
 Received event H245_EVENT_MSD while at state H245_MS_NONE
 Oct 21 16:25:16.657: //51/8043215D0100/H323/cch323_run_h245_ms_sm: Sent
 MSD Request
 Oct 21 16:25:16.657: //51/8043215D0100/H323/h245_ms_set_new_state:
 Changing from H245_MS_NONE state to H245_MS_OUTGOING_WAIT state

 does anyone have any pointers?

 thanks
 Samson


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Re: [OSL | CCIE_Voice] CME as SRST help.

2013-10-17 Thread Ramcharan Arya
Hi Bill,

I believe this might be related to a bug with using octo lines in CME SRST.

Come out of SRST and reload the router this might resolve the issue.

Regards,
Ramcharan Arya CCIE # 28926 ( Voice/Routing  Switching)




On Thu, Oct 17, 2013 at 9:34 AM, Bill Hatcher wchatc...@gmail.com wrote:

 It’s not working!!  Can anyone see something I may be doing wrong? My PRI
 and CUE register, I can even see SIP MWI being sent, but my phones will not
 register.  They worked when I was using call-manager-fallback though so I
 know my SRST configuration is correct on the CallManager.



 telephony-service

  srst mode auto-provision all

  srst ephone description auto provisioned ephone  : Oct 17 2013 14:47:07

  srst dn line-mode octo

  max-ephones 4

  max-dn 4

  ip source-address 10.10.202.1 port 2000

  max-conferences 12 gain -6

  transfer-system full-consult

  create cnf-files version-stamp 7960 Oct 17 2013 16:07:18



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 www.PlatinumPlacement.com

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Re: [OSL | CCIE_Voice] ccie written

2013-10-07 Thread Ramcharan Arya
Hi Karen,

When I passed my CCIE voice written exam after that my CCIE RS
automatically got renewed.


Regards,
Ramcharan Arya CCIE # 28926 ( Voice/Routing  Switching)





On Mon, Oct 7, 2013 at 12:19 PM, Karen Johnson karen.johnson...@yahoo.cawrote:

 hi Arya and all,

 when you have 2 ccie specialization, do you need to write WRITTEN exam
 for both or just one ?

 K

   *From:* Ramcharan Arya ramcharan.a...@gmail.com
 *To:* Josh Petro josh.pe...@gmail.com
 *Cc:* ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com;
 sanity insanity networksanitytoinsan...@gmail.com
 *Sent:* Wednesday, October 2, 2013 6:59:08 PM
 *Subject:* Re: [OSL | CCIE_Voice] Presence - on hook and off hook status

   Hi Josh,

 I do not believe it is related to vmware environment. I am assuming your
 CUPS is integrated with CUCM using SIP trunk.

 Can you enable SIP debug level to detail and run collect SIP debug logs (
 on primary call processing engine i.e. Sub) and check SIP logs  why there
 is delay in status.?

 Im my home lab I never had this issue it works almost instantly my CUPS
 client is installed on UCCX server.

 Regards,
 Ramcharan Arya CCIE # 28926 ( Voice/RS)



  On Wed, Oct 2, 2013 at 7:47 PM, Josh Petro josh.pe...@gmail.com wrote:

 I have a huge delay in my presence updates on my system. Im assuming
 thatapos;s from CUPS being installed in my lab vmware environment though.
 Try to go off hook or call a pstn number and let it sit for 1-2 minutes. If
 anyone knows how to fix the lag, please let me know. Im assuming its
 related to vmware.
 Josh
  On Oct 1, 2013 11:20 AM, Ovidiu Popa ovi.p...@gmail.com wrote:

  Hi

 You could try a reboot of the CUPS server. Worked for me a couple of
 times...

 Cheers,
 Ovidiu


 On Tue, Oct 1, 2013 at 4:43 PM, Martin Sloan martinsloa...@gmail.comwrote:

  Hi MJ,

 Is the end user assigned on the line level of the hard phone?  That
 assignment is unique per line appearance so if you make the association on
 the CUPC device it does not automatically populate to the hard phone/any
 other line appearance.  When the phone goes off-hook CUCM checks the end
 user assignment for that appearance and if there is an end user assigned it
 check whether that end user is assigned CUP licensing to decide if the
 publish message is sent over the CUPS SIP trunk.

 BR,
 Marty


  On Tue, Oct 1, 2013 at 10:29 AM, sanity insanity 
 networksanitytoinsan...@gmail.com wrote:

 Hello all,

 I have configured presence and both softphone and deskphone modes , IM
 and voicemail is working fine on the clients

 However I have a question when I lift the handset of the phone ( hard
 phone ) that is assoicated
 with the CUPC clients . I see that the presence status does not show  
 On the phone   and does not turn yellow.

 I have tried reseting my sip trunk pointing to the presence server yet I
 see the same issue.

 Please let me know what can be done to fix this ?  Also is this  a major
 issue ?

 -MJ

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 Are you a CCNP or CCIE and looking for a job? Check out
 http://www.platinumplacement.com/



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

 Are you a CCNP or CCIE and looking for a job? Check out
 http://www.platinumplacement.com/



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

 Are you a CCNP or CCIE and looking for a job? Check out
 http://www.platinumplacement.com/


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

 Are you a CCNP or CCIE and looking for a job? Check out
 http://www.platinumplacement.com/



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

 Are you a CCNP or CCIE and looking for a job? Check out
 www.PlatinumPlacement.com



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

Are you a CCNP or CCIE and looking for a job? Check out 
www.PlatinumPlacement.com

Re: [OSL | CCIE_Voice] Issues in opening the workbook and accessing my account

2013-10-05 Thread Ramcharan Arya
Hi,

I am able to access my account and able to open workbook successfully.

Have you updated your Adobe reader recently if yes then you might have to
install correct fileopen plug-in.

Thanks,
Ramcharan Arya CCIE # 28926(Voice/Routing  Switching)



On Sat, Oct 5, 2013 at 8:04 AM, Vignesh Sethuraman
sethuvign...@gmail.comwrote:

 Dear All,

 I am facing issue in opening the PDF workbook and accessing my account in
 ipexpert today.

 I tried to send e-mail to supp...@ipexpert.com but still waiting for the
 answer.

 Just wanted to ensure if this problem exists only for me or to every
 ipexpert customer.

 Thanks,
 Viki

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Re: [OSL | CCIE_Voice] Presence - on hook and off hook status

2013-10-02 Thread Ramcharan Arya
Hi Josh,

I do not believe it is related to vmware environment. I am assuming your
CUPS is integrated with CUCM using SIP trunk.

Can you enable SIP debug level to detail and run collect SIP debug logs (
on primary call processing engine i.e. Sub) and check SIP logs  why there
is delay in status.?

Im my home lab I never had this issue it works almost instantly my CUPS
client is installed on UCCX server.

Regards,
Ramcharan Arya CCIE # 28926 ( Voice/RS)



On Wed, Oct 2, 2013 at 7:47 PM, Josh Petro josh.pe...@gmail.com wrote:

 I have a huge delay in my presence updates on my system. Im assuming
 that's from CUPS being installed in my lab vmware environment though. Try
 to go off hook or call a pstn number and let it sit for 1-2 minutes. If
 anyone knows how to fix the lag, please let me know. Im assuming its
 related to vmware.
 Josh
 On Oct 1, 2013 11:20 AM, Ovidiu Popa ovi.p...@gmail.com wrote:

 Hi

 You could try a reboot of the CUPS server. Worked for me a couple of
 times...

 Cheers,
 Ovidiu


 On Tue, Oct 1, 2013 at 4:43 PM, Martin Sloan martinsloa...@gmail.comwrote:

 Hi MJ,

 Is the end user assigned on the line level of the hard phone?  That
 assignment is unique per line appearance so if you make the association on
 the CUPC device it does not automatically populate to the hard phone/any
 other line appearance.  When the phone goes off-hook CUCM checks the end
 user assignment for that appearance and if there is an end user assigned it
 check whether that end user is assigned CUP licensing to decide if the
 publish message is sent over the CUPS SIP trunk.

 BR,
 Marty


 On Tue, Oct 1, 2013 at 10:29 AM, sanity insanity 
 networksanitytoinsan...@gmail.com wrote:

 Hello all,

 I have configured presence and both softphone and deskphone modes , IM
 and voicemail is working fine on the clients

 However I have a question when I lift the handset of the phone ( hard
 phone ) that is assoicated
 with the CUPC clients . I see that the presence status does not show
   On the phone   and does not turn yellow.

 I have tried reseting my sip trunk pointing to the presence server yet
 I see the same issue.

 Please let me know what can be done to fix this ?  Also is this  a
 major issue ?

 -MJ

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Re: [OSL | CCIE_Voice] unity connection - transfer option for users.

2013-10-02 Thread Ramcharan Arya
Hi Marty,

In order to preserve Original calling party TON you have to consider
existing route pattern should not override  so two possible ways to achieve
this you can try to use a translation patter which evaluation prior to
route pattern. Let us assume you prefix some additional character and
create clng party x-formation pattern and DDI -predot what was prefix in TP
and set appropriate plan and type. Use separate pt/css for clng party
x-formation pattern.

Another option is using application dial-rule can also use for this.

Regards,
Ramcharan Arya CCIE # 28926 (Voice/Routing  Switching)


On Wed, Oct 2, 2013 at 12:53 PM, Martin Sloan martinsloa...@gmail.comwrote:

 Yeah, I wasn't sure on that one either and had to test it out.  I can't
 recall what the exact requirement, if any, for calling party TON was on the
 'practice test' that I had with a similar task but I'm thinking the only
 way to properly set the calling TON would be with Xforms on the port level
 since it could be any number on the PSTN phone, even the number you're
 trying to dial out to from VM.  They'd have to be very specific Xforms
 though since it could potentially override the current dial-plan
 manipulations in RL and RP if general masks are used like 10 X's.


 On Wed, Oct 2, 2013 at 1:21 PM, Justin Carney 
 justin.s.car...@gmail.comwrote:

 I wasn't sure RDNIS would matter here but figured I would throw it out
 there anyway (as it applies when redirecting TO CUC).  It seems the unity
 service parameter mentioned earlier obviates the need to use RDNIS.

 With the option you proposed on creating a new RP/RL just for this
 requirement I would just set the digit manipulation/TON on the RL to
 whatever you see inbound from that specific PSTN ANI to HQ - unless the
 question told you what the expected outbound ANI/TON should be.  Another
 option would be to compare the original PSTN number with the destination
 PSTN and set to local if same NPA, LD if different NPA, or international
 different country codes.  If it comes in unknown/unknown then send it back
 out that way.


 On Wed, Oct 2, 2013 at 12:48 PM, Martin Sloan martinsloa...@gmail.comwrote:

 The RDNIS shouldn't be a factor here.  I just labbed this up and there
 is no Redirecting Number IE in the ISDN messages for this scenario.  It's
 more of a straight dial from Unity.

 I think the places to be checked are:

 CUCM service parameter
 Call Routing Path

 Whatever Route Pattern - Route List is being used needs to have the
 Use Calling Party's External Phone Number Mask checked and no masking
 being done below, like truncating the calling party number to 7 digits if
 that was part of the requirement for the sites local PSTN dialing.  I
 recommend partitioning out a new pattern that matches the number you're
 trying to dial and handling the digit manipulation separately from the rest
 of the dial plan to keep it conceptually simple, but not necessarily
 'cleaner'.  Kind of along the lines of keeping AAR, CFUR, SNR separate.

 As for the calling party TON on this, your guess is as good as mine.  If
 the task doesn't specifically ask to set the calling party TON and it says
 to use any line from the PSTN phone, what do you do?

 Marty


 On Wed, Oct 2, 2013 at 12:14 PM, Justin Carney 
 justin.s.car...@gmail.com wrote:

 What do you see on the voice gateway for ANI/DNIS of the two separate
 calls - inbound (PSTN to Phone (which is then CFNA to CUC)) and outbound
 (CUC to PSTN alt dest 9515)?

 Take a look your gateways settings for Redirecting Number IE Delivery
 (RDNIS) for both inbound/outbound.  These checkboxes are adjacent to the
 Display IE Delivery (which is usually turned on).

 To test and understand the behavior of these settings I would recommend
 ticking these boxes on/off and retrying your inbound/outbound calls in this
 (and other) scenario.  As a test try setting up a call such as PSTN  SA
 phone  CFA to a different PSTN number and look at the q931 debugs for
 ANI/DNIS/RDNIS.

 I haven't tested this recently and not sure if it applies in your
 stated scenario but try checking the box on SA gateway for the outbound
 RDNIS.  This should allow CUC to send 3 IE out to the PSTN - the original
 ANI (PSTN caller), the redirecting number/RDNIS (would expect this to be
 either the DN of original called IP phone or possibly the vm port/pilot's
 DN/EPNM) and lastly the DNIS should be 9515.

 For a different scenario with SB in SRST - when a call to a SB phone
 does CFNA to vm you will send it out the PSTN to HQ vm pilot and on SA gw
 you need to allow (check) the inbound RDNIS.  In this case the IE at SA
 router is ANI-original caller, DNIS-DID of vm pilot, RDNIS-the SB phone's
 DN.



 -Justin


 On Wed, Oct 2, 2013 at 10:56 AM, ramesh rameshdol...@rediffmail.comwrote:

 hi Martin,

 I have done that ( preserve original calling number ) in unity
 however it does not preserve the correct calling number for pstn callers .
 Also the type and plan of the called

Re: [OSL | CCIE_Voice] Unity express

2013-09-19 Thread Ramcharan Arya
Can you send below command output from CUE CLI.

trace networking smtp all



Thanks,
Ramcharan Arya CCIE # 28926 (Voice/Routing  Switching)




On Thu, Sep 19, 2013 at 12:35 PM, Dharambir kumar varma 
dharambi...@gmail.com wrote:

 Hi All,

 users are not getting voice mail notification on email..
 please guide...
 --
  Regards,
  Dharambir Kumar
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Re: [OSL | CCIE_Voice] Unity express

2013-09-19 Thread Ramcharan Arya
You need to figure out how to access CUE using CLI.

ssh/telnet into your router
type below command ( depend up on module)

service-module service-engine x/y session

once you have to CUE using CLI then run debug command.






On Thu, Sep 19, 2013 at 3:18 PM, Dharambir kumar varma 
dharambi...@gmail.com wrote:

 sir i dont have access on CME ...
 but we can acess only cue through GUI...
 is there any other way???
 i am confused about IMAP..it shows '' no active IMAP session

 but i have configured all parameters on CUE
 Lke SMTP setting /Domain setting/User notification enable/user email
 address.
 is anything requiredto configure on exchange server side...

 please suggest where can i find info.

 Thanks


 On 9/20/13, Ramcharan Arya ramcharan.a...@gmail.com wrote:
  Can you send below command output from CUE CLI.
 
  trace networking smtp all
 
 
 
  Thanks,
  Ramcharan Arya CCIE # 28926 (Voice/Routing  Switching)
 
 
 
 
  On Thu, Sep 19, 2013 at 12:35 PM, Dharambir kumar varma 
  dharambi...@gmail.com wrote:
 
  Hi All,
 
  users are not getting voice mail notification on email..
  please guide...
  --
   Regards,
   Dharambir Kumar
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 please
  visit www.ipexpert.com
 
  Are you a CCNP or CCIE and looking for a job? Check out
  www.PlatinumPlacement.com
 
 


 --
  Regards,
  Dharambir Kumar

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[OSL | CCIE_Voice] CCIE collaboration written - official study material availability

2013-08-08 Thread Ramcharan Arya
Hi,

Can someone please tell me about timeline when collaboration study material
will be available.

Regards,
Ramcharan Arya
CCIE # 28926 ( Voice/Routing  Switching)
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Re: [OSL | CCIE_Voice] SIP Early offer

2013-08-07 Thread Ramcharan Arya
Just go through this SRND article this might be useful.

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/7x/trunks.html#wp1045957

If you have IPX 5 lab handbooks CCIE-Voice-5-Lab-Handbook-
-Workbook-Lab-5.pdf sec 4.5 SIP Trunk Troubleshooting Vik has covered  SIP
trunk troubleshooting explanation in great detail.

HTH

Thanks,
Ramcharan Arya- CCIE # 28926 ( Voice/Routing  Switching)



On Wed, Aug 7, 2013 at 9:34 PM, Karen Johnson karen.johnson...@yahoo.cawrote:

 IOS mtp, i understand the explicit mtp from ucm.

 but this must used IOS mtp , because if I do not put MGRL that only
 contacin IOS MTP and not check MTP in trunk, i will not work as Early.

 K

   *From:* Rafael Chavantes raf...@chavantes.com
 *To:* Karen Johnson karen.johnson...@yahoo.ca
 *Cc:* ccie_voice@onlinestudylist.com
 *Sent:* Wednesday, August 7, 2013 2:11:28 PM
 *Subject:* Re: [OSL | CCIE_Voice] SIP Early offer

  Are you using mtp on ios or on cucm?
 Remember that for early offer you can use the software mtp on cucm
 Also remember that if cucm mtps are not associated to any Mrg, they reside
 on the none mrg, and be used even if not explicity selected on any mrg
 Em 07/08/2013 15:19, Karen Johnson karen.johnson...@yahoo.ca escreveu:


 all,

 when i check MTP in SIP trunk and use MRGL, call become SIP Early offer.
 However why if I do  sh sccp connection  , I did not see SIP Early offer
 use my MTP resource at all?

 K

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Re: [OSL | CCIE_Voice] !!End users permissions and groups!!

2013-08-01 Thread Ramcharan Arya
Hi,

IMO you may  want to avoid giving AXL Admin permission to end users.

Regards,
Ramcharan Arya - CCIE # 28926 (Voice/Routing  Switching)



On Thu, Aug 1, 2013 at 7:45 PM, Drake J jdrake...@gmail.com wrote:

 Hello All,

 In the callmanger If the end users are already created such as HQ1, HQ2.,
 HQ3, SB1 and so on ... and if we add all  roles  and and groups to them
 under permissions  then will this be wrong or should not be done ? Could
 someone comment on this? Are we allowed to  do this?

 -Drake

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Re: [OSL | CCIE_Voice] NO Extension in CME-SRST

2013-08-01 Thread Ramcharan Arya
Check your WAN QOS  one-way compression might be causing this.

Regards,
Ramcharan Arya - CCIE # 28926 ( Voice/Routing  Switching)



On Thu, Aug 1, 2013 at 8:27 PM, Amdo Ngawa datapack...@gmail.com wrote:

 I am having similar issue. No resolution so far? Will read and google.

 Thanks


 On Thu, May 16, 2013 at 4:21 PM, CISCO CCIE VOICE 
 ccievoic...@gmail.comwrote:

 HI experts,

 When the Phones are in CME SRST its does not shows Extension on Phone
 display this  happens when  i change the name +86223033001 to SiteC Phone 1
 under Ephone-dn's

 Thnks






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[OSL | CCIE_Voice] uccx script : get reporting statistics playing negative value

2013-07-11 Thread Ramcharan Arya
Hi,

My UCCX script playing negating output value from get reporting statistics.


Can someone please give expert advice what could be wrong in uccx script
which result into negative output.

Thanks  Regards,
Ramcharan Arya
CCIE # 28926 (RS)
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Re: [OSL | CCIE_Voice] AAR and uccx

2013-07-10 Thread Ramcharan Arya
Hi Piyush,

I assigned HQ device pool and location setting on CTI route point and CTI
ports to HQ.

