[OSL | CCIE_Voice] AAR Configuration

2013-11-04 Thread CISCO CCIE VOICE
Hi Guys,

i am trying to configure AAR between 2 sites, when i assign AAR-GROUP and
AAR-CSS on device pool it does not take effect rather i have to apply it
each phone device and GW inorder for it to work.Is there any thing i am
missing

Thanks
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Re: [OSL | CCIE_Voice] AAR Configuration

2013-11-04 Thread CISCO CCIE VOICE
Hi Justin,

Thanks for your reply i will try that and wll update you on ths...

Thanks



On Mon, Nov 4, 2013 at 4:54 PM, Justin Carney justin.s.car...@gmail.comwrote:

 The aar-group setting on device pool does NOT get pushed to all devices in
 the device pool, while the aar-css does.

 My strategy is to set aar-css at the dev pool and to manually set the
 aar-group on EVERY device (phone, gw, vm or cti port, etc) and EVERY
 line/dn.  This take no thought while provisioning thing and when I get to
 an aar question the only thing to build is the rlist (maybe, if an existing
 doesnt match exactly) and route pattern.

 That said my strategy is slightly overprovisioned to save time.  I did
 thorough testing and came up with the minimum config for aar:
 1. The calling entity must have the AAR-CSS on the DEVICE (there is no aar
 css field on a dn, it only exists on a device/port/gw)
 2. The calling entity must have the AAR-GROUP set on Either Device *OR*
 Line/DN
 3. The called/target LINE/DN must have the AAR-GROUP.  (this makes sense,
 as you call a dn and you don't care which device(s) have a line appearance
 for this dn.) if the called DN doesn't have the aar-group it will NOT work,
 regarless of whether the device where the dn is assigned has the aar-group

 In summary, my strategy pit the group every where and the css on devices
 and I don't have to memorize the minimum req in the lab - or more
 importantly I don't revisit config pahes just to setup aar.

 Hope this helps...

 -Justin

  On Nov 4, 2013 6:57 AM, CISCO CCIE VOICE ccievoic...@gmail.com wrote:

 Hi Guys,

 i am trying to configure AAR between 2 sites, when i assign AAR-GROUP and
 AAR-CSS on device pool it does not take effect rather i have to apply it
 each phone device and GW inorder for it to work.Is there any thing i am
 missing

 Thanks


 ___
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 www.PlatinumPlacement.com


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[OSL | CCIE_Voice] max calls and busy-trigger using ephone-template

2013-08-03 Thread CISCO CCIE VOICE
Hi Experts,

I am not able to restrict max-call-per-button 2 and busy-trigger-per-button
1 using ephone-template

ephone-dn-template  1
 call-forward busy 2220
 call-forward noan 2220 timeout 20
 huntstop channel 1

 ephone-template  1
 softkeys remote-in-use  Newcall CBarge
 max-calls-per-button 2
 busy-trigger-per-button 1

SiteB-RTR(config)#do sh run | s telep
telephony-service
 sdspfarm units 2
 sdspfarm tag 1 SB-CONF
 srst mode auto-provision none
 srst ephone template 1
 srst dn template 1
 srst dn line-mode octo
 max-ephones 10
 max-dn 10 no-reg
 ip source-address 177.2.11.1 port 2000
 timeouts interdigit 3
 system message fallback mode
 time-zone 8
 time-format 24
 voicemail 2220
 max-conferences 8 gain -6
 call-forward pattern .T
 transfer-system full-consult
 transfer-pattern .T

thanks
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[OSL | CCIE_Voice] Channel Selection Order

2013-07-03 Thread CISCO CCIE VOICE
Hi Guys,

I am trying to understand the Channel selection order, i know that if
TELCo is using bottom Up then  i have to use opposite Direction i.e Top
Down .My Telco is uses last channel for IN-bound calls and set Top down on
my side for OUT-bound calls but still i can see that its taking bottom up
for out bound calls,so did i miss any thing ? can any one share there
knowledge and experience as to how i can configure it


Thanks
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Re: [OSL | CCIE_Voice] Channel Selection Order

2013-07-03 Thread CISCO CCIE VOICE
Thanks for ur reply,Usually the order configured in Cisco CallManager for
outbound calls is the opposite of the Telco for inbound calls in order to
increase the likelihood of available channels.


On Wed, Jul 3, 2013 at 12:29 PM, LorenzLGRC lorenzl...@gmail.com wrote:

 Hello,
 as far as i know the B channel selection order is not bound to your telco
 configuration.
 You can set top-down or bottom-up with no issue at all.

 hth
 lorenz


 On Wed, Jul 3, 2013 at 10:58 AM, CISCO CCIE VOICE 
 ccievoic...@gmail.comwrote:

 Hi Guys,

 I am trying to understand the Channel selection order, i know that if
 TELCo is using bottom Up then  i have to use opposite Direction i.e Top
 Down .My Telco is uses last channel for IN-bound calls and set Top down on
 my side for OUT-bound calls but still i can see that its taking bottom up
 for out bound calls,so did i miss any thing ? can any one share there
 knowledge and experience as to how i can configure it


 Thanks


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Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 89, Issue 4

2013-07-01 Thread CISCO CCIE VOICE
Hi Michal,

As i can see that Clock source line command which is by default  enable
further can u give an example for both User Side and Network Side
 means can u past the commands that indicates both user and Network Side

Thanks



On Mon, Jul 1, 2013 at 7:00 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
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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: How to configure Clocking for GW's
   (michael.se...@compucom.com)


 --

 Message: 1
 Date: Mon, 1 Jul 2013 15:09:31 +
 From: michael.se...@compucom.com
 To: ccie_voice@onlinestudylist.com
 Subject: Re: [OSL | CCIE_Voice] How to configure Clocking for GW's
 Message-ID:

 E88AFBC60579234C886F57D49AC243A937026EA9@SPW099EXM06.compucom.local
 Content-Type: text/plain; charset=us-ascii

 How to set clocking for PSTN Router and Branch Routers Example using MGCP
 gateway:
 !
 !
 Question:
  Take clocking for Layer 1 from Network side. --Means PSTN is Network
 Side
  Your PRI clocking of layer 2 should be user side.  --Means Branch takes
 clock from PSTN
 !
 !
 PSTN ROUTER
  network-clock-participate wic 1
 controller T1 0/1/0
  clock source internal
  linecode B8ZS
  framing ESF
  pri-group timeslots 1-24
  clock source internal Makes PSTN the Network Side
  description ** T1 PRI Voice Connection To:  S0/1/0 BR1-RTR **
 !
 !
 interface Serial0/1/0:23
  description ** T1 VOICE CONNECTION TO S0/1/0 BR1-RTR **
  no ip address
  encapsulation hdlc
  isdn switch-type primary-ni
  isdn protocol-emulate network Makes PSTN the Network Side
  isdn incoming-voice voice
  isdn negotiate-bchan resend-setup
  isdn outgoing display-ie
  isdn outgoing ie redirecting-number
  no cdp enable
 !
 !
 PSTN#show controller t1 0/1/0
 T1 0/1/0 is up.
  Framing is ESF, Line Code is B8ZS, Clock Source is Internal.
 !
 !
 BRANCH ROUTERS USING T1  MGCP
  isdn switch-type primary-ni
  network-clock-participate wic 0
  network-clock-select 1 T1 0/1/0
 controller T1 0/0/0
  pri-group timeslots 1-24 service mgcp
  clock source line Makes Router the user Side gets clock from PSTN
  linecode B8ZS
  framing ESF
  description ==VOICE PRI==
 !
 !
 interface Serial0/1/0:23
  no ip address
  encapsulation hdlc
  isdn switch-type primary-ni
  isdn incoming-voice voice
  isdn bind-l3 ccm-manager
  isdn send-alerting
  isdn sending-complete
  isdn outgoing display-ie
  isdn outgoing ie redirecting-number
  no cdp enable
 !
 !
 BR1-RTR#show controller t1 0/1/0
 T1 0/1/0 is up.
  Framing is ESF, Line Code is B8ZS, Clock Source is Line.
 !
 !
 !Hope this helps clarify the clocking issues and configuration.
 !
 !
 Michael Sears, CCIE(V)#38404
 Designing and Implementing Cisco Unified Communications on Unified
 Computing Systems






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 End of CCIE_Voice Digest, Vol 89, Issue 4
 *

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Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 89, Issue 4

2013-07-01 Thread CISCO CCIE VOICE
Thank You Michael, Question asked about L1 should me NETWORK SIDE and L2
USER SIDE in that case

controller t1 0/0/0
clock source line -L1 Network Side (This is by default
enable no need to add it)

and for L2: USER SIDE do we need to add any additional commands under
serial interface 0/0/0:23 ?

Thanks



On Mon, Jul 1, 2013 at 9:29 PM, Sears, Michael (msears) 
michael.se...@compucom.com wrote:

  NETWORK SIDE:

 !
 PSTN ROUTER
  network-clock-participate wic 1
 controller T1 0/1/0
  clock source internal
  linecode B8ZS
  framing ESF
  pri-group timeslots 1-24
  clock source internal Makes PSTN the Network Side
  description ** T1 PRI Voice Connection To:  S0/1/0 BR1-RTR **
 !
 !
 interface Serial0/1/0:23
  description ** T1 VOICE CONNECTION TO S0/1/0 BR1-RTR **
  no ip address
  encapsulation hdlc
  isdn switch-type primary-ni
  isdn protocol-emulate network Makes PSTN the Network Side
  isdn incoming-voice voice
  isdn negotiate-bchan resend-setup
  isdn outgoing display-ie
  isdn outgoing ie redirecting-number
  no cdp enable
 !
 !
 PSTN#show controller t1 0/1/0
 T1 0/1/0 is up.
  Framing is ESF, Line Code is B8ZS, Clock Source is Internal.
 !
 USER SIDE

 BRANCH ROUTERS USING T1  MGCP
  isdn switch-type primary-ni
  network-clock-participate wic 0
  network-clock-select 1 T1 0/1/0
 controller T1 0/0/0
  pri-group timeslots 1-24 service mgcp
  clock source line Makes Router the user Side gets clock from PSTN
  linecode B8ZS
  framing ESF
  description ==VOICE PRI==
 !
 !
 interface Serial0/1/0:23
  no ip address
  encapsulation hdlc
  isdn switch-type primary-ni
  isdn incoming-voice voice
  isdn bind-l3 ccm-manager
  isdn send-alerting
  isdn sending-complete
  isdn outgoing display-ie
  isdn outgoing ie redirecting-number
  no cdp enable
 !
 !
 BR1-RTR#show controller t1 0/1/0
 T1 0/1/0 is up.
  Framing is ESF, Line Code is B8ZS, Clock Source is Line.
 !

