Re: [OSL | CCIE_Voice] Vol2 Lab10 Q 4.6 - Access VM from the phone

2010-11-10 Thread Roger Carpio
Hello David,

I think this might help you accomplish your goal:

http://www.cisco.com/en/US/docs/voice_ip_comm/cupa/visual_voicemail/7.0/english/install/guide/install_phones.html#wp1056189

Regards,
Roger Carpio.

On Wed, Nov 10, 2010 at 8:25 AM, David A david.a...@gmail.com wrote:

 Hi Tamer,

 I am able to access voicemail using messages button. I was trying to
 configure the services button similar to CUE voiceview but on Unity
 Connection

 Thanks,
 DA

 On 11/10/10, rsmail...@solcon.nl rsmail...@solcon.nl wrote:
  Tamer,
 
  try dialing the pilot point directly.
  for example dial 5600 (VM pilot)
  Do you get the unity prompt or not ?
  then check the VM pilot and VM profile settings.
  make sure the phone uses the VM profile.
 
  also if you press the voicemail button, and nothing happens.
  the service doesn't exist or no VM profile configured for that phone.
 
  regards, Ron
 
 
  Hi Tamer,
 
  Any idea what the service url is to configure on CUCM for Voice View
  Access on Unity Connection
 
 
  Thanks,
  DA
 
  On 11/10/10, Tamer Ismail tih...@gmail.com wrote:
  David,
  Yes it should works.
  Check if you have voicemail service in phone services tab, and check
  what is
  your default voicemail profile.
 
  Tamer,
 
  -Original Message-
  From: ccie_voice-boun...@onlinestudylist.com
  [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of David A
  Sent: Wednesday, November 10, 2010 6:15 AM
  To: ccie_voice@onlinestudylist.com
  Subject: [OSL | CCIE_Voice] Vol2 Lab10 Q 4.6 - Access VM from the phone
 
  Hi All,
 
  The question requires that all VM users should be able to log onto
  Voicemail system via the services button on phone.
 
  This works fine for CUE access.
 
  Can it be done on the Unity Connection. I couldnt find anything. Is it
  possible?
 
 
  Thanks,
  DA
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Re: [OSL | CCIE_Voice] Vol2 Lab10 Q 4.6 - Access VM from the phone

2010-11-10 Thread Roger Carpio
David,

Is the Service Category * setup as Java MIDLet or XML Service?

Regards,
Roger Carpio.

On Wed, Nov 10, 2010 at 9:26 AM, David A david.a...@gmail.com wrote:

 Thanks Roger,

 I followed the document. When I test the url

 http://x.x.x.x/midlets/VisualVoicemail/VisualVoicemail.jad

 with my unity connection ip addr. I get an error -NOT Found. Do I need
 a seperate license?

 Thanks for the help.

 DA

 On 11/10/10, Roger Carpio roger.car...@gmail.com wrote:
  Hello David,
 
  I think this might help you accomplish your goal:
 
 
 http://www.cisco.com/en/US/docs/voice_ip_comm/cupa/visual_voicemail/7.0/english/install/guide/install_phones.html#wp1056189
 
  Regards,
  Roger Carpio.
 
  On Wed, Nov 10, 2010 at 8:25 AM, David A david.a...@gmail.com wrote:
 
  Hi Tamer,
 
  I am able to access voicemail using messages button. I was trying to
  configure the services button similar to CUE voiceview but on Unity
  Connection
 
  Thanks,
  DA
 
  On 11/10/10, rsmail...@solcon.nl rsmail...@solcon.nl wrote:
   Tamer,
  
   try dialing the pilot point directly.
   for example dial 5600 (VM pilot)
   Do you get the unity prompt or not ?
   then check the VM pilot and VM profile settings.
   make sure the phone uses the VM profile.
  
   also if you press the voicemail button, and nothing happens.
   the service doesn't exist or no VM profile configured for that phone.
  
   regards, Ron
  
  
   Hi Tamer,
  
   Any idea what the service url is to configure on CUCM for Voice View
   Access on Unity Connection
  
  
   Thanks,
   DA
  
   On 11/10/10, Tamer Ismail tih...@gmail.com wrote:
   David,
   Yes it should works.
   Check if you have voicemail service in phone services tab, and check
   what is
   your default voicemail profile.
  
   Tamer,
  
   -Original Message-
   From: ccie_voice-boun...@onlinestudylist.com
   [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of David
 A
   Sent: Wednesday, November 10, 2010 6:15 AM
   To: ccie_voice@onlinestudylist.com
   Subject: [OSL | CCIE_Voice] Vol2 Lab10 Q 4.6 - Access VM from the
   phone
  
   Hi All,
  
   The question requires that all VM users should be able to log onto
   Voicemail system via the services button on phone.
  
   This works fine for CUE access.
  
   Can it be done on the Unity Connection. I couldnt find anything. Is
 it
   possible?
  
  
   Thanks,
   DA
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   visit www.ipexpert.com
  
  
  
  
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 please
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Re: [OSL | CCIE_Voice] The simplest config for CBarge

2010-11-13 Thread Roger Carpio
Hello,

Do you have more details about this issue? It wouldn´t hurt if you add some
sdi traces to your problem description. Can you see CUCM looking for CFB
resources in the Default List?

Regards.
Roger Carpio.

On Sat, Nov 13, 2010 at 4:52 AM, Miron Kobelski findko...@gmail.com wrote:

 Hello,
 Hi,

 I'm playing with CBarge on a shared line between HQ phone 2 and BR1 phone
 2. The proctor guide specifies that 2 dspfarm conferencing profiles are
 needed (1 on HQ router and 1 on BR1 router), each associated to separate
 MRG/MRGL, which is later applied to both phones.

 I made 2 observations:
 1) when I tried not to use MRGLs and leave both CFBs in the default MRGL,
 CBarge didn't work. why is that? The media resources not associated with any
 MRG should be available to everybody.

 2) when I configured only one CFB (no matter whether on HQ or BR1 router)
 and put it in a single MRG/MRGL associated with both phones, it worked. That
 would suggest we need only 1 CFB.

 Could anybody confirm these observations? I'm especially interested in
 explaining question 1. Is there anything in the documentation explaining
 this behaviour?

 regards
 kobel

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Re: [OSL | CCIE_Voice] ? wild card....

2010-11-14 Thread Roger Carpio
Mudassar,

Which wildcard did you use? 91X will block any number from 91[0 thru 9].
Actually this wouldn't let you type any 4th digit since it is the closest
match when dialing a 900 numbers.

To block 900 numbers; I would use the device-line CSS approach with RP
91900XXX in partition blocked assigned to the IP phones line CSS.

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/7x/dialplan.html#wp1150997

Regards,
Roger Carpio.

On Sun, Nov 14, 2010 at 8:41 AM, Mudassar Khalid mudas...@hotmail.comwrote:

  Hi Experts,
 while practicing volume 1 lab 5.9: block 91900? pattern for all phones. ?
 wild card has no effect if the number is not consecutive
 zeros after 91900 digit string.
 CUCM help says: The route pattern 91X? routes or blocks all numbers in
 the range 91 through 919.
 But I am not able to hit this pattern unless I dial consecutive zeros.
 Would anybody highlight its usage?

 Thanks,
 Mudassar

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Re: [OSL | CCIE_Voice] Mobile Voice Access

2011-01-05 Thread Roger Carpio
Hello ShinGei,

Thanks for the info. In the end, MGCP will not make it for us [?]

Regards,
Roger Carpio.

On Sun, Jan 2, 2011 at 9:34 AM, ShinGei Yong shingei.y...@gmail.com wrote:

 Hi Roger,

 MVA on a MGCP control gateway is possible.In fact,that is a coexisting of
 both
 MGCP and H323 on the same gateway,but you could not used a MGCP
 control PRI for MVA.

 you may refer to Netpro for own interest.
 https://supportforums.cisco.com/thread/2005673

 Shingei.

 On Sun, Jan 2, 2011 at 11:16 PM, Shrini linuxbos...@gmail.com wrote:

  Hi ShinGei , bkvalentine, Rogers et al

 I remember it was successful last time when I configured it another lab
 when HQ was h323.

 Now I was confused around dial-peers hence had the question.

 I will give a try now with MGCP + H323 on HQ and it should work.

 Thanks all.
 Shrini

  *From:* ShinGei Yong [mailto:shingei.y...@gmail.com]
 *Sent:* Sunday, January 02, 2011 6:42 AM
 *To:* Shrini; ccie_voice@onlinestudylist.com
 *Subject:* Re: [OSL | CCIE_Voice] Mobile Voice Access

 Hi Shrini,

 What about your UCM configuration?
 1. is your H323 GW registered with UCM?
 2. what is your dialing behavior internally?4 or 10?if is 4,
 then your in  outbound dp should be 4 digit patten as well
 instead of 10.

 Please provide more info

 Shingei

 On Sun, Jan 2, 2011 at 4:13 PM, Shrini linuxbos...@gmail.com wrote:

  Hi Experts,

 *Wish you all a Happy and Prosperous New Year 2011*

 First question this year :-)

 HQ Site is MGCP.

 When I call HQ Phone 5002 , HQ PSTN is ringing  --  all is good.

 I have configured MVA number 5999 in service parameters and
 Media Resources -- MVA -- 5999 / PT-INTERNAL / English

 on router.

 application
   service cmm http://177.1.10.10:8080/ccmivr/pages/IVRMainpage.vxml
   !
 !
 dial-peer voice 5999 pots
  service cmm
  incoming called-number 2123945999
  no digit-strip

 Also on CUCM :


 But when I call 2123945999 from HQ PSTN 2123942123 I am getting fast busy
 tone. It is not invoking the vxml script.