Following is isdn debug output when calling from 4002 to 4000 ( during AAR)

HQ(config)#
Jul 10 22:15:08.149: ISDN Se0/0/0:23 Q931: TX - SETUP pd = 8  callref =
0x000F
Bearer Capability i = 0x8090A2
Standard = CCITT
Transfer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA98397
Exclusive, Channel 23
Calling Party Number i = 0x1181, '85224044011'
Plan:ISDN, Type:International
Called Party Number i = 0x91, '85224044102'
Plan:ISDN, Type:International
Jul 10 22:15:08.201: ISDN Se0/0/0:23 Q931: RX - CALL_PROC pd = 8  callref
= 0x800F
Channel ID i = 0xA98397
Exclusive, Channel 23
HQ(config)#
Jul 10 22:15:08.329: ISDN Se0/0/0:23 Q931: RX - ALERTING pd = 8  callref =
0x800F
Progress Ind i = 0x8088 - In-band info or appropriate now available
Jul 10 22:15:08.341: ISDN Se0/0/0:23 Q931: TX - DISCONNECT pd = 8  callref
= 0x000F
Cause i = 0x8290 - Normal call clearing

Site C phone 2 Line 2 4102 ring and call disconnect in 2 sec.

with below error message

Jul 10 22:16:48.524: ISDN Se0/0/0:15 Q931: TX - SETUP pd = 8  callref =
0x0048
Sending Complete
Bearer Capability i = 0x8090A2
Standard = CCITT
Transfer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA9838B
Exclusive, Channel 11
Display i = 'SiteC Phone 2'
Calling Party Number i = 0x1183, '+85224044002'
Plan:ISDN, Type:International
Called Party Number i = 0x91, '85224044010'
Plan:ISDN, Type:International
Jul 10 22:16:48.524: ISDN Se0/0/0:15 Q931: TX - RELEASE pd = 8  callref =
0x80D5
Jul 10 22:16:48.780: ISDN Se0/0/0:15 Q931: RX - STATUS pd = 8  callref =
0x00D5
Cause i = 0x80E202 - Message not compatible with call state or not
implemented
Call State i = 0x0C
Jul 10 22:16:48.780: ISDN Se0/0/0:15 Q931: RX - RELEASE pd = 8  callref =
0x8047
Jul 10 22:16:48.780: ISDN Se0/0/0:15 Q931: RX - RELEASE_COMP pd = 8
callref = 0x00D5

Basically when calling from HQ PH1 to 4000 ( in AAR) it work without any
issue.


Thanks,
Ramcharan Arya



On Tue, Jul 9, 2013 at 10:37 PM, jainpiyush2...@ymail.com wrote:

 Hello Ram,

 You can assign Hq device pool and location setting to cti route point and
 cti ports..
 And assign site c device pool and location to site c phones...

 Regards,
 Piyush Jain

 Sent from my android device.




 -Original Message-
 From: Ramcharan Arya ramcharan.a...@gmail.com
 To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
 Sent: Tue, 09 Jul 2013 7:59 AM
 Subject: [OSL | CCIE_Voice] AAR and uccx

 Hi,
 I have a CTI Route point 4000 and two CTI port 410x and 410y



 SiteC Phone 1 and SC Phone 2 are in CSQ which is assign to application and
 associated with trigger.

 Due to  RSVP when call exceed ip rsvp bandwidth call  from uccx to site
 Phones should use AAR and to over PSTN.

 My doubts are .

 What should be local and device pool of CTI ports so it should work in AAR
 when PSTN caller make call to CTI route point number 4000.

 Can someone please advice about this.

 Thanks,
 Ramcharan Arya


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[OSL | CCIE_Voice] AAR and uccx

2013-07-08 Thread Ramcharan Arya
Hi,
I have a CTI Route point 4000 and two CTI port 410x and 410y



SiteC Phone 1 and SC Phone 2 are in CSQ which is assign to application and
associated with trigger.

Due to  RSVP when call exceed ip rsvp bandwidth call  from uccx to site
Phones should use AAR and to over PSTN.

My doubts are .

What should be local and device pool of CTI ports so it should work in AAR
when PSTN caller make call to CTI route point number 4000.

Can someone please advice about this.

Thanks,
Ramcharan Arya
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[OSL | CCIE_Voice] CME Date-format in SRST show mm/dd/yy

2013-06-19 Thread Ramcharan Arya
Hi,
I have some issue with date-format for SC phones.

On CUCM date-format is mm-dd-yy

When site C is in SRST mode. phone date-format change to mm/dd/yy.

I have specify  date-format  in telephony-service

date-format mm-dd-yy

I have recreate CNF files for phones phone in SRST always show date format

can someone please suggest what is the correct way to keep date-formate
mm-dd-yy while phones are in SRST mode.

Thanks  Regards,
Ramcharan Arya
CCIE # 28926 ( RS)
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[OSL | CCIE_Voice] CUCM Privacy button display as 'Private' Is this a bug.?

2013-06-19 Thread Ramcharan Arya
Hello,

When I assign a privacy button on HQ phone. my display show private not
privacy as per label.Is this a cosmetic bug in CUCM.?

Can someone please confirm.

Thanks  Regards,
Ramcharan Arya
CCIE # 28926 (RS)
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Re: [OSL | CCIE_Voice] H323 fast start question

2013-06-03 Thread Ramcharan Arya
According my interpretation If requirement say collecting CUCM traces
then you might want to think about using RTMT tool for logs.

Debug h225 ans1 command you can run output on IOS router which does not
gives an output from CUCM trace.


On Mon, Jun 3, 2013 at 8:06 PM, singh singh8...@in.com wrote:


 HI All,

 If we are required to setup fast between H323 gateway and callmanger and
 indicate the ...

 a) Inbound fast start to CUCM
 b) Outbound fast start from CUCM


 Can we therefore just collect debug h225 asn1 and indicate the above ?
 or is it required to demonstrate this by collecting CUCM traces ?


 -singh


 Get Yourself a cool, short *@in.com* Email ID 
 now!http://www3.in.com/sso/commonregister.php?ref=INutm_source=inviteutm_medium=outgoing

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Re: [OSL | CCIE_Voice] Cisco: Provide a reasonable transition path from CCIE Voice to CCIE Collaboration

2013-05-31 Thread Ramcharan Arya
I signed as well.

Thanks,
Ramcharan Arya CCIE # 28926 ( RS)



On Fri, May 31, 2013 at 9:09 PM, Ikenna Izugbokwe 
ikenna.izugbo...@gmail.com wrote:

 Done.

 Ikenna Izugbokwe
 Former - CCIE #36,472 (Voice)


 On Fri, May 31, 2013 at 8:40 PM, Tian id21...@gmail.com wrote:

 DONE…

 ** **

 *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
 ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Edgar Feliz
 *Sent:* Friday, May 31, 2013 1:36 PM
 *To:* Suresh Bhandari
 *Cc:* ccie_voice@onlinestudylist.com
 *Subject:* Re: [OSL | CCIE_Voice] Cisco: Provide a reasonable transition
 path from CCIE Voice to CCIE Collaboration

 ** **

 DONE...My wife also posted it on FB.

 ** **

 On Fri, May 31, 2013 at 12:40 AM, Suresh Bhandari bring...@gmail.com
 wrote:

 *Cisco: Provide a reasonable transition path from CCIE Voice to CCIE
 Collaboration *- Sign the Petition!

 For the interested candidates...

 Please join this campaign: http://chn.ge/17A0zXE 

 I already did. Now its your turn.

 Initially shared by Martin Sloan (martinsloa...@gmail.com)
 


 --
 Suresh Bhandari


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 ** **

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 --
 I

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[OSL | CCIE_Voice] Unity Connection Restriction Tables setting for External Transfer

2013-05-20 Thread Ramcharan Arya
Hello,
I need clarification regarding when call goes into HQ Phone  and press key
1 to transfer call to PSTN Line 1.

Since this is external number 2123942123. Since this is external call and
Unity connection is using Default system transfer.

Do I need to remove block pattern from restriction table.

Please let me know your feedback/suggestions.

Thanks  Regards,
Ramcharan Arya
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Re: [OSL | CCIE_Voice] 5 Lab Handbook Lab 4 task 7.3 : Caller hear Ringback when agent phone ringing

2013-05-05 Thread Ramcharan Arya
Hi,
I have select codec g729 in IP voice streaming media apps service parameter and 
then restart IP voice streaming service on pub and sub.

Also on CTI port need to add network on hold music source.

If you watch vik's voice troubleshooting video it is covered also audio on 
demand Amy has also explained about this in great detail.

In vol 2 lab 7 is another good example.

Thanks,
Ramcharan Arya

Sent from my iPhone

On May 5, 2013, at 7:01 PM, Hesham Abdelkereem heshamcentr...@gmail.com wrote:

 Hi Ramarchan,
 
 I wonder if you have solved the issue as I am looking forward how we 
 accomplish that.
 If you solved it please share your solution if you don't mind.
 
 
 Thanks,
 Hesham
 
 
 On 27 April 2013 18:19, Ramcharan Arya ramcharan.a...@gmail.com wrote:
 Hi,
 
 I have configure ringback as per solutions guide when agent phone is ringing 
 caller hear tone of hold. I am using music on hold audio source 2 
 ringback2.wav file
 
 ringback file is upload in PUB and SUB. IP voice streaming media app service 
 restarted
 
 Network on hold is set as per solutions guide.PSTN caller and HQ caller hear 
 tone on hold.
 
 Can someone please suggest approach how to troubleshoot this issue.
 
 Thanks  Regards,
 Ramcharan Arya
 
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Re: [OSL | CCIE_Voice] rtp location start time

2013-05-01 Thread Ramcharan Arya
Food menu is not good. There were very  little options for vegetarians. I
was not happy at all.




On Wed, May 1, 2013 at 9:41 PM, Pavan K pav.c...@gmail.com wrote:

 I second that. Excellent experience at RTP. Wasn't impressed with the
 proctors at SJC.
 On May 1, 2013 7:52 PM, STEPHEN FREEBERG sfreeberg...@gmail.com wrote:

 RTP exams begin at 7:15 am, You should arrive no later than 7:00 am.

 David Blair is the proctor and in my opinion is the best proctor I have
 encountered at Cisco.

 Steve

 On May 1, 2013, at 5:06 PM, Karen Johnson karen.johnson...@yahoo.ca
 wrote:

 hi,

 anyone can share what start time in RTP and proctors there?

 tks
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[OSL | CCIE_Voice] 5 Lab Handbook Lab 4 task 7.3 : Caller hear Ringback when agent phone ringing

2013-04-27 Thread Ramcharan Arya
Hi,

I have configure ringback as per solutions guide when agent phone is
ringing caller hear tone of hold. I am using music on hold audio source 2
ringback2.wav file

ringback file is upload in PUB and SUB. IP voice streaming media app
service restarted

Network on hold is set as per solutions guide.PSTN caller and HQ caller
hear tone on hold.

Can someone please suggest approach how to troubleshoot this issue.

Thanks  Regards,
Ramcharan Arya
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Re: [OSL | CCIE_Voice] CME BACD dial-peer

2013-04-23 Thread Ramcharan Arya
Please remove port 1820 from VoIP service it will work

Sent from my iPhone

On Apr 23, 2013, at 9:54 PM, Martin Sloan martinsloa...@gmail.com wrote:

 Rafael,
 
 Thanks for the assist.  I've attached the sh run and debug.
 
 Marty
 
 
 On Tue, Apr 23, 2013 at 10:10 PM, Rafael Chavantes raf...@chavantes.com 
 wrote:
 Hello Martin,
 Can you please send the sh run and the output for debug ccapi inout?
 
 
 On Tuesday, April 23, 2013, Martin Sloan wrote:
 Hello experts,
 
 I'm having some trouble with the BACD dial-peer in vol 1 WB section 9.2.  
 I'm following the CUCME BACD 'tcl in flash mem' guide as recommended by Vik 
 but I'm getting fast busy when dialing into the aa.  If I bring up a POTS 
 dial-peer for PSTN-AA or if I modify session target to ras in the voip 
 dial-peer for IP Phone-AA, the aa works so I at least know that part is 
 solid. 
 
 Here's the dial-peer config I'm using that isn't working:
 
 dial-peer voice 222 voip
  service aa
  destination-pattern 3500
  session target ipv4:10.10.110.3
  incoming called-number 3500
  dtmf-relay h245-alphanumeric
  codec g711ulaw
  no vad
 
 Also:
  allow-connections h323 to h323
  allow-connections h323 to sip
  allow-connections sip to h323
  allow-connections sip to sip
 
 And a snip from the end of a voip dialpeer debug:
 
 Apr 24 00:52:24.603: //-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
 Apr 24 00:52:24.603: //-1//DPM/dpAssociateIncomingPeerCore:
Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=222
 Apr 24 00:52:24.623: //-1//DPM/dpAssociateIncomingPeerCore:
Calling Number=3006, Called Number=, Voice-Interface=0x0,
Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search 
 Type=PEER_TYPE_VOICE,
Peer Info Type=DIALPEER_INFO_SPEECH
 Apr 24 00:52:24.623: //-1//DPM/dpAssociateIncomingPeerCore:
Result=Success(0) after DP_MATCH_ORIGINATE; Incoming 
 Dial-peer=40001-dial-peer)#
 
 I see in the debug it matches 222 as the inbound dial-peer but then it also 
 matches 40001 for the SIP CME phone.  Any help is much appreciated.
 
 Marty
 
 
 -- 
 Rafael Chavantes
 
 BR2_sh_run.txt
 BACD_ccapi_inout.txt
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Re: [OSL | CCIE_Voice] OWLE Lab 1 Task 2.4 No Service Configured

2013-04-22 Thread Ramcharan Arya
Hi Edgar,

I tried this solution in my pod today. Message  No Service configured
show under directory. Earlier requirement was to display this message as
soon as directory button depress.

Can you write down detail steps may be I am doing something wrong.

Thanks  regards,
Ramcharan Arya
CCIE # 28926 (RS)



On Wed, Apr 17, 2013 at 7:19 PM, Edgar Feliz ejzi...@gmail.com wrote:

 much easier way is to go to go to CCX create a file in notepad that says
 No Services configured save it as scph1.html  etcset you enterprise
 parameter to both...on your phone set it to external and on site c ph1
  Directories url enter. http://ip address of CCX/scphn1.html


 On Wed, Apr 17, 2013 at 2:57 PM, Ramcharan Arya 
 ramcharan.a...@gmail.comwrote:

 Thank you William for detail steps. I will try this afternoon in my pod.

 In summary : For No service configuration on phone I should use solution
 describe either in OWLE lab 4 solutions guide or below describe in email.

 Regards,
 Ramcharan Arya
 CCIE # 28926


 On Wed, Apr 17, 2013 at 9:09 AM, William Bell b...@ucguerrilla.comwrote:

 Right. The order you place the MenuItem child nodes in the XML file will
 be the order they are presented on the phone.

 What I do when practicing lab 4 was as follows:

 *CCX*

 1. RDP to CCX
 2. From CCX browse to xml directory URL  (copied from Ent Params in CUCM)
 3. Copy XML from CCX browser to notepad


 *CUCM* / *CLI*

 1. Go to CLI on CUCM and run query

 admin:*run sql select name,urltemplate from telecasterservice*
 nameurltemplate
 ===
 
 Missed CallsApplication:Cisco/MissedCalls
 Received Calls  Application:Cisco/ReceivedCalls
 Placed CallsApplication:Cisco/PlacedCalls
 Intercom Calls  Application:Cisco/IntercomCalls
 Personal Directory  Application:Cisco/PersonalDirectory
 Corporate Directory Application:Cisco/CorporateDirectory
 Voicemail   Application:Cisco/Voicemail

 *Back to CCX*
 *
 *
 In CCX you will edit the XML you have in notepad. You will create
 MenuItem child nodes using the above output. The SQL name maps to the XML
 Name field and the urltemplate maps to the URL field.

 Build your menu accordingly.

 You will likely need to create more than one XML file to service from
 CCX. One for normal phones and one for the phone that is being
 customized. The nature of the customization will vary based on lab
 requirement.

 Save files to c:\inetpub\wwwroot\

 *Back to CUCM (cli)*

 Now you want to disable the various directories. I wouldn't delete them
 and I sure as heck wouldn't muddle through the web interface to toggle the
 services as enabled/disable. So, I use SQL to disable the services:

 run sql update telecasterservice set enabled = 'f' where name like '%d
 Calls' or name like '%Directory'

 The above query will disable: Missed Calls, Received Calls, Placed
 Calls, Personal Directory, and Corporate Directory.


 *
 *
 *Back to CUCM (web)*
 *
 *
 You want to set the services provisioning from Internal to Both. You
 can do this from Ent Params, via common phone profile, or directly on each
 phone (via BAT). I use Ent Params because you will also need to remove the
 default Directories URL.

 Restart the phones.



 -Bill
 --
 William Bell, CCIE #38914
 blog: http://ucguerrilla.com
 twitter: @ucguerrilla




 On Apr 17, 2013, at 9:25 AM, Ramcharan Arya wrote:

 Your Bill you are correct OWLE Lab4 solution seems to be more
 appropriate for this task.

 So when I create XML file I can keep the same priority of services  as
 show in telecasterservices table to keep directory structure identical.

 Thanks  Regards,
 Ramcharan Arya
 CCIE # 28926 (RS)



 On Wed, Apr 17, 2013 at 6:56 AM, William Bell b...@ucguerrilla.comwrote:

 Ramcharan,

 The only alternate solution is that you create two XML files that you
 host from the UCCX web server. One XML file is a complete replica of the
 Directory menu so that your normal phones have a normal experience. The
 other would display a status message of services disabled.

 Look at OWLE Lab 4, question 2.2 for a question that is similar. Tinker
 with it a bit and you can map it into your alternate solution.

 -Bill



 On Apr 16, 2013, at 11:16 PM, Ramcharan Arya wrote:

 Hi Vik,

 Can you please suggest about alternate solutions  because current
 solution break voicemail service on the phone.

 Thanks  Regards,
 Ramcharan Arya



 On Tue, Apr 16, 2013 at 12:10 PM, Ramcharan Arya 
 ramcharan.a...@gmail.com wrote:

 Thanks Bill I will check archive for this discussion.


 On Tue, Apr 16, 2013 at 12:00 PM, Bill Lake whl...@gmail.com wrote:

 Look for a detailed discussion on this very issue around November
 19th 2012



 Sent from my iPhone

 On Apr 16, 2013, at 11:25 AM, Ramcharan Arya 
 ramcharan.a...@gmail.com wrote:

 Hi Bill

 I breaks voicemail service on that phone. So what is the alternative
 solutions for this task.?

 Thanks

Re: [OSL | CCIE_Voice] OWLE Lab 1 Task 2.4 No Service Configured

2013-04-17 Thread Ramcharan Arya
Your Bill you are correct OWLE Lab4 solution seems to be more appropriate
for this task.

So when I create XML file I can keep the same priority of services  as show
in telecasterservices table to keep directory structure identical.

Thanks  Regards,
Ramcharan Arya
CCIE # 28926 (RS)



On Wed, Apr 17, 2013 at 6:56 AM, William Bell b...@ucguerrilla.com wrote:

 Ramcharan,

 The only alternate solution is that you create two XML files that you host
 from the UCCX web server. One XML file is a complete replica of the
 Directory menu so that your normal phones have a normal experience. The
 other would display a status message of services disabled.

 Look at OWLE Lab 4, question 2.2 for a question that is similar. Tinker
 with it a bit and you can map it into your alternate solution.

 -Bill



 On Apr 16, 2013, at 11:16 PM, Ramcharan Arya wrote:

 Hi Vik,

 Can you please suggest about alternate solutions  because current solution
 break voicemail service on the phone.

 Thanks  Regards,
 Ramcharan Arya



 On Tue, Apr 16, 2013 at 12:10 PM, Ramcharan Arya ramcharan.a...@gmail.com
  wrote:

 Thanks Bill I will check archive for this discussion.


 On Tue, Apr 16, 2013 at 12:00 PM, Bill Lake whl...@gmail.com wrote:

 Look for a detailed discussion on this very issue around November 19th
 2012



 Sent from my iPhone

 On Apr 16, 2013, at 11:25 AM, Ramcharan Arya ramcharan.a...@gmail.com
 wrote:

 Hi Bill

 I breaks voicemail service on that phone. So what is the alternative
 solutions for this task.?

 Thanks,
 Ramcharan Arya
 CCIE # 28926 (RS)



 On Tue, Apr 16, 2013 at 11:22 AM, William Bell b...@ucguerrilla.comwrote:

 Correct. If you convert a phone to using External URLs only then you
 will be breaking Messages button. You can apply a URL to the messages
 button but it doesn't restore the behavior you are looking for because the
 VM services URL that I have seen used/reference is a XML phone menu object.
 Which means you get a menu.

 So, what you are seeing is expected per the requirements of the OWLE
 lab.


 -Bill
  --
 William Bell, CCIE #38914
 blog: http://ucguerrilla.com
 twitter: @ucguerrilla




 On Apr 16, 2013, at 11:42 AM, Ramcharan Arya wrote:

 Hi,

 When I use proposed solutions according to solution guide for disabling
 directory service it works but voicemail service also stop working on the
 same phone.

 Can someone please test this and let me know if anyone had similar
 problem.