 Michael Sears, CCIE(V)#38404

 “Designing and Implementing Cisco Unified Communications on Unified
 Computing Systems”

 ** **

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Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 89, Issue 4

2013-07-01 Thread CISCO CCIE VOICE
Yes, i am saying Layer 1 and Layer 2 of ISO  Model


On Mon, Jul 1, 2013 at 10:53 PM, Sears, Michael (msears) 
michael.se...@compucom.com wrote:

  When you say L1 and L2 are you talking about layer 1 and 2 of the OSI
 Model?

 ** **

 Michael Sears, CCIE(V)#38404

 “Designing and Implementing Cisco Unified Communications on Unified
 Computing Systems”

 ** **

 *From:* CISCO CCIE VOICE [mailto:ccievoic...@gmail.com]
 *Sent:* Monday, July 01, 2013 12:58 PM
 *To:* Sears, Michael (msears)
 *Cc:* ccie_voice@onlinestudylist.com
 *Subject:* Re: CCIE_Voice Digest, Vol 89, Issue 4

 ** **

 Thank You Michael, Question asked about L1 should me NETWORK SIDE and L2
 USER SIDE in that case 

 ** **

 controller t1 0/0/0

 clock source line -L1 Network Side (This is by default
 enable no need to add it)  

 ** **

 and for L2: USER SIDE do we need to add any additional commands under
 serial interface 0/0/0:23 ?

 ** **

 Thanks

 ** **

 ** **

 On Mon, Jul 1, 2013 at 9:29 PM, Sears, Michael (msears) 
 michael.se...@compucom.com wrote:

 NETWORK SIDE:

 !
 PSTN ROUTER
  network-clock-participate wic 1
 controller T1 0/1/0
  clock source internal
  linecode B8ZS
  framing ESF
  pri-group timeslots 1-24
  clock source internal Makes PSTN the Network Side
  description ** T1 PRI Voice Connection To:  S0/1/0 BR1-RTR **
 !
 !
 interface Serial0/1/0:23
  description ** T1 VOICE CONNECTION TO S0/1/0 BR1-RTR **
  no ip address
  encapsulation hdlc
  isdn switch-type primary-ni
  isdn protocol-emulate network Makes PSTN the Network Side
  isdn incoming-voice voice
  isdn negotiate-bchan resend-setup
  isdn outgoing display-ie
  isdn outgoing ie redirecting-number
  no cdp enable
 !
 !
 PSTN#show controller t1 0/1/0
 T1 0/1/0 is up.
  Framing is ESF, Line Code is B8ZS, Clock Source is Internal.
 !

 USER SIDE

 BRANCH ROUTERS USING T1  MGCP
  isdn switch-type primary-ni
  network-clock-participate wic 0
  network-clock-select 1 T1 0/1/0
 controller T1 0/0/0
  pri-group timeslots 1-24 service mgcp
  clock source line Makes Router the user Side gets clock from PSTN
  linecode B8ZS
  framing ESF
  description ==VOICE PRI==
 !
 !
 interface Serial0/1/0:23
  no ip address
  encapsulation hdlc
  isdn switch-type primary-ni
  isdn incoming-voice voice
  isdn bind-l3 ccm-manager
  isdn send-alerting
  isdn sending-complete
  isdn outgoing display-ie
  isdn outgoing ie redirecting-number
  no cdp enable
 !
 !
 BR1-RTR#show controller t1 0/1/0
 T1 0/1/0 is up.
  Framing is ESF, Line Code is B8ZS, Clock Source is Line.
 !

 Michael Sears, CCIE(V)#38404

 “Designing and Implementing Cisco Unified Communications on Unified
 Computing Systems”

  

 ** **

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Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 88, Issue 131

2013-06-26 Thread CISCO CCIE VOICE
Hi Todd,

Please add the below commands and let me know the result...

voice service voip
 allow-connections h323 to h323
 h323
  emptycapability
  h225 id-passthru
  h225 connect-passthru

Thanks



On Wed, Jun 26, 2013 at 5:38 AM, 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: Codec and CAC section (Karen Johnson)
2. Re: quto qos voip trust in WAN (Karen Johnson)
3. Five-Lab, Self-Study Challenge -- Lab #2 Gatekeeper
   (Todd Carswell)


 --

 Message: 1
 Date: Tue, 25 Jun 2013 09:12:49 -0700 (PDT)
 From: Karen Johnson karen.johnson...@yahoo.ca
 To: Amit Sharma aryan231...@gmail.com
 Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
 Subject: Re: [OSL | CCIE_Voice] Codec and CAC section
 Message-ID:
 1372176769.52990.yahoomail...@web163901.mail.gq1.yahoo.com
 Content-Type: text/plain; charset=iso-8859-1

 tks Amit, remembered seeing we do not need in DP. can someone confirm?



 
 From: Amit Sharma aryan231...@gmail.com
 To: Karen Johnson karen.johnson...@yahoo.ca
 Sent: Monday, June 24, 2013 10:49:38 PM
 Subject: Re: [OSL | CCIE_Voice] Codec and CAC section



 no...
 u have to add in both sides...!
 dont remove from DP...!



 On Mon, Jun 24, 2013 at 6:54 PM, Karen Johnson karen.johnson...@yahoo.ca
 wrote:

 hi amit,
 ?
 do you mean i should take Location out from DP? and just put in phones?
 ?
 tks
 K
 
 
 From: Amit Sharma aryan231...@gmail.com
 To: Karen Johnson karen.johnson...@yahoo.ca
 Sent: Monday, June 24, 2013 1:25:43 AM
 Subject: Re: [OSL | CCIE_Voice] Codec and CAC section
 
 
 
 u have to add 2 way sessions...
 and also add the location that should be apply to end phones...of hq and
 site-c.
 
 
 
 On Mon, Jun 24, 2013 at 9:07 AM, Karen Johnson karen.johnson...@yahoo.ca
 wrote:
 
 hi amit.
 ?
 why it need 8 instead of 4?? question ask for 4 g729 call only.
 ?
 tks
 
 
 From: Amit Sharma aryan231...@gmail.com
 To: Karen Johnson karen.johnson...@yahoo.ca
 Sent: Sunday, June 23, 2013 11:53:56 PM
 Subject: Re: [OSL | CCIE_Voice] Codec and CAC section
 
 
 
 dude you have to use this line:
 
 
 max session software 8..
 
 
 this i think u missed and used 4 on behalf of 8...
 
 
 
 
 
 On Sun, Jun 23, 2013 at 9:26 PM, Karen Johnson 
 karen.johnson...@yahoo.ca wrote:
 
 hi folks,
 ?
 can anyone share experience on what to check on?this section , I got 0
 for few attempt.
 ?
 Here is what I did :
 ?
 UCM
 =
 ?
 - service parameter : no G722 and ILBC?
 - Enterprise parameter G711 intra, G729 inter
 - Region : HQ? SB?? SC,?? HQ-HQ : G711? , SB-SB? G711, SC-SC : g711??
 (rest? relation : G729)
 ? and assign tp DP
 - Location : HQ? and SC? : mandatory , assign to DP
 - MRGL HQ -- MRG-- MTP from HQ??? ?? same for SC?? , assign to DP
 ?
 router HQ and SC
 =
 ?
 - dspfarm profile 3 mtp
 codec pass-through
 codec g729r8
 rsvp
 maximum sessions software 4 (as they asked 4 session of g729)
 associate application SCCP
 
 - interface Serial0/0/0.1 point-to-point
 frame-relay interface-dlci 102
 ip rsvp bandwidth 112
 
 verification
 =
 - call hq to hq, sb sb : g711, inter site phone and GW : g729
 - sh ip rsvp reservation : 40 k (ring) , and 24 k (connect)
 
 
 question:
 
 - did i miss something critical that cause the mark to be 0 ?
 ?
 ?
 ?
 ?
 
 ___
 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/
 
 
 
 --
 
 Thanks  Regard'sAmit Sharma
 
 
 
 
 
 
 
 --
 
 Thanks  Regard'sAmit Sharma
 
 
 
 


 --

 Thanks  Regard'sAmit Sharma
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 --

 Message: 2
 Date: Tue, 25 Jun 2013 09:13:47 -0700 (PDT)
 From: Karen Johnson karen.johnson...@yahoo.ca
 To: Edgar Feliz ejzi...@gmail.com
 Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
 Subject: Re: [OSL | CCIE_Voice] quto qos voip trust in WAN
 Message-ID:
 1372176827.46135.yahoomail...@web163901.mail.gq1.yahoo.com
 Content-Type: text/plain; charset=iso-8859-1

 hi edgar,
 ?
 I mean for WAN qos on Sb site in lab exam. do we need auto qos voip? or
 auto qos 

Re: [OSL | CCIE_Voice] B-ACD

2013-06-24 Thread CISCO CCIE VOICE
Thanks somphol i wll try that and let you know


On Sun, Jun 23, 2013 at 11:56 AM, Somphol Boonjing somp...@gmail.comwrote:

 Remove all of the param under the service did the trick,

 Say you have this in your running config

 application
  service app-b-acd
   param number-of-hunt-grps 2
   param aa-hunt1 
   param aa-hunt2 1222
   param queue-len 15
   param queue-manager-debugs 1
 !

 Then,

 application
 service app-b-acd
 no param number-of-hunt-grps 2
 no param aa-hunt1 
 no param aa-hunt2 1222
 no param queue-len 15
 no param queue-manager-debugs 1

 Once there is no param set for the service, it will be removed from the
 running-config.

 ---
 Detail trace below:
 ---

 Branch2#show run | begin application
 application
  service app-b-acd
   param queue-len 15
   param aa-hunt1 
   param queue-manager-debugs 1
   param aa-hunt2 1222
   param number-of-hunt-grps 2
  !
 !