 What am I doing wrong here ?

 TIA
 Shrini






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Re: [OSL | CCIE_Voice] Mobile Voice Access

2011-01-05 Thread Roger Carpio
Hello,

Did you add this gateway as H323 in CUCM as well? Features and Services
guide does not seem to mention MGCP for this feature.

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_0_1/ccmfeat/fsmobmgr.html#wp1206437

Were you specifically asked to have MVA and MGCP without H323?

Regards,
Roger Carpio.

On Sun, Jan 2, 2011 at 2:13 AM, Shrini linuxbos...@gmail.com wrote:

  Hi Experts,

 *Wish you all a Happy and Prosperous New Year 2011*

 First question this year :-)

 HQ Site is MGCP.

 When I call HQ Phone 5002 , HQ PSTN is ringing  --  all is good.

 I have configured MVA number 5999 in service parameters and
 Media Resources -- MVA -- 5999 / PT-INTERNAL / English

 on router.

 application
   service cmm http://177.1.10.10:8080/ccmivr/pages/IVRMainpage.vxml
   !
 !
 dial-peer voice 5999 pots
  service cmm
  incoming called-number 2123945999
  no digit-strip

 Also on CUCM :


 But when I call 2123945999 from HQ PSTN 2123942123 I am getting fast busy
 tone. It is not invoking the vxml script.


 What am I doing wrong here ?

 TIA
 Shrini






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Re: [OSL | CCIE_Voice] T1 Pri Issue

2011-01-06 Thread Roger Carpio
Hello Deepak,

Have you ran the debug isdn q921 on this GW? If so; do you see messages
going back and forth between the RTR and the PSTN?
Could you please attach the configuration of T1 controller 0/0/0 as well?

Regards,
Roger Carpio.

On Thu, Jan 6, 2011 at 3:24 PM, Deepak sidana sidana_dee...@yahoo.comwrote:

 Hi All,

 I am trying to connect the T1 from Br1-RTR to PSTN-WAN. Only when i use
 service mgcp, under controller, layer 2 isdn staus as TEI_ASSIGNED.

 At PSTN-WAN Router, i am using isdn protocol-emulate network under
 s0/0/0:23

 Branch1 Config:-

 BR1-RTR#sh isdn sta
 Global ISDN Switchtype = primary-net5
 ISDN Serial0/0/0:23 interface
  dsl 0, interface ISDN Switchtype = primary-net5
 Layer 1 Status:
  ACTIVE
 Layer 2 Status:
  TEI = 0, Ces = 1, SAPI = 0, State = TEI_ASSIGNED

 !
 controller T1 0/0/1
  framing esf
  linecode b8zs
  cablelength long 0db
 !
 interface Serial0/0/0:23
  no ip address
  encapsulation hdlc
  isdn switch-type primary-net5
  isdn bind-l3 ccm-manager
  isdn incoming-voice voice
  no cdp enable

 Please share you experince, if some one faced the same issue.

 ThanksRgds
 Deepak Sidana.


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Re: [OSL | CCIE_Voice] GK-Controlled Trunk in CUCM

2011-01-17 Thread Roger Carpio
Michael,

You do not need to add the GW to CUCM; otherwise this will not be scalable
at all. I had the same confusion but then I got it to work without adding my
GW (CME) to CUCM.

Regards,
Roger Carpio.

On Mon, Jan 17, 2011 at 2:16 PM, Michael Luo hout...@gmail.com wrote:

 I was trying to understand the Gatekeeper-controlled Trunk in CUCM.

 If I configured the Gatekeeper-controlled Trunk, do I still have to
 configured the H323 gateways?

 Consider this scenario:

 GW --- GK --- CUCM

 If there's a call from GW to CUCM, my understanding was:

 1) GW sends ARQ to GK.
 2) GK resolves the called number and sends ACF to GW.
 3) GW will try to communicate with CUCM.

 In step 3, if GW was not configured in CUCM, the call will fail.

 Did I understand this wrong?

 Thanks!
 Michael


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Re: [OSL | CCIE_Voice] GK-Controlled Trunk in CUCM

2011-01-17 Thread Roger Carpio
Michael,

All I can say is that I had the same behavior in the past; now I can get it
to work without adding the CME to CUCM. I know that by default for security
reasons CUCM will deny connections from unknown sources; however to me the
gatekeeper controlled trunk seems like an exception. In regards to the CSS;
you use the Trunk's CSS for incoming calls.

If you wish I can take some traces from my lab and share them with you for
further analysis.

Regards,
Roger Carpio.

On Mon, Jan 17, 2011 at 5:40 PM, Michael Luo hout...@gmail.com wrote:

 Hi Roger,

 I was hoping the same as you.  But I couldn't figure out who to get it
 worked without adding the H323 GW (CME) to CUCM.

 From my observation, adding H323 GW (CME) to CUCM is needed because:

 1) For security reason, CUCM won't accept H323 connection from an unknown
 source (ie. not configured in CUCM).
 2) Without the GW configured in CUCM, there's no way to configure in-bound
 call parameters, such as Calling Search Space, Significant Digits, etc.

 So I guess without the GW being configured on CUCM, you can make outbound
 calls (from CUCM to CME).  But you cannot make inbound calls (from CME to
 CUCM).

 Michael

 On Mon, Jan 17, 2011 at 4:24 PM, Roger Carpio roger.car...@gmail.comwrote:

 Michael,

 You do not need to add the GW to CUCM; otherwise this will not be scalable
 at all. I had the same confusion but then I got it to work without adding my
 GW (CME) to CUCM.

 Regards,
 Roger Carpio.

 On Mon, Jan 17, 2011 at 2:16 PM, Michael Luo hout...@gmail.com wrote:

 I was trying to understand the Gatekeeper-controlled Trunk in CUCM.

 If I configured the Gatekeeper-controlled Trunk, do I still have to
 configured the H323 gateways?

 Consider this scenario:

 GW --- GK --- CUCM

 If there's a call from GW to CUCM, my understanding was:

 1) GW sends ARQ to GK.
 2) GK resolves the called number and sends ACF to GW.
 3) GW will try to communicate with CUCM.

 In step 3, if GW was not configured in CUCM, the call will fail.

 Did I understand this wrong?

 Thanks!
 Michael


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Re: [OSL | CCIE_Voice] Cisco 79XX remote control

2011-01-18 Thread Roger Carpio
Hello,

What does this have to do with the CCIE Lab preparation? I bet you'll get a
lot of answers if you google it.

Regards,
Roger Carpio.

On Tue, Jan 18, 2011 at 1:44 AM, haroon javed harooon.ja...@gmail.comwrote:


 Does anyone know of any preferably software that will allow you to mange
 the Cisco 79xx series of phones remotely.  We have a large amount of
 phones that are staticly assinged and are moving to a new CM server with a
 different IP.  The CM and TFTP entries specifcly need to be change to
 reflect the new IP of the CM server.


 Thanks


 --

 Regards,

 *Haroon Javed*

 *Telecom Engineer*

 Cell: +92 (321) 8430260




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Re: [OSL | CCIE_Voice] MGCP configuration at cisco.com

2011-01-21 Thread Roger Carpio
Hello,

You're following the configuration example of a T1 CAS while you're
configuring a T1 PRI.

Regards,
Roger Carpio.


On Thu, Jan 20, 2011 at 10:27 PM, linuxboss.9 linuxbos...@gmail.com wrote:

 Do we need to add ccm-manager mgcpapp under incoming dial-peer of MGCP


 http://www.cisco.com/en/US/tech/tk1077/technologies_configuration_example09186a008022eaa3.shtml

 I configure like this below

 mgcp
 mgcp call-agent 177.1.10.10 service mgcp
 mgcp bind control source-interface g0/0.250
 mgcp bind media source-interface g0/0.250
 mgcp dtmf-relay voip codec all mode out-of-band

 ccm-manager redundant-host 177.1.10.10
 ccm-manager switchback immediate
 ccm-manager mgcp

 under serial0/0/0:23
 isdn bind-l3 ccm-manager

 And it worked all times with any issues. Just looking at Cisco
 documentation it is different.

 Please advice
 Thanks
 Shrini

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Re: [OSL | CCIE_Voice] RisDC VM usage incrementing

2011-02-01 Thread Roger Carpio
Open a case with Cisco TAC please for further investigation, this is a CCIE
study list.

Regards,
Roger Carpio.

On Tue, Feb 1, 2011 at 3:46 AM, Ashar Siddiqui siddas...@gmail.com wrote:

 Hello,



 I have a customer where RisDC service is using some 350MB Virtual memory
 but it increments with passage of time till it gets to 950MB (within a
 month).

 As soon as it hit 950MB mark, RTMT starts throwing alerts as VM falls below
 30%.

 First we thought it’s tomcat which is using excessive memory but then after
 a bit of investigation found it’s using normal VM (~940MB).



 I have searched over the internet but that did not help. Customer wants to
 know why it increase as there are only 1000 users.

 Has anyone come across this issue of RisDC? Can it be a memory leak issue?