 Thanks  Regards,
 Ramcharan Arya
 CCIE # 28926 (RS)

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

 Are you a CCNP or CCIE and looking for a job? Check out
 www.PlatinumPlacement.com



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

 Are you a CCNP or CCIE and looking for a job? Check out
 www.PlatinumPlacement.com




 --
 William Bell, CCIE #38914
 blog: http://ucguerrilla.com
 Follow me on twitter @ucguerrilla





___
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www.ipexpert.com

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www.PlatinumPlacement.com

Re: [OSL | CCIE_Voice] OWLE Lab 1 Task 2.4 No Service Configured

2013-04-17 Thread Ramcharan Arya
Thank you William for detail steps. I will try this afternoon in my pod.

In summary : For No service configuration on phone I should use solution
describe either in OWLE lab 4 solutions guide or below describe in email.

Regards,
Ramcharan Arya
CCIE # 28926


On Wed, Apr 17, 2013 at 9:09 AM, William Bell b...@ucguerrilla.com wrote:

 Right. The order you place the MenuItem child nodes in the XML file will
 be the order they are presented on the phone.

 What I do when practicing lab 4 was as follows:

 *CCX*

 1. RDP to CCX
 2. From CCX browse to xml directory URL  (copied from Ent Params in CUCM)
 3. Copy XML from CCX browser to notepad


 *CUCM* / *CLI*

 1. Go to CLI on CUCM and run query

 admin:*run sql select name,urltemplate from telecasterservice*
 nameurltemplate
 ===
 
 Missed CallsApplication:Cisco/MissedCalls
 Received Calls  Application:Cisco/ReceivedCalls
 Placed CallsApplication:Cisco/PlacedCalls
 Intercom Calls  Application:Cisco/IntercomCalls
 Personal Directory  Application:Cisco/PersonalDirectory
 Corporate Directory Application:Cisco/CorporateDirectory
 Voicemail   Application:Cisco/Voicemail

 *Back to CCX*
 *
 *
 In CCX you will edit the XML you have in notepad. You will create MenuItem
 child nodes using the above output. The SQL name maps to the XML Name
 field and the urltemplate maps to the URL field.

 Build your menu accordingly.

 You will likely need to create more than one XML file to service from CCX.
 One for normal phones and one for the phone that is being customized. The
 nature of the customization will vary based on lab requirement.

 Save files to c:\inetpub\wwwroot\

 *Back to CUCM (cli)*

 Now you want to disable the various directories. I wouldn't delete them
 and I sure as heck wouldn't muddle through the web interface to toggle the
 services as enabled/disable. So, I use SQL to disable the services:

 run sql update telecasterservice set enabled = 'f' where name like '%d
 Calls' or name like '%Directory'

 The above query will disable: Missed Calls, Received Calls, Placed Calls,
 Personal Directory, and Corporate Directory.


 *
 *
 *Back to CUCM (web)*
 *
 *
 You want to set the services provisioning from Internal to Both. You can
 do this from Ent Params, via common phone profile, or directly on each
 phone (via BAT). I use Ent Params because you will also need to remove the
 default Directories URL.

 Restart the phones.



 -Bill
 --
 William Bell, CCIE #38914
 blog: http://ucguerrilla.com
 twitter: @ucguerrilla




 On Apr 17, 2013, at 9:25 AM, Ramcharan Arya wrote:

 Your Bill you are correct OWLE Lab4 solution seems to be more appropriate
 for this task.

 So when I create XML file I can keep the same priority of services  as
 show in telecasterservices table to keep directory structure identical.

 Thanks  Regards,
 Ramcharan Arya
 CCIE # 28926 (RS)



 On Wed, Apr 17, 2013 at 6:56 AM, William Bell b...@ucguerrilla.comwrote:

 Ramcharan,

 The only alternate solution is that you create two XML files that you
 host from the UCCX web server. One XML file is a complete replica of the
 Directory menu so that your normal phones have a normal experience. The
 other would display a status message of services disabled.

 Look at OWLE Lab 4, question 2.2 for a question that is similar. Tinker
 with it a bit and you can map it into your alternate solution.

 -Bill



 On Apr 16, 2013, at 11:16 PM, Ramcharan Arya wrote:

 Hi Vik,

 Can you please suggest about alternate solutions  because current
 solution break voicemail service on the phone.

 Thanks  Regards,
 Ramcharan Arya



 On Tue, Apr 16, 2013 at 12:10 PM, Ramcharan Arya 
 ramcharan.a...@gmail.com wrote:

 Thanks Bill I will check archive for this discussion.


 On Tue, Apr 16, 2013 at 12:00 PM, Bill Lake whl...@gmail.com wrote:

 Look for a detailed discussion on this very issue around November 19th
 2012



 Sent from my iPhone

 On Apr 16, 2013, at 11:25 AM, Ramcharan Arya ramcharan.a...@gmail.com
 wrote:

 Hi Bill

 I breaks voicemail service on that phone. So what is the alternative
 solutions for this task.?

 Thanks,
 Ramcharan Arya
 CCIE # 28926 (RS)



 On Tue, Apr 16, 2013 at 11:22 AM, William Bell b...@ucguerrilla.comwrote:

 Correct. If you convert a phone to using External URLs only then you
 will be breaking Messages button. You can apply a URL to the messages
 button but it doesn't restore the behavior you are looking for because the
 VM services URL that I have seen used/reference is a XML phone menu 
 object.
 Which means you get a menu.

 So, what you are seeing is expected per the requirements of the OWLE
 lab.


 -Bill
  --
 William Bell, CCIE #38914
 blog: http://ucguerrilla.com
 twitter: @ucguerrilla




 On Apr 16, 2013, at 11:42 AM, Ramcharan Arya wrote:

 Hi,

 When I use proposed solutions according to solution guide for
 disabling directory service

[OSL | CCIE_Voice] OWLE Lab 1 Task 2.4 No Service Configured

2013-04-16 Thread Ramcharan Arya
Hi,

When I use proposed solutions according to solution guide for disabling
directory service it works but voicemail service also stop working on the
same phone.

Can someone please test this and let me know if anyone had similar problem.

Thanks  Regards,
Ramcharan Arya
CCIE # 28926 (RS)
___
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www.ipexpert.com

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www.PlatinumPlacement.com

Re: [OSL | CCIE_Voice] OWLE Lab 1 Task 2.4 No Service Configured

2013-04-16 Thread Ramcharan Arya
Hi Bill

I breaks voicemail service on that phone. So what is the alternative
solutions for this task.?

Thanks,
Ramcharan Arya
CCIE # 28926 (RS)



On Tue, Apr 16, 2013 at 11:22 AM, William Bell b...@ucguerrilla.com wrote:

 Correct. If you convert a phone to using External URLs only then you will
 be breaking Messages button. You can apply a URL to the messages button but
 it doesn't restore the behavior you are looking for because the VM services
 URL that I have seen used/reference is a XML phone menu object. Which means
 you get a menu.

 So, what you are seeing is expected per the requirements of the OWLE lab.


 -Bill
 --
 William Bell, CCIE #38914
 blog: http://ucguerrilla.com
 twitter: @ucguerrilla




 On Apr 16, 2013, at 11:42 AM, Ramcharan Arya wrote:

 Hi,

 When I use proposed solutions according to solution guide for disabling
 directory service it works but voicemail service also stop working on the
 same phone.

 Can someone please test this and let me know if anyone had similar problem.

 Thanks  Regards,
 Ramcharan Arya
 CCIE # 28926 (RS)

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

 Are you a CCNP or CCIE and looking for a job? Check out
 www.PlatinumPlacement.com



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

Are you a CCNP or CCIE and looking for a job? Check out 
www.PlatinumPlacement.com

Re: [OSL | CCIE_Voice] OWLE Lab 1 Task 2.4 No Service Configured

2013-04-16 Thread Ramcharan Arya
Thanks Bill I will check archive for this discussion.


On Tue, Apr 16, 2013 at 12:00 PM, Bill Lake whl...@gmail.com wrote:

 Look for a detailed discussion on this very issue around November 19th 2012



 Sent from my iPhone

 On Apr 16, 2013, at 11:25 AM, Ramcharan Arya ramcharan.a...@gmail.com
 wrote:

 Hi Bill

 I breaks voicemail service on that phone. So what is the alternative
 solutions for this task.?

 Thanks,
 Ramcharan Arya
 CCIE # 28926 (RS)



 On Tue, Apr 16, 2013 at 11:22 AM, William Bell b...@ucguerrilla.comwrote:

 Correct. If you convert a phone to using External URLs only then you will
 be breaking Messages button. You can apply a URL to the messages button but
 it doesn't restore the behavior you are looking for because the VM services
 URL that I have seen used/reference is a XML phone menu object. Which means
 you get a menu.

 So, what you are seeing is expected per the requirements of the OWLE lab.


 -Bill
  --
 William Bell, CCIE #38914
 blog: http://ucguerrilla.com
 twitter: @ucguerrilla




 On Apr 16, 2013, at 11:42 AM, Ramcharan Arya wrote:

 Hi,

 When I use proposed solutions according to solution guide for disabling
 directory service it works but voicemail service also stop working on the
 same phone.

 Can someone please test this and let me know if anyone had similar
 problem.

 Thanks  Regards,
 Ramcharan Arya
 CCIE # 28926 (RS)

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

 Are you a CCNP or CCIE and looking for a job? Check out
 www.PlatinumPlacement.com



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

 Are you a CCNP or CCIE and looking for a job? Check out
 www.PlatinumPlacement.com


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

Are you a CCNP or CCIE and looking for a job? Check out 
www.PlatinumPlacement.com

Re: [OSL | CCIE_Voice] OWLE Lab 1 Task 2.4 No Service Configured

2013-04-16 Thread Ramcharan Arya
Hi Vik,

Can you please suggest about alternate solutions  because current solution
break voicemail service on the phone.

Thanks  Regards,
Ramcharan Arya



On Tue, Apr 16, 2013 at 12:10 PM, Ramcharan Arya
ramcharan.a...@gmail.comwrote:

 Thanks Bill I will check archive for this discussion.


 On Tue, Apr 16, 2013 at 12:00 PM, Bill Lake whl...@gmail.com wrote:

 Look for a detailed discussion on this very issue around November 19th
 2012



 Sent from my iPhone

 On Apr 16, 2013, at 11:25 AM, Ramcharan Arya ramcharan.a...@gmail.com
 wrote:

 Hi Bill

 I breaks voicemail service on that phone. So what is the alternative
 solutions for this task.?

 Thanks,
 Ramcharan Arya
 CCIE # 28926 (RS)



 On Tue, Apr 16, 2013 at 11:22 AM, William Bell b...@ucguerrilla.comwrote:

 Correct. If you convert a phone to using External URLs only then you
 will be breaking Messages button. You can apply a URL to the messages
 button but it doesn't restore the behavior you are looking for because the
 VM services URL that I have seen used/reference is a XML phone menu object.
 Which means you get a menu.

 So, what you are seeing is expected per the requirements of the OWLE
 lab.


 -Bill
  --
 William Bell, CCIE #38914
 blog: http://ucguerrilla.com
 twitter: @ucguerrilla




 On Apr 16, 2013, at 11:42 AM, Ramcharan Arya wrote:

 Hi,

 When I use proposed solutions according to solution guide for disabling
 directory service it works but voicemail service also stop working on the
 same phone.

 Can someone please test this and let me know if anyone had similar
 problem.

 Thanks  Regards,
 Ramcharan Arya
 CCIE # 28926 (RS)

 ___
 For more information regarding industry leading CCIE Lab training,
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Re: [OSL | CCIE_Voice] CME - GK - UCM bandwidth

2013-04-08 Thread Ramcharan Arya
Here is bug id

CSCsl74701 Bug Details ARQ requests 1280 when no regions are defined to use
g711

Regards,
Ramcharan Arya
CCIE # 28926 (RS)



On Mon, Apr 8, 2013 at 9:02 AM, Suresh Bhandari bring...@gmail.com wrote:

 Will check and let you know.

 BTW you mentioned a known bug can you please post a link of it?

 Thanks.


 On Mon, Apr 8, 2013 at 7:41 PM, Mohamed Gazzaz mgaz...@hotmail.comwrote:

 Hi Suresh,

 I think you are hitting a known bug . Please go to service parameters --
 call manager  and change the following to G729

 Intraregion Audio Codec Default: G729

 Regards,
 Mohamed Gazzaz

 --
 Date: Mon, 8 Apr 2013 19:02:14 +0545
 From: bring...@gmail.com
 To: ccie_voice@onlinestudylist.com
 Subject: [OSL | CCIE_Voice] CME - GK - UCM bandwidth


 Hello Experts!

 I have HQ as the GK and CME, CUCM publisher and subscriber are registered
 to GK in single zone. Further, GK Trunk is in separate region/DP with
 hard-coded G.729 codec with other regions/itself as well.

 When I call from HQ to CME side, and check sh gatek call it shows that
 the call is consuming 16K bandwidth, which is expected.

 The dial-peer to CME has g729r8 as codec (the default one).

 Even then, the same command displays that the bandwidth consumed is 128K.

 Any thoughts on what probably I missed?
 --
 Suresh Bhandari

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 regarding industry leading CCIE Lab training, please visit
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 --
 Suresh Bhandari

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[OSL | CCIE_Voice] Lab 1: Task 3.1 : 5 Lab Handbook - workbook lab

2013-04-04 Thread Ramcharan Arya
Hi,
I have question regarding User-side Q.921/Q.931 Signalling

when I configure my gateway will full T1 PRI

command isdn protocol-emulate user does not appear under D channel
configuration.

interface Serial0/0/0:23
 no ip address
 encapsulation hdlc
 isdn switch-type primary-ni
 isdn incoming-voice voice
 isdn map address 911 plan isdn type unknown
 isdn map address [2-9]..$ plan isdn type subscriber
 isdn map address 1[2-9]..[2-9]..$ plan isdn type national
 isdn outgoing display-ie
 isdn outgoing ie redirecting-number
 no cdp enable

Do we need to add this command manual.? WB says it is by default.

Please let us know your thoughts

Thanks  Regards,
Ramcharan Arya
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Re: [OSL | CCIE_Voice] Lab Exam Speed Strategy

2013-04-03 Thread Ramcharan Arya
Hi Bill,

Thank you very much for nice writeup on strategy.

This is really helpful for CCIE vocie lab aspirants.  Do you have any
strategy how many notepad sessions to keep open simultaneously.

How to arrange SecureCRT sessions screen, online lab webpage, and notepad
on 32 screen.

I am still practice same method which I learn during RS bootcamp with
Marko.If you have any better approach please share.


Regards,
Ramcharan Arya
CCIE # 28926 (RS)



On Wed, Apr 3, 2013 at 10:57 AM, William Bell b...@ucguerrilla.com wrote:

 I have had this as a draft for a few days. Just too busy to finish it
 until now. So, some of my thoughts are redundant to what others have said.
 Hopefully that isn't a bad thing.

 Timing is definitely a critical aspect of the exam. I know I have areas
 where I am slower than I should be. I suspect most people do. Most of my
 comments herein are based on my self-study practice labs. I have taken the
 lab a couple of times but most of the tinkering I have done with my method
 is during self-study. When I sit for the real lab, I don't tinker. I go
 with whatever method I have been practicing. So, that is suggestion #1:
 Don't tinker on lab day, stick to your guns and don't 2nd guess your method.

 Going back to the OP, I believe you should look at the bright side. Your
 statement ...I seemed to keep moving forward... is key. The fact you were
 able to avoid a stall is important. I believe controlling this exam is
 about rhythm and finding what config approach helps you establish a
 sustainable and consistent rhythm. Speed on any individual task is critical
 but rhythm is king in my opinion.

 Like others (most?), I follow the device-based approach. It has been
 around since pre 3.0 blueprint (contrary to popular opinion) and is a
 proven strategy. However, I have found that you will need to customize that
 approach to suit your needs. For me, it is about managing the transitions.
 Again, I believe focusing on establishing and maintaining a rhythm is
 absolutely key. Smoothing the transitions and/or stacking tasks that help
 ease transitions is important. Also, you won't maintain the same rhythm
 throughout the exam. Some tasks you will bang out (or should) very fast.
 Others, you will need to pay close attention to what you are doing.

 So, suggestion #2 is find your rhythm.

 Establishing your rhythm is a product of repetition. Practice, practice,
 practice as Mr. Sears puts it. You may also need some face time with the
 real lab to help you come into your rhythm. For example, some of the weak
 spots I had (or maybe still have) and adjustments I made.


 1. Transitioning from read-through to config. The read through is/was the
 worst for me. Most people I have spoken with (who have passed or come
 close) are able to get through the read-through in 30 minutes. Some say
 less. I was taking a whole lot more time than 30 minutes. My budget for
 this task is 30m today.

 Adjustments I made:


- Dial Plan. I was building out my dial plan (on paper) during the
read through. My logic was that you have to do it at some point, just do
it now. The flaw with that logic is that to establish a good rhythm, you
need to avoid lingering on a task for too long. I decided that I would
focus on getting the tasks mapped out as quick as possible and the task of
mapping out a dial plan could wait. So, I added a section in my table to
track the DP-related tasks by task ID (e.g. 4.1) only.
- Skim. I already know that I am going to do a thorough read on the
questions at least once (during config) and likely twice (during
validation). No sense in reading the question in detail 3 times. So, I
mainly focus on what devices/apps are affected by the question and put the
ID in the table.


 How I use this task:

- I build a table (like the dev-based approach table) to track tasks
- I build a table to track what the PSTN wants to see for off net
calls (this is key because I can build an entire h323 config just on that
info)
- I track phone/user features/buttons/etc. This goes to speed when
customizing phones
- I build a basic.txt text file to capture IP addresses, vlans,
interface assignments, etc. I keep this very short. Note that with the new
exam, the instructions are electronic. This kinda de-emphasizes the need to
store some of this info.



 2. Where to stick in QoS. QoS is clearly infrastructure and all of the
 action happens in the router config. However, for me there is a slow down
 if I try to stick QoS config in with banging out vlan, ntp, dhcp, rsvp,
 mrm, h323, etc. I can do all of the other IOS configs very quickly in
 notepad. QoS is noticeably slower for me.

 Adjustment I made:

- I avoid auto qos on the LAN switch if I can. I have found the LAN
QoS questions to be incomplete. Meaning, they don't reflect reality in
any way. So, if I can I just bang out the minimum config w/o auto-qos and I
do

[OSL | CCIE_Voice] CLI Command to monitor Directed call Park in CUCM

2013-04-03 Thread Ramcharan Arya
Hi,

Is there any  command to check Directed call Park status on CUCM using CLI.?

Please let me know.

Thanks  Regards,
Ramcharan Arya
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Re: [OSL | CCIE_Voice] Lab Exam Speed Strategy

2013-04-02 Thread Ramcharan Arya
Hi Mike,

Thank you for sharing great information.

Can you share some detail about approach and sequence to follow like
Infrastructure, gateway configuration, QoS and SRST, Presence . Unity, UCCX
etc.

When did you do SRST testing in the middle or at the end of the lab.?

Please share your experience.

Thanks  Regards,
Ramcharan Arya
CCIE # 28926 ( Routing  Switching)



On Tue, Apr 2, 2013 at 12:26 PM, michael.se...@compucom.com wrote:

 It took me 4 attempts to pass the lab.  Actually the first three attempts
 helped to develop a strategy for passing.  The proctor in RTP, David,
 thought me something, don't look at a no pass as a failure but a as
 learning experience.  After my third attempt I couldn't stand to see
 another fail on the score report.  I took 45 days, doing two labs a day
 following the same strategy.

 If your typing skills are below 70 words/minute or less or you are hunt
 and peck typist take a typing class won't hurt have to type fast.

 Briefly read the entire lab and absorb as much as possible 5 to 10 minutes
 maximum regarding CUCM and gateway, QoS, etc.

 Perform all your switch and gateway configurations first including
 everything so you don't have to revisit them.  Write all configuration for
 SW and Gateways in notepad prior to putting into devices and same to
 desktop, leave them there when leaving the lab.  Copy all the
 customization's you'll need and put in notepad and put on desktop, i.e.,
 media resources, dial-peer, other customizations.  Don't type and memorize
 things you can obtain from links copy from links and edit

 1.) Configure the SW first and take what configure you can from there
 and move onto R1.
 2.) Configure R1 and take configuration from there to R2 and edit and
 add additional configuration.
 3.) Configure R2 including SRST/GK/Dial-peers/MVA/everything.  Move
 configure from R2 and R1 to R3 and edit.
 4.) Configure R3/CUE/Presence/SRST using configuration from R1 and R2
 that's reusable.
 5.) Don't type the same thing twice.
 5.) Now move to CUCM.  You should have a pretty good idea of what you
 will need from reading lab.
 6.) Open browser to CUCM Pub, Sub, Unity.  Add ntp and any required
 customizations
 7.) Configure CUCM moving from left to right, save phones for last.
 8.) Configure UNITY and all voicemail customization
 9.) Configure UCCX script and record prompts unless they are
 pre-recorded for you.
 10.)Configure Presence if you have it on your lab.
 11.)Need at least three hours to test and validate.
 12.)Make every attempt to complete lab before lunch.
 13.)Feel good at lunch relax forget the lab
 14.)Get your score report that says PASS.
 15.)Preform Troubleshooting as you are most comfortable with I saved
 it for last.