 Branch2(config)#application
 Branch2(config-app)# service app-b-acd
 Branch2(config-app-param)#no  param queue-len 15
 Warning: parameter queue-len has not been registered under app-b-acd
 namespace
 Branch2(config-app-param)#no  param aa-hunt1 
 Warning: parameter aa-hunt1 has not been registered under app-b-acd
 namespace
 Branch2(config-app-param)#
 Branch2(config-app-param)#do show run | begin application
 application
  service app-b-acd
   param queue-manager-debugs 1
   param aa-hunt2 1222
   param number-of-hunt-grps 2
  !
 !

 Branch2(config-app-param)#no param queue-manager-debugs 1
 Warning: parameter queue-manager-debugs has not been registered under
 app-b-acd namespace
 Branch2(config-app-param)#no param aa-hunt2 1222
 Warning: parameter aa-hunt2 has not been registered under app-b-acd
 namespace
 Branch2(config-app-param)#no param number-of-hunt-grps 2
 Warning: parameter number-of-hunt-grps has not been registered under
 app-b-acd namespace
 Branch2(config-app-param)#do show run | begin application
  associate application SCCP
 !
 dspfarm profile 5 conference
  codec g711ulaw
  codec g711alaw
  codec g729ar8
  codec g729abr8


 On Sun, Jun 23, 2013 at 12:20 AM, Bill Lake whl...@gmail.com wrote:

 Try doing all command not just these

 Sent from my iPhone

 On Jun 22, 2013, at 6:51 AM, CISCO CCIE VOICE ccievoic...@gmail.com
 wrote:

 Thanks Bill for your reply,

  I have done no service app-b-acd and no service app-b-acd-aa but showing
 all those commands in  Running configuration

 thanks



 On Sat, Jun 22, 2013 at 1:48 PM, Bill Lake whl...@gmail.com wrote:

 If it is showing up in the running configuration, then you most likely
 see something like below, the best way to remove this is to no the commands

 Or to have done a Archive or copy of the config before you apply it.
 then restore that config as the startup and reboot.

 application

 * service app-b-acd *

   param number-of-hunt-grps 2

   param aa-hunt2 

   param aa-hunt3 1222

   param queue-len 15

   param queue-manager-debugs 1

 !

 * service app-b-acd-aa *

   paramspace english index 1

   paramspace english language en

   paramspace english location flash:

   param service-name app-b-acd

   param handoff-string app-b-acd-aa

   param aa-pilot 8005550123

   param welcome-prompt _bacd_welcome.au

   param number-of-hunt-grps 2

   param dial-by-extension-option 1

   param second-greeting-time 60

   param call-retry-timer 15

   param max-time-call-retry 700

   param max-time-vm-retry 2

   param voice-mail 5003

 !

 dial-peer voice 222 voip

  service app-b-acd-aa

  destination-pattern 8005550123

  session target ipv4:192.168.1.1

  incoming called-number 8005550123

  dtmf-relay h245-alphanumeric

  codec g711ulaw

  no vad



 On Sat, Jun 22, 2013 at 2:25 AM, Somphol Boonjing somp...@gmail.comwrote:

 That one is the embedded one so you actually can not remove it.
 However, you can simply ignore it and use one that is external script.

 So, if you have the external BACD script, you can use it instead of the
 embedded one.

 Branch2#show flash | inc bacd
  107   30421bacd/app-b-acd-3.0.0.2.tcl
  108   55599bacd/app-b-acd-aa-3.0.0.2.tcl

 application
  service *funnyqueue flash:/bacd/*app-b-acd-3.0.0.2.tcl
 -- you can you whatever name you like, in this
 case funnyqueue
 -- point the script to the script with correct
 path
. (detail remove for brevity)...

  !

  service* funnyaa  flash:/bacd/app-b-acd-aa-3.0.0.2.tcl *
 -- you can you whatever name you like, in this
 case funnyaa
 -- point the script to the script with correct
 path
. (detail remove for brevity).
param service-name *funnyqueue* -- refer to your queue application
 name
param handoff-string *funnyaa*
. (detail remove for brevity).

 !

 dial-peer voice 222 voip
  service *funnyaa*   -- refer to your AA application name.
. (detail remove for brevity

Re: [OSL | CCIE_Voice] B-ACD

2013-06-22 Thread CISCO CCIE VOICE
application
no service app-b-acd
no service app-b-acd-aa




On Sat, Jun 22, 2013 at 9:06 AM, Somphol Boonjing somp...@gmail.com wrote:

 Hi,

 Are you able to show part of the configuration that you have tried to
 remove from the running configuration?

 --Somphol


 On Sat, Jun 22, 2013 at 1:00 AM, CISCO CCIE VOICE 
 ccievoic...@gmail.comwrote:

 Hi,

 I am trying to Remove B-ACD configuration but still showing in the
 running configuration i have restarted the router but no look any guess?


 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



___
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www.PlatinumPlacement.com

Re: [OSL | CCIE_Voice] B-ACD

2013-06-22 Thread CISCO CCIE VOICE
Hi Boonjing,

thanks for ur reply,what if i  have the requirement to use  only embedded
B-A-CD and if any reason i have to remove it then what i have to do ?



On Sat, Jun 22, 2013 at 10:25 AM, Somphol Boonjing somp...@gmail.comwrote:

 That one is the embedded one so you actually can not remove it.   However,
 you can simply ignore it and use one that is external script.

 So, if you have the external BACD script, you can use it instead of the
 embedded one.

 Branch2#show flash | inc bacd
  107   30421bacd/app-b-acd-3.0.0.2.tcl
  108   55599bacd/app-b-acd-aa-3.0.0.2.tcl

 application
  service *funnyqueue flash:/bacd/*app-b-acd-3.0.0.2.tcl
 -- you can you whatever name you like, in this
 case funnyqueue
 -- point the script to the script with correct
 path
. (detail remove for brevity)...

  !

  service* funnyaa  flash:/bacd/app-b-acd-aa-3.0.0.2.tcl *
 -- you can you whatever name you like, in this
 case funnyaa
 -- point the script to the script with correct
 path
. (detail remove for brevity).
param service-name *funnyqueue* -- refer to your queue application
 name
param handoff-string *funnyaa*
. (detail remove for brevity).

 !

 dial-peer voice 222 voip
  service *funnyaa*   -- refer to your AA application name.
. (detail remove for brevity)...
 !

 To remove it from the running config, then you can,

 application
  no service *funnyqueue flash:/bacd/*app-b-acd-3.0.0.2.tcl
  no service* funnyaa  flash:/bacd/app-b-acd-aa-3.0.0.2.tcl*

 Ref:
 http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/bacd/configuration/guide/40bacd.html#wp1026305

 Compared to using the embedded one below:

 application
  service app-b-acd   -- you can't change the name of the embedded BACD
 Queue script
. (detail remove for brevity)...
  !

  service app-b-acd-aa   -- you can't change the name of the embedded BACD
 AA script
. (detail remove for brevity).
param service-name app-b-acd -- refer to the embedded BACD Queue script
param handoff-string app-b-acd-aa
. (detail remove for brevity).
 !

 dial-peer voice 222 voip
  service app-b-acd-aa   -- refer to the name of the embedded BACD AA
 script
. (detail remove for brevity)...
 !

 Ref: Embedded Call-Queue and AA Tcl Scripts: Example
 http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/bacd/configuration/guide/40bacd.html



 On 22/06/2013 4:18 PM, CISCO CCIE VOICE ccievoic...@gmail.com wrote:

 application
 no service app-b-acd
 no service app-b-acd-aa




 On Sat, Jun 22, 2013 at 9:06 AM, Somphol Boonjing somp...@gmail.comwrote:

 Hi,

 Are you able to show part of the configuration that you have tried to
 remove from the running configuration?

 --Somphol


 On Sat, Jun 22, 2013 at 1:00 AM, CISCO CCIE VOICE ccievoic...@gmail.com
  wrote:

 Hi,

 I am trying to Remove B-ACD configuration but still showing in the
 running configuration i have restarted the router but no look any guess?


 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




___
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] B-ACD

2013-06-22 Thread CISCO CCIE VOICE
Thanks Bill for your reply,

 I have done no service app-b-acd and no service app-b-acd-aa but showing
all those commands in  Running configuration

thanks



On Sat, Jun 22, 2013 at 1:48 PM, Bill Lake whl...@gmail.com wrote:

 If it is showing up in the running configuration, then you most likely see
 something like below, the best way to remove this is to no the commands

 Or to have done a Archive or copy of the config before you apply it.  then
 restore that config as the startup and reboot.

 application

 * service app-b-acd *

   param number-of-hunt-grps 2

   param aa-hunt2 

   param aa-hunt3 1222

   param queue-len 15

   param queue-manager-debugs 1

 !

 * service app-b-acd-aa *

   paramspace english index 1

   paramspace english language en

   paramspace english location flash:

   param service-name app-b-acd

   param handoff-string app-b-acd-aa

   param aa-pilot 8005550123

   param welcome-prompt _bacd_welcome.au

   param number-of-hunt-grps 2

   param dial-by-extension-option 1

   param second-greeting-time 60

   param call-retry-timer 15

   param max-time-call-retry 700

   param max-time-vm-retry 2

   param voice-mail 5003

 !

 dial-peer voice 222 voip

  service app-b-acd-aa

  destination-pattern 8005550123

  session target ipv4:192.168.1.1

  incoming called-number 8005550123

  dtmf-relay h245-alphanumeric

  codec g711ulaw

  no vad



 On Sat, Jun 22, 2013 at 2:25 AM, Somphol Boonjing somp...@gmail.comwrote:

 That one is the embedded one so you actually can not remove it.
 However, you can simply ignore it and use one that is external script.

 So, if you have the external BACD script, you can use it instead of the
 embedded one.

 Branch2#show flash | inc bacd
  107   30421bacd/app-b-acd-3.0.0.2.tcl
  108   55599bacd/app-b-acd-aa-3.0.0.2.tcl

 application
  service *funnyqueue flash:/bacd/*app-b-acd-3.0.0.2.tcl
 -- you can you whatever name you like, in this
 case funnyqueue
 -- point the script to the script with correct
 path
. (detail remove for brevity)...