 Please find below the server tech specs:



 admin:show hardware

 HW Platform   : 7835I2
 Processors: 1
 Type  : Family: Xeon
 CPU Speed : 2333
 Memory: 2048 MBytes
 Object ID : 1.3.6.1.4.1.9.1.585
 OS Version: UCOS 4.0.0.0-37
 Serial Number :

 RAID Version  :
 RAID Firmware Version:  5.2-0 (15432)
 RAID BIOS Version:  5.2-0 (15432)

 BIOS Information  :
 1.17

 RAID Details  :
 Controllers found: 1

  admin:show version active
 *Active Master Version: 7.1.5.11901-1*
 Active Version Installed Software Options:
 dp-ffr.3-1-15.GB.cop
 cm-locale-en_GB-7.1.5.1000-1.cop
 cm-locale-cs_CZ-7.1.5.1000-1.cop
 localeinstaller

 admin:show memory size
 This command is not supported by this server model.

  admin:show memory
   show memory count
   show memory modules
   show memory size

 admin:show memory count
 This command is not supported by this server model.

 Thanks

 Ash



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Re: [OSL | CCIE_Voice] CUPC - issue with switching between softphone and deskphone modes

2011-02-05 Thread Roger Carpio
I have seen the same exact issue, other times the same configuration works
just fine. I have never collected logs when the issue happened to me though.

Regards,
Roger Carpio.

On Sat, Feb 5, 2011 at 10:43 AM, Miron Kobelski findko...@gmail.com wrote:

 Hello,

 I've configured softphone and deskphone modes for CUPC. Both work fine, I
 can make calls, etc.
 The issue is that I can't easily switch between them. When I'm in softphone
 mode and click deskphone mode, nothing happens.
 In order to force change of mode, I need to restart CUPC each time. After
 restart it's in deskphone mode. The same in the other direction (deskhone 
 softphone).
 Has anybody seen this? I was wondering if this is some obvious
 misconfiguration or a bug...

 regards
 kobel

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Re: [OSL | CCIE_Voice] Cisco SIP phone not registering in SRST

2011-02-08 Thread Roger Carpio
telephony-service instead and call-manager-fallback are not used for SIP
SRST.

I would try removing the number from the voice register pools. I've had the
same error message in the past and not with SIP SRST but with SIP CME. The
problem was related to the DN.

voice register pool  1
 id network 10.2.3.0 mask 255.255.0.0
 codec g711ulaw

voice register pool  2
 id network 10.2.3.0 mask 255.255.0.0
 codec g711ulaw

Have you tried this?

Regards,
Roger Carpio.


On Tue, Feb 8, 2011 at 6:59 PM, Shrini linuxbos...@gmail.com wrote:

  Can you try using telephony-service instead of call-manager-fallback.

 On 2/8/2011 4:00 PM, Chris wrote:

 I am having issues with registering my sip phone during SRST my SCCP phone
 registers just fine. Is it possible to register both SCCP and SIP phones in
 SRST?
 The phone does not register and I receive the error:

 .Feb  8 23:51:11.575: VOICE_REG_POOL: Register request for (3001) from
 (10.2.3.145)
 .Feb  8 23:51:11.575: VOICE_REG_POOL: Contact doesn't match any pools

 Here is my SIP SRST config the complete config is attached:
 voice service voip
  allow-connections h323 to h323
  allow-connections h323 to sip
  allow-connections sip to h323
  allow-connections sip to sip
  no supplementary-service h450.2
  no supplementary-service h450.3
  h323
   call start slow
  sip
   bind control source-interface FastEthernet0/0.302
   bind media source-interface FastEthernet0/0.302
   registrar server
 voice register global
  max-dn 144
  max-pool 24
 voice register pool  1
  id network 10.2.3.0 mask 255.255.0.0
  number 1 3001
  codec g711ulaw
 voice register pool  2
  id network 10.2.3.0 mask 255.255.0.0
  number 1 30..
  codec g711ulaw


 Thanks,
 Chris Lundi


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Re: [OSL | CCIE_Voice] High Traffic Try Again Later

2011-02-14 Thread Roger Carpio
7.1.5 is not the version running in the CCIE Lab. Open a TAC case for
further research.

On Mon, Feb 14, 2011 at 10:08 AM, Tatiana Oliveira tati_olivei...@yahoo.com
 wrote:

 I´m receiving High Traffic Try Again Later on Cisco 7965 Phones and I´m
 using the 7.1.5.1-12. Do you have any idea about how to solve that?



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Re: [OSL | CCIE_Voice] Hardware VPN

2011-02-17 Thread Roger Carpio
I use a cheap 831, IP phones with power bricks setup as daisy chain and it
has worked just fine so far.

Regards,
Roger Carpio.

On Thu, Feb 17, 2011 at 8:37 AM, Pithog Oil pithog...@yahoo.com wrote:


 Hi Experts,

 i need assisitance on what device will be the most economical and most
 appropriate for using IPEXPERT's Hardware VPN.

 This is what i think please let me know if its a good idea

 1, 800 series router with VPN capabilities

 2, POE 12 port switch

 3, 3 7965G iP PHONES

 I hope i am not leaving anything out.


 Pithog Oil


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Re: [OSL | CCIE_Voice] OT : Missing normal softkeys from 9971.

2011-03-16 Thread Roger Carpio
9971 are not evaluated in CCIE Lab as far as I know therefore, this is not
the right place for this question. Try doing some google or post this
question at Cisco Support forums.

http://www.cisco.com/en/US/docs/voice_ip_comm/cuipph/9971_9951_8961/8_5/english/user_guide/book_911/callingfeatures_rt.html#wp1027556

Regards,
Roger Carpio.

On Wed, Mar 16, 2011 at 1:41 PM, ccielab...@gmail.com wrote:

 I'm hoping i missed something simple.
 I just got a 9971 registered on a CUCM 7.x server.

 it works great but I noticed there are no available softkeys for hold,
 park , etc during a call.

 It's my first SIP phone I'm using, so is there something I'm missing
 regarding supplemental services?
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Re: [OSL | CCIE_Voice] WB II, Lab 10, Calling Party Localization

2011-03-27 Thread Roger Carpio
Hello George,

Very rarely I've seen replication status 2 and weird behaviors happen. Have
you checked replication anywhere else? CLI / Unified Reporting?

If you've done this many times and you're sure the configuration has been
done correctly; drop this replication and reset it cluster wide. I hope it
helps.

Regards,
Roger Carpio.

On Sun, Mar 27, 2011 at 8:28 AM, George Goglidze gogli...@gmail.com wrote:

 Hi all,

 I was wondering if anyone has seen this before.

 There is a requirement to show in Missed/Received directory numbers in E164
 format, but in alerting/connected state it should be localized for user.
 Subscriber to 7digit, national to 10digit, international to 011

 I have configured correct prefixes on the incoming gateways, so
 Missed/Received directory is fine! It shows correct values with the +sign
 E164 format.

 And I have configured pt-norm-hq-ani, pt-norm-br1-ani, pt-norm-br2-ani,
 which are in their corresponding css-norm-hq-ani, css-norm-br1-ani,
 css-norm-br2-ani
 and I've applied them to the corresponding Device Pools, in Calling Party
 Transformation CSS.

 But when the call comes in, I still get in alerting/connected state E164
 format.

 All my devices phones and gateways are registered to Sub first, and in
 failed to Publisher.

 If I change this and I register all the devices to the Publisher, instead
 of a subscriber, changing it in Callmanager group configuration, then it all
 works as expected.
 Therefore I thought maybe Sub is not replicating correctly, but I've
 checked and the replication Status is 2! which means it's replicating fine.
 I have even restarted the Subscriber but it didn't fix the issue.

 I am out of ideas, and I've done it many times in prior Labs, and it has
 always worked fine!
 I've searched through CCM Service Parameters, and couldn't find anything
 that could affect this behaviour.

 If anyone has seen this, please let me know...

 Regards,

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Re: [OSL | CCIE_Voice] Local Gateway/Standard Local Route Group

2011-03-27 Thread Roger Carpio
How many sites are supposed to use this RP? If only one site is meant to use
it; SLRG will have an extra step (assigning the RG to the DP).

Regards,
Roger Carpio.

On Sun, Mar 27, 2011 at 9:22 AM, Rogers Ochieng rogersochi...@gmail.comwrote:

 If i have a requirement to only use local gateway for a route pattern, is
 it advisable to use a specific route list pointing to the route group with
 the gateway or a route list pointing to a SLRG will do?


 Rogers

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Re: [OSL | CCIE_Voice] One Week Lab Experience - April 11 thru 15 - Who will attend?

2011-04-01 Thread Roger Carpio
Pablo,

I'll see you there :-)

Regards,
Roger Carpio.

On Fri, Apr 1, 2011 at 1:39 PM, Pablo Meneses pmenese...@gmail.com wrote:

 Hi there,

 I am just wondering how will attend to April 19th's One Week Lab Experience
 boot camp in San Jose for CCIE Voice?

 Regards.

 -Pablo Meneses.

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Re: [OSL | CCIE_Voice] One Week Lab Experience - April 11 thru 15 - Who will attend?

2011-04-01 Thread Roger Carpio
I think Pablo got confused; title and message body don't match... I'll see
you there on April 11th

Regards,
Roger Carpio.

On Fri, Apr 1, 2011 at 7:14 PM, Steve Denney (stdenney)
stden...@cisco.comwrote:

 I’ll be there as well (assuming you meant April 11th and not 19th) :)



 cheers, sd



 *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
 ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Roger Carpio
 *Sent:* Friday, April 01, 2011 7:46 PM
 *To:* Pablo Meneses
 *Cc:* ccie_voice@onlinestudylist.com
 *Subject:* Re: [OSL | CCIE_Voice] One Week Lab Experience - April 11 thru
 15 - Who will attend?