 There was a guy walking down the street in NYC and he recognized a famous
 pianist.  He stopped him and ask him How do you get to Carnegie Hall.  The
 pianist replied Practice, Practice, Practice.

 Michael Sears, CCIE(V)#38404
 Cisco Certified Unfied Communications Computing Systems Specialist
 E911 Specialist
 Designing and Implementing Cisco Unified Communications on Unified
 Computing Systems


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Re: [OSL | CCIE_Voice] HQ and Branch1 phones cannot call CUE

2013-03-20 Thread Ramcharan Arya
Hi,

You have to add on your CME side.

voice service voip
allow connection h t h
allow connection h t  s
allow connection s t h

Thanks,
Ramcharan Arya





On Wed, Mar 20, 2013 at 1:16 PM, Vikky Kumar vikkyne...@gmail.com wrote:

 Hi Experts,

 I configured branch 2 CME/CUE working normal for Voice mails.

 CUE is registered with CUCM but I call not call CUE(6220)  from HQ and
 Branch1 but branch2 phones 4001 and 4002 working normal/ringing from every
 where.

 FYI. I have also configured CAC between on Br2 site - HQ site

 Please hel.

 Regards

 Vikky


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Re: [OSL | CCIE_Voice] HQ and Branch1 phones cannot call CUE

2013-03-20 Thread Ramcharan Arya
Hi,

You need to add codec g729r8 in your transcoder profile.

Codec Configuration
 Codec : g711ulaw, Maximum Packetization Period : 30
 Codec : g711alaw, Maximum Packetization Period : 30
 Codec : g729ar8, Maximum Packetization Period : 60
 Codec : g729abr8, Maximum Packetization Period : 60


Thanks,
Ramcharan Arya


On Wed, Mar 20, 2013 at 2:46 PM, Vikky Kumar vikkyne...@gmail.com wrote:

 Hi Experts,

 I have configured :

 1. transcoder on CUCM for HQ  BR2 (Cisco IOS Enhanced Media Termination
 Point )
 and also Router of  BR2 (dspfarm profile 2 transcode , max sess 4 .. )

 2. above hq-xcode,, sc-xcode is registered   included in respective  mrg
  mrgl  DP

 3. voice service voip , allow connection h t h,allow connection h t  s,
 allow connection s t h

 When call from HQ i can notice randomly (6221, 6222, 6223) , very very
 small ring/tone, Display Ring out with busy tone

 Please advise...

 [
 Router BR2 sh dspfarm all 

 Dspfarm Profile Configuration

  Profile ID = 2, Service = TRANSCODING, Resource ID = 1
  Profile Description :
  Profile Service Mode : Non Secure
  Profile Admin State : UP
  Profile Operation State : ACTIVE
  Application : SCCP   Status : ASSOCIATED
  Resource Provider : FLEX_DSPRM   Status : UP
  Number of Resource Configured : 4
  Number of Resource Available : 4
  Codec Configuration
  Codec : g711ulaw, Maximum Packetization Period : 30
  Codec : g711alaw, Maximum Packetization Period : 30
  Codec : g729ar8, Maximum Packetization Period : 60
  Codec : g729abr8, Maximum Packetization Period : 60

 ]


 Regards,

 Vikky



 On Wed, Mar 20, 2013 at 9:54 PM, Ramcharan Arya 
 ramcharan.a...@gmail.comwrote:

 Hi,

 You have to add on your CME side.

 voice service voip
 allow connection h t h
 allow connection h t  s
 allow connection s t h

 Thanks,
 Ramcharan Arya





 On Wed, Mar 20, 2013 at 1:16 PM, Vikky Kumar vikkyne...@gmail.comwrote:

  Hi Experts,

 I configured branch 2 CME/CUE working normal for Voice mails.

 CUE is registered with CUCM but I call not call CUE(6220)  from HQ and
 Branch1 but branch2 phones 4001 and 4002 working normal/ringing from every
 where.

 FYI. I have also configured CAC between on Br2 site - HQ site

 Please hel.

 Regards

 Vikky


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Re: [OSL | CCIE_Voice] IntegratedServicesEngine bound to Loopback can ping but can't http to it?

2013-03-19 Thread Ramcharan Arya
Jason,

Can you share your full Branch2 configuration here.?

Thanks,
Ramcharan Arya
CCIE # 28926 (RS)


On Tue, Mar 19, 2013 at 6:31 PM, Andy Thanh tanthanh2...@gmail.com wrote:

 Did you advertise this network through ospf? This is for simple command:

 router ospf 1
  network 0.0.0.0 255.255.255.255 area 0


 On Wed, Mar 20, 2013 at 5:11 AM, Sergey Heyphets ser...@heyphets.comwrote:

 Jason,


 is your route to *1*10.32.60.253 ?


 On Tue, Mar 19, 2013 at 4:56 PM, Jason Aarons scubajas...@gmail.comwrote:

  ** **

 What are some troubleshooting shooting steps for a
 CUE/IntegratedServicesEngine that is using ip unnumbered loopback0?

 ** **

 I can ping the ISE locally and from remote sites, but I can’t get the
 http web page for the ISE  to show (CME web page is good). I can ping
 10.32.60.253 from another site but web page doesn’t come up in Internet
 Explorer.

 ** **

 I can session to ISE  (show tech page is correct), I have reset ISE, I
 have done a restore factory default. I have reloaded the router and the ISE.
 

 ** **

 What else can I do?  If I switch it to another interface, reload, etc it
 works fine (ping, web page, phones pilot number works fine). What are some
 CUE/ISE debug commands I should run to see if http traffic is making it to
 the CUE/ISE?

 ** **

 ** **

 Branch2#

 Int loopback0 

Ip address 10.32.60.254 255.255.255.0

Ip ospf network point-to-point

 !

 Int integrated-services-engine 1/0

 Ip unnumbered loopback0

Ip ospf network point-to-point

 Service-module ip address 10.32.60.253

 Service-module ip default-gateway 10.32.60.254

 Keepalive 10

 No shut

 !

 Ip route 110.32.60.253 255.255.255.255 integrated-service-engine 1/0

 ** **

 Seems to me I’m missing something when using loopbacks for the CUE/ISE.*
 ***

 ** **

 -jason

 ** **

 Could be some typo’s above this is from memory of my home lab, I’m not
 at home right now.

 ** **

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Re: [OSL | CCIE_Voice] IntegratedServicesEngine bound to Loopback can ping but can't http to it?

2013-03-19 Thread Ramcharan Arya
Did he enable ip http server  also ip http path flash/gui

I suggest he need to xtract tar file and make sure all files are there in
flash/gui directory



On Tue, Mar 19, 2013 at 7:33 PM, Bill whl...@gmail.com wrote:

   Ip ospf network point-to-point

 And he said he can ping and session to it,  web interface is not working
 though?


 Sent from my iPad

 On Mar 19, 2013, at 6:31 PM, Andy Thanh tanthanh2...@gmail.com wrote:

 Did you advertise this network through ospf? This is for simple command:

 router ospf 1
  network 0.0.0.0 255.255.255.255 area 0

 On Wed, Mar 20, 2013 at 5:11 AM, Sergey Heyphets ser...@heyphets.comwrote:

 Jason,


 is your route to *1*10.32.60.253 ?


 On Tue, Mar 19, 2013 at 4:56 PM, Jason Aarons scubajas...@gmail.comwrote:

  ** **

 What are some troubleshooting shooting steps for a
 CUE/IntegratedServicesEngine that is using ip unnumbered loopback0?

 ** **

 I can ping the ISE locally and from remote sites, but I can’t get the
 http web page for the ISE  to show (CME web page is good). I can ping
 10.32.60.253 from another site but web page doesn’t come up in Internet
 Explorer.

 ** **

 I can session to ISE  (show tech page is correct), I have reset ISE, I
 have done a restore factory default. I have reloaded the router and the ISE.
 

 ** **

 What else can I do?  If I switch it to another interface, reload, etc it
 works fine (ping, web page, phones pilot number works fine). What are some
 CUE/ISE debug commands I should run to see if http traffic is making it to
 the CUE/ISE?

 ** **

 ** **

 Branch2#

 Int loopback0 

Ip address 10.32.60.254 255.255.255.0

Ip ospf network point-to-point

 !

 Int integrated-services-engine 1/0

 Ip unnumbered loopback0

Ip ospf network point-to-point

 Service-module ip address 10.32.60.253

 Service-module ip default-gateway 10.32.60.254

 Keepalive 10

 No shut

 !

 Ip route 110.32.60.253 255.255.255.255 integrated-service-engine 1/0

 ** **

 Seems to me I’m missing something when using loopbacks for the CUE/ISE.*
 ***

 ** **

 -jason

 ** **

 Could be some typo’s above this is from memory of my home lab, I’m not
 at home right now.

 ** **

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Re: [OSL | CCIE_Voice] SRST transfer system and pattern

2013-02-16 Thread Ramcharan Arya
Hi,

As per cisco CME design guide these commands are necessary. Please refer
cisco CME SRND.

http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/srnd/design/guide/clproc.html#wp1068396

Regards,
Ramcharan Arya
CCIE # 28926 ( RS)


On Fri, Feb 15, 2013 at 4:51 PM, Cory Gray corygray22...@hotmail.comwrote:

 I have had several conversations with people on this.  Everyone can easily
 make SRST work but scoring points seems to be the trickiest thing in the
 lab.  So I do not think anyone knows for sure what should or should not be
 on the “template”  I have never scored any points there so I cannot give an
 OPINION on what should or should not be there.  People say they score
 points and then go with the same template on the next lab and get 0 so it
 is a mystery.  People can share templates without breaking NDA since the
 question is never discussed.  Getting the question right is the easy part!
 

 ** **

 *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
 ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Pixar Perfect
 *Sent:* Friday, February 15, 2013 5:26 PM
 *To:* CCIE Voice OSL
 *Subject:* [OSL | CCIE_Voice] SRST transfer system and pattern

 ** **

 transfer-system full-consultdo we need to specify this? I thought by
 default it is wnabled but I read on voiceie forum someone scored nothing on
 SRST adn the only conclusion was the transfersystem consult was missing.
 Any thoughts?

 ** **

  srst mode auto-provision all

  srst ephone description SRST-EPHONES-CME  

  srst dn template 1

  srst dn line-mode octo

  max-ephones 10

  max-dn 10 preference 2 no-reg both

  ip source-address 10.10.1.13 SiteC Loopback  port 2000

  time-zone 42

  max-conferences 8 gain -6

  call-forward pattern .T

  time-webedit 

 * transfer-system full-consult*

 * transfer-pattern .T*

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Re: [OSL | CCIE_Voice] Site A to Site C can't leave vm on CUE

2013-02-02 Thread Ramcharan Arya
Hi,
Check your WAN QoS It could be related to one way RTP compression.

Regards
Ramcharan Arya

On Sat, Feb 2, 2013 at 12:37 PM, Cory Gray corygray22...@hotmail.comwrote:

 

 *You need to transfer h323 to sip on SiteC*

 Voice services voip

 Allow connections h323 to sip

 ** **

 *Make sure your Site C Gwy register to Site A GK as H323-GWY instead of
 VOIP-GWY*

 No gateway

 Gateway 

 ** **

 You do need a transcoder and it looks like you have it configured mostly
 correct

 You do not need ephone 3

 Also it is a good practice to use version 6 or higher for sccp ccm
 configuration or higher

 Also use bind interface X under sccp ccm group

 You should also not need “mailbox-selection last-redirect-num”

 ** **

 ** **

 *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
 ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Joe Fearday
 *Sent:* Saturday, February 02, 2013 11:28 AM
 *To:* ccie_voice@onlinestudylist.com
 *Subject:* [OSL | CCIE_Voice] Site A to Site C can't leave vm on CUE

 ** **

 Site A to Site C can't leave vm on CUE 

  

 Site A to Site C configured for g729
 Site A routes to Site C using gk in region C
 PSTN can leave vm on CUE as can other site C phones
 I am thinking this is a site C transcoding problem
 CUE only supports g711; right?
 Site C transcoding detailed below along with ephone:

  

 sccp local Vlan102
 sccp ccm 10.196.102.1 identifier 1 version 3.1
 sccp
 !
 sccp ccm group 1
  associate ccm 1 priority 1
  associate profile 1 register sctrans
 !
 dspfarm profile 1 transcode
  codec g711ulaw
  codec g711alaw
  codec g729ar8
  codec g729abr8
  codec g729r8
  maximum sessions 12
  associate application SCCP

  telephony-service
  sdspfarm units 1
  sdspfarm transcode sessions 20
  sdspfarm tag 1 sctrans

  

 ephone  3
 codec g729r8 dspfarm-assist

  

 dial-peer voice 9000 voip
  mailbox-selection last-redirect-num
  destination-pattern 3180  (CUE)
  session protocol sipv2
  session target ipv4:10.196.102.2
  dtmf-relay sip-notify
  codec g711ulaw
  no vad

 Any assistance is greatly apprecaited.
 Joe Fearday

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Re: [OSL | CCIE_Voice] Failed Voice Lab

2013-01-31 Thread Ramcharan Arya
Hi Ben,

I feel bad about this news  you did not score passing marks in lab exam.
Can you please share your experience  without breaking NDA.

At least this will be useful for others who are preparing for lab.

Thanks  Regards,
Ramcharan Arya
CCIE # 28926

On Thu, Jan 31, 2013 at 1:39 PM, Ben John benjoh...@hotmail.com wrote:

  Guys,
 i took my lab last week in SJ and i failed again that was my fourth
 attempt. i got everything working tested and i was confident i will get it
 this time but when i got my result i was disappointed i don't feel like
 going back again i feel like i am not spending enough times with my family
 and it cost me almost 2K for each attempt. i have seen all the labs as far
 as i know i don't if i do go back which one i will get and there is no
 guaranty i will pass it looks like i am doing the way cisco wants it. i am
 really down right the good thing is i can do Voice.

 Thanks,

 Ben

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Re: [OSL | CCIE_Voice] CME 12.4T and max-dn X no-reg

2013-01-30 Thread Ramcharan Arya
Hi Jason,

If you put command max-dn 30 no-reg both you need to reboot the router then
phone will not register into Gatekeeper with alias.

Regards,
Ramcharan Arya
CCIE # 28926 ( RS)



On Wed, Jan 30, 2013 at 7:49 PM, Jason Aarons scubajas...@gmail.com wrote:

 Is there a known bug with max-dn 30 no-reg in that ephone-dns still
 register to gatekeeper? Anyone got a bug id?  

 ** **

 Pretty clear it don’t work, you have to put no-reg in the ephone-dn itself.
 

 ** **

 -jason

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[OSL | CCIE_Voice] Question on Auto Qos WAN QoS

2013-01-28 Thread Ramcharan Arya
Hi,

I have noticed when I run AutoQoS on sub-interface it automatically add
encapsulation CISCO on Inteface-dlci

Config Prior to AutoQoS:

interface Serial0/1/0
 no ip address
 encapsulation frame-relay IETF
 no frame-relay inverse-arp
 frame-relay lmi-type ansi
!
interface Serial0/1/0.1 point-to-point
 ip address 10.10.112.2 255.255.255.0
 ip ospf mtu-ignore
 snmp trap link-status
 frame-relay interface-dlci 102
---

Configuration After AutoQoS

interface Serial0/1/0.1 point-to-point
 ip address 10.10.112.2 255.255.255.0
 ip ospf mtu-ignore
 snmp trap link-status
 frame-relay interface-dlci 102 CISCO - Encapsulation Cisco
class AutoQoS-FR-Se0/1/0-1
  auto qos voip trust

Prior to auto configuration on my primary interface frame-relay
encapsulation is IETF.

I think encapsulation should be match after autoqos.

Thanks  Regards,
Ramcharan Arya
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Re: [OSL | CCIE_Voice] Workbook 1 Lab 5a

2013-01-28 Thread Ramcharan Arya
You might have assigned  gateway directly under route-pattern.

If you remove gateway from route-pattern then you should be to add into
route-group/route-list.

Regards,
Ramcharan Arya


On Mon, Jan 28, 2013 at 4:42 PM, Nathan Silvers silver...@gmail.com wrote:

 Has anyone seen when trying to create a route group the gateways do not
 all appear as available devices, they appear when configuring route
 patterns but not under route groups.

 --
 The biggest mistake people make in life is not trying to make a living at
 doing what they most enjoy.

 - Malcolm Forbes

 Nathan Silvers

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Re: [OSL | CCIE_Voice] Workbook 1 Lab 5a

2013-01-28 Thread Ramcharan Arya
Yes, This pre-configuration could be part of  troubleshooting.


On Mon, Jan 28, 2013 at 5:10 PM, Nathan Silvers silver...@gmail.com wrote:

 hahaha nice.. gotta love how detailed the solutions guide is at times...
 hey its all pre-configured but you have to remove this configuration before
 you can do the configuration you we tell you to do.. thanks guys!


 On Mon, Jan 28, 2013 at 3:05 PM, Ramcharan Arya 
 ramcharan.a...@gmail.comwrote:

 You might have assigned  gateway directly under route-pattern.

 If you remove gateway from route-pattern then you should be to add into
 route-group/route-list.

 Regards,
 Ramcharan Arya


 On Mon, Jan 28, 2013 at 4:42 PM, Nathan Silvers silver...@gmail.comwrote:

 Has anyone seen when trying to create a route group the gateways do not
 all appear as available devices, they appear when configuring route
 patterns but not under route groups.

 --
 The biggest mistake people make in life is not trying to make a living
 at doing what they most enjoy.

 - Malcolm Forbes

 Nathan Silvers

 ___
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 please visit www.ipexpert.com

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 www.PlatinumPlacement.com





 --
 The biggest mistake people make in life is not trying to make a living at
 doing what they most enjoy.

 - Malcolm Forbes

 Nathan Silvers

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Re: [OSL | CCIE_Voice] [CCIE VOICE - Answer the Dial-Plan questions.

2013-01-25 Thread Ramcharan Arya
Hi ,

I suggest you work on VOL 1 Lab 5A  5C this will help you understanding
dialplan.Both labs are very good for call routing section.


Regards,
Ramcharan Arya
CCIE # 28926

On Fri, Jan 25, 2013 at 8:58 AM, ie ravindra ieravin...@gmail.com wrote:

 .
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Re: [OSL | CCIE_Voice] Custom Background image for phones registered to cme

2013-01-20 Thread Ramcharan Arya
Hi,

Another option is there is a FTP server available to download CUE license.

You can copy backgroud image on FTP server ( I believe it is available on
pc given for testing).

ip ftp username ipexpert
ip ftp password ipexpert

copy ftp flash:

tclsh
puts [open flash:Desktops/320x196x4/List.xml w+] {
CiscoIPPhoneImageList
ImageItem Image=”TFTP:Desktops/320x196x4/TN-greenbaypackers.png”
URL=”TFTP:Desktops/320x196x4/greenbaypackers.png”/
/CiscoIPPhoneImageList}
tclq

tftp-server flash:Desktops/320x196x4/TN-greenbaypackers.png
tftp-server flash:Desktops/320x196x4/greenbaypackers.png
tftp-server flash:Desktops/320x196x4/List.xml

do wr mem

Thanks,
Ramcharan Arya
CCIE # 28926 (Routing  Switching)


On Sun, Jan 20, 2013 at 7:22 PM, sanity insanity 
networksanitytoinsan...@gmail.com wrote:

 What is the easiest way of getting a customized background image and
 configuring it for cme  ip phones . Given that there is no TFTP server , we
 just need to use the callmanger . Please illustrate the steps...
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Re: [OSL | CCIE_Voice] Gatekeeper in Single zone : Call Processing engine prioritize

2013-01-09 Thread Ramcharan Arya
Hello Steffen,

One of the requirement of this task:

you are not allowed to use zone prefixes in this task. What are the other
way to achieve this.