  !

  service* funnyaa  flash:/bacd/app-b-acd-aa-3.0.0.2.tcl *
 -- you can you whatever name you like, in this
 case funnyaa
 -- point the script to the script with correct
 path
. (detail remove for brevity).
param service-name *funnyqueue* -- refer to your queue application
 name
param handoff-string *funnyaa*
. (detail remove for brevity).

 !

 dial-peer voice 222 voip
  service *funnyaa*   -- refer to your AA application name.
. (detail remove for brevity)...
 !

 To remove it from the running config, then you can,

 application
  no service *funnyqueue flash:/bacd/*app-b-acd-3.0.0.2.tcl
  no service* funnyaa  flash:/bacd/app-b-acd-aa-3.0.0.2.tcl*

 Ref:
 http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/bacd/configuration/guide/40bacd.html#wp1026305

 Compared to using the embedded one below:

 application
  service app-b-acd   -- you can't change the name of the embedded BACD
 Queue script
. (detail remove for brevity)...
  !

  service app-b-acd-aa   -- you can't change the name of the embedded
 BACD AA script
. (detail remove for brevity).
param service-name app-b-acd -- refer to the embedded BACD Queue
 script
param handoff-string app-b-acd-aa
. (detail remove for brevity).
 !

 dial-peer voice 222 voip
  service app-b-acd-aa   -- refer to the name of the embedded BACD AA
 script
. (detail remove for brevity)...
 !

 Ref: Embedded Call-Queue and AA Tcl Scripts: Example
 http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/bacd/configuration/guide/40bacd.html



 On 22/06/2013 4:18 PM, CISCO CCIE VOICE ccievoic...@gmail.com wrote:

 application
 no service app-b-acd
 no service app-b-acd-aa




 On Sat, Jun 22, 2013 at 9:06 AM, Somphol Boonjing somp...@gmail.comwrote:

 Hi,

 Are you able to show part of the configuration that you have tried to
 remove from the running configuration?

 --Somphol


 On Sat, Jun 22, 2013 at 1:00 AM, CISCO CCIE VOICE 
 ccievoic...@gmail.com wrote:

 Hi,

 I am trying to Remove B-ACD configuration but still showing in the
 running configuration i have restarted the router but no look any guess?


 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




 ___
 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

[OSL | CCIE_Voice] B-ACD issue

2013-06-21 Thread CISCO CCIE VOICE
HI experts,

I am trying to remove B-ACD cnofigs from the router but still show in
running config,i have reloaded the router but still no luck any guess ?

Thks
___
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[OSL | CCIE_Voice] B-ACD

2013-06-21 Thread CISCO CCIE VOICE
Hi,

I am trying to Remove B-ACD configuration but still showing in the running
configuration i have restarted the router but no look any guess?


thanks
___
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www.ipexpert.com

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Re: [OSL | CCIE_Voice] BACD Timer

2013-06-18 Thread CISCO CCIE VOICE
Thanks Martin, i went through that doc but still its not clear to me the
purpose of using it and how does it effect my B-ACD script


On Tue, Jun 18, 2013 at 2:13 AM, Martin Sloan martinsloa...@gmail.comwrote:

 I got the info below from this guide -
 http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/bacd/configuration/guide/40bacd.html#wp1026305

 It has good examples you can copy/past/edit.  I believe 'Call-Queue and AA
 Tcl Scripts in Flash Memory: Example' is the best one to use.


 http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/bacd/configuration/guide/40bacd.html#wp1026305

 Step 29

 *param* *max-time-call-retry* *seconds *
 Example:

 Router(config-app-param)# param max-time-call-retry 700

 (Optional) Sets the maximum amount of time for the call-retry timer. This
 is the maximum period of time for which a call can stay in a call queue and
 retry to connect with a hunt group before the call is sent to an alternate
 destination number.

 •*seconds*—Maximum period of time, in seconds. The range is from 30 to
 3600. The default is 600.


 On Mon, Jun 17, 2013 at 5:11 PM, CISCO CCIE VOICE 
 ccievoic...@gmail.comwrote:

 hi experts,

 I am trying to understand timer for *param* *max-time-call-retry can
 anyone share there knowledge about how does it effect the bacd script and
 the purpose of this field*
 *
 *
 *Thnks*
 *  *

 ___
 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] BACD Timer

2013-06-17 Thread CISCO CCIE VOICE
hi experts,

I am trying to understand timer for *param* *max-time-call-retry can
anyone share there knowledge about how does it effect the bacd script and
the purpose of this field*
*
*
*Thnks*
*  *
___
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[OSL | CCIE_Voice] UCCX Native codec g729

2013-06-16 Thread CISCO CCIE VOICE
Hi,

How can i verify that UCCX is using G729 codec native

Thnajs
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[OSL | CCIE_Voice] UCCX Native Codec G729

2013-06-16 Thread CISCO CCIE VOICE
Hi,

How can i verify that UCCX is using G729 codec native

Thanks
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Re: [OSL | CCIE_Voice] UCCX Native codec g729

2013-06-16 Thread CISCO CCIE VOICE
Thanks  for your reply its fine that we have select G729 codec under System
Parametes but my question was is there a way  to test which shows UCCX is
using g729 codec native


On Sun, Jun 16, 2013 at 1:15 PM, OSL StudyList collaboration.c...@gmail.com
 wrote:

 Go to CCX admin and look under System and select System Parameters
 —
 Sent from Mailbox https://www.dropbox.com/mailbox for iPad


 On Sun, Jun 16, 2013 at 3:37 AM, CISCO CCIE VOICE 
 ccievoic...@gmail.comwrote:

 Hi,

 How can i verify that UCCX is using G729 codec native

 Thnajs



___
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Re: [OSL | CCIE_Voice] UCCX Native Codec G729

2013-06-16 Thread CISCO CCIE VOICE
After i configure UCCX with codec g729 under service parameter,if i call to
UCCX trigger number let say 5000 from HQ or Branch 1 or PSTN how do i know
that this call has been using g729 uccx native codec not that from region


On Sun, Jun 16, 2013 at 2:41 PM, Somphol Boonjing somp...@gmail.com wrote:

 Hi,

 I think the easiest way is to check UCCX Service Parameters under System
 menu.

 --Somphol.


 --Somphol


 On Sun, Jun 16, 2013 at 7:36 PM, CISCO CCIE VOICE 
 ccievoic...@gmail.comwrote:

 Hi,

 How can i verify that UCCX is using G729 codec native

 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



___
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Re: [OSL | CCIE_Voice] UCCX Native Codec G729

2013-06-16 Thread CISCO CCIE VOICE
Thanks Bill,but if i Press question mark 2 time thn i can see its taking
g729 codec that fine but ths result can be due to Region not from UCCX G729
Codec


On Sun, Jun 16, 2013 at 3:37 PM, Bill Talley bill.tal...@aos5.com wrote:

  On the IP phone, press the question mark button twice.

  On the vgw, 'show call active voice compact'.

   *Bill Talley*
 UC Systems Consultant

  *Alexander Open Systems, Inc*
 913.307.2330 (scheduling) | 913.744.3219 (direct)
 Web http://www.aos5.com/ | Request Support http://www.aos5.com/support
  | 
 Facebookhttp://www.facebook.com/pages/Alexander-Open-Systems-AOS/109484829074064
  | LinkedIn http://www.linkedin.com/company/aos




  Sent from an Apple iOS device with very tiny touchscreen input keys.
  Please excude my typtos.

 On Jun 16, 2013, at 8:14 AM, CISCO CCIE VOICE ccievoic...@gmail.com
 wrote:

   After i configure UCCX with codec g729 under service parameter,if i
 call to UCCX trigger number let say 5000 from HQ or Branch 1 or PSTN how do
 i know that this call has been using g729 uccx native codec not that from
 region


 On Sun, Jun 16, 2013 at 2:41 PM, Somphol Boonjing somp...@gmail.comwrote:

 Hi,

  I think the easiest way is to check UCCX Service Parameters under
 System menu.

  --Somphol.


 --Somphol


  On Sun, Jun 16, 2013 at 7:36 PM, CISCO CCIE VOICE ccievoic...@gmail.com
  wrote:

  Hi,

  How can i verify that UCCX is using G729 codec native

  Thanks


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 please visit www.ipexpert.com

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 www.PlatinumPlacement.com



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 CONFIDENTIALITY NOTICE: This electronic mail transmission (including any
 accompanying attachments) is intended solely for its authorized
 recipient(s), and may contain confidential and/or legally privileged
 information. If you are not an intended recipient, or responsible for
 delivering some or all of this transmission to an intended recipient, be
 aware that any review, copying, printing, distribution, use or disclosure
 of the contents of this message is strictly prohibited. If you have
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 Although this e-mail and any attachments are believed to be free of any
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Re: [OSL | CCIE_Voice] UCCX Native Codec G729

2013-06-16 Thread CISCO CCIE VOICE
I have Branch 2 site which contains Transcoder moreover all my agents are
in Branch-2 site i have region matrix between other region


On Sun, Jun 16, 2013 at 4:54 PM, Bill Talley bill.tal...@aos5.com wrote:

  Do you have a transcoder setup and being used for the call?  If so,
 check the SCCC connections.  Otherwise you would get a busy signal if
 region said g729 and CCX was set for g711.






  Sent from an Apple iOS device with very tiny touchscreen input keys.
  Please excude my typtos.

 On Jun 16, 2013, at 9:41 AM, CISCO CCIE VOICE ccievoic...@gmail.com
 wrote:

   Thanks Bill,but if i Press question mark 2 time thn i can see its
 taking g729 codec that fine but ths result can be due to Region not from
 UCCX G729 Codec


 On Sun, Jun 16, 2013 at 3:37 PM, Bill Talley bill.tal...@aos5.com wrote:

  On the IP phone, press the question mark button twice.

  On the vgw, 'show call active voice compact'.

   *Bill Talley*
 UC Systems Consultant

  *Alexander Open Systems, Inc*
 913.307.2330 (scheduling) | 913.744.3219 (direct)
 Web http://www.aos5.com/ | Request Supporthttp://www.aos5.com/support
  | 
 Facebookhttp://www.facebook.com/pages/Alexander-Open-Systems-AOS/109484829074064
  | LinkedIn http://www.linkedin.com/company/aos




  Sent from an Apple iOS device with very tiny touchscreen input keys.
  Please excude my typtos.