 Pablo,

 I'll see you there :-)

 Regards,
 Roger Carpio.

 On Fri, Apr 1, 2011 at 1:39 PM, Pablo Meneses pmenese...@gmail.com
 wrote:

 Hi there,

 I am just wondering how will attend to April 19th's One Week Lab Experience
 boot camp in San Jose for CCIE Voice?

 Regards.

 -Pablo Meneses.

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Re: [OSL | CCIE_Voice] lab 5 every location with SIP TRUNK

2011-04-03 Thread Roger Carpio
Jonny.

I know how can you pass this exam. Stop wasting your time and ours looking
for a way to cheat and get to study.

Regards,
Roger Carpio.

On Sun, Apr 3, 2011 at 11:30 AM, Jonny Mendas jonny.men...@hotmail.comwrote:

  ok sry justin if u feel thats ok.. i think i got more tensed bec of 3
 attempts now no money.

 but lab is so hard. i do not know how we can pas!


 Thankyou



 --
 From: jbr...@tsginc.biz

 To: ccie_voice@onlinestudylist.com
 Date: Sun, 3 Apr 2011 15:47:27 +
 Subject: Re: [OSL | CCIE_Voice] lab 5 every location with SIP TRUNK


  Can a moderator remove Johnny from this list?  This is a clear violation
 of the NDA and I’m sure no one on this list wants anything to do with him.



 Justin



 *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
 ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Jonny Mendas
 *Sent:* Sunday, April 03, 2011 5:50 AM
 *To:* ccie_voice@onlinestudylist.com
 *Subject:* [OSL | CCIE_Voice] lab 5 every location with SIP TRUNK



 LAB 5 update


 Presence
 IPPM needs to be configured for one of the phones

 One of the HQPhones need to be set up in softphone mode on the test pC
 =
 IPCC was full page

 IPPA service needs to be configured and assigned to a couple of lines 3101
 and 2102
 ===
 AAR

 Configure AAR for calls between SiteC and HQ

 ==
 RINGLIST

 

 SIP TRUNK

 Need to create a SIP trunk to an IP and write a note


 Trust me guys i was unable to remember it ... otherwise i would have shared
 it.. full lab..
 If anyone got this lab we can work out and make full lab!!!

 Or anyone has real labs i heard they also released full lab 5

 Please guys i need to pass anyhow now i
 heard it is over everywhere!! means every location.

 Please PM for the same.

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Re: [OSL | CCIE_Voice] Certificates for Cisco Presence Server

2011-04-06 Thread Roger Carpio
Did you read this on some Cisco documentation or are you just following what
you were told? The latter won't let you go further down CCIE's path. Check
the deployment guide and if you confirm there is a certificate requirement
to get Presence server to work for your CCIE Lab, please let me know it.

http://docwiki.cisco.com/wiki/Cisco_Unified_Presence,_Release_7.x_--_Deployment_of_Cisco_Unified_Presence

Regards,
Roger Carpio.

On Wed, Apr 6, 2011 at 12:47 AM, lee huntl huntc...@gmail.com wrote:

 Dear all,

 I am having trouble to get the Cisco Presence server to work.  I was told I
 need to install a *root certificate* (*.cer) onto the Cisco Presence
 server for it to work.  But other than this root Certificate from the CA
 server, is there any other certificate I need in order to get it to work??

 Thanks,
 Hunt
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Re: [OSL | CCIE_Voice] Help with SIP Firmware

2011-04-13 Thread Roger Carpio
Auto register it to CUCM.

On Wed, Apr 13, 2011 at 9:24 PM, Mann Chaddha mann.chad...@gmail.comwrote:

 Hi All

 I converted my 7961 phone to SIP Load for a Lab but am not able to revert
 the Phone to SCCP Load. For changing any network settings on the phone, like
 TFTP etc, the phone prompts for Authorization with Username  Password.

 Has anyone faced this issue? Can anyone suggest how should I revert the
 phone back to factory defaults or procedure to edit network settings?

 A big thanks in advance.
 .
 Mann

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Re: [OSL | CCIE_Voice] Voice mail in CUPS

2011-04-16 Thread Roger Carpio
Before adding the user to UC; did you configure the Allow Users to Access
Voice Mail Using an IMAP Client option in COS?

On Sat, Apr 16, 2011 at 6:32 PM, Erwan Erwan e_er...@yahoo.com wrote:

 hi all,

 can someone advice, what i miss in CUPS voicemail.

 I kept geeting this error  from Show Server Health  in CUPC client

 *Failed to Connect - Invalid Credentials or Account Locked*
 **
 I verified user name and password for voicemail is working in Phone itself

 tks

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Re: [OSL | CCIE_Voice] Gatekeeper question

2011-05-30 Thread Roger Carpio
CCM service Device Name of GK-controlled Trunk That Will Use Port 1720;
just type the trunk name.

Rgs,
Roger Carpio.

On Mon, May 30, 2011 at 11:39 AM, Randall Crumm rrcr...@yahoo.com wrote:

 Hi,
 How do I get gk-trunk_1 and 2  to repflect port 1720?

 Thanks,
 Randall

 HQ-RTR(config)#do show gatekeeper gw
 GATEWAY TYPE PREFIX TABLE
 =
 Prefix: 3*
   Zone HQ master gateway list:
 10.10.110.3:1720 BR2-RTR

 Prefix: 1#*
   Zone HQ master gateway list:
 10.10.210.10:40081 GK-TRUNK_1
 10.10.210.11:33277 GK-TRUNK_2
   Zone HQ prefix 1... priority gateway list(s):
Priority 5:
 10.10.210.10:40081 GK-TRUNK_1
 10.10.210.11:33277 GK-TRUNK_2
   Zone HQ prefix 5... priority gateway list(s):
Priority 5:
 10.10.210.10:40081 GK-TRUNK_1
 10.10.210.11:33277 GK-TRUNK_2



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Re: [OSL | CCIE_Voice] Gatekeeper question--RC

2011-05-30 Thread Roger Carpio
That should do it. If they do not show the requested port after adding the
trunk name to the service parameter I would try resetting the trunk;
resetting CCM service, gatekeeper shut/no shut etc.

Rgs,
Roger Carpio.

On Mon, May 30, 2011 at 1:36 PM, Randall Crumm rrcr...@yahoo.com wrote:


 Thanks for all the answers.

 I did search the services for 1720 and entered gk-trunk.

 I'm working on WB2 lab 2 and the output from the PG shows port 1720 and my
 output shows 40081 and 33277 respectively below.

 The questions in the WB has output and says the port should show 1720
 no 40081 and 33277. This is what I am trying to do. The PG did not specify
 anything specific as well.

 Should I not worry about this?

 Randall




 --
 *From:* Roger Carpio roger.car...@gmail.com
 *To:* Randall Crumm rrcr...@yahoo.com
 *Cc:* Online Study ccie_voice@onlinestudylist.com
 *Sent:* Monday, May 30, 2011 12:07 PM
 *Subject:* Re: [OSL | CCIE_Voice] Gatekeeper question

 CCM service Device Name of GK-controlled Trunk That Will Use Port 1720;
 just type the trunk name.

 Rgs,
 Roger Carpio.

 On Mon, May 30, 2011 at 11:39 AM, Randall Crumm rrcr...@yahoo.com wrote:

 Hi,
 How do I get gk-trunk_1 and 2  to repflect port 1720?

 Thanks,
 Randall

 HQ-RTR(config)#do show gatekeeper gw
 GATEWAY TYPE PREFIX TABLE
 =
 Prefix: 3*
   Zone HQ master gateway list:
 10.10.110.3:1720 BR2-RTR

 Prefix: 1#*
   Zone HQ master gateway list:
 10.10.210.10:40081 GK-TRUNK_1
 10.10.210.11:33277 GK-TRUNK_2
   Zone HQ prefix 1... priority gateway list(s):
Priority 5:
 10.10.210.10:40081 GK-TRUNK_1
 10.10.210.11:33277 GK-TRUNK_2
   Zone HQ prefix 5... priority gateway list(s):
Priority 5:
 10.10.210.10:40081 GK-TRUNK_1
 10.10.210.11:33277 GK-TRUNK_2



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Re: [OSL | CCIE_Voice] CUE/CME

2011-05-30 Thread Roger Carpio
Look at the folder's name in flash memory where GUI files are stored. It
must match what you defined under ip http path flash:/GUI and btw is case
sensitive :-P

Rgs,
Roger Carpio.

On Mon, May 30, 2011 at 2:08 PM, Randall Crumm rrcr...@yahoo.com wrote:

 HI,
 Here is a dumb question, but causing me issues.

 My cue is stable and I can log in to the web page with the user from the
 cue cli.

 Now when I start the initialization wizard I keep getting login to CME as
 administrator failed. check CME config.

 Am I suppose to use the user under tele-service? In my case admin/c1sc0123?

 snip-its

 ip http server
 ip http authentication local
 no ip http secure-server
 ip http path flash:/GUI


 telephony-service
  max-ephones 7
  max-dn 7
  ip source-address 10.10.110.3 port 2000
  voicemail 3600
  max-conferences 8 gain -6
  call-forward pattern .T
  web admin system name admin password c1sc0123
  dn-webedit
  transfer-system full-consult
  transfer-pattern .
  transfer-pattern .T


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Re: [OSL | CCIE_Voice] frame-relay traffic-shaping on the test

2011-06-01 Thread Roger Carpio
If you're not been provided with a value to Site C; then 1544 is acceptable.
That's the one I use.