Regards,
Ramcharan Arya

How should we send call to SUB vs PUB in this given situation and we are
not allow to

On Wed, Jan 9, 2013 at 12:14 AM, Steffen Bruening stbruen...@gmail.comwrote:

 You can define a gw-priority for the trunks to each endpoint.

 Am Mittwoch, 9. Januar 2013 schrieb Ramcharan Arya :

 Hello,

 I have a small query about Gatekeeper in single zone configuration it is
 prefer PUB as call processing ( as per ARQ message).

 if lab requirement says make subscriber as call processing engine then it
 does not meet requirement.

 Does any one know the answer what is the solution of this tricky
 situation.

 Regards,
 Ramcharan Arya
 CCIE # 28926


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[OSL | CCIE_Voice] Gatekeeper in Single zone : Call Processing engine prioritize

2013-01-08 Thread Ramcharan Arya
Hello,

I have a small query about Gatekeeper in single zone configuration it is
prefer PUB as call processing ( as per ARQ message).

if lab requirement says make subscriber as call processing engine then it
does not meet requirement.

Does any one know the answer what is the solution of this tricky situation.

Regards,
Ramcharan Arya
CCIE # 28926
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[OSL | CCIE_Voice] Any good blogs link on DB Replication Troubleshooting ?

2013-01-05 Thread Ramcharan Arya
Hello,

I am running on db replication issues very often.

PUB  Sub status are in these combinations 2:0, 3:0, 3:3, 4:4,

Is any these structured procedure to fix these different state of db
replications.

I have search IPX blogs but could not find information I was looking for.

Can you someone please share detail procedures about how to fix DB
replication between in following state:

Thanks  Regards,
Ramcharan Arya
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[OSL | CCIE_Voice] OWLE Lab 4 : Task 6.3 :Voicemail Forward to Alternate extension

2013-01-04 Thread Ramcharan Arya
Hello,

Have issue with particular task:

When someone call ext 2002 ( Gwashington) during voicemail greeting when
press 9 then call should forward to external number 4678124 ( PSTN Line).

This is working as expected.

Below is debug isqn q931 output:

HQ-RTR(config)#
Jan  4 23:09:57.642: ISDN Se0/0/0:23 Q931: TX - DISCONNECT pd = 8  callref
= 0x0004
Cause i = 0x8290 - Normal call clearing
Jan  4 23:09:57.654: ISDN Se0/0/0:23 Q931: RX - RELEASE pd = 8  callref =
0x8004
Jan  4 23:09:57.686: ISDN Se0/0/0:23 Q931: TX - RELEASE_COMP pd = 8
callref = 0x0004
Jan  4 23:12:44.994: ISDN Se0/0/0:23 Q931: TX - SETUP pd = 8  callref =
0x0005
Bearer Capability i = 0x8090A2
Standard = CCITT
Transfer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA98388
Exclusive, Channel 8
Display i = 'VoiceMail'
Called Party Number i = 0xC1, '4678124'
Plan:ISDN, Type:Subscriber(local)
Jan  4 23:12:45.042: ISDN Se0/0/0:23 Q931: RX - CALL_PROC pd = 8  callref
= 0x8005
Channel ID i = 0xA98388
Exclusive, Channel 8
Jan  4 23:12:45.058: ISDN Se0/0/0:23 Q931: RX - ALERTING pd = 8  callref =
0x8005
Progress Ind i = 0x8188 - In-band info or appropriate now available
Jan  4 23:12:55.886: ISDN Se0/0/0:23 Q931: TX - DISCONNECT pd = 8  callref
= 0x0005
Cause i = 0x8290 - Normal call clearing

This is no sending any calling number when Unity make call to external
phone. This does not meet Telco requirements.

Is there any way to fix this problem when call goes out to external number
from PSTN it should show calling number in isdn q931. Any thought.?

Thanks  Regards,
Ramcharan Arya
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Re: [OSL | CCIE_Voice] OWLE Lab 4 : Task 6.3 :Voicemail Forward to Alternate extension

2013-01-04 Thread Ramcharan Arya
Thank you everyone for your swift response on this.

Imaging If this site becomes as SRST. Does this service parameter can still
meet telco requirement.

If this is H.323 gateway then it is easy to fix in IOS gateway.

Regards,
Ramcharan Arya



On Fri, Jan 4, 2013 at 8:36 PM, Cory Gray corygray22...@hotmail.com wrote:

 Without that parameter you should at least see the DN of the Unity Port as
 the calling number.  Are you doing a SIP integration?  I am not sure what
 would be the calling number in that situation.

 Sent from my iPhone

 On Jan 4, 2013, at 9:14 PM, William Bell b...@ucguerrilla.com wrote:

 Check CUCM service parameters Display Original Calling Number on Transfer
 from Unity. Set this to true.

 HTH.

 -Bill
 --
 William Bell
 blog: http://ucguerrilla.com
 twitter: @ucguerrilla



 On Jan 4, 2013, at 6:22 PM, Ramcharan Arya wrote:

 Hello,

 Have issue with particular task:

 When someone call ext 2002 ( Gwashington) during voicemail greeting when
 press 9 then call should forward to external number 4678124 ( PSTN Line).

 This is working as expected.

 Below is debug isqn q931 output:

 HQ-RTR(config)#
 Jan  4 23:09:57.642: ISDN Se0/0/0:23 Q931: TX - DISCONNECT pd = 8
 callref = 0x0004
 Cause i = 0x8290 - Normal call clearing
 Jan  4 23:09:57.654: ISDN Se0/0/0:23 Q931: RX - RELEASE pd = 8  callref =
 0x8004
 Jan  4 23:09:57.686: ISDN Se0/0/0:23 Q931: TX - RELEASE_COMP pd = 8
 callref = 0x0004
 Jan  4 23:12:44.994: ISDN Se0/0/0:23 Q931: TX - SETUP pd = 8  callref =
 0x0005
 Bearer Capability i = 0x8090A2
 Standard = CCITT
 Transfer Capability = Speech
 Transfer Mode = Circuit
 Transfer Rate = 64 kbit/s
 Channel ID i = 0xA98388
 Exclusive, Channel 8
 Display i = 'VoiceMail'
 Called Party Number i = 0xC1, '4678124'
 Plan:ISDN, Type:Subscriber(local)
 Jan  4 23:12:45.042: ISDN Se0/0/0:23 Q931: RX - CALL_PROC pd = 8  callref
 = 0x8005
 Channel ID i = 0xA98388
 Exclusive, Channel 8
 Jan  4 23:12:45.058: ISDN Se0/0/0:23 Q931: RX - ALERTING pd = 8  callref
 = 0x8005
 Progress Ind i = 0x8188 - In-band info or appropriate now
 available
 Jan  4 23:12:55.886: ISDN Se0/0/0:23 Q931: TX - DISCONNECT pd = 8
 callref = 0x0005
 Cause i = 0x8290 - Normal call clearing

 This is no sending any calling number when Unity make call to external
 phone. This does not meet Telco requirements.

 Is there any way to fix this problem when call goes out to external number
 from PSTN it should show calling number in isdn q931. Any thought.?

 Thanks  Regards,
 Ramcharan Arya


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 visit www.ipexpert.com

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Re: [OSL | CCIE_Voice] CUCM-CUPS integration

2013-01-02 Thread Ramcharan Arya
check your inbound ACL and proxy domain setting.


On Wed, Jan 2, 2013 at 10:54 AM, Suresh Bhandari bring...@gmail.com wrote:

 Using UCM / UCx 7 and UPS 8.0 in VM in home lab.

 UCM and UCx integrated successfully using SCCP.

 UCM has everything configured, including SIP Trunk, required Users etc.

 Configured CUPS, and I can see that the users created in UCM are populated
 in CUPS.

 DiagnosticsSystem Troubleshooter shows that things required for basic
 integration are okay, a snapshot is attached (1.png).

 All *Feature Services *except three are up and running. Services not
 running are:

 Cisco UP XCP XMPP Federation Connection Manager
 Cisco UP XCP Message Archiver
 Cisco UP XCP Directory Service

 Have yet to go detail on the purposes of these services.

 All *Network Services *are up and running.

 Now my problem is that when I check the CUPS server in Cluster Topology,
 all the services for this server seems UNKNOWN state, snapshot attached
 (2.png). I am trying to connect to the CUPC in Deskphone mode, but... :(

 Does anyone faced this issue earlier? Please help.

 --
 Suresh Bhandari

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[OSL | CCIE_Voice] OWLE Lab 4 : Task 8.1 : Directed call park retrieval

2013-01-02 Thread Ramcharan Arya
Hi,

Task requirement is below:

OWLE Lab 4 : Task 8.1
User at Site A phone 2 can transfer an active call to extension 2456.
Ensure that the user can retrieve the call by dialing *2456.

I have configured directed call park and call is successfully park bu
unable to retrieve by dialing *2456.

When I press *2456 is play reorder tone.

Assuming enterprise parameters are setup correctly. Please suggest how this
can be achive.

Regards,
Ramcharan Arya
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Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 82, Issue 54

2012-12-28 Thread Ramcharan Arya
Hi MJ

Can you try dtmf-relay sip-notify on CUE dial-peer.

Thanks,
Ramcharan Arya


On Fri, Dec 28, 2012 at 7:37 PM, ccie_voice-requ...@onlinestudylist.comwrote:

 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. Re: [OSL | CCIE_Security] The only guy who is CCIE in 6
   (Sheraz Sheraz)
2. CUE with CUCM (CCIEing)
3. Re: CUE with CUCM (Hesham Abdelkereem)
4. Re: CUE with CUCM (Bill Lake)
5. Re: In cme -srst mode the calls to CUE do not work (Amaia Lesta)


 --

 Message: 1
 Date: Sat, 29 Dec 2012 00:07:43 +0500
 From: Sheraz Sheraz she...@live.com
 To: Jay McMickle jay.mcmic...@yahoo.com
 Cc: ccie...@onlinestudylist.com ccie...@onlinestudylist.com,
 ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com,
 ccie...@onlinestudylist.com ccie...@onlinestudylist.com,
 ccie_secur...@onlinestudylist.com
 ccie_secur...@onlinestudylist.com,
 ccie_wirel...@onlinestudylist.com
 ccie_wirel...@onlinestudylist.com
 Subject: Re: [OSL | CCIE_Voice] [OSL | CCIE_Security] The only guy who
 is  CCIE in 6
 Message-ID: snt002-w2070ce9856bd78ed8376be7f0...@phx.gbl
 Content-Type: text/plain; charset=iso-8859-1

 Yes dear i meant to say all of 6 and active! not all of 8. the most
 hardest CCIE track is CCIE data center according to my observation let see
 how much CCIE gona come out with CCIE data center.

 Regards
 Sheraz

 CC: ccie...@onlinestudylist.com; ccie_secur...@onlinestudylist.com;
 ccie...@onlinestudylist.com; ccie_wirel...@onlinestudylist.com;
 ccie_voice@onlinestudylist.com
 From: jay.mcmic...@yahoo.com
 Subject: Re: [OSL | CCIE_Security] The only guy who is CCIE in 6
 Date: Fri, 28 Dec 2012 08:21:49 -0600
 To: she...@live.com

 That's awesome, but one question. You stated all 6, and maybe I took out
 of context, but isn't there 7/8 IE tracks (if you count both SP tracks).
 See here that Cisco states 8 active IE tracks.
 http://www.cisco.com/web/learning/le3/ccie/track_comparison/index.html
 Is Roman the only person to hold 6? I thought someone had 7?

 Regards,Jay McMickle- CCIE #35355 (RS)Sent from my iPhone 5
 On Dec 28, 2012, at 6:59 AM, Sheraz Sheraz she...@live.com wrote:




  Roman Rodichev 6x CCIE #7927 ( yes six ). Roman is the first person in
  the world to hold all 6 active CCIE certifications!!! He is also the
 instructor, content developer, and owner of ieMentor

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 www.PlatinumPlacement.com
 -- next part --
 An HTML attachment was scrubbed...
 URL:
 /archives/ccie_voice/attachments/20121229/1a158df6/attachment-0001.html

 --

 Message: 2
 Date: Sat, 29 Dec 2012 01:15:31 +0300
 From: CCIEing aboaz...@gmail.com
 To: ccie_voice@onlinestudylist.com
 Subject: [OSL | CCIE_Voice] CUE with CUCM
 Message-ID:
 
 caehpkkkws_fvonxva52l4_mfuoef9mq8svwm3zxelv7nbpi...@mail.gmail.com
 Content-Type: text/plain; charset=iso-8859-1

 Hi All,

 I want to make sure that the CUE integration with CUCM is included of the
 LAB exam , right ?


 Thanks
 -- next part --
 An HTML attachment was scrubbed...
 URL:
 /archives/ccie_voice/attachments/20121229/db361223/attachment-0001.html

 --

 Message: 3
 Date: Fri, 28 Dec 2012 14:34:02 -0800
 From: Hesham Abdelkereem heshamcentr...@gmail.com
 To: CCIEing aboaz...@gmail.com
 Cc: ccie_voice@onlinestudylist.com
 Subject: Re: [OSL | CCIE_Voice] CUE with CUCM
 Message-ID: c1dce626-bc34-4e9a-92d2-7a95adf9d...@gmail.com
 Content-Type: text/plain; charset=iso-8859-1

 Yes it's but not necessarily all the attempts you will do.
 But yes CUE with CUCM is included but not in all the models just for your
 information
 On Dec 28, 2012, at 2:15 PM, CCIEing aboaz...@gmail.com wrote:

  Hi All,
 
  I want to make sure that the CUE integration with CUCM is included of
 the LAB exam , right ?
 
 
  Thanks
 
 
  ___
  For more information regarding industry leading CCIE Lab training,
 please visit www.ipexpert.com
 
  Are you a CCNP or CCIE and looking for a job? Check out
 www.PlatinumPlacement.com



 --

 Message: 4
 Date: Fri, 28 Dec 2012 17:35:22 -0600
 From

[OSL | CCIE_Voice] OWLE : lab 4 : Task 6.1 : Unable to set default pin

2012-12-23 Thread Ramcharan Arya
Hello,

I have trouble with below task in Unity connection. When I import these
users I am unable to set their PIN 12345. It is showing error message.

Import users “gwashington” and “jadams” into the Unity Connection server
ensuring the PIN is “12345”.

I already change password length and uncheck trivial password option.

Can someone please suggest fix/workaround of this issue.

Regards,
Ramcharan Arya
CCIE # 28926 ( RS)
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[OSL | CCIE_Voice] CUCM slow while changing service parameter

2012-12-22 Thread Ramcharan Arya
Hello,

My CUCM response is very slow when I update Cisco Callmanager service
paramter. It is running on vm.

Can you please suggest is there any setting on vm to improve the
performance.

Rest everything works normal.

Please suggest if you have any technical detail to share about this.

Regards,
Ramcharan Arya
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Re: [OSL | CCIE_Voice] CUCM slow while changing service parameter

2012-12-22 Thread Ramcharan Arya
Yes MOH is configured as per task.

Is there any workaroun/solution to fix this problem.

Regards,
Ramcharan Arya

On Sat, Dec 22, 2012 at 9:04 AM, scubajasona scubajas...@gmail.com wrote:

 Is multicast moh enabled? Seems enabling multicast slows my vmware
 workstation down. Floods 239.x.x.x nonstop.
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Re: [OSL | CCIE_Voice] CUCM slow while changing service parameter

2012-12-22 Thread Ramcharan Arya
Hi Bill,

This is an ESXi 5.1 think deployment.

Regards,
Ramcharan Arya



On Sat, Dec 22, 2012 at 11:33 AM, Bill Lake whl...@gmail.com wrote:

 Are you running this in VM inside an OS or ESXi



 On Sat, Dec 22, 2012 at 7:56 AM, Ramcharan Arya 
 ramcharan.a...@gmail.comwrote:

 Hello,

 My CUCM response is very slow when I update Cisco Callmanager service
 paramter. It is running on vm.

 Can you please suggest is there any setting on vm to improve the
 performance.

 Rest everything works normal.

 Please suggest if you have any technical detail to share about this.

 Regards,
 Ramcharan Arya


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

 Are you a CCNP or CCIE and looking for a job? Check out
 www.PlatinumPlacement.com



___
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[OSL | CCIE_Voice] Is there any issue with ipexpert.com?

2012-12-20 Thread Ramcharan Arya
Hello,

Can someone please check ipexpert.com it seems site is not working. I am
unable to open pdf files and watch streaming videos.

My Internet connection is good because youtube is working fine.

Regards,
Ramcharan Arya
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Re: [OSL | CCIE_Voice] Is there any issue with ipexpert.com?

2012-12-20 Thread Ramcharan Arya
Thank you for checking. Yes I can see site is up now. I did notify support
via email prior to sending email out to group.

Regards,
Ramcharan Arya


On Thu, Dec 20, 2012 at 6:57 AM, Chrysostomos Christofi 
ch.christ...@logicom.net wrote:

  Hi

 ** **

 Its working ok

 ** **

 But if you have an issue send an email to support 

 ** **

 *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
 ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Ramcharan Arya
 *Sent:* Πέμπτη, 20 Δεκεμβρίου 2012 2:02 μμ
 *To:* ccie_voice@onlinestudylist.com
 *Subject:* [OSL | CCIE_Voice] Is there any issue with ipexpert.com?

 ** **

 Hello,

 Can someone please check ipexpert.com it seems site is not working. I am
 unable to open pdf files and watch streaming videos.

 My Internet connection is good because youtube is working fine.

 Regards,
 Ramcharan Arya

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Re: [OSL | CCIE_Voice] VOL1 : Lab 9 Task 2: BACD Problem

2012-12-20 Thread Ramcharan Arya
Hi,

Finally my problem is fixed. In VOL 1 Lab 4 I change H323 port 1820 and I
did not reload my router after lab 4.

When Cisco TAC run command

debug h245 asn1

TAC  found ISDN cause code and trace the problem source. After change
port 1720 under voice service voip it fixed the problem.


Regards,
Ramcharan Arya
CCIE # 28926(RS)


On Mon, Dec 17, 2012 at 6:43 PM, Ramcharan Arya ramcharan.a...@gmail.comwrote:

 Hello Bill,

 I have bacd tcl file into flash memory.

 Brach2#show flash | inc tcl
 118  25804 Dec 15 2012 18:45:58 bacdprompts/app-b-acd-2.1.2.2.tcl
 119  36667 Dec 15 2012 18:46:14 bacdprompts/app-b-acd-aa-2.1.2.2.tcl
 Brach2#

 when I call from ext 3001 to aa pilot ext 3500 it is playing reorder tone
 with unknown number.