 On Jun 16, 2013, at 8:14 AM, CISCO CCIE VOICE ccievoic...@gmail.com
 wrote:

   After i configure UCCX with codec g729 under service parameter,if i
 call to UCCX trigger number let say 5000 from HQ or Branch 1 or PSTN how do
 i know that this call has been using g729 uccx native codec not that from
 region


 On Sun, Jun 16, 2013 at 2:41 PM, Somphol Boonjing somp...@gmail.comwrote:

 Hi,

  I think the easiest way is to check UCCX Service Parameters under
 System menu.

  --Somphol.


 --Somphol


  On Sun, Jun 16, 2013 at 7:36 PM, CISCO CCIE VOICE 
 ccievoic...@gmail.com wrote:

  Hi,

  How can i verify that UCCX is using G729 codec native

  Thanks


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 please visit www.ipexpert.com

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 www.PlatinumPlacement.com



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 CONFIDENTIALITY NOTICE: This electronic mail transmission (including any
 accompanying attachments) is intended solely for its authorized
 recipient(s), and may contain confidential and/or legally privileged
 information. If you are not an intended recipient, or responsible for
 delivering some or all of this transmission to an intended recipient, be
 aware that any review, copying, printing, distribution, use or disclosure
 of the contents of this message is strictly prohibited. If you have
 received this electronic mail message in error, please delete it from your
 system without copying it, and contact sender immediately by Reply e-mail,
 or by calling 913-307-2300, so that our address records can be corrected.

 Although this e-mail and any attachments are believed to be free of any
 virus or other defect that might negatively affect any computer system into
 which it is received and opened, it is the responsibility of the recipient
 to ensure that it is virus free and no responsibility is accepted by the
 sender for any loss or damage arising in any way in the event that such a
 virus or defect exists.




 CONFIDENTIALITY NOTICE: This electronic mail transmission (including any
 accompanying attachments) is intended solely for its authorized
 recipient(s), and may contain confidential and/or legally privileged
 information. If you are not an intended recipient, or responsible for
 delivering some or all of this transmission to an intended recipient, be
 aware that any review, copying, printing, distribution, use or disclosure
 of the contents of this message is strictly prohibited. If you have
 received this electronic mail message in error, please delete it from your
 system without copying it, and contact sender immediately by Reply e-mail,
 or by calling 913-307-2300, so that our address records can be corrected.

 Although this e-mail and any attachments are believed to be free of any
 virus or other defect that might negatively affect any computer system into
 which it is received and opened, it is the responsibility of the recipient
 to ensure that it is virus free and no responsibility is accepted by the
 sender for any loss or damage arising in any way in the event that such a
 virus or defect exists.


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Re: [OSL | CCIE_Voice] UCCX Native Codec G729

2013-06-16 Thread CISCO CCIE VOICE
Thanks bill got your point


On Sun, Jun 16, 2013 at 5:13 PM, Bill Talley bill.tal...@aos5.com wrote:

Any transcoder for ccx would be at HQ unless the ccx server is not in
 HQ.   looks like a few people have forwarded viable suggestions so you
 should be able to determine what you're asking fairly quickly.  Good luck.


  Sent from an Apple iOS device with very tiny touchscreen input keys.
  Please excude my typtos.

 On Jun 16, 2013, at 10:06 AM, CISCO CCIE VOICE ccievoic...@gmail.com
 wrote:

   I have Branch 2 site which contains Transcoder moreover all my agents
 are in Branch-2 site i have region matrix between other region


 On Sun, Jun 16, 2013 at 4:54 PM, Bill Talley bill.tal...@aos5.com wrote:

  Do you have a transcoder setup and being used for the call?  If so,
 check the SCCC connections.  Otherwise you would get a busy signal if
 region said g729 and CCX was set for g711.






  Sent from an Apple iOS device with very tiny touchscreen input keys.
  Please excude my typtos.

 On Jun 16, 2013, at 9:41 AM, CISCO CCIE VOICE ccievoic...@gmail.com
 wrote:

   Thanks Bill,but if i Press question mark 2 time thn i can see its
 taking g729 codec that fine but ths result can be due to Region not from
 UCCX G729 Codec


 On Sun, Jun 16, 2013 at 3:37 PM, Bill Talley bill.tal...@aos5.comwrote:

  On the IP phone, press the question mark button twice.

  On the vgw, 'show call active voice compact'.

   *Bill Talley*
 UC Systems Consultant

  *Alexander Open Systems, Inc*
 913.307.2330 (scheduling) | 913.744.3219 (direct)
 Web http://www.aos5.com/ | Request Supporthttp://www.aos5.com/support
  | 
 Facebookhttp://www.facebook.com/pages/Alexander-Open-Systems-AOS/109484829074064
  | LinkedIn http://www.linkedin.com/company/aos




  Sent from an Apple iOS device with very tiny touchscreen input keys.
  Please excude my typtos.

 On Jun 16, 2013, at 8:14 AM, CISCO CCIE VOICE ccievoic...@gmail.com
 wrote:

   After i configure UCCX with codec g729 under service parameter,if i
 call to UCCX trigger number let say 5000 from HQ or Branch 1 or PSTN how do
 i know that this call has been using g729 uccx native codec not that from
 region


 On Sun, Jun 16, 2013 at 2:41 PM, Somphol Boonjing somp...@gmail.comwrote:

 Hi,

  I think the easiest way is to check UCCX Service Parameters under
 System menu.

  --Somphol.


 --Somphol


  On Sun, Jun 16, 2013 at 7:36 PM, CISCO CCIE VOICE 
 ccievoic...@gmail.com wrote:

  Hi,

  How can i verify that UCCX is using G729 codec native

  Thanks


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Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 87, Issue 56

2013-05-21 Thread CISCO CCIE VOICE
Hi experts,

i am also searching  for that solution thanks i will also try that and let
u know, which command i need to run on SB gateway to get the output,do i
need call start fast under voice service voip in SB gateway ?


Thanks



On Tue, May 21, 2013 at 11:41 AM, 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: CCIE_Voice Digest, Vol 87, Issue 55 (Mohammed Ameenullah)


 --

 Message: 1
 Date: Tue, 21 May 2013 11:41:39 +0300
 From: Mohammed Ameenullah ameen...@gmail.com
 To: ccie_voice@onlinestudylist.com
 Cc: jainpiyush2...@ymail.com
 Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 87, Issue 55
 Message-ID:
 
 caau-xfjcw1d08ac3hydp2npcgy_jzq_au8kcsulzeeh549z...@mail.gmail.com
 Content-Type: text/plain; charset=iso-8859-1

 Hi Piyush,

 i have tried with with G711ulaw on SB gateway its working fine for me with
 redundant to HQ call routing here what i have done

 I have created MTP and Xcode on Site B router

 sccp ccm group 1
 ass ccm 1 prio 1
 ass ccm 2 prio 2
 ass pro 1 reg SB-XCODE
 ass pro 2 reg SB-MTP


 dspfarm pro 1 trans
 max sess 4
 ass app sccp
 no shut

 dspfram pro 2 mtp
 codec g711ulaw
 max sess soft 8
 ass app sccp
 no shut

 and on CUCM u have to create MRG n MRGL and assign ths MRGL to SB Gateway
 and check MTP required in CUCM Gateway page

 you can try ths configuration and let me know ur feedback ...







 On Tue, May 21, 2013 at 10:08 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. (no subject) (ie ravindra)
 2. (MGCP Teardown) (ie ravindra)
 3. Re: (no subject) (Shabeeb Mohammed)
 4. h323 Fast start configuration (Piyush Jain)
 
 
  --
 
  Message: 1
  Date: Tue, 21 May 2013 04:26:42 +0530
  From: ie ravindra ieravin...@gmail.com
  To: CCIE Study ccie_voice@onlinestudylist.com
  Subject: [OSL | CCIE_Voice] (no subject)
  Message-ID:
  
  cabadenywv08supms0meguj29ihtrbt+enrvpouse56vcyxu...@mail.gmail.com
  Content-Type: text/plain; charset=iso-8859-1
 
  Dear All,
 
  Whats is the real meaning of MGCP tear down. Is it means dropping a call
 or
  , What  ? thanks for your valuable input.
 
  Ravi,
  -- next part --
  An HTML attachment was scrubbed...
  URL:
  /archives/ccie_voice/attachments/20130521/91fde8a8/attachment-0001.html
 
  --
 
  Message: 2
  Date: Tue, 21 May 2013 04:41:48 +0530
  From: ie ravindra ieravin...@gmail.com
  To: CCIE Study ccie_voice@onlinestudylist.com
  Subject: [OSL | CCIE_Voice] (MGCP Teardown)
  Message-ID:
  CABADEnz40hKrB=bWpHQA8eNomj3v6nU=
  57ufmvq+hnvmmm-...@mail.gmail.com
  Content-Type: text/plain; charset=iso-8859-1
 
  Dear All,
 
  Whats is the real meaning of MGCP tear down. Is it means dropping a call
 or
  , What  ? thanks for your valuable input.
 
  Ravi,
  -- next part --
  An HTML attachment was scrubbed...
  URL:
  /archives/ccie_voice/attachments/20130521/c6e48219/attachment-0001.html
 
  --
 
  Message: 3
  Date: Tue, 21 May 2013 10:36:49 +0530
  From: Shabeeb Mohammed shabeebc...@gmail.com
  To: ie ravindra ieravin...@gmail.com
  Cc: ccie_voice@onlinestudylist.com
  ccie_voice@onlinestudylist.com
  Subject: Re: [OSL | CCIE_Voice] (no subject)
  Message-ID:
  
  caojzbky0gwx7s_2udiduvw6euuqsfxfuwzm6638mtg1a+pn...@mail.gmail.com
  Content-Type: text/plain; charset=iso-8859-1
 
  Hey ravi,
 
  I believe it means that mgcp packets was disrupted in between
 transmission
  resulting in packet loss etc. This error implies that there is a bug in
 the
  ios. Try upgrading the ios and check
 
  Regards
  Shabeeb
  On 21 May 2013 04:28, ie ravindra ieravin...@gmail.com wrote:
 
   Dear All,
  
   Whats is the real meaning of MGCP tear 

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

2013-05-17 Thread CISCO CCIE VOICE
Hi,

Command button is showing in the configuration but the extension is
not appearing on the phone

thnks



On Fri, May 17, 2013 at 1:31 AM, Vignesh Sethuraman
sethuvign...@gmail.comwrote:

 Please check if the command button x:x is available or not. It might have
 got removed.