On Wed, Jun 1, 2011 at 7:23 AM, adam compton com...@gmail.com wrote:

 So I have a question as it pertains to the test.  If I use auto qos on the
 Serial interfaces of the HQ and SiteB routers, frame-relay traffic shaping
 will be enabled.  I know it will use the bandwidth that you configure for
 shaping.  I also understand on the HQ router, the link to SiteC would
 default to 56k in frame-relay traffic shaping.  Let's assume nothing is
 asked for QOS on the SiteC connection.

 My instinct is to configure the bandwith on the SITE C link to 1544 so that
 the link will not be affected.  Am I using the correct value for this
 scenario?

 Adam Compton

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Re: [OSL | CCIE_Voice] Called party transformation mask vs. route pattern/route list manipulation

2011-06-02 Thread Roger Carpio
I would say it all depends on the dial plan. Both can pretty much accomplish
the same; the problem at least for me is, how complex it gets when TEHO,
backup GWs and calling/called party presentation comes into play. In my
opinion CdPT patterns are scalable and they help you reduce your dial plan
however the CCIE does not care about scalability, just functionality. Get
used to both approaches; old school and the one introduced in CUCM 7; then
you should be able to come with an action plan when your sitting in the lab.

Regards,
Roger Carpio.

On Wed, Jun 1, 2011 at 7:31 PM, adam compton com...@gmail.com wrote:

 The list is probably tired of hearing from me tonight :)

 So when should you choose Called party transformation mask over route
 pattern/route list manipulation.

 I ran into a question with Device mobility that required called party
 transformation, but I can't identify any other scenarios where i would
 absolutely want to use called party transformation.

 Does it make plus dialing manipulation easier?

 Regards,
 Adam Compton

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Re: [OSL | CCIE_Voice] DHCP Debug in CUCM

2011-06-02 Thread Roger Carpio
John,

No debugs in CUCM for DHCP but you still have the Cisco DHCP Monitor Service
Traces and packet capture.

There is no way to bind the MAC of a device to an IP address as far as I can
tell; there is no option in GUI to accomplish this.

Regards,
Roger Carpio.

On Thu, Jun 2, 2011 at 8:45 AM, John Anzar janzar1...@live.com wrote:


 Hi Guys

 I have a very basic question about DHCP. most of the time this works with
 out any issue.

 Is there anyway that we can DHCP debug in CUCM ?? Can we get the mac biind
 with the IP form CUCM ?

 Best Regards
 J anzar

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Re: [OSL | CCIE_Voice] SRST reference necessary?

2011-06-11 Thread Roger Carpio
That's your SRST reference.

On Sat, Jun 11, 2011 at 6:15 PM, Greg cnn...@gmail.com wrote:

 If I have only SCCP phones, do I still need to configure SRST reference in
 CUCM?

 I thought I could just specify the default router as SRST in device pool.

 Was that correct?

 Thanks!
 Greg

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Re: [OSL | CCIE_Voice] QOS LAN-WAN

2011-06-11 Thread Roger Carpio
Hummm it all depends; real world or CCIE lab? Are you sending packets
already tagged to the switch? Then you might just honor it and pass it thru
or have some policing based on the tags, like guarantee some bandwidth for
voice but drop COS to 0 when exceeded. I think you have some options to play
with based on whatever you want to accomplish.

On the CCIE test you'll be provided with the requirements on that port; if
any.

Saludos,
Roger Carpio.
On Sat, Jun 11, 2011 at 6:40 PM, Santiago Figueroa sfigue...@mnet.com.mxwrote:

   Hello All,



 What QOS most to be configured at port F0/0 that connect gateway WAN with
 switch LAN. In other Word what QOS applies to port Sw that connect Gateway
 WAN, How to be classify the input/output to port the goes to LAN



 Flow QOS



 Ipphone--port Sw--SW--SWportGW-fa0/0GW(here is doubt)-WAN





 Thanks.





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Re: [OSL | CCIE_Voice] MLF or FR12 ONLY between HQ and SB - what to do with BR2

2011-06-15 Thread Roger Carpio
The map-class is the workaround and as far as I know the bandwidth command
wont work.

Regards,
Roger Carpio.

On Tue, Jun 14, 2011 at 6:02 PM, Bill Lake whl...@gmail.com wrote:

 I think if you set the bandwidth to greater than 768 you will not have this
 applied to the interface for BR2

 On Tue, Jun 14, 2011 at 5:17 PM, Roig Borrell, Francesc Xavier 
 francesc.ro...@tecnocom.es wrote:

  Hi all,



 If in the exam in Qos WAN exercise were asked to configure MLF or FR12
 between HQ and SB, supposing there is only one physical interface in HQ, the
 traffic shaping applied will affect the BR2 subintertace  appliying a
 default mincir 28000 which is a problem for more than one call





 HQ#sh frame-relay pvc 102



 cir 56000 bc 7000  be 0 byte limit 875interval 125


   mincir 28000 byte increment 875   Adaptive Shaping none

   pkts 533   bytes 168144pkts delayed 526   bytes delayed
 166528

   shaping active

   traffic shaping drops 0

   Queueing strategy: fifo

   Output queue 39/40, 56 drop, 531 dequeued

 HQ#



 Should we create this specific class map for BR2 as a workaround, or
 shouldn’t we do anything because nothing has been required for BR2 and we
 would breaking a requirement?



 map-class frame-relay BR2

  frame-relay cir 1536000

  frame-relay bc 15360

  frame-relay be 0

  frame-relay mincir  1536000





 Which would be the right answer? What do you think?



 Thanks in advance! Regards

 Francesc







 Francesc Xavier Roig Borrell

 Network Senior Consultant

 Integración de Sistemas y Tecnología

 C/Santander, 49-51

 Barcelona 08020

 Tel. Fijo: (+34) 934965108 ext 64697

 Tel. Móvil / Fax: (+34) 647 32 21 55 (24697) / (+34) 934965161

 email: francesc.ro...@tecnocom.es

 http://www.tecnocom.es



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Re: [OSL | CCIE_Voice] CCS

2011-06-17 Thread Roger Carpio
How is this related to the CCIE Lab preparation? I bet there is a sql query
to retrieve that information but I don't see how can this help us in the
lab.

Regards,
Roger Carpio.

On Fri, Jun 17, 2011 at 3:44 PM, Santiago Figueroa sfigue...@mnet.com.mxwrote:

  Hello all

 ** **

 Canyone tell me if exist any command from CLI in CUCM to shown me that
 include within a Calling Search space? What Partitions are in that CCS?***
 *

 ** **

 ** **

 ** **

 ** **

 Thanks.

 ** **

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Re: [OSL | CCIE_Voice] (no subject)

2011-06-19 Thread Roger Carpio
Santiago,

This is a question I would expect from an end user; not someone who's
preparing for the CCIE Lab. Please do your homework; get detailed sdi/sdl
traces, app logs, packet captures and if those don't give you the answer;
forward them to me and I'll take a look at them.

Please don't take it personal, we've all went thru basic configuration
issues when we started but let's just try to keep the distribution list at
expert level as much as possible. Perhaps if you ask for help telling us
what you have done; you wouldn't receive an answer like mine.

Finally, from my experience; look at the back of the IP phone and make sure
you are adding the correct phone model in CUCM. An App log will easily show
you this mismatch if that were the case.

Regards,
Roger Carpio.

On Sun, Jun 19, 2011 at 10:49 AM, Santiago Figueroa
sfigue...@mnet.com.mxwrote:

  Hello I have a doubt what is reason the ip phone 7941 do not registered
 to cucm the status reject, I am sure the MAC and phone type in the cucm is
 add correctly.

 ** **

 Thanks.

 ** **

 ** **

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Re: [OSL | CCIE_Voice] E164 with sip trunk

2011-06-19 Thread Roger Carpio
Hello,

I am not sure I am following you, so the call uses a SIP trunk from CUCM to
?? Then there is an MGCP call? Where do we loose the E164 number?

Regards,
Roger Carpio.

On Sun, Jun 19, 2011 at 12:19 AM, donny f f.faraday...@gmail.com wrote:

 hi ,

 how do we make call using SIP trunk, so it will show  +E164  in PSTN phone
 ?  this is MGCP call

 tks

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Re: [OSL | CCIE_Voice] about problem creating SIP profile in CME

2011-06-19 Thread Roger Carpio
Not sure if this will help but have you deleted the profile 1, make sure the
file is gone and then configure it again?

Regards,
Roger Carpio.

On Sat, Jun 18, 2011 at 6:31 PM, Marco zurdo zurdosp...@gmail.com wrote:



 Hi Mailers i have a trouble creating the SIP profile for pool 1 i changed
 the id mac with the correct one, but  i created the profile  in the service
 global, i check the tftp bind  and i see the *cnf with the id mac that i
 change

 show voice register tftp-bind
 tftp-server syncinfo.xml url flash:/syncinfo.xml
 tftp-server SIPDefault.cnf url flash:/SIPDefault.cnf
 tftp-server softkeyDefault_kpml.xml url flash:/softkeyDefault_kpml.xml
 tftp-server softkeyDefault.xml url flash:/softkeyDefault.xml
 tftp-server softkey1_kpml.xml url flash:/softkey1_kpml.xml
 tftp-server softkey1.xml url flash:/softkey1.xml
 tftp-server cme_dialplan_1.xml url flash:/cme_dialplan_1.xml
 tftp-server SIP003094C423D4.cnf url flash:/SIP003094C423D4.cnf  (i change
  this)
 tftp-server SIP000D299D348C.cnf url flash:/SIP000D299D348C.cnf



  but whe i trie to see the cnf file this i get thie error :%Error opening
 flash:/SIP003094c423d4.cnf (File not found)


 any ideas.