 --

 Dec 18 00:43:21.285: //-1/C08E1E428089/DPM/dpMatchPeersCore:
Calling Number=, Called Number=3500, Peer Info Type=DIALPEER_INFO_SPEECH
 Dec 18 00:43:21.285: //-1/C08E1E428089/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=3500
 Dec 18 00:43:21.285: //-1/C08E1E428089/DPM/dpMatchPeersCore:
Result=Success(0) after DP_MATCH_DEST
 Dec 18 00:43:21.285: //-1/C08E1E428089/DPM/dpMatchPeersMoreArg:
Result=SUCCESS(0)
List of Matched Outgoing Dial-peer(s):
  1: Dial-peer Tag=222
 Dec 18 00:43:21.285: //-1/C08E1E428089/DPM/dpMatchPeersCore:
Calling Number=, Called Number=3500, Peer Info Type=DIALPEER_INFO_SPEECH
 Dec 18 00:43:21.285: //-1/C08E1E428089/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=3500
 Dec 18 00:43:21.285: //-1/C08E1E428089/DPM/dpMatchPeersCore:
Result=Success(0) after DP_MATCH_DEST
 Dec 18 00:43:21.285: //-1/C08E1E428089/DPM/dpMatchPeersMoreArg:
Result=SUCCESS(0)
List of Matched Outgoing Dial-peer(s):
  1: Dial-peer Tag=222
 Dec 18 00:43:21.285: //-1//DPM/dpMatchPeersCore:
Calling Number=3500, Called Number=3500, Peer Info
 Type=DIALPEER_INFO_SPEECH
 Dec 18 00:43:21.285: //-1//DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=3500
 Dec 18 00:43:21.285: //-1//DPM/dpMatchPeersCore:
Result=Success(0) after DP_MATCH_DEST
 Dec 18 00:43:21.285: //-1//DPM/dpMatchPeersMoreArg:
Result=SUCCESS(0)
List of Matched Outgoing Dial-peer(s):
  1: Dial-peer Tag=222
 Dec 18 00:43:21.285: //-1//DPM/dpMatchPeersCore:
Calling Number=3500, Called Number=3500, Peer Info
 Type=DIALPEER_INFO_SPEECH
 Dec 18 00:43:21.285: //-1//DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=3500
 Dec 18 00:43:21.285: //-1//DPM/dpMatchPeersCore:
Result=Success(0) after DP_MATCH_DEST
 Dec 18 00:43:21.285: //-1//DPM/dpMatchPeersMoreArg:
Result=SUCCESS(0)
List of Matched Outgoing Dial-peer(s):
  1: Dial-peer Tag=222
 Dec 18 00:43:21.285: //-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
 Dec 18 00:43:21.285: //-1//DPM/dpAssociateIncomingPeerCore:
Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=222
 Dec 18 00:43:21.289: //-1//DPM/dpMatchPeersCore:
Calling Number=, Called Number=3500, Peer Info Type=DIALPEER_INFO_SPEECH
 Dec 18 00:43:21.289: //-1//DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=3500
 Dec 18 00:43:21.289: //-1//DPM/dpMatchPeersCore:
Result=Success(0) after DP_MATCH_DEST
 Dec 18 00:43:21.289: //-1//DPM/dpMatchPeersMoreArg:
Result=SUCCESS(0)
List of Matched Outgoing Dial-peer(s):
  1: Dial-peer Tag=222
 Dec 18 00:43:21.289: //-1/C08E1E428089/DPM/dpMatchPeersCore:
Calling Number=, Called Number=3500, Peer Info Type=DIALPEER_INFO_SPEECH
 Dec 18 00:43:21.289: //-1/C08E1E428089/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=3500
 Dec 18 00:43:21.289: //-1/C08E1E428089/DPM/dpMatchPeersCore:
Result=Success(0) after DP_MATCH_DEST
 Dec 18 00:43:21.289: //-1/C08E1E428089/DPM/dpMatchPeersMoreArg:
Result=SUCCESS(0)
List of Matched Outgoing Dial-peer(s):
  1: Dial-peer Tag=222

 I used new translation rule you provide me it is very nice way to strip
 DNIS  :-)

 I will continue to troubleshoot this until I fixed it. May be I can open a
 TAC case later if it unresolved.

 Thanks,
 Ramcharan Arya


 On Mon, Dec 17, 2012 at 5:10 PM, Bill Lake whl...@gmail.com wrote:

 This should help you avoid the dual results (one time getting +343500 and
 one time getting 3500



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[OSL | CCIE_Voice] VOL1 : Lab 9 Task 2: BACD Problem

2012-12-17 Thread Ramcharan Arya
=Success(0) after DP_MATCH_DEST
Dec 17 16:27:23.404: //-1//DPM/dpMatchPeersMoreArg:
   Result=SUCCESS(0)
   List of Matched Outgoing Dial-peer(s):
 1: Dial-peer Tag=222
Dec 17 16:27:23.404: //-1/78A3AD858034/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=3500, Peer Info Type=DIALPEER_INFO_SPEECH
Dec 17 16:27:23.404: //-1/78A3AD858034/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=3500
Dec 17 16:27:23.404: //-1/78A3AD858034/DPM/dpMatchPeersCore:
   Result=Success(0) after DP_MATCH_DEST
Dec 17 16:27:23.404: //-1/78A3AD858034/DPM/dpMatchPeersMoreArg:
   Result=SUCCESS(0)
   List of Matched Outgoing Dial-peer(s):
 1: Dial-peer Tag=222
Dec 17 16:27:23.412: ISDN Se0/0/0:15 Q931: TX - CALL_PROC pd = 8  callref
= 0x80D4
Channel ID i = 0xA98381
Exclusive, Channel 1
Dec 17 16:27:23.420: ISDN Se0/0/0:15 Q931: TX - DISCONNECT pd = 8  callref
= 0x80D4
Cause i = 0x80A6 - Network out of order
Dec 17 16:27:23.432: ISDN Se0/0/0:15 Q931: RX - RELEASE pd = 8  callref =
0x00D4
Cause i = 0x82E4 - Invalid information element contents


Can you please suggest what might be causing this problem.

Thanks  Regards,
Ramcharan Arya
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Re: [OSL | CCIE_Voice] VOL1 : Lab 9 Task 2: BACD Problem

2012-12-17 Thread Ramcharan Arya
Hello Bill,

I have bacd tcl file into flash memory.

Brach2#show flash | inc tcl
118  25804 Dec 15 2012 18:45:58 bacdprompts/app-b-acd-2.1.2.2.tcl
119  36667 Dec 15 2012 18:46:14 bacdprompts/app-b-acd-aa-2.1.2.2.tcl
Brach2#

when I call from ext 3001 to aa pilot ext 3500 it is playing reorder tone
with unknown number.

--

Dec 18 00:43:21.285: //-1/C08E1E428089/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=3500, Peer Info Type=DIALPEER_INFO_SPEECH
Dec 18 00:43:21.285: //-1/C08E1E428089/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=3500
Dec 18 00:43:21.285: //-1/C08E1E428089/DPM/dpMatchPeersCore:
   Result=Success(0) after DP_MATCH_DEST
Dec 18 00:43:21.285: //-1/C08E1E428089/DPM/dpMatchPeersMoreArg:
   Result=SUCCESS(0)
   List of Matched Outgoing Dial-peer(s):
 1: Dial-peer Tag=222
Dec 18 00:43:21.285: //-1/C08E1E428089/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=3500, Peer Info Type=DIALPEER_INFO_SPEECH
Dec 18 00:43:21.285: //-1/C08E1E428089/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=3500
Dec 18 00:43:21.285: //-1/C08E1E428089/DPM/dpMatchPeersCore:
   Result=Success(0) after DP_MATCH_DEST
Dec 18 00:43:21.285: //-1/C08E1E428089/DPM/dpMatchPeersMoreArg:
   Result=SUCCESS(0)
   List of Matched Outgoing Dial-peer(s):
 1: Dial-peer Tag=222
Dec 18 00:43:21.285: //-1//DPM/dpMatchPeersCore:
   Calling Number=3500, Called Number=3500, Peer Info
Type=DIALPEER_INFO_SPEECH
Dec 18 00:43:21.285: //-1//DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=3500
Dec 18 00:43:21.285: //-1//DPM/dpMatchPeersCore:
   Result=Success(0) after DP_MATCH_DEST
Dec 18 00:43:21.285: //-1//DPM/dpMatchPeersMoreArg:
   Result=SUCCESS(0)
   List of Matched Outgoing Dial-peer(s):
 1: Dial-peer Tag=222
Dec 18 00:43:21.285: //-1//DPM/dpMatchPeersCore:
   Calling Number=3500, Called Number=3500, Peer Info
Type=DIALPEER_INFO_SPEECH
Dec 18 00:43:21.285: //-1//DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=3500
Dec 18 00:43:21.285: //-1//DPM/dpMatchPeersCore:
   Result=Success(0) after DP_MATCH_DEST
Dec 18 00:43:21.285: //-1//DPM/dpMatchPeersMoreArg:
   Result=SUCCESS(0)
   List of Matched Outgoing Dial-peer(s):
 1: Dial-peer Tag=222
Dec 18 00:43:21.285: //-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
Dec 18 00:43:21.285: //-1//DPM/dpAssociateIncomingPeerCore:
   Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=222
Dec 18 00:43:21.289: //-1//DPM/dpMatchPeersCore:
   Calling Number=, Called Number=3500, Peer Info Type=DIALPEER_INFO_SPEECH
Dec 18 00:43:21.289: //-1//DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=3500
Dec 18 00:43:21.289: //-1//DPM/dpMatchPeersCore:
   Result=Success(0) after DP_MATCH_DEST
Dec 18 00:43:21.289: //-1//DPM/dpMatchPeersMoreArg:
   Result=SUCCESS(0)
   List of Matched Outgoing Dial-peer(s):
 1: Dial-peer Tag=222
Dec 18 00:43:21.289: //-1/C08E1E428089/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=3500, Peer Info Type=DIALPEER_INFO_SPEECH
Dec 18 00:43:21.289: //-1/C08E1E428089/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=3500
Dec 18 00:43:21.289: //-1/C08E1E428089/DPM/dpMatchPeersCore:
   Result=Success(0) after DP_MATCH_DEST
Dec 18 00:43:21.289: //-1/C08E1E428089/DPM/dpMatchPeersMoreArg:
   Result=SUCCESS(0)
   List of Matched Outgoing Dial-peer(s):
 1: Dial-peer Tag=222

I used new translation rule you provide me it is very nice way to strip
DNIS  :-)

I will continue to troubleshoot this until I fixed it. May be I can open a
TAC case later if it unresolved.

Thanks,
Ramcharan Arya


On Mon, Dec 17, 2012 at 5:10 PM, Bill Lake whl...@gmail.com wrote:

 This should help you avoid the dual results (one time getting +343500 and
 one time getting 3500

___
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www.ipexpert.com

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www.PlatinumPlacement.com

Re: [OSL | CCIE_Voice] VOL1 : Lab 9 Task 2: BACD Problem

2012-12-17 Thread Ramcharan Arya
Hi Everyone,

I am sending my Branch2 router configuration can someone please try this
config in your router. I am trying to eliminate if my .tcl script file
might have some problem.

my IOS ver : System image file is
flash:c2800nm-adventerprisek9_ivs_li-mz.124-22.T5.bin

Many Thanks,

Regards
Ramcharan Arya


On Mon, Dec 17, 2012 at 6:43 PM, Ramcharan Arya ramcharan.a...@gmail.comwrote:

 Hello Bill,

 I have bacd tcl file into flash memory.

 Brach2#show flash | inc tcl
 118  25804 Dec 15 2012 18:45:58 bacdprompts/app-b-acd-2.1.2.2.tcl
 119  36667 Dec 15 2012 18:46:14 bacdprompts/app-b-acd-aa-2.1.2.2.tcl
 Brach2#

 when I call from ext 3001 to aa pilot ext 3500 it is playing reorder tone
 with unknown number.

 --

 Dec 18 00:43:21.285: //-1/C08E1E428089/DPM/dpMatchPeersCore:
Calling Number=, Called Number=3500, Peer Info Type=DIALPEER_INFO_SPEECH
 Dec 18 00:43:21.285: //-1/C08E1E428089/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=3500
 Dec 18 00:43:21.285: //-1/C08E1E428089/DPM/dpMatchPeersCore:
Result=Success(0) after DP_MATCH_DEST
 Dec 18 00:43:21.285: //-1/C08E1E428089/DPM/dpMatchPeersMoreArg:
Result=SUCCESS(0)
List of Matched Outgoing Dial-peer(s):
  1: Dial-peer Tag=222
 Dec 18 00:43:21.285: //-1/C08E1E428089/DPM/dpMatchPeersCore:
Calling Number=, Called Number=3500, Peer Info Type=DIALPEER_INFO_SPEECH
 Dec 18 00:43:21.285: //-1/C08E1E428089/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=3500
 Dec 18 00:43:21.285: //-1/C08E1E428089/DPM/dpMatchPeersCore:
Result=Success(0) after DP_MATCH_DEST
 Dec 18 00:43:21.285: //-1/C08E1E428089/DPM/dpMatchPeersMoreArg:
Result=SUCCESS(0)
List of Matched Outgoing Dial-peer(s):
  1: Dial-peer Tag=222
 Dec 18 00:43:21.285: //-1//DPM/dpMatchPeersCore:
Calling Number=3500, Called Number=3500, Peer Info
 Type=DIALPEER_INFO_SPEECH
 Dec 18 00:43:21.285: //-1//DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=3500
 Dec 18 00:43:21.285: //-1//DPM/dpMatchPeersCore:
Result=Success(0) after DP_MATCH_DEST
 Dec 18 00:43:21.285: //-1//DPM/dpMatchPeersMoreArg:
Result=SUCCESS(0)
List of Matched Outgoing Dial-peer(s):
  1: Dial-peer Tag=222
 Dec 18 00:43:21.285: //-1//DPM/dpMatchPeersCore:
Calling Number=3500, Called Number=3500, Peer Info
 Type=DIALPEER_INFO_SPEECH
 Dec 18 00:43:21.285: //-1//DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=3500
 Dec 18 00:43:21.285: //-1//DPM/dpMatchPeersCore:
Result=Success(0) after DP_MATCH_DEST
 Dec 18 00:43:21.285: //-1//DPM/dpMatchPeersMoreArg:
Result=SUCCESS(0)
List of Matched Outgoing Dial-peer(s):
  1: Dial-peer Tag=222
 Dec 18 00:43:21.285: //-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
 Dec 18 00:43:21.285: //-1//DPM/dpAssociateIncomingPeerCore:
Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=222
 Dec 18 00:43:21.289: //-1//DPM/dpMatchPeersCore:
Calling Number=, Called Number=3500, Peer Info Type=DIALPEER_INFO_SPEECH
 Dec 18 00:43:21.289: //-1//DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=3500
 Dec 18 00:43:21.289: //-1//DPM/dpMatchPeersCore:
Result=Success(0) after DP_MATCH_DEST
 Dec 18 00:43:21.289: //-1//DPM/dpMatchPeersMoreArg:
Result=SUCCESS(0)
List of Matched Outgoing Dial-peer(s):
  1: Dial-peer Tag=222
 Dec 18 00:43:21.289: //-1/C08E1E428089/DPM/dpMatchPeersCore:
Calling Number=, Called Number=3500, Peer Info Type=DIALPEER_INFO_SPEECH
 Dec 18 00:43:21.289: //-1/C08E1E428089/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=3500
 Dec 18 00:43:21.289: //-1/C08E1E428089/DPM/dpMatchPeersCore:
Result=Success(0) after DP_MATCH_DEST
 Dec 18 00:43:21.289: //-1/C08E1E428089/DPM/dpMatchPeersMoreArg:
Result=SUCCESS(0)
List of Matched Outgoing Dial-peer(s):
  1: Dial-peer Tag=222

 I used new translation rule you provide me it is very nice way to strip
 DNIS  :-)

 I will continue to troubleshoot this until I fixed it. May be I can open a
 TAC case later if it unresolved.

 Thanks,
 Ramcharan Arya


 On Mon, Dec 17, 2012 at 5:10 PM, Bill Lake whl...@gmail.com wrote:

 This should help you avoid the dual results (one time getting +343500 and
 one time getting 3500



=~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2012.12.17 18:59:22 =~=~=~=~=~=~=~=~=~=~=~=
term len 0
Brach2#sh run
Building configuration...


Current configuration : 15223 bytes
!
! Last configuration change at 01:57:14 CET Tue Dec 18 2012
!
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname Brach2
!
boot-start-marker

Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 82, Issue 15

2012-12-13 Thread Ramcharan Arya
Hi,

Also check igmp snooping on LAN switch.

Regards,
Ramcharan Arya
CCIE # 28926 ( RS)



On Thu, Dec 13, 2012 at 1:23 PM, ccie_voice-requ...@onlinestudylist.comwrote:

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


 Today's Topics:

1. Blueprint Change (Michael Davis)
2. Re: Blueprint Change (Marko Milivojevic)
3. Purchasing updated rea l labs LAB --7 (james jacob)
4. Re: Hyphothetical Voice Infrastructure Issue Question
   (Mark Thrash (marthras))
5. Re: Hyphothetical Voice Infrastructure Issue Question
   (Mark Thrash (marthras))


 --

 Message: 1
 Date: Thu, 13 Dec 2012 09:20:17 -0800 (PST)
 From: Michael Davis michaeldavis1...@yahoo.com
 To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
 Subject: [OSL | CCIE_Voice] Blueprint Change
 Message-ID:
 1355419217.99823.yahoomail...@web126004.mail.ne1.yahoo.com
 Content-Type: text/plain; charset=iso-8859-1

 The Blueprint for voice will change, and I was also told soon. When I took
 my first attempt at my CCIE voice this week, I was told it weill not be as
 quickly as you would think by the proctor. Vik, on the other hand, is
 convinced it will sooner that latter.
 ?
 You just have to keep checking the Cisco web site for updates.
 ?
 Michael
 -- next part --
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 /archives/ccie_voice/attachments/20121213/eb4ecd13/attachment-0001.html

 --

 Message: 2
 Date: Thu, 13 Dec 2012 10:09:59 -0800
 From: Marko Milivojevic mar...@ipexpert.com
 To: Michael Davis michaeldavis1...@yahoo.com
 Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
 Subject: Re: [OSL | CCIE_Voice] Blueprint Change
 Message-ID:
 CAGDYm0wjgb53h8C5JcCF9jvZaLVJ08ZBnbu=
 gtuknepshhc...@mail.gmail.com
 Content-Type: text/plain; charset=UTF-8

 Remember that even when they announce the new blueprint, it will take
 6 months before it goes active.

 --
 Marko Milivojevic - CCIE #18427 (SP RS)
 Senior CCIE Instructor - IPexpert

 On Thu, Dec 13, 2012 at 9:20 AM, Michael Davis
 michaeldavis1...@yahoo.com wrote:
  The Blueprint for voice will change, and I was also told soon. When I
 took
  my first attempt at my CCIE voice this week, I was told it weill not be
 as
  quickly as you would think by the proctor. Vik, on the other hand, is
  convinced it will sooner that latter.
 
  You just have to keep checking the Cisco web site for updates.
 
  Michael
 
  ___
  For more information regarding industry leading CCIE Lab training, please
  visit www.ipexpert.com
 
  Are you a CCNP or CCIE and looking for a job? Check out
  www.PlatinumPlacement.com


 --

 Message: 3
 Date: Fri, 14 Dec 2012 00:38:04 +0530
 From: james jacob jamesja...@gmail.com
 To: ccie_voice@onlinestudylist.com
 Subject: [OSL | CCIE_Voice] Purchasing updated rea l labs LAB --7
 Message-ID:
 CADbpF_Re0pjPt60iW2bEqX=
 919ok-npr9jdbseyylrlt9ej...@mail.gmail.com
 Content-Type: text/plain; charset=iso-8859-1

 Guys i am purchasing real lab please let us know if anyone here like to
 share the cost so that we can get it in cheaper rate

 Below is the link

 http://www.c e r t k n o w le d g
 e.com/ipb/topic/24-ccievoicelabscom-real-labs/page__st__140

 Email me if anyone is interested further (jamesja...@gmail.com) on google
 talk

 thanks
 -- next part --
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 /archives/ccie_voice/attachments/20121214/ac385bb7/attachment-0001.html

 --

 Message: 4
 Date: Thu, 13 Dec 2012 18:44:17 +
 From: Mark Thrash (marthras) marth...@cisco.com
 To: Michael Davis michaeldavis1...@yahoo.com,
 ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
 Subject: Re: [OSL | CCIE_Voice] Hyphothetical Voice Infrastructure
 Issue Question
 Message-ID:
 079142a0a27a8641bdac42dbd9eccdda15515...@xmb-rcd-x12.cisco.com
 Content-Type: text/plain; charset=us-ascii

 Verified CDP ver 2 running?
 __
 Mark Thrash
 marth...@cisco.commailto:marth...@cisco.com
 NCE, Collaboration and Unified Communications Practice
 MSTM, MCSE, CCIE R/S 2405

 office  408-894-2086
 mobile 918-671-3237
 __
 The information transmitted is intended only for the person or entity to
 which it is addressed

[OSL | CCIE_Voice] RSVP with Moh between HQPh1 and BR1Ph1

2012-12-05 Thread Ramcharan Arya
Hello,

I have little issue with my task. I have enable RSVP call-agent between HQ
Ph1 and BR1 Phone.

Show ip rsvp reservation showing active call at 24K for G729r8 codec.

For BR1 I am using SUB as Multicast MOH audio source. It is part of
Device-pool IOS which all region setting G711 codec.

When I press hold key on HQ Ph1 :

1. Show ip rsvp res still showing once active call at 24K
2. On CUCM CLI SUb is showing as Multicast  stream
3. On BR1 I hear tone on Hold ( No music on hold without multicast)

Does it mean I need to add codec pass-through in my SCCP profile for MTP.?

Please suggest.

Regards,
Ramcharan Arya
___
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[OSL | CCIE_Voice] CUCM TFTP service restart from CLI

2012-12-03 Thread Ramcharan Arya
Hello,

Can you someone please tell me command syntax how to restart Cisco Tftp
service from CUCM CLI.?