 On Thu, May 16, 2013 at 10:21 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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 visit www.ipexpert.com

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[OSL | CCIE_Voice] NO Extension in CME-SRST

2013-05-16 Thread CISCO CCIE VOICE
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] Sending Calling Name to PSTN

2013-05-07 Thread CISCO CCIE VOICE
Hi Experts,

How can i send Calling name to PSTN over MGCP and H323 Gateway?

thanks
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[OSL | CCIE_Voice] Unable to play welcome Prompt

2013-05-07 Thread CISCO CCIE VOICE
Hi Experts,

When i dial UCCX Trigger i am not able to play custom welcome prompt rather
its playing MOH.


thanks
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[OSL | CCIE_Voice] IPCC Problem

2013-04-30 Thread CISCO CCIE VOICE
Hi Experts,

I am getting an error i am sorry we r currently facing system problm and r
unable t process ur call during CSQ

Thanks
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Re: [OSL | CCIE_Voice] IPCC Problem

2013-04-30 Thread CISCO CCIE VOICE
Hi,
whn i call frm PSTN to 2022400 it play the  Thank you for calling then
Thank You for waiting, we wll connect you with our Agent HQ1  then i am
getting MOH audio after that call will goes to Site B Phone 1 and play the
prompt as Thank You for waiting, we wll connect you with our Agent
SiteB1 and then i am able hear MOH audio .After that i getting i am sorry
we r currently facing system problm and r unable t process ur call





On Tue, Apr 30, 2013 at 9:26 AM, Hesham Abdelkereem 
heshamcentr...@gmail.com wrote:

 Thats definitely a wrong script was made however ,
 If you want to update the script make the following steps

 1-Edit the script on the UCCX Editor Application
 2-Save As anything (the old name same as listed on UCCX Admin)
 3-Validate the script on UCCX Editor and make sure its successfully
 validated
 4-Then go to UCCX Admin  Script Management - Upload the Script
 -- Overwrite the Script -- Then Refresh The Script
 5-Refresh the Application Script

 Then you should be good to go

 Thanks,
 Hesham


 On 29 April 2013 23:03, CISCO CCIE VOICE ccievoic...@gmail.com wrote:

 Hi Experts,

 I am getting an error i am sorry we r currently facing system problm and
 r unable t process ur call during CSQ

 Thanks


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Re: [OSL | CCIE_Voice] IPCC Problem

2013-04-30 Thread CISCO CCIE VOICE
I am facing the same problem when is use  default icd.aef script
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Re: [OSL | CCIE_Voice] SIP Delay Vs Early Offer

2013-04-29 Thread CISCO CCIE VOICE
Thanks Robert


On Sun, Apr 28, 2013 at 5:11 PM, Robert Thomas tho...@gmail.com wrote:

 The description is not entirely correct.

 Delayed Offer:  The UAC SDP is relayed on the ACK Message

 INVITE -
 100 Trying -
 200OK - With SDP
 ACK - With SDP

 Early Offer

 INVITE -  With SDP
 100 Trying -
 200OK - With SDP
 ACK -


 The SDP is moved from the ACK to the INVITE, This is why is Early Offer.

 The 200OK should always contain SDP from the UAS.  Only Exception might be
 when the SDP is sent on a provisional 1XX Message, like a 180 Ringing or
 183 Session Progress.

 In that case is refered to as Early Media instead of early offer.

 So saying on delayed offer the SDP is on the 200OK, is incorrect since,
 both scenarios have SDP on the 200OK, The difference is the variation
 between the INVITE and the ACK.

 This link has further description.
 http://www.iplogos.fr/English-Resources/Focus/sip-early-media-early-offer-en.html



 On Sun, Apr 28, 2013 at 2:51 AM, CISCO CCIE VOICE 
 ccievoic...@gmail.comwrote:

 Thank you suresh i will check ths on my pod...


 On Sun, Apr 28, 2013 at 11:26 AM, Suresh Bhandari bring...@gmail.comwrote:

 Content Length 0 means, there will be an OK followed with SDP So
 basically both are same.


 On Sun, Apr 28, 2013 at 1:56 PM, CISCO CCIE VOICE ccievoic...@gmail.com
  wrote:

 Hi Suresh, Thanks for you reply, If SIP 200 OK message has media then
 its DELAYED OFFER and SIP INVITE Message with content-length 0 also means
 the DELAYED OFFER so which one i have to consider ?

 thanks



 On Sun, Apr 28, 2013 at 10:59 AM, Suresh Bhandari 
 bring...@gmail.comwrote:

 If your SIP Invite message is containing media information, it is an
 early offer. If the response OK message has media, then it is delayed 
 offer.

 I prefer checking Content-Length field to find out if its an Early
 or Delayed offer. Value of 0 mean Delayed offer, non-zero value means it 
 is
 including SDP message and so its an early offer.

 HTH


 On Sun, Apr 28, 2013 at 1:15 PM, CISCO CCIE VOICE 
 ccievoic...@gmail.com wrote:

 HI Experts,

 Which SIP Message contains Delay Offer and Early Offer message ? i i
 am bit confuse as some documents says SIP INVITE message contains both
 Delay and Early Offer and some documents say its SIP 200 OK  message


 Thanks


 ___
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 please visit www.ipexpert.com

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 --
 Suresh Bhandari





 --
 Suresh Bhandari



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 --
 Robert Thomas Zamora
 tho...@gmail.com +50689389544
 http://cr.linkedin.com/pub/robert-thomas-zamora/29/913/8a8
 CCNP, CCNP Voice

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[OSL | CCIE_Voice] MGCP Auto vs Manual configuration

2013-04-28 Thread CISCO CCIE VOICE
Hi Experts

Which method to be use in CCIE Voice lab exam in order to configure MGCP
Auto or Manually.


Thanks
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Re: [OSL | CCIE_Voice] MGCP Auto vs Manual configuration

2013-04-28 Thread CISCO CCIE VOICE
HI Thanks,

but if suppose i want to use 6 channel from T1 PRI  on HQ Router and 12
channel of E1 PRI from Branch-2 router,so how can i busy out channels under
service parameter ? and how does CUCM knows about which router interface to
busy out as i have two site with different fractional PRI

Any reference or document on ths will be g8rt help...

thanks


On Sun, Apr 28, 2013 at 9:46 AM, Mohamed Gazzaz mgaz...@hotmail.com wrote:

 Pick up whatever method you are comfortable with and
 practice, practice. practice ..

 Try both methods and see which one is faster for you.

 --
 Date: Sun, 28 Apr 2013 09:10:30 +0300
 From: ccievoic...@gmail.com
 To: ccie_voice@onlinestudylist.com
 Subject: [OSL | CCIE_Voice] MGCP Auto vs Manual configuration


 Hi Experts

 Which method to be use in CCIE Voice lab exam in order to configure MGCP
 Auto or Manually.


 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

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[OSL | CCIE_Voice] SIP Delay Vs Early Offer

2013-04-28 Thread CISCO CCIE VOICE
HI Experts,

Which SIP Message contains Delay Offer and Early Offer message ? i i am bit
confuse as some documents says SIP INVITE message contains both Delay and
Early Offer and some documents say its SIP 200 OK  message


Thanks
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Re: [OSL | CCIE_Voice] SIP Delay Vs Early Offer

2013-04-28 Thread CISCO CCIE VOICE
Hi Suresh, Thanks for you reply, If SIP 200 OK message has media then its
DELAYED OFFER and SIP INVITE Message with content-length 0 also means the
DELAYED OFFER so which one i have to consider ?

thanks



On Sun, Apr 28, 2013 at 10:59 AM, Suresh Bhandari bring...@gmail.comwrote:

 If your SIP Invite message is containing media information, it is an early
 offer. If the response OK message has media, then it is delayed offer.

 I prefer checking Content-Length field to find out if its an Early or
 Delayed offer. Value of 0 mean Delayed offer, non-zero value means it is
 including SDP message and so its an early offer.

 HTH


 On Sun, Apr 28, 2013 at 1:15 PM, CISCO CCIE VOICE 
 ccievoic...@gmail.comwrote:

 HI Experts,

 Which SIP Message contains Delay Offer and Early Offer message ? i i am
 bit confuse as some documents says SIP INVITE message contains both Delay
 and Early Offer and some documents say its SIP 200 OK  message


 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




 --
 Suresh Bhandari

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Re: [OSL | CCIE_Voice] MGCP Auto vs Manual configuration

2013-04-28 Thread CISCO CCIE VOICE
HI Thanks Again, So i can use Fractional T1/E1 PRI with Automatic
MGCP Configuration?


On Sun, Apr 28, 2013 at 10:56 AM, Suresh Bhandari bring...@gmail.comwrote:

 Regarding your question about busyout channels in CUCM, you will use the
 MGCP gateway address as appeared in your CUCM gateway page and busyout
 channels using 1.

 You can find a good example by clicking the parameter.
 For your reference, for say your SA site, the SP Change B-Channel
 Maintenance Status 1 will be S0/SU0/DS1-0@*SA-RTR* =  0011  
   to busyout channels 7-23.

 Notice the boldface text that represents your gateway.

 Further, you have to check Enable status Poll in your gateway page.

 Hope this helps.


 On Sun, Apr 28, 2013 at 12:45 PM, CISCO CCIE VOICE 
 ccievoic...@gmail.comwrote:

 HI Thanks,

 but if suppose i want to use 6 channel from T1 PRI  on HQ Router and 12
 channel of E1 PRI from Branch-2 router,so how can i busy out channels under
 service parameter ? and how does CUCM knows about which router interface to
 busy out as i have two site with different fractional PRI

 Any reference or document on ths will be g8rt help...

 thanks


 On Sun, Apr 28, 2013 at 9:46 AM, Mohamed Gazzaz mgaz...@hotmail.comwrote:

 Pick up whatever method you are comfortable with and
 practice, practice. practice ..