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Re: [OSL | CCIE_Voice] How to avoid phone firmware upgrade in the lab

2011-06-21 Thread Roger Carpio
If the previous candidate using your lab spot changed the protocol as
requested, let's say his lab had SIP and now yours is SCCP; there is nothing
you can do to avoid this. It is not necessarily a waste of time; unless you
keep staring at the phones LOL

Regards,
Roger Carpio.

On Tue, Jun 21, 2011 at 2:16 PM, Pithog Oil pithog...@yahoo.com wrote:

 Hi Experts,

 what should i do so as to avoid phone firmware upgrade at the lab, i have
 discovered that some times i unknowingly have a phone firmware upgrade from
 one version to another.

 Kind assist it is usually a waste of time.

 Pithog Oil

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Re: [OSL | CCIE_Voice] How to avoid phone firmware upgrade in the lab

2011-06-21 Thread Roger Carpio
Dave,

Nobody can guarantee that phones will be restored to a pre-decided firmware
version, while proctors definitely restore the lab to an almost blank
config; restoring the firmware requires some config and they just won't do
it for you. Just like CUE, it might be licensed for CME and your lab
requests JTAPI integration but fear not; if you have to do the conversion;
all you need is to know how to do it; files will be provided... Easy as pie.

Regards,
Roger Carpio.

On Tue, Jun 21, 2011 at 8:25 PM, CCIE Voice ccvoic...@gmail.com wrote:

 Do you say that Lab will not be restored to a pre-decided configurations on
 all devices before begining of each test? OR in initial lab setup , phones
 not having access to CUCM / CME for firmware changes?

 Dave

 On Wed, Jun 22, 2011 at 2:21 AM, Roger Carpio roger.car...@gmail.comwrote:

 If the previous candidate using your lab spot changed the protocol as
 requested, let's say his lab had SIP and now yours is SCCP; there is nothing
 you can do to avoid this. It is not necessarily a waste of time; unless you
 keep staring at the phones LOL

 Regards,
 Roger Carpio.

   On Tue, Jun 21, 2011 at 2:16 PM, Pithog Oil pithog...@yahoo.comwrote:

Hi Experts,

 what should i do so as to avoid phone firmware upgrade at the lab, i have
 discovered that some times i unknowingly have a phone firmware upgrade from
 one version to another.

 Kind assist it is usually a waste of time.

 Pithog Oil

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Re: [OSL | CCIE_Voice] Cisco Contcat Center Express VM

2011-06-22 Thread Roger Carpio
http://lmgtfy.com/?q=UCCX+OS

I would suggest you to use google first and if you can't find it anywhere
then let us know...

Regards,
Roger Carpio.

On Wed, Jun 22, 2011 at 3:41 PM, shelesh shel...@hotmail.com wrote:

 I need help to get Cisco Contact center express 7 virtual machine so I can
 work in my home lab, I have Cisco Contact center express 7 installer but I
 don't have OS installer, If someone can help me to get direct Virtual
 Machine (VM) that will be fine.

 -Original Message-
 From: Ravindra Lakpriya [mailto:lakpr...@gmail.com]
 Sent: Wednesday, June 22, 2011 11:38 PM
 To: shelesh
 Cc: ccie_voice@onlinestudylist.com
 Subject: Re: [OSL | CCIE_Voice] Cisco Contcat Center Express VM

 Whats the issue you are facing ?


 On Wed, Jun 22, 2011 at 11:32 PM, shelesh shel...@hotmail.com wrote:
  Hi,
 
 
 
  I need your guys help to get Cisco Contact center express VM machine, for
  CCIE study.
 
  Can anyone help me to get this.
 
 
 
  Raj.
 
  __ Information from ESET NOD32 Antivirus, version of virus
 signature
  database 6033 (20110411) __
 
  The message was checked by ESET NOD32 Antivirus.
 
  http://www.eset.com
 
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  visit www.ipexpert.com
 
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  www.PlatinumPlacement.com
 



 --
 Ravindra Lakpriya


 __ Information from ESET NOD32 Antivirus, version of virus
 signature
 database 6033 (20110411) __

 The message was checked by ESET NOD32 Antivirus.

 http://www.eset.com



 __ Information from ESET NOD32 Antivirus, version of virus
 signature
 database 6033 (20110411) __

 The message was checked by ESET NOD32 Antivirus.

 http://www.eset.com


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Re: [OSL | CCIE_Voice] How do i change a Phone registered to CUCM from SIP to SCCP

2011-06-23 Thread Roger Carpio
In regards to the SIP to SCCP conversion; the easiest way is to enable auto
registration making sure the protocol setup for this is SCCP. Obviously
there shouldn't be any record for this IP phone in CUCM.

In regards to the other question, first you need to understand why does the
IP phone request the image; then you can come up with a solution.
Troubleshooting Cisco IP Phone Registration Problems with Cisco CallManager
3.x and 4.x
http://www.cisco.com/en/US/products/hw/phones/ps379/products_tech_note09186a008009485a.shtml#topic1

Do not worry about the version; IP phone registration is pretty much the
same. Once you understand how this works; you won't ever have to ask this
question.

There are other good documents at cisco.com with much more information than
this; I'll recommend you to take a look at them if you still have additional
questions.

Regards,
Roger Carpio.

On Thu, Jun 23, 2011 at 11:11 AM, Pithog Oil pithog...@yahoo.com wrote:

 Hi Experts,

 How do i change a phone registered to CUCM from SIP to SCCP, also there are
 situations where the firmware on the phone is different from the one the
 tftp server is giving me , that means an upgrade will start taking place,
 how do i ensure the phone receives IP address without upgrading its FIRMWARE
 .

 Kind Regards

 Sola Ayoade


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Re: [OSL | CCIE_Voice] Urgent priority check

2011-06-26 Thread Roger Carpio
I agree with Randall, if you hit # the call will match 9011!# right away, no
need to use urgent priority.

On Sun, Jun 26, 2011 at 4:07 PM, Randall Saborio ill2...@gmail.com wrote:

 Still. I just tried it on my lab with patterns:
 9011!
 9011!#

 And 9011# routed immediately without using urgent priority. That is
 because ! excludes # as a digit. From the help, it says:
 The exclamation point (!) wildcard matches one or more digits in the range
 0 through 9..


 On Sun, Jun 26, 2011 at 2:37 PM, Victor Malyuga 
 victor_maly...@yahoo.comwrote:

 I said international pattern ending with #

 try 9011! and 9011!# without urgent priority. You will have to wait for
 interdigit timeout for 9011#

 --- On *Sun, 26/6/11, Randall Saborio ill2...@gmail.com* wrote:


 From: Randall Saborio ill2...@gmail.com
 Subject: Re: [OSL | CCIE_Voice] Urgent priority check
 To: Victor Malyuga victor_maly...@yahoo.com
 Cc: ccie_voice@onlinestudylist.com, donny f f.faraday...@gmail.com
 Date: Sunday, 26 June, 2011, 20:58


 I disagree, it is not needed for pattern ending in #.

 Usually it is for patterns starting in \+ AND ending in ! and 9911 as
 exception.
 A few more special cases maybe.

 On Sun, Jun 26, 2011 at 12:10 PM, Victor Malyuga 
 victor_maly...@yahoo.comhttp://uk.mc290.mail.yahoo.com/mc/compose?to=victor_maly...@yahoo.com
  wrote:

   international  pattern ending with # sign.

 --- On *Sun, 26/6/11, donny f 
 f.faraday...@gmail.comhttp://uk.mc290.mail.yahoo.com/mc/compose?to=f.faraday...@gmail.com
 * wrote:


 From: donny f 
 f.faraday...@gmail.comhttp://uk.mc290.mail.yahoo.com/mc/compose?to=f.faraday...@gmail.com


 Subject: [OSL | CCIE_Voice] Urgent priority check
 To: 
 ccie_voice@onlinestudylist.comhttp://uk.mc290.mail.yahoo.com/mc/compose?to=ccie_voice@onlinestudylist.com
 Date: Sunday, 26 June, 2011, 16:08


  hi all,

 for Route Pattern in lab.   when we need to check  urgent priority.
 I only use for 911 and 9911.

 do we need to use for  AAR  RP : \+.1   or other Route Pattern ?  what is
 the best practice

 tks

 -Inline Attachment Follows-

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 --
 Randall da ill Saborio
 CCIE Voice Wannabe #10054675811




 --
 Randall da ill Saborio
 CCIE Voice Wannabe #10054675811


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Re: [OSL | CCIE_Voice] FRF.12 question

2011-06-26 Thread Roger Carpio
Well you might want to check what happened to that interface. Run the show
traffic-shape; if it looks fine to you, then leave it like that :)

http://www.cisco.com/en/US/docs/ios/12_2/qos/command/reference/qrfcmd11.html#wp1019473

Regards,
Roger Carpio.