Thanks  Regards,
Ramcharan Arya
CCIE # 28926 (RS)
___
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Re: [OSL | CCIE_Voice] CME 7.0.1 Call Park presence monitoring/BLF Speed dial

2012-11-19 Thread Ramcharan Arya
Hello,

Need help on setting up presence in CME for call park. I have enable
presence and BLF Speeddial when I park call BFL does not glow.

Can you please suggest  any other way to monitor call park slot.

Thanks,
Ramcharan Arya


On Mon, Nov 12, 2012 at 2:13 PM, Ramcharan Arya ramcharan.a...@gmail.comwrote:

 Hello,

 I have am working on CME call pickup excercise. I have testing call park
 and pickup, When I press pickup software I am getting  getting fast  tone.


 Here is my configuration.


 ephone-dn  17 octo-line
 number 4731
 label 4731
 description BR2 Ph1
 !
 ephone-dn  18 octo-line
 number 4732
 label 4732
 description BR2 Ph2
 !
 ephone-dn  19  octo-line
 number 4733
 label 4733
 Description BR2 Ph3

 !
 ephone-template  1
  softkeys idle  Newcall Pickup Redial Dnd
  softkeys connected  Hold Park Trnsfer Endcall Flash

 ephone  15
 ephone-template 1
 device-security-mode none
 mac-address 0CD9.9690.6FC2
 type 7962
 button  1:17
 !
 ephone  17
 ephone-template 1
 device-security-mode none
 mac-address 0CD9.9690.6CA8
 type 7962
 button  1:18
 !
 ephone  18
 ephone-template 1
 device-security-mode none
 mac-address 34BD.C82D.A1D7
 type 7962
 button  1:19
 !
 ephone-dn  16
  number 4735
  park-slot timeout 20 limit 2 recall
  description park slot for All Users
 !
 telephony-service
 call-park system application
 !


 Testing procedure:

 1. Call from ext 4731 to ext 4732
 2. On ext 4732 answer the phone and put call on park
 3. When I press pickup softkey to retrieve call on park is not working
 getting busy tone.


 Can you please tell me if I am doing something wrong.

 Thanks,
 Ramcharan Arya
 CCIE # 28926(RS)

___
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www.ipexpert.com

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www.PlatinumPlacement.com

Re: [OSL | CCIE_Voice] CME 7.0.1 Call Park presence monitoring/BLF Speed dial

2012-11-19 Thread Ramcharan Arya
Thank you !

Yes you are right  BLS speed for call park slot feature is in CME 7.1.
However I am running CME 7.0.1  in my lab rack that is why it did not work.

Regards,
Ramcharan Arya


On Mon, Nov 19, 2012 at 6:30 PM, William Bell b...@ucguerrilla.com wrote:

 IIRC, using BLF speed dial to monitor call park slot status was introduced
 in CME 7.1.  You can use monitor on a line. I just tested this and it
 appears to work as desired. Here is a sample:


 ephone-dn  6
  number 1001 no-reg primary
  park-slot
  allow watch
 !
 ephone  1
  device-security-mode none
  description SiteC Phone 1
  mac-address ..
  ephone-template 2
  type 7961
  no dnd feature-ring
  button  1f1 2c3,4 3m6
 !


 With the above config, if I initiate a Call Park from any station on the
 CME button 3 on ephone 1 lights up. The button can also be used to speed
 dial the call park slot.


 HTH.

 -Bill

 --
 William Bell
 blog: http://ucguerrilla.com
 twitter: @ucguerrilla



 On Nov 19, 2012, at 5:25 PM, Ramcharan Arya wrote:

 Hello,

 Need help on setting up presence in CME for call park. I have enable
 presence and BLF Speeddial when I park call BFL does not glow.

 Can you please suggest  any other way to monitor call park slot.

 Thanks,
 Ramcharan Arya


 On Mon, Nov 12, 2012 at 2:13 PM, Ramcharan Arya 
 ramcharan.a...@gmail.comwrote:

 Hello,

 I have am working on CME call pickup excercise. I have testing call park
 and pickup, When I press pickup software I am getting  getting fast  tone.


 Here is my configuration.


 ephone-dn  17 octo-line
 number 4731
 label 4731
 description BR2 Ph1
 !
 ephone-dn  18 octo-line
 number 4732
 label 4732
 description BR2 Ph2
 !
 ephone-dn  19  octo-line
 number 4733
 label 4733
 Description BR2 Ph3

 !
 ephone-template  1
  softkeys idle  Newcall Pickup Redial Dnd
  softkeys connected  Hold Park Trnsfer Endcall Flash

 ephone  15
 ephone-template 1
 device-security-mode none
 mac-address 0CD9.9690.6FC2
 type 7962
 button  1:17
 !
 ephone  17
 ephone-template 1
 device-security-mode none
 mac-address 0CD9.9690.6CA8
 type 7962
 button  1:18
 !
 ephone  18
 ephone-template 1
 device-security-mode none
 mac-address 34BD.C82D.A1D7
 type 7962
 button  1:19
 !
 ephone-dn  16
  number 4735
  park-slot timeout 20 limit 2 recall
  description park slot for All Users
 !
 telephony-service
 call-park system application
 !


 Testing procedure:

 1. Call from ext 4731 to ext 4732
 2. On ext 4732 answer the phone and put call on park
 3. When I press pickup softkey to retrieve call on park is not working
 getting busy tone.


 Can you please tell me if I am doing something wrong.

 Thanks,
 Ramcharan Arya
 CCIE # 28926(RS)


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

 Are you a CCNP or CCIE and looking for a job? Check out
 www.PlatinumPlacement.com



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

Are you a CCNP or CCIE and looking for a job? Check out 
www.PlatinumPlacement.com

[OSL | CCIE_Voice] Vol1 - Lab 6-A SRST Testing correct procedure

2012-11-15 Thread Ramcharan Arya
Hi All,
I am working on Vol-1 Lab 6A High availability. As per DSG instruction it
is describe to stop CCM services on Pub  Sub.  However 5 Lab Handbook lab
4 has different approach by adding static route on Branch A/Site B.

If later apporach is appropriate then these workbook need updates so
students do not follow inappropriate procedure.

Another query about SRST which option is appropriate ccm-manager-fallback
or telephony-services.


Please share your inputs on this.

Thanks,
Ramcharan Arya
CCIE # 28926(RS)
___
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www.ipexpert.com

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www.PlatinumPlacement.com

[OSL | CCIE_Voice] CME 7.0.1 Call Park Call Pickup issue

2012-11-12 Thread Ramcharan Arya
Hello,

I have am working on CME call pickup excercise. I have testing call park
and pickup, When I press pickup software I am getting  getting fast  tone.


Here is my configuration.


ephone-dn  17 octo-line
number 4731
label 4731
description BR2 Ph1
!
ephone-dn  18 octo-line
number 4732
label 4732
description BR2 Ph2
!
ephone-dn  19  octo-line
number 4733
label 4733
Description BR2 Ph3

!
ephone-template  1
 softkeys idle  Newcall Pickup Redial Dnd
 softkeys connected  Hold Park Trnsfer Endcall Flash

ephone  15
ephone-template 1
device-security-mode none
mac-address 0CD9.9690.6FC2
type 7962
button  1:17
!
ephone  17
ephone-template 1
device-security-mode none
mac-address 0CD9.9690.6CA8
type 7962
button  1:18
!
ephone  18
ephone-template 1
device-security-mode none
mac-address 34BD.C82D.A1D7
type 7962
button  1:19
!
ephone-dn  16
 number 4735
 park-slot timeout 20 limit 2 recall
 description park slot for All Users
!
telephony-service
call-park system application
!


Testing procedure:

1. Call from ext 4731 to ext 4732
2. On ext 4732 answer the phone and put call on park
3. When I press pickup softkey to retrieve call on park is not working
getting busy tone.


Can you please tell me if I am doing something wrong.

Thanks,
Ramcharan Arya
CCIE # 28926(RS)
___
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www.ipexpert.com

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www.PlatinumPlacement.com

[OSL | CCIE_Voice] CCIE training + Lab expenses while US Tax returns filling

2012-11-01 Thread Ramcharan Arya
Hello,

I have a stupid question which I need your thoughts.?

How do we consider these  self paid amount towards CCIE training and
Lab exam fee while filling US tax returns.

Please let me if anyone have done this while filling tax returns in past years.

Regards,
Ramcharan Arya
CCIE # 28926 (RS)
___
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[OSL | CCIE_Voice] Unable to call AVT from PSTN 911

2012-10-25 Thread Ramcharan Arya
Hello,

I am working on a task trying to call AVT extension  4005 from PSTN
911. I am able to call voice mail and auto-attendent but when I call
AVT number I get re-order tone.

Has anyone try AVT setup on CUE. Please let me know.

I got another problem with MWI lamp is not working when use left voice
mail to extension.


telephony-service
no auto-reg-ephone
  max-ephones 52
 max-dn 192
 ip source-address 10.10.128.1 port 2000
 voicemail 4004
 max-conferences 8 gain -6
 moh music-on-hold.au
 multicast moh 239.23.4.10 port 2000
 web admin system name Admin password admin
 dn-webedit
 time-webedit
 transfer-system full-consult
!
ephone-dn  98
 number 4010
 mwi off
!
!
ephone-dn  99
 number 4011
 mwi on

!
dial-peer voice 98 voip
 description MWI OFF
 incoming called-number 4010
 codec g711ulaw
 no vad
!
dial-peer voice 99 voip
 description MWI ON
 incoming called-number 4011
 codec g711ulaw
 no vad
!
dial-peer voice 4004 voip
 destination-pattern 4004
 session protocol sipv2
 session target ipv4:10.10.128.2
 dtmf-relay sip-notify
 codec g711ulaw
 no vad
!
dial-peer voice 4005 voip
   ** Description AVT ***
 destination-pattern 4005
 session protocol sipv2
 session target ipv4:10.10.128.2
 dtmf-relay sip-notify
 codec g711ulaw
 no vad

Can you please  suggest is something is missing in my configuration.

Thanks  Regards,
Ramcharan Arya
___
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Re: [OSL | CCIE_Voice] Question of Home lab virtual server setup

2012-10-22 Thread Ramcharan Arya
Hi Everyone,

Thank you very much for all your suggestions and inputs.

Finally I have decided to build this server with ESXI 5.1. I initially
ran into couple of issues.

1. Got Pink screen error message : execute disable/no execute cpu
feature to be enable
2.  vSphere client was unable to connect The underlying correction was
closed: could not establish trust relatoinship for SSL/TLS secure
channel

Both issues are fixed now. Tomorrow I will continue to setup VMs.

Finally I should be able to start this lab for Vol1 I will use
proctorlabs once I am approaching  otwards my lab dates.

Thanks once again for all your help.

Regards,
Ramcharan Arya
CCIE # 28926 ( Routing  Switching)


On Mon, Oct 22, 2012 at 7:04 PM, Craig Hill (crahill) crah...@cisco.com wrote:
 The versions on the lab are not officially supported by Cisco in VMWare, but 
 this link will give a good guideline on the requirements if you follow the 
 8.0 trains. Can you run CUP and UCXN on 80 Gig drives in VMWare? Probably, 
 but I have had very few issues following the guidelines here. Also, if you 
 are like me , you may want to take  multiple snapshots as you work  through 
 the scenarios for quick restoration,  and this will consume space as well.

 http://docwiki.cisco.com/wiki/Unified_Communications_Virtualization_Downloads_(including_OVA/OVF_Templates)


 Following these recommendations, you will be tight on space, and I would 
 again recommend using thin provisioning to help offset some of this.


 Craig





 -Original Message-
 From: Bill [mailto:whl...@gmail.com]
 Sent: Monday, October 22, 2012 5:40 PM
 To: Craig Hill (crahill)
 Cc: Ramcharan Arya; ccie_voice@onlinestudylist.com
 Subject: Re: [OSL | CCIE_Voice] Question of Home lab virtual server setup

 Most Cisco UCM servers only require a 72 or 80 GB HDD so 400 should cover you 
 2 UCM (pub  sub) CUCx, CUPS, and UCCX.

 Bill


 On Oct 22, 2012, at 1:22 PM, Craig Hill (crahill) crah...@cisco.com wrote:

 You are a little light on HD space, to run everything. I would go with ESXi 
 5 and use thin provisioning. I think you will get better performance if you 
 use ESXi vs Workstation. Until I got my hands on a UCS server, I built a 
 home based PC with 2 TB of drive space and 16 gig of RAM, and an adaptec 
 controller card. This ran 5 VM's pretty well and I had no issues with Audio.


 Craig



 -Original Message-
 From: ccie_voice-boun...@onlinestudylist.com
 [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Ramcharan
 Arya
 Sent: Sunday, October 21, 2012 10:27 PM
 To: ccie_voice@onlinestudylist.com
 Subject: [OSL | CCIE_Voice] Question of Home lab virtual server setup

 Hello,

 Apologies for sending this message to the list.

 I need your advice regarding my home lab.

 My CUCM was running on a desktop which crashed now I got new HP DL 360
 G5 server it is 2x QUAD-cpu  XEON 2.00  Gig hz.
 This server has 16 Gig RAM.

 Currently server is configured with RAID 5 so disk spare is showing
 400 Gig but I want to change to RAID 0 so I can have full 590 Gig disk space.

 I have two choices:

 1. Install Windows 2008 R2 enterprise edition 64 bit OS eval 180 days.
 Then install
 VMware  Workstations  7.0 and create 4-5 virtual machines.

 2. Install ESXI 5.1 which is free and manage Virtual environment using
 vSphere client

 Because I want to take snapshot of these VM for these virtual machine on 
 regular basis.


 This is my last attempt to build this server otherwise I will start using 
 proctorlab rack if it does not work.
 Please suggest and send your response directly.

 Has anyone setup Unifiedfx lab edition. In my setup Audio is not
 working. Please share your advice

 Thanks  Regards,
 Ramcharan Arya
 CCIE # 28926
 ___
 For more information regarding industry leading CCIE Lab training,
 please visit www.ipexpert.com

 Are you a CCNP or CCIE and looking for a job? Check out
 www.PlatinumPlacement.com
 ___
 For more information regarding industry leading CCIE Lab training,
 please visit www.ipexpert.com

 Are you a CCNP or CCIE and looking for a job? Check out
 www.PlatinumPlacement.com
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Are you a CCNP or CCIE and looking for a job? Check out 
www.PlatinumPlacement.com


[OSL | CCIE_Voice] Question of Home lab virtual server setup

2012-10-21 Thread Ramcharan Arya
Hello,

Apologies for sending this message to the list.

I need your advice regarding my home lab.

My CUCM was running on a desktop which crashed now I got new HP DL 360
G5 server it is 2x QUAD-cpu  XEON 2.00  Gig hz.
This server has 16 Gig RAM.

Currently server is configured with RAID 5 so disk spare is showing
400 Gig but I want to change to RAID 0 so I can have full 590 Gig disk
space.

I have two choices:

1. Install Windows 2008 R2 enterprise edition 64 bit OS eval 180 days.
Then install
VMware  Workstations  7.0 and create 4-5 virtual machines.

2. Install ESXI 5.1 which is free and manage Virtual environment using
vSphere client

Because I want to take snapshot of these VM for these virtual machine
on regular basis.


This is my last attempt to build this server otherwise I will start
using proctorlab rack if it does not work.
Please suggest and send your response directly.

Has anyone setup Unifiedfx lab edition. In my setup Audio is not
working. Please share your advice

Thanks  Regards,
Ramcharan Arya
CCIE # 28926
___
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www.ipexpert.com

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[OSL | CCIE_Voice] IOS Version in LAB according to blueprint

2012-09-18 Thread Ramcharan Arya
Hello,

Can someone please confirm what is the exact version in lab according
to Blueprint.I checked blueprint but it does not specify proper
details about IOS version.

On my home lab cisco 2821 router IOS version is
c2800nm-adventerprisek9_ivs-mz.124-20.T6.bin

My CME ver is 7.0.1

Please let me know if this is the correct IOS  version for router.

Thank you,
Ramcharan Arya
CCIE # 28926 (Routing  Switching)
___
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Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 79, Issue 32

2012-09-07 Thread Ramcharan Arya
Hello,

Enable Debug and trace on CME and CUE module and send output so I can
take a look at it.

Command on CME :
debug ip http authentication

Command On CUE :

Cue# trace webinterface all

Cue# show trace buffer tail

http://www.cisco.com/en/US/docs/voice_ip_comm/unity_exp/design/CP_CIPExpress/cipce19.html#wp1015605

Thanks,
Ramcharan Arya
CCIE # 28926 ( RS


On Fri, Sep 7, 2012 at 1:12 PM,  ccie_voice-requ...@onlinestudylist.com wrote:
 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
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 When replying, please edit your Subject line so it is more specific
 than Re: Contents of CCIE_Voice digest...


 Today's Topics:

1. Re: CUE GUI login (Dane Warner)


 --

 Message: 1
 Date: Fri, 7 Sep 2012 14:11:58 -0400
 From: Dane Warner dwar...@epochuniversal.com
 To: Dan Quinlan (daquinla) daqui...@cisco.com, Kevin Spicer
 ke...@kevinspicer.co.uk
 Cc: Online Study \(ccie_voice@onlinestudylist.com\)
 ccie_voice@onlinestudylist.com
 Subject: Re: [OSL | CCIE_Voice] CUE GUI login
 Message-ID:
 
 839DE40FFC6A1443A1376AC8917475B028BC15058B@EXMBX01.connectwise.hosted

 Content-Type: text/plain; charset=us-ascii

 Still no dice either way. Trying 10.10.110.3 doesn't work at all.

 Thanks for the suggestions.

 Dane Warner, CCVP
 Sr. Network Engineer
 Epoch Universal, Inc.
 (909)226-0755
 [cid:image001.jpg@01CD8CE9.989B36F0]

 Confidentiality Notice: This email and the information contained within it, 
 to include any attachment(s) is for the express use of Epoch Universal, Inc 
 and the person(s) and entities to which it is addressed.  It contains 
 confidential and privileged information.  As such it is intended solely for 
 the person(s) and entities to which it is addressed.  If you are not the 
 intended recipient, you have received this transmittal in error and are 
 obligated to delete this email and all of its attachments immediately.  Any 
 disclosure, re-creation or distribution of this transmittal or its contents 
 is strictly prohibited.  Please notify the sender immediately of your receipt 
 of this transmittal.  Please accept our sincere apologies for any 
 inconvenience.

 From: Dan Quinlan (daquinla) [mailto:daqui...@cisco.com]
 Sent: Friday, September 07, 2012 11:03 AM
 To: Kevin Spicer
 Cc: Dane Warner; Online Study (ccie_voice@onlinestudylist.com)
 Subject: Re: [OSL | CCIE_Voice] CUE GUI login

 Try putting a / after GUI in ip http path:

 ip http path flash:GUI/
 DQ
 d...@cisco.commailto:d...@cisco.com

 Sent from my iPhone

 On Sep 7, 2012, at 1:54 PM, Kevin Spicer 
 ke...@kevinspicer.co.ukmailto:ke...@kevinspicer.co.uk wrote:

 What happens if you use 10.10.110.3 in the url?

 On 7 Sep 2012 18:10, Dane Warner 
 dwar...@epochuniversal.commailto:dwar...@epochuniversal.com wrote:
 All,

 I'm in WB2 Lab 4, have done a factory restore on the CUE module, gone through 
 initial setup successfully.
 When I try to login to the GUI via http://10.10.202.2/Web/index.html, it says 
 'waiting for 
 10.10.202.2/Web/Common/HomePage.dohttp://10.10.202.2/Web/Common/HomePage.do'
  and nothing shows up, blank page. Both IE and Firefox.

 ip http server
 ip http authentication local
 no ip http secure-server
 ip http path flash:GUI

 telephony-service
 sdspfarm units 1
 sdspfarm transcode sessions 3
 sdspfarm tag 1 br2-xcoder
 conference hardware
 srst mode auto-provision dn
 em logout 0:0 0:0 0:0
  max-ephones 3
 max-dn 6 no-reg
 ip source-address 10.10.110.3 port 2000
 system message Your current options
 load 7960-7940 P00308000500
 load 7962 SCCP42.8-3-3S
 time-zone 21
 time-format 24
 date-format dd-mm-yy
 voicemail 4600
 max-conferences 8 gain -6
 call-forward pattern 
 web admin system name admin password cisco
 dn-webedit
  time-webedit
  transfer-system full-consult
 transfer-pattern 
 create cnf-files version-stamp 7960 Sep 05 2012 21:26:56

 should be all I need.