 Try both methods and see which one is faster for you.

 --
 Date: Sun, 28 Apr 2013 09:10:30 +0300
 From: ccievoic...@gmail.com
 To: ccie_voice@onlinestudylist.com
 Subject: [OSL | CCIE_Voice] MGCP Auto vs Manual configuration


 Hi Experts

 Which method to be use in CCIE Voice lab exam in order to configure MGCP
 Auto or Manually.


 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



 ___
 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




 --
 Suresh Bhandari

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Re: [OSL | CCIE_Voice] SIP Delay Vs Early Offer

2013-04-28 Thread CISCO CCIE VOICE
Thank you suresh i will check ths on my pod...


On Sun, Apr 28, 2013 at 11:26 AM, Suresh Bhandari bring...@gmail.comwrote:

 Content Length 0 means, there will be an OK followed with SDP So
 basically both are same.


 On Sun, Apr 28, 2013 at 1:56 PM, CISCO CCIE VOICE 
 ccievoic...@gmail.comwrote:

 Hi Suresh, Thanks for you reply, If SIP 200 OK message has media then its
 DELAYED OFFER and SIP INVITE Message with content-length 0 also means the
 DELAYED OFFER so which one i have to consider ?

 thanks



 On Sun, Apr 28, 2013 at 10:59 AM, Suresh Bhandari bring...@gmail.comwrote:

 If your SIP Invite message is containing media information, it is an
 early offer. If the response OK message has media, then it is delayed offer.

 I prefer checking Content-Length field to find out if its an Early or
 Delayed offer. Value of 0 mean Delayed offer, non-zero value means it is
 including SDP message and so its an early offer.

 HTH


 On Sun, Apr 28, 2013 at 1:15 PM, CISCO CCIE VOICE ccievoic...@gmail.com
  wrote:

 HI Experts,

 Which SIP Message contains Delay Offer and Early Offer message ? i i am
 bit confuse as some documents says SIP INVITE message contains both Delay
 and Early Offer and some documents say its SIP 200 OK  message


 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




 --
 Suresh Bhandari





 --
 Suresh Bhandari

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[OSL | CCIE_Voice] ISDN No alerting Message

2013-04-27 Thread CISCO CCIE VOICE
HI experts,

When Branch-2 MGCP router is in CME SRST mode ,i am unable to see Alerting
message on inbound call from PSTN,moreover when i call from  PSTN my call
is directly connected and there is no ring back tone ,Can any one help to
fix the issue.


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 = 0xA18381
Preferred, Channel 1
Progress Ind i = 0x8183 - Origination address is non-ISDN
Display i = 'Emergency 911/999'
Calling Party Number i = 0x0180, '911'
Plan:ISDN, Type:Unknown
Called Party Number i = 0x81, '24044001'
Plan:ISDN, Type:Unknown
ISDN Se0/0/0:15 Q931: Received SETUP  callref = 0x808D callID = 0x000D
switch = primary-net5 interface = User
ISDN Se0/0/0:15 Q931: TX - CALL_PROC pd = 8  callref = 0x808D
Channel ID i = 0xA98381
Exclusive, Channel 1
ISDN Se0/0/0:15 Q931: TX - CONNECT pd = 8  callref = 0x808D
ISDN Se0/0/0:15 Q931: RX - CONNECT_ACK pd = 8  callref = 0x008D
Branch2(config-if)#
%ISDN-6-CONNECT: Interface Serial0/0/0:0 is now connected to 911 N/A
%ISDN-6-CONNECT: Interface Serial0/0/0:0 is now connected to 911 N/A
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[OSL | CCIE_Voice] H323 and SLRG

2013-04-10 Thread CISCO CCIE VOICE
Hi Guys,

Can i group H323 GW and SLRG in one Route List ? it that the best practice
?can any one provide me with the reference for best practice to which
protocol in can group together  like h323, mgcp,sip trunk,GK Trunk ect in
to Route List.

Thanks
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[OSL | CCIE_Voice] E1 Framing

2013-04-02 Thread CISCO CCIE VOICE
Hi Guys,

Regarding  E1 Framing if the question states the below

FOR E1 use the following:

switch type:net5
framing :CRC
linecode :HDB3

then do i need to use the NO-CRC or CRC ?
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[OSL | CCIE_Voice] IE display vs display IE

2013-04-01 Thread CISCO CCIE VOICE
Hi guys,

what is the difference between IE display and display IE command

interface Serial0/1/0:15
 no ip address
 encapsulation hdlc
 isdn switch-type primary-net5
 isdn incoming-voice voice
 isdn bind-l3 ccm-manager
 isdn outgoing *display-ie VS ie dislay*
 isdn outgoing ie redirecting-number
 no cdp enable
!

Thanks
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Re: [OSL | CCIE_Voice] IE display vs display IE

2013-04-01 Thread CISCO CCIE VOICE
Thanks All


On Mon, Apr 1, 2013 at 2:48 PM, ie ravindra ieravin...@gmail.com wrote:

 I have seen this also. There are three commands. isdn  outgoing display-ie
 / isdn outgoing ie display both commands are same as I heard. but there is
 a another command called facility ie. don't know what the use is. I have
 read in many document to use facility ie if  display ie or ie display not
 worked.


 On Mon, Apr 1, 2013 at 5:06 PM, Suresh Bhandari bring...@gmail.comwrote:

 The help output displays the same thing:

 *isdn outgoing ?*
display-ie DISPLAY IE in outgoing ISDN messages is allowed

 and
 *isdn outgoing ie ?
 *
...
displayDISPLAY IE in outgoing ISDN messages is allowed
...

 so this may be otherwise option for the same display-ie.

 Any other views?


 On Mon, Apr 1, 2013 at 5:06 PM, CISCO CCIE VOICE 
 ccievoic...@gmail.comwrote:

 Hi guys,

 what is the difference between IE display and display IE command

 interface Serial0/1/0:15
  no ip address
  encapsulation hdlc
  isdn switch-type primary-net5
  isdn incoming-voice voice
  isdn bind-l3 ccm-manager
  isdn outgoing *display-ie VS ie dislay*
  isdn outgoing ie redirecting-number
  no cdp enable
 !

 Thanks

 ___
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 please visit www.ipexpert.com

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 www.PlatinumPlacement.com




 --
 Suresh Bhandari

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[OSL | CCIE_Voice] H323 One Way Audio Troubleshooting

2013-03-19 Thread CISCO CCIE VOICE
Hi Experts,

Can any one share there knowledge and experience on how to troubleshoot
one-way audio when the call is answer from PSTN phone which messages do i
need to look at on RTMT and which traces do i need to enable on CUCM to
check the One way audio problem ..

Thanks
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[OSL | CCIE_Voice] CFUR

2013-03-18 Thread CISCO CCIE VOICE
Hi experts,

i have been trying to test the Call forward unregister whn the call arrive
on destination phone its show on the display as FORWARD,FOR and BY on the
phone screen so is there a way to display FROM on the destination phone
screen

thanks
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[OSL | CCIE_Voice] CUCM DHCP issue

2013-03-16 Thread CISCO CCIE VOICE
Hi experts,

i am facing a problem with CUCM DHCP ,my phones are taking an ip address as
first IP from starting range  and so on actually it has to take last ip
first and so on

eg: CUCM DHCP range :177.1.11.10 -177.1.11.30
  MY phones getting an ip as 177.1.11.10

thanks in advance
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[OSL | CCIE_Voice] Directory folder on Router

2013-03-12 Thread CISCO CCIE VOICE
Hi Experts,

can any one help me I want to create directory folder on router without
formatting flash  when i use mkdir command its saying that invalid input

thnks
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Re: [OSL | CCIE_Voice] Directory folder on Router

2013-03-12 Thread CISCO CCIE VOICE
THANKS CORY AND BELL REALLY APPRECIATED

On Tue, Mar 12, 2013 at 6:06 PM, Cory Gray corygray22...@hotmail.comwrote:

 I tried this for hours and there is no way (that I could find).  You must
 format flash to get the command


 http://www.cisco.com/en/US/docs/routers/access/1800/1841/software/configuration/guide/b_cflash.html#wp23144
 

 ** **

 ** **

 *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
 ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *CISCO CCIE VOICE
 *Sent:* Tuesday, March 12, 2013 10:51 AM
 *To:* ccie_voice@onlinestudylist.com
 *Subject:* [OSL | CCIE_Voice] Directory folder on Router

 ** **

 Hi Experts,

 ** **

 can any one help me I want to create directory folder on router without
 formatting flash  when i use mkdir command its saying that invalid input *
 ***

 ** **

 thnks

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[OSL | CCIE_Voice] H323 Trunk to PSTN

2013-03-11 Thread CISCO CCIE VOICE
HI experts,

If the question ask to configure H323 Trunk to backbone router then in that
do i need to configure it as H323 Gateway or H323 Non -GK controlled trunk?

Thnks
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[OSL | CCIE_Voice] H323 Trunk Debugs in RTMT

2013-03-10 Thread CISCO CCIE VOICE
Hi Experts,

Can anyone explain me how to collect h323 trunk debugs from RTMT ...

Thanks
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[OSL | CCIE_Voice] SIP and H323 Trunk to PSTN

2013-03-08 Thread CISCO CCIE VOICE
HI Guys,

Can any one Share H323 and SIP Trunk configuration that need to be done on
PSTN Router in order for it to work properly ...


Thanks
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Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 85, Issue 11

2013-03-06 Thread CISCO CCIE VOICE
Phone NTP reference is used for phones that uses SIP protocol to get there
time ...