On Sun, Jun 26, 2011 at 4:34 PM, Cristobal Priego cristobalpri...@gmail.com
 wrote:

 hello all

 when you configure FRF.12 manually on your seial interfaces on HQ and BR2
 on the HQ router where the same physical interface is used to connect BR1
 and BR2,  BR1 link isn't affected at all because traffic shaping isn't
 enabled on the physical interface, correct ?
 so i can pretty much ignore that link in regards to a basic QoS config if
 not needed

 thanks


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Re: [OSL | CCIE_Voice] Passed

2011-06-29 Thread Roger Carpio
Excellent George!!! Shall we celebrate with a Bacon  Maple Sundae?

Congratulations :-)

On Wed, Jun 29, 2011 at 1:14 PM, Ki Wi kiwi.vo...@gmail.com wrote:

 What kind of technical problem you faced and managed to get a voucher?

 My router rebooted halfway due to traceback consider technical problem ?
 Haha... I didn't get any extra time for it and even spend extra time to run
 back and fore to inform the proctor  =[ bad luck

 Sent from my iPhone
 Pls pardon my fat fingers.

 On Jun 29, 2011, at 9:44 PM, George Goglidze gogli...@gmail.com wrote:

  Hi all,
 
  As the subject suggests, it's official, I'm dual CCIE #19926 RS and
 Voice starting yesterday.
 
  Finally, it's a relief, no more studying late, spending weekends on a
 computer, making calls, my neighbours think I escaped a psychiatric, after
 hearing voices at night TEST VOICEMAIL FOR HQ PHONE 1 and my personal
 favourite YOUR POSITION IN QUEUE IS :)
 
  I would like to transmit a very special thank you to IPExpert, and
 especially Vik Malhi. He's definitely made difference for me as a trainer.
 Just when you think you know it all about something, he would come up with
 something to prove me wrong, to show me the gaps in my knowledge I didn't
 know existed.
 
  Thank you Vik!
 
  Thanks to everyone on this forum too, there are many good people on the
 forum, with big knowledge, and more importantly willing to share it. It was
 big fun, I enjoyed the process a lot.
 
  By the way, I made it technically on my first attempt. Well, I payed once
 only, although I went to exam 3 times.
  1st time, technical problems, Cisco gave me free retake voucher, 2nd
 time, again technical problems, again free voucher.
  To be honest after so many free retakes, I'm not even sure if I really
 passed, or I was costing them too much so they decided to give it to me :-),
 good samaritans these cisco guys.
 
  Wish you all good luck, and don't get frustrated, and most importantly
 have fun.
 
  Regards,
 
 
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Re: [OSL | CCIE_Voice] Can i Daisy Chain phones of Different device pool

2011-07-01 Thread Roger Carpio
That shouldn't be a problem... Device Pool has nothing to do with daisy
chain unless you're testing device mobility. In that situation it all
depends if the switchports support dot1q so that PC port and Switch port
reside on different VLANs.


On Thu, Jun 30, 2011 at 8:42 AM, Pithog Oil pithog...@yahoo.com wrote:

   Hi Experts,

 Please i will like to know if i can daisy chain phones of different Device
 pools.

 Thanks for you assistance.

 Regards
 Pithog oil

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Re: [OSL | CCIE_Voice] UCCX server install issue VMWARE

2011-07-17 Thread Roger Carpio
Hi Michael,

I had the same problem, try searching the web for W2K3 WA. Once you have
done this, make sure to get a snapshot of your clean installation so you
don't have to do it again in case you are practicing integration. I think
since validation can't validate your server; you still have to activate it.

Regards,
Roger Carpio.

On Sun, Jul 17, 2011 at 5:48 PM, Michael Dietrich mdd2...@gmail.com wrote:

 Installing on ESXi

 I have installed base MS Server 2003 from Cisco IPT OS Disk
 Win SVr 2003 Teleco Svr App SW 3.0 1-4 CPU
 Product Key : QW438-46P6G-MHXTB-99XGG-XX7CG

 using as base OS for UCCX 7 install for CCIE-V home lab

 Problem is when I try to activate the server key (microsoft) it will not
 activate...I have internet access on the 2003 Server VM
 Anyone else run into this issue?

 thanks, Michael


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Re: [OSL | CCIE_Voice] E1 Controller Card

2011-08-16 Thread Roger Carpio
Are you able to type the card type e1 command?

Regards,
Roger Carpio.

On Tue, Aug 16, 2011 at 9:03 AM, Rafay Aslam rafayc...@gmail.com wrote:

 Hi
 I have my CCIE Voice lab @ home, last night when I booted my router, my E1
 controller card did not show up in my Br2 Configuration, nor I can see any
 lights on controller itself, show inventory does show I have E1 Controller
 card but it disappear from my configuration, I can not even do controller E1
 XXXx
 I have installed the card in another port, no luck! I am thinking my card
 is dead, its time to buy another one, but if you have any troubleshooting
 steps I am all ears.
 Regards


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Re: [OSL | CCIE_Voice] CFwdALL on SRST - CME / CUE

2011-08-16 Thread Roger Carpio
Don't take it personal; he just asked you to do what you were supposed to do
from the beginning. Want better answers? That's simple, ask better
questions. The same way you felt with his answer I felt with your question.
We're all looking for the same goal; nobody is testing people here, that's
the Lab proctor's task.

Do you have any debug output so we can share knowledge while keeping a
focused troubleshooting? Guessing will lead to an useless email chain and
unfixed problem. In other words, help us help you.

If you do not know what debugs you should run; well ask for them and we will
be happy to provide you with the document you need to read.

Regards,
Roger Carpio.

On Tue, Aug 16, 2011 at 2:34 PM, Wael Agina waelag...@gmail.com wrote:

 Hi Randall,

   Thanks for reply, however it would be nice if you use some kind words.
 this is a sharing info and knowledge exchange forum /group.

 If you know an answer you are welcome and appreciated to share, if you want
 to test me or other members this is not the place.

 Wish you the best.

 Regards,
 Wael Agina

 On Tue, Aug 16, 2011 at 7:00 PM, Randall Saborio ill2...@gmail.comwrote:

 I think Amit is on the right track.

 Wael, how do you assume that CUE is not sending the original DNIS
 information?
 What type of debug do you use to confirm or reject your assumption?

 Not sure how long you have been studying, but you have to practice on
 troubleshooting as well.

 I don't want to give you the answer here, but let's use this example to
 exercise on what would you check?

 Can you share the output that you would use?


 On Tue, Aug 16, 2011 at 7:49 AM, Wael Agina waelag...@gmail.com wrote:

 Hi Amit,
  Thanks.

   I tried it, but not working.
 I think CME is not fwding the original dnis info to the cue, so it plays
 greeting of last called number.

 Is there any other idea ?

 Regards,
 Wael Agina

 On Tue, Aug 16, 2011 at 3:02 PM, amit batra batraji...@yahoo.comwrote:

 try voicemail mailbox-selection original-called command on the config
 mode of CUE module..

 Regards
 Amit

 --
 *From:* Wael Agina waelag...@gmail.com
 *To:* ccie_voice@onlinestudylist.com
 *Sent:* Tuesday, August 16, 2011 10:58 PM
 *Subject:* [OSL | CCIE_Voice] CFwdALL on SRST - CME / CUE

 Dear All,

I have BR2 in SRST  abd all voicemail is OK, CUE is working fine with
 BR2 in SRST.
 All phone2 br2 ph1, ph2 4001/4002 can send and recieve vm messages.

 However i set CFA on 4001 to CFwdALL to 4002, now when call from pstn,
 the cue run vm greeting of 4002 not 4001.
 ALso if 4002 is CFwdALL to 4001, any caller to 4002 have call fwd to
 4001 and hears 4001 VM greeting, not 4002.

 I think we should have some kind of options on CUE to take care of
 redirecting number as i need to hear the redirecting VM greeting ?
 Any Idea ?


 --

 Thanks and Best Regards,
 Wael Agina

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 --

 Thanks and Best Regards,
 Wael Agina

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 --
 Randall da ill Saborio
 CCIE Voice Wannabe #10054675811




 --

 Thanks and Best Regards,
 Wael Agina

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Re: [OSL | CCIE_Voice] CME Speed Dials

2011-08-16 Thread Roger Carpio
Dear Deepak,

If the following link does not help you, try posting your question at
Cisco's support community. This is the wrong place.

http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/admin/configuration/guide/cmespeed.html#wp1044566

Regards,
Roger Carpio.

On Tue, Aug 16, 2011 at 2:39 PM, Deepak sidana sidana_dee...@yahoo.comwrote:

 Dear All,

 One of my client is using CME 7.0 they need that the phone user can manage
 their speed dial from the phone.

 I have checked the CME admin guide, but could't find much info.
 Do we need to create any specific url to acheive this.

 below is the config of my  telephony-services.

 telephony-service
  em logout 0:0 0:0 0:0
  max-ephones 10
  max-dn 14
  ip source-address 192.168.1.10 port 2000
  url directories http://192.168.1.10/localdirectory
  max-conferences 8 gain -6
  transfer-system full-consult
  directory entry 1 1001 name user1
  create cnf-files version-stamp Jan 01 2002 00:00:00

 ThanksRgds
 Deepak Sidana.

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Re: [OSL | CCIE_Voice] Update - STill not working --- Re: CFwdALL on SRST - CME / CUE

2011-08-20 Thread Roger Carpio
Nice output. As you can see CME is sending an INVITE to CUE with Diversion
4002 which is the last redirecting number. Would you please try this command
on dial-peer sending the call to CUE *mailbox-selection* *orig-called-num *and
let us know how it goes?

http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/command/reference/cme_m1ht.html#wp1029669

Regards,
Roger Carpio.