 Any ideas?

 Thanks,

 Dane Warner, CCVP
 Sr. Network Engineer
 Epoch Universal, Inc.
 (909)226-0755tel:%28909%29226-0755
 image001.jpg

 Confidentiality Notice: This email and the information contained within it, 
 to include any attachment(s) is for the express use of Epoch Universal, Inc 
 and the person(s) and entities to which it is addressed.  It contains 
 confidential and privileged information.  As such it is intended solely for 
 the person(s) and entities to which it is addressed.  If you are not the 
 intended recipient, you have received this transmittal in error and are 
 obligated to delete this email and all of its attachments immediately.  Any 
 disclosure, re-creation

[OSL | CCIE_Voice] Pub Sub VM snapshot issue with cluster

2012-09-05 Thread Ramcharan Arya
Hello Everyone,

I have take a VM snapshot of my CUCM  PUB  Sub. After completion of
first lab when I restart VM from snaptop cluster is not working
properly.
Can someone please advice me some best practice how to preserve VM
status so I can use  CUCM for next lab practice.

Another query : For Branch 2 router I am using cisco 3725 router with
NM-HDV for T1 PRI with PVDM-16.I have VWIC2-2MFT1/E1 controller it is
not compatible any suggestion how can I make it work.?

Thank you in advance.

Regards,
Ramcharan Arya
CCIE # 28926 ( RS)
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Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 79, Issue 8

2012-09-04 Thread Ramcharan Arya
Hi,

I checked https://www.ipexpert.com/Cisco/CCIE/Voice/Bootcamps their
website also advertising CCIE Voice version 4.0.

When I click on one of the link its showing version 3.0.

 My guess is version 4.0 is coming soon because ipexpert start
updating their website.


Regards,
Ramcharan Arya
CCIE # 28926 (RS)




On Mon, Sep 3, 2012 at 7:55 PM,  ccie_voice-requ...@onlinestudylist.com wrote:
 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

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 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. Re: CCIE Blueprint version change (Dan Quinlan (daquinla))
2. Voicemail SRST (Justin Barksdale)
3. UCCX (Bill Lake)
4. Re: Voicemail SRST (Dan Quinlan (daquinla))


 --

 Message: 1
 Date: Tue, 4 Sep 2012 00:12:54 +
 From: Dan Quinlan (daquinla) daqui...@cisco.com
 To: Bikramjit Singh biksinghc...@gmail.com
 Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
 Subject: Re: [OSL | CCIE_Voice] CCIE Blueprint version change
 Message-ID: 7c6fde39-79f8-48b9-a096-caed1d166...@cisco.com
 Content-Type: text/plain; charset=us-ascii

 I have no inside info on this, but given that it wasn't announced at Cisco 
 LIVE! this year, I'd expect that the next revision, if it happens in the next 
 12 months, will be 2900 and 3900 series gateways with PVDM3's, 9.x trains of 
 software, and will include video endpoints of some fashion (8900 or 9900 
 series phones, perhaps the Jabber for Windows Client, maybe even a 
 VCS-registered endpoint such as an EX unit or the Jabber Video client). As 
 for timing, Cisco LIVE! next year seems logical to me.

 Again, this is all a guess and not based on any knowledge or fact.

 DQ
 d...@cisco.com

 Sent from my iPhone

 On Sep 3, 2012, at 1:51 PM, Bikramjit Singh biksinghc...@gmail.com wrote:

 Hi Folks,

 Is there anyone who has a logical prediction on when is the blueprint about 
 to change?
 Also, what kind of changes are we expecting; both from software and hardware 
 (voice gateways, pvdm, etc..) perspective.

 Thanks!
 ___
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 visit www.ipexpert.com

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 www.PlatinumPlacement.com


 --

 Message: 2
 Date: Mon, 3 Sep 2012 20:15:49 -0400
 From: Justin Barksdale jus...@barksdale.net
 To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
 Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
 Subject: [OSL | CCIE_Voice] Voicemail SRST
 Message-ID: b824a61d-8404-461f-86c5-1dd6b6298...@barksdale.net
 Content-Type: text/plain;   charset=us-ascii

 Steve,

 Voicemail while in SRST can be accomplished using the redirecting number in 
 order to reach the required mailbox.  Alternate extension are not required.

 Justin Barksdale
 CCIE #29866


 Sent from my iPhone 4.

 On Sep 3, 2012, at 8:00 PM, ccie_voice-requ...@onlinestudylist.com wrote:

 Voicemail SRST


 --

 Message: 3
 Date: Mon, 3 Sep 2012 19:41:48 -0500
 From: Bill Lake whl...@gmail.com
 To: CCIE Study ccie_voice@onlinestudylist.com
 Subject: [OSL | CCIE_Voice] UCCX
 Message-ID:
 cadpb93on1m2ybbrofyktjw4zp-t4afi+h_fk7-j4kfaggjo...@mail.gmail.com
 Content-Type: text/plain; charset=iso-8859-1

 Trying to install UCCX on laptop to practice scripting but while I
 installed Server 2003 and did the registry update, it says

 Installation of Cisco Unified Contact Center Express cannot be performed
 on the current version of MCS OS Service Release.  Please upgrade the OS
 image version to 2003.1.5a or higher and try again

 Does anyone know how to overcome this issue. I tried the following
 [image: Inline image 1]

 Bill
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 --

 Message: 4
 Date: Tue, 4 Sep 2012 00:55:08 +
 From: Dan Quinlan (daquinla) daqui...@cisco.com
 To: Justin Barksdale jus...@barksdale.net
 Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
 Subject: Re: [OSL | CCIE_Voice] Voicemail SRST
 Message-ID: 4519d20e-485f-4fdc-91a6-333f97671...@cisco.com
 Content

[OSL | CCIE_Voice] Question about PSTN phone registration?

2012-09-04 Thread Ramcharan Arya
Hello,

I  have a question about  PSTN WAN router physical connectivity.
According to ipexpert workbook vol 1 configurations PSTN WAN router is
suppose to be connect on Port 2 on HQ-3750 Switch on VLAN 10.

On PSTN WAN Router Fa0/0 IP is 10.10.100.2

On HQ-3750  switch;

interface Vlan10
 ip address 10.10.100.3 255.255.255.0
!
interface FastEthernet1/0/2
 switchport access vlan 10
 switchport mode access
 switchvoice vlan 20
 spanning-tree portfastport


Can someone confirm to me.?

Does any one know the IP address of the PSTN Phone.? I am assuming
there is some DHCP Pool for PSTN Phone which is not  shown in the
configuration.

It would be really nice to have physical connection diagram for voice
v rack similar to RS
This will help me to setup my rack.

Thanks  Regards,
Ramcharan Arya
CCIE # 28926 ( RS)
___
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[OSL | CCIE_Voice] cisco 3725 for PSTN WAN

2012-08-31 Thread Ramcharan Arya
Hi Everybody,

I am building home lab for voice track. I have Cisco 3725 router with
12.4 Advance IP service IOS with PVDM2-32  and 2 VWIC-2MFT1/E1
controllers.

Is this router appropriate for PSTN WAN router.

Additional question for PSTN Phone is Cisco 7942 phone sufficient for
testing. Because I have 2x7960  4x7942 phone.

Please let me know your inputs/feedback.

Have a nice long weekend !!

Thanks  Best Regards,
Ramcharan Arya
CCIE # 28926 ( Routing  Switching)
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[OSL | CCIE_Voice] Home Lab Setup Question : Single PoE Switch with multiple VLANs (VRF-Lite/Private VLAN) for HQ, BR1, BR2 and PSTN Phone

2012-08-29 Thread Ramcharan Arya
Hello,

I am setting up my home lab. I have single 3750 PoE switch. Is it OK
to use one single switch with multiple VLAN or vrf-lite for HQ, BR1,
BR2 and PSTN Phone.

I understand setup will limit me for QoS  implementation.

Please share you input if someone has done similar setup for home lab
using single PoE switch.

Thanks  Regards,
Ramcharan Arya
CCIE # 28926 ( RS)
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Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 78, Issue 87

2012-08-28 Thread Ramcharan Arya
Hi Jason,

Congratulations !!!

Thank you very much for sharing valuable information.

Regards,
Ramcharan Arya
CCIE # 28926 ( Routing  Switching)


On Tue, Aug 28, 2012 at 8:24 AM,
ccie_voice-requ...@onlinestudylist.com wrote:
 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. Re: SLRG is same as LRG??? (Jason Murray)
2. CCIE Passed! (Jason Murray)
3. Re: CCIE Passed! (Dan Quinlan (daquinla))


 --

 Message: 1
 Date: Tue, 28 Aug 2012 08:23:08 -0400
 From: Jason Murray murr...@usa.com
 To: Krishna vinayak_...@yahoo.com,Online Study
 ccie_voice@onlinestudylist.com
 Subject: Re: [OSL | CCIE_Voice] SLRG is same as LRG???
 Message-ID: 20120828122308.203...@gmx.com
 Content-Type: text/plain; charset=utf-8

 LRG is only found mentioned in the Device Pool configuration. And the only 
 way you can use that setting is if you use SLRG. So that is how those two 
 correlate to each other.

 - Original Message -
 From: Krishna
 Sent: 08/27/12 11:11 PM
 To: Online Study
 Subject: Re: [OSL | CCIE_Voice] SLRG is same as LRG???

  hi folks,

 i was wondering slrg and lrg are same in terms of functionality... can both 
 viewed as same ..??? please advise me on this query...

 thank you
 krishna.
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 --

 Message: 2
 Date: Tue, 28 Aug 2012 09:13:43 -0400
 From: Jason Murray murr...@usa.com
 To: Online Study ccie_voice@onlinestudylist.com
 Subject: [OSL | CCIE_Voice] CCIE Passed!
 Message-ID: 20120828131344.203...@gmx.com
 Content-Type: text/plain; charset=utf-8

 Hello everyone just wanted to send a note out that I have passed my CCIE last 
 week #36408!! First and foremost I want to thank God for the ability to 
 retain as much information as I had to take in. Definitely a lot of prayers 
 were answered last week. Next I have to thank my family for sure for putting 
 up with me for the last year (esp these last 4 months of not having my full 
 attention almost daily). They have sacrificed with me. It will be nice not 
 getting up pretty much every morning including Saturdays and turning on my 
 home lab. Will enjoy some peace and quiet of not having the hum of servers 
 and routers running in my office. I wanted to say thanks to Vik and the 
 ipexpert crew. I used their BLS product for about a year now. I had my own 
 equipment so it was easy for me to turn it on almost everyday and run through 
 the walkthroughs and everything until I had it down. Then a month out I went 
 to the One Week Lab Experience which I think put it all together and 
 finalized my 
 st
  udies. I would like to have said I passed my first time, as I did come very 
 close I didn't but the next 30 days I worked with Vik to help understand 
 where I was needing improvements and the next time I went in I passed. Vik 
 went above and beyond in helping me understand it. He truly loves what he 
 does and enjoys helping people. If he didn't he would have put up with all my 
 emails. I know he's especially glad I passed so he doesn't have to be 
 bothered by me anymore lol :). Also I did use another product some (Cisco 
 360) which helped to see the grading process using their graded assessments. 
 But as far as actual study material I used ipexpert exclusively.

  So to help those out on this same journey as I was I thought I would put my 
 strategy out there so it may help someone else.

  My strategy was using the device based approach (search google and you will 
 find a nice video on this) where I made a table and read through the entire 
 guide and put each question under the device it corresponded to. Once that 
 was complete I started with SiteC and did that router config along with CUE. 
 Then moved to SB, complete, then check CUE and finish. Then to SA RTR finish 
 that, check CUE if needed. Then finish the 3750 config. On the routers if I 
 happened to be doing QOS or RSVP or something I would configure on the router 
 I was on and then I would hop over to the router it was going to and finish 
 that particular config on it. Then come back and finish up on the router I 
 was on. Oh and the first time I hit the NTP config I would hop over to CUCM 
 and configure it and then ssh into it and run a utils dbreplication repair 
 all since NTP does have a tendency to mess up replication

[OSL | CCIE_Voice] Unable to logon into Call Manager after installation

2012-08-27 Thread Ramcharan Arya
Hello,

I am starting for my CCIE Voice track. As a part of this preperation I
am building home lab for CCIE voice. I have installed Call Manager
7.0.1 on Dell 780 Optiplex dekstop running windows 7 64-bit  8 Gig
RAM,  Running VM workstation 7.0. I have create a VM  and call manager
installed successfully.

When I restart my virtual machine I am unable to logon into CLI using
admin userid and password. I am typing correct userid and password. I
am able to ping Virtual machine IP address but unanle to open http
url.?

Can someone please assist me if you have installed Call manager on virtual.

Thanks  Regards,
Ramcharan Arya
CCIE # 28926 ( Routing  Switching)
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[OSL | CCIE_Voice] CME CUE with L3 IP phone subnet on switch

2012-07-23 Thread Ramcharan Arya
Hi,

I am working on a problem where CME router is configure as
routing-on-a-stick with 4 different VLAN due to performance problem I
want to move layer-3 routing on cisco 3650 switch.

Here is my configuration on CME and L3 switch. Everything is working
fine. External called is able to record voice-mail and phone and able
to access mailbox on CUE.
--

Please suggest if you see anything  drawback with this configurations.

interface GigabitEthernet0/1
Description ==L3 connection to switch port Gi0/1
 ip address 172.25.169.1 255.255.255.192
 duplex auto
 speed auto
!

interface Loopback1
 ip address 172.30.23.37 255.255.255.252
!

interface Service-Engine0/0
 ip unnumbered Loopback1
 service-module ip address 172.30.23.38 255.255.255.252
 service-module ip default-gateway 172.30.23.37
!
ip route 172.30.23.38 255.255.255.255 Service-Engine0/0

!
router eigrp 64831
 passive-interface default
 no passive-interface GigabitEthernet0/1
 network 172.25.169.1 0.0.0.0
 network 172.30.23.37 0.0.0.0
 auto-summary
!
telephony-service
 max-ephones 58
 max-dn 150
 ip source-address 172.30.23.37 port 2000

!
dial-peer voice 4701 voip
 destination-pattern 4701
 session protocol sipv2
 session target ipv4:172.30.23.37
 dtmf-relay h245-alphanumeric
 codec g711ulaw
 no vad

---

on 3560 Switch
--

ip dhcp excluded-address 172.25.169.65 172.25.169.67
!
ip dhcp pool iphone
   network 172.25.169.64 255.255.255.192
   default-router 172.25.169.65
   option 150 ip 172.30.23.37
!
interface Vlan70
 ip address 172.25.169.65 255.255.255.192
end
!
interface GigabitEthernet0/17
 switchport mode access
 switchport voice vlan 70
 spanning-tree portfast

!

router eigrp 64831
no auto-summary
 network 172.25.169.2 0.0.0.0
 network 172.25.169.65 0.0.0.0
!
--

cmerouter#sh cdp neigh | inc Gig 0/1
Switch   Gig 0/1176 R S I WS-C3560G Gig 0/1

!

Thanks  Regards,
Ramcharan Arya
CCIE # 28926
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Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 68, Issue 121. ?;4

2011-10-15 Thread Ramcharan Arya

Qq/Rrwecgfhojuupop

Sent from my iPhone

On Oct 15, 2011, at 2:27 AM, ccie_voice-requ...@onlinestudylist.com wrote:

 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. Re: MOH multicast (Ashraf Ayyash)
   2. MVA Configuration Doubts. (Yash)
   3. Re: IP Blue (running multiple instances) (Ashraf Ayyash)
   4. Re: thoughts on plan / type / name requirementsforani / dnis
  (Kshitij Singhi)
 
 
 --
 
 Message: 1
 Date: Fri, 14 Oct 2011 18:55:30 -0700
 From: Ashraf Ayyash ash.ayy...@gmail.com
 To: Mohammed Al baqari baqari.voic...@gmail.com
 Cc: ccie_voice@onlinestudylist.com
 Subject: Re: [OSL | CCIE_Voice] MOH multicast
 Message-ID:
CAEW==nvQTQxpg4=c-OXGUbKFG9Xbe3f=cadiqtpom3xhs-h...@mail.gmail.com
 Content-Type: text/plain; charset=iso-8859-1
 
 hello All ,
 
 this is called MMOH Snooping and the ccm is a  victim , he have no idea
 about what you do in the Multicast moh stream , so the answer here is not
 fully accurate , but you will see the counter increament in the show perf
 query command because the ccm think he is streaming the moh and he is but we
 are stopping the stream from reaching the phone .
 
 the fact is you are configueding the router to stream MOH from the flash of
 the router and you are using the same exact ip and port of what the ccm
 using to stream the MOH , so the Phone will request to have MOH stearmed and
 you will see the ccm respoding with skinny message to listen to the ip and
 the port you have configued on the ccm
 so the phone will try to subscibe to ths stream  which is already played
 from the flash and so you will get the mmoh from the flash , at this point
 of time , the ccm think he is streaming the moh from himself and thats why
 we calling this
 snooping .
 
 you can do deb ephone moh shwich will show you the stream and the ip add
 that you flood the music to by command the multicast and you can collect
 sniffer trace from the phone to see the multicast traffic as well .
 
 Ash
 
 On Fri, Oct 14, 2011 at 2:50 PM, Mohammed Al baqari 
 baqari.voic...@gmail.com wrote:
 
 You are requesting MoH server from CUCM so the count will be increased in
 CUCM.
 
 However, you are hearing the stream from BR1 flash . This is continuously
 played and once the channel is opened on phone by CUCM to get the multicast
 stream you will hear the multicast coming from flash
 
 
 
 
 On Fri, Oct 14, 2011 at 11:44 PM, Ccie Voice v.c...@yahoo.com wrote:
 
  But I am using router flash memory Not CUCM
 
  --
 *From: *Mohammed Al baqari baqari.voic...@gmail.com;
 *To: *Ccie Voice v.c...@yahoo.com;
 *Cc: *CCIE Study ccie_voice@onlinestudylist.com;
 *Subject: *Re: [OSL | CCIE_Voice] MOH multicast
 *Sent: *Fri, Oct 14, 2011 3:24:18 PM
 
  use the CLI command [show perf query class Cisco MOH Device] on CUCM
 CLI interface. The count as below should be increased. This will reflect in
 CUCM and not on BR1 router.
 
 admin:show perf query class Cisco MOH Device
 ==query class :
 
 - Perf class (Cisco MOH Device) has instances and values:
MOH_2   - MOHHighestActiveResources  = 1
MOH_2   - MOHMulticastResourceActive = 1
MOH_2   - MOHMulticastResourceAvailable  = 25
MOH_2   - MOHOutOfResources  = 0
MOH_2   - MOHTotalMulticastResources = 25
MOH_2   - MOHTotalUnicastResources   = 250
MOH_2   - MOHUnicastResourceActive   = 0
MOH_2   - MOHUnicastResourceAvailable= 250
 
 On Fri, Oct 14, 2011 at 10:14 AM, Ccie Voice v.c...@yahoo.com wrote:
 
 Hi everybody,
 I am trying to configure Multicast MOH in Branch 1 router, I added the
 file to router flash and I configure CUCM to use multicast.
 
 when I am calling from one phone to another in BR1 I am able to hear the
 file that I uploaded to router flash memory. but if I tried to use the
 following command:
 
 sho ccm-manager music-on-hold
 the result:
 Current active multicast sessions : 0
 
 is it using Multicast?
 how I can make sure that I am using multicast not unicast?
 
 
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 www.PlatinumPlacement.com http://www.platinumplacement.com/
 
 
 
 
 ___
 For more information regarding industry leading 

[OSL | CCIE_Voice] 4-port HWIC-ESW - Phone can't power-on

2011-08-19 Thread Ramcharan Arya
Hi,

I have cisco 2821 router and 4-port HWIC-WSP PoE card. Ip phone connected to
one of the port on switch module but it is unable to power on.

Here is module information:

Router#sh inv | inc 4
NAME: 4 Port FE Switch on Slot 0 SubSlot 0, DESCR: 4 Port FE Switch
PID: HWIC-4ESW , VID: V01 , SN: FOC122434S2
NAME: WIC/VIC/HWIC 0 Power Daughter Card, DESCR: 4-Port HWIC-ESW Power
Daughter Card
PID: ILPM-4, VID: V01 , SN: FOC122710SV


Can you someone please share your configure of your router for comparison.

Thanks,
Ramcharan Arya
___
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