On Tue, Mar 5, 2013 at 1:15 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. Phone NTP Reference Vs NTP Reference (CCIEing)
2. Re: Phone NTP Reference Vs NTP Reference (Jason Lee)
3. Re: Phone NTP Reference Vs NTP Reference (Cory Gray)
4. Re: Phone NTP Reference Vs NTP Reference (CCIEing)
5. AAR HELP (CISCO CCIE VOICE)
6. Re: AAR HELP (Jamie Parr (jamparr))
7. Re: AAR HELP (CISCO CCIE VOICE)


 --

 Message: 1
 Date: Tue, 5 Mar 2013 00:15:37 +0300
 From: CCIEing aboaz...@gmail.com
 To: ccie_voice@onlinestudylist.com
 Subject: [OSL | CCIE_Voice] Phone NTP Reference Vs NTP Reference
 Message-ID:
 CAEHpKk+Lkk6bve2AygUMTSEH=-
 fjd++6ubszgxyhqepodbb...@mail.gmail.com
 Content-Type: text/plain; charset=iso-8859-1

 Hello All,

 The following question cross my mind while doing the NTP configuration
 stuff..

 What is the difference between the Phone NTP reference configuration in the
 CCM Web administration page
 and
 The NTP reference on the OS Administration page??

 does the 1st one for the endpoints where the 2nd one is for the CUCM
 itself?

 Thanks
 -- next part --
 An HTML attachment was scrubbed...
 URL:
 /archives/ccie_voice/attachments/20130305/981b33f7/attachment-0001.html

 --

 Message: 2
 Date: Mon, 4 Mar 2013 16:48:44 -0500
 From: Jason Lee jas7...@gmail.com
 To: CCIEing aboaz...@gmail.com
 Cc: ccie_voice ccie_voice@onlinestudylist.com
 Subject: Re: [OSL | CCIE_Voice] Phone NTP Reference Vs NTP Reference
 Message-ID:
 
 capf5dnpx9u2jpau501_xkedvcnemmtysrouh6+obvyfg0lg...@mail.gmail.com
 Content-Type: text/plain; charset=iso-8859-1

 The Phone NTP Reference is used for SIP endpoints.  SIP endpoints store a
 NTP server address internally and they use the Phone NTP Reference
 parameter to obtain that information.  This parameter is not required for
 SCCP endpoints.

 The second is for the CUCM server.

 You were pretty much spot on with your guess.

 HTH,

 Jason


 On Mon, Mar 4, 2013 at 4:15 PM, CCIEing aboaz...@gmail.com wrote:

  Hello All,
 
  The following question cross my mind while doing the NTP configuration
  stuff..
 
  What is the difference between the Phone NTP reference configuration in
  the CCM Web administration page
  and
  The NTP reference on the OS Administration page??
 
  does the 1st one for the endpoints where the 2nd one is for the CUCM
  itself?
 
  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
 
 -- next part --
 An HTML attachment was scrubbed...
 URL:
 /archives/ccie_voice/attachments/20130304/a63b29ef/attachment-0001.html

 --

 Message: 3
 Date: Mon, 4 Mar 2013 16:48:55 -0500
 From: Cory Gray corygray22...@hotmail.com
 To: CCIEing aboaz...@gmail.com
 Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
 Subject: Re: [OSL | CCIE_Voice] Phone NTP Reference Vs NTP Reference
 Message-ID: bay404-eas56c8a84c06f6f0e59b0d5cfb...@phx.gbl
 Content-Type: text/plain; charset=us-ascii

 Phone ntp reference is for SIP phones only

 Sent from my iPhone

 On Mar 4, 2013, at 4:42 PM, CCIEing aboaz...@gmail.com wrote:

  Hello All,
 
  The following question cross my mind while doing the NTP configuration
 stuff..
 
  What is the difference between the Phone NTP reference configuration in
 the CCM Web administration page
  and
  The NTP reference on the OS Administration page??
 
  does the 1st one for the endpoints where the 2nd one is for the CUCM
 itself?
 
  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: Tue, 5 Mar 2013 01:05:22 +0300
 From: CCIEing aboaz...@gmail.com
 To: Cory Gray corygray22...@hotmail.com, Suresh Bhandari
 bring...@gmail.com
 Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
 Subject: Re: [OSL | CCIE_Voice] Phone NTP Reference Vs NTP

[OSL | CCIE_Voice] VM in SRST

2013-03-06 Thread CISCO CCIE VOICE
HI All

When the Branch-1 is in SRST Proper Mode (call-manager-fallback)  When i am
trying t press voice mail button its playing system greeting not the user
greeting ,its there any thing i need do on CUCM in order to play user
greeting when the phones are in SRST..


Thanks
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[OSL | CCIE_Voice] AAR HELP

2013-03-05 Thread CISCO CCIE VOICE
 Hi experts,

Can any one help me to understand the below Question

Call from HQ to Branch-2 should invoke AAR base on RSVP bandwidth,If
HQPH1 calls BR2PH1 then it should display E.164 number and Network
congestion Re-routing Message on Phone
 So ,my doubt is that do i need to use AAR and location  on VM Pilot
Number,VM Ports ,CUE-PORTS and CTI-Route Point etc


Thanks
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Re: [OSL | CCIE_Voice] AAR HELP

2013-03-05 Thread CISCO CCIE VOICE
Hi Jamie,

Thanks for your reply,Actually the question does not provide much
information its just say that Call From HQ to BRABCH-2 so is it safe to use
on AAR Group/CSS as well as location on Voice Mail configuration let me try
these

Thanks


On Tue, Mar 5, 2013 at 12:13 PM, Jamie Parr (jamparr) jamp...@cisco.comwrote:

  I configure AAR on all of these. Applying AAR to the device pool will
 mean only giving all the relevant extensions the AAR group, not much extra
 config as you have to visit all those pages anyway. Better safe than sorry
 

 ** **

 *Jamie Parr*
 Engineer - IT
 jamp...@cisco.com
 Phone: *+44 20 8824 2641*
 Mobile: *+44 7590622049*

 ** **

 *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
 ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *CISCO CCIE VOICE
 *Sent:* 05 March 2013 08:13
 *To:* ccie_voice@onlinestudylist.com
 *Subject:* [OSL | CCIE_Voice] AAR HELP

 ** **

 Hi experts,

 Can any one help me to understand the below Question

 Call from HQ to Branch-2 should invoke AAR base on RSVP bandwidth,If
 HQPH1 calls BR2PH1 then it should display E.164 number and Network
 congestion Re-routing Message on Phone
  So ,my doubt is that do i need to use AAR and location  on VM Pilot
 Number,VM Ports ,CUE-PORTS and CTI-Route Point etc


 Thanks

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[OSL | CCIE_Voice] Using BAT in LAB

2013-02-26 Thread CISCO CCIE VOICE
Hi,
Is it safe to use BAT in real lab exam?

thanks
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Re: [OSL | CCIE_Voice] Using BAT in LAB

2013-02-26 Thread CISCO CCIE VOICE
Thanks Mark for quick reply...

On Tue, Feb 26, 2013 at 11:50 PM, Mark Thrash (marthras) marth...@cisco.com
 wrote:

   Oh yea

 __

 Mark Thrash

 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 and may contain confidential and/or privileged
 material. Any review, retransmission, dissemination or other use of, or
 taking of any action in reliance upon, this information by persons or
 entities other than the intended recipient is prohibited. If you received
 this in error, please contact the sender and delete the material from any
 computer.

 __

 ** **

   From: CISCO CCIE VOICE ccievoic...@gmail.com
 Date: Tuesday, February 26, 2013 2:44 PM
 To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
 Subject: [OSL | CCIE_Voice] Using BAT in LAB

   Hi,
 Is it safe to use BAT in real lab exam?

 thanks


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[OSL | CCIE_Voice] MWI in SRST

2013-02-24 Thread CISCO CCIE VOICE
HI,

Does MWI works on SRST Proper mode (call-manager-fallback).?

Thanks
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[OSL | CCIE_Voice] Dail Plan Consideration in SRST Mode

2013-02-24 Thread CISCO CCIE VOICE
Hi

Can any one help me with Dial Plan consideration when calling from HQ Site
to Branch 1 Site,following what i have configure.But the problem is that on
B1PH1 screen  its showing  as below

*From +14082021001
Forward by:2001*


*HQ SITE:*

*Extension Range*:1XXX

*Partition:*Branch_1_SRST_PT
*CSS :*Branch_1_SRST_CSS--contains-Branch_1_SRST_PT

*B1PH1*:i have assign CFUR as 2001 with CSS:Branch_1_SRST_CSS


*Route Pattern:* 2XXX--Branch_1_SRST_PT
*Route List  :* Standard Local Route Group
*Prefix :*91972303



*BRANCH 1 SITE:*

*Extension Range :*2XXX

dial-peer voice 10 pots
destination-pattern 1...
perfix 14082021
port 0/0/0:23

thanks
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Re: [OSL | CCIE_Voice] Dail Plan Consideration in SRST Mode

2013-02-24 Thread CISCO CCIE VOICE
Thanks i will configure as mention and let you know...



On Sun, Feb 24, 2013 at 4:18 PM, Cory Gray corygray22...@hotmail.comwrote:

 You probably have redirecting number outbound checked on your site a
 gateway.  Uncheck it, reset your gateway and let us know

 Sent from my iPhone

 On Feb 24, 2013, at 6:45 AM, CISCO CCIE VOICE ccievoic...@gmail.com
 wrote:

 Hi

 Can any one help me with Dial Plan consideration when calling from HQ Site
 to Branch 1 Site,following what i have configure.But the problem is that on
 B1PH1 screen  its showing  as below

 *From +14082021001
 Forward by:2001*


 *HQ SITE:*

 *Extension Range*:1XXX

 *Partition:*Branch_1_SRST_PT
 *CSS :*
 Branch_1_SRST_CSS--contains-Branch_1_SRST_PT

 *B1PH1*:i have assign CFUR as 2001 with CSS:Branch_1_SRST_CSS


 *Route Pattern:* 2XXX--Branch_1_SRST_PT
 *Route List  :* Standard Local Route Group
 *Prefix :*91972303



 *BRANCH 1 SITE:*

 *Extension Range :*2XXX

 dial-peer voice 10 pots
 destination-pattern 1...
 perfix 14082021
 port 0/0/0:23

 thanks

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[OSL | CCIE_Voice] SRST DIAL PLAN

2013-02-23 Thread CISCO CCIE VOICE
HI

When in SRST either SRST Proper or CME as SRST ,when i call from HQ to
branch office do i need to assign unregistered DN in order to have ext to
ext calling working case of WAN fails
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