On Sat, Aug 20, 2011 at 7:40 AM, Wael Agina waelag...@gmail.com wrote:

 Dears,

   I repeated the scenario and used the command voicemail mailbox-selection
 original-called on CUE, but still CUE running the 4002 vm not called no 4001

 below is deb isd and deb ccsip message
 Call is 911 --- 4001  CFWDALL --- 4002  CUE VM

 *Is there any command needed on CME to forward orginal called no. to CUE*
 I dont see CME forward the original numer 4001 to CUE, it just forward new
 called redirected no 4002.

 SC-R3#
 Aug 20 11:34:04.273: ISDN Se0/0/0:15 Q931: RX - SETUP pd = 8  callref =
 0x008D
 Bearer Capability i = 0x8090A3
 Standard = CCITT
 Transfer Capability = Speech
 Transfer Mode = Circuit
 Transfer Rate = 64 kbit/s
 Channel ID i = 0xA98381
 Exclusive, Channel 1
 Progress Ind i = 0x8583 - Origination address is non-ISDN
 Display i = 'Emergency Services'
 Calling Party Number i = 0x0080, '911'
 Plan:Unknown, Type:Unknown
 Called Party Number i = 0x80, '24044001'
 Plan:Unknown, Type:Unknown
 Aug 20 11:34:04.273: ISDN Se0/0/0:15 Q931: Received SETUP  callref = 0x808D
 callID = 0x000D switch = primary-ni interface = User
 SC-R3#
 Aug 20 11:34:04.289: ISDN Se0/0/0:15 Q931: TX - CALL_PROC pd = 8  callref
 = 0x808D
 Channel ID i = 0xA98381
 Exclusive, Channel 1
 Aug 20 11:34:04.313: ISDN Se0/0/0:15 Q931: TX - ALERTING pd = 8  callref =
 0x808D
 Progress Ind i = 0x8188 - In-band info or appropriate now available

 SC-R3#
 SC-R3#
 SC-R3#
 SC-R3#
 Aug 20 11:34:13.325: //-1//SIP/Msg/ccsipDisplayMsg:
 Sent:
 INVITE sip:4500@10.10.202.2:5060 SIP/2.0
 Date: Sat, 20 Aug 2011 11:34:13 GMT
 Call-Info: sip:10.10.202.1:5060
 ;method=NOTIFY;Event=telephone-event;Duration=2000
 Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE,
 NOTIFY, INFO, REGISTER
 From: Emergency Services sip:911@10.10.202.1;tag=8DE55C-118B
 Allow-Events: kpml, telephone-event
 Supported: 100rel,timer,resource-priority,replaces,sdp-anat
 Min-SE:  1800
 Diversion: sip:4002@10.10.202.1
 ;privacy=off;reason=no-answer;counter=1;screen=no
 Remote-Party-ID: Emergency Services sip:911@10.10.202.1
 ;party=calling;screen=no;privacy=off
 Cisco-Guid: 701090686-3394703840-2166858313-2897629234
 Timestamp: 1313840053
 Content-Length: 187
 User-Agent: Cisco-SIPGateway/IOS-12.x
 To: sip:4500@10.10.202.2
 Contact: sip:911@10.10.202.1:5060
 Expires: 180
 Content-Disposition: session;handling=required
 Content-Type: application/sdp
 Call-ID: 29CA67A6-CA5711E0-812AA249-ACB65032@10.10.202.1
 Via: SIP/2.0/UDP 10.10.202.1:5060;branch=z9hG4bK46C74
 CSeq: 101 INVITE
 Max-Forwards: 70

 v=0
 o=CiscoSystemsSIP-GW-UserAgent 3375 837 IN IP4 10.10.202.1
 s=SIP Call
 c=IN IP4 10.10.202.1
 t=0 0
 m=audio 17030 RTP/AVP 0
 c=IN IP4 10.10.202.1
 a=rtpmap:0 PCMU/8000
 a=ptime:20

 Aug 20 11:34:13.333: //-1//SIP/Msg/ccsipDisplayMsg:
 Received:
 SIP/2.0 100 Trying
 Via: SIP/2.0/UDP 10.10.202.1:5060;branch=z9hG4bK46C74
 To: sip:4500@10.10.202.2;tag=cuef2922481
 From: Emergency Services sip:911@10.10.202.1;tag=8DE55C-118B
 Call-ID: 29CA67A6-CA5711E0-812AA249-ACB65032@10.10.202.1
 CSeq: 101 INVITE
 Content-Length: 0
 Timestamp: 1313840053
 Contact: sip:4500@10.10.202.2:5060


 Aug 20 11:34:13.385: //-1//SIP/Msg/ccsipDisplayMsg:
 Received:
 SIP/2.0 180 Ringing
 Via: SIP/2.0/UDP 10.10.202.1:5060;branch=z9hG4bK46C74
 To: sip:4500@10.10.202.2;tag=cuef2922481
 From: Emergency Services sip:911@10.10.202.1;tag=8DE55C-118B
 Call-ID: 29CA67A6-CA5711E0-812AA249-ACB65032@10.10.202.1
 CSeq: 101 INVITE
 Content-Length: 0
 Contact: sip:4500@10.10.202.2:5060


 Aug 20 11:34:13.409: //-1//SIP/Msg/ccsipDisplayMsg:
 Received:
 SIP/2.0 200 Ok
 Via: SIP/2.0/UDP 10.10.202.1:5060;branch=z9hG4bK46C74
 To: sip:4500@10.10.202.2;tag=cuef2922481
 From: Emergency Services sip:911@10.10.202.1;tag=8DE55C-118B
 Call-ID: 29CA67A6-CA5711E0-812AA249-ACB65032@10.10.202.1
 CSeq: 101 INVITE
 Content-Length: 176
 Contact: sip:4500@10.10.202.2:5060
 Content-Type: application/sdp
 Call-Info: sip:10.10.202.2:5060
 ;method=NOTIFY;Event=telephone-event;Duration=2000
 Allow-Events: telephone-event

 v=0
 o=CiscoSystemsSIP-Workflow-App-UserAgent 3234 3234 IN IP4 10.10.202.2
 s=SIP Call
 c=IN IP4 10.10.202.2
 t=0 0
 m=audio 16896 RTP/AVP 0
 a=rtpmap:0 pcmu/8000
 a=ptime:20

 Aug 20 11:34:13.417: //-1//SIP/Msg/ccsipDisplayMsg:
 Sent:
 ACK sip:4500@10.10.202.2:5060 SIP/2.0
 Date: Sat, 20 Aug 2011

Re: [OSL | CCIE_Voice] SIP Dial Rules and +Dialing

2011-08-21 Thread Roger Carpio
John,

I can't find the plus sign as a valid character; what if you just use * as a
wildcard?

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_0_1/ccmcfg/b03sdial.html#wp1052731

Regards,
Roger Carpio.

On Sun, Aug 21, 2011 at 2:19 PM, John McGaughey (jomcgaug) 
jomcg...@cisco.com wrote:

 Do SIP Rules support +Dialing?  For example if I have the following SIP
 Dial Rule:

 ** **

 Pattern - \+1212…….

 Timeout – 0

 ** **

 On a SIP phone missed calls shows +12123945001.  If I try to dial this
 number I have to wait for the interdigit timeout.  It seems to ignore the
 \+1212……. pattern I created.

 ** **

 All my other non +Dial patterns work just fine (i.e. 5…).

 ** **

 Is this normal or have I got something mis-configured?

 ** **

 John

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Re: [OSL | CCIE_Voice] lab6 new lab

2011-08-29 Thread Roger Carpio
Today's moral after reading this message.. Lab + # in subject = just a waste
of time or aka troll. Please do not use the study list to share this
information; you're just not doing any good by letting us know how big the
test was or how complex you believe it is. I already know how big and
complex they are (I've tried it twice). What you can learn from your
experience is that you are just not prepared to take it.

On Mon, Aug 29, 2011 at 3:06 PM, jason jaso...@rediffmail.com wrote:

 hi guys,

 I failed my lab and got lab 6 it is really bad feelings for me, because i
 went for the first attempt and got new lab and this time again new lab
 ):

 Very upset for the same, lot of new features lab is so big i am sure no one
 can complete it for sure,

 Best of luck guys for the attempt

 regards
 Jason


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Re: [OSL | CCIE_Voice] paging with CallManager 7

2011-08-29 Thread Roger Carpio
Don't you worry boys... I know if we behave Santa Claus will program it for
us... LOL

On Mon, Aug 29, 2011 at 12:56 PM, Ashraf Ayyash ash.ayy...@gmail.comwrote:

 the release still coming over  in the few next weeks,the release notes for
 that version is not yet published .


 Ash

 On Mon, Aug 29, 2011 at 7:19 PM, Mark Reed marklr...@gmail.com wrote:

 Where did you see that?  I looked at the 8.6 release notes and don't see
 that feature.


 On Mon, Aug 29, 2011 at 3:20 AM, Ashraf Ayyash ash.ayy...@gmail.comwrote:

 Hello Erwan ,

 Paging Feature will be available on the latest 8.6 CCM Version  ,

 Ash

 On Mon, Aug 29, 2011 at 8:38 AM, Erwan Erwan e_er...@yahoo.com wrote:

  hi all,

 does call manager 7 have paging feature  we can use ? for basic paging

 tks

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 --
 Thanks,

 Mark L Reed
 Home: 260-637-1585



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