Re: [OSL | CCIE_Voice] DTMF Issue with SIP TRUNK (CUCM and CUE)

2014-01-30 Thread Vignesh Sethuraman
Hello Somphol/Justin,

I have resolved the issue by adding the command no supplementary-service
sip moved-temporarily.

Thanks a lot Somphol for pointing the document to me.

Thank you Justin for providing me the inputs.

Regards,
Viki









On Thu, Jan 30, 2014 at 6:15 AM, Justin Carney justin.s.car...@gmail.comwrote:

 I concur with Somphol's suggestion and that mtp shouldn't be required.

 You stated you can record the voicemail but I don't see the sdspfarm tag
 1 BR2-IOS-XCODE command under telephony-service.  Is your transcoder
 showing its registered with show sccp command?  I'm guessing that it is
 registered else you wouldn't be getting to cue using g729 that is coming
 over the wan (maybe the tag command just got lost on the copy/paste of the
 config to the email?).

 (Also for the sccp config you're missing the same tag command for the cfb
 and the conference hardware command.  You have the sccp ccm pointing to
 the cucm ip after cme, are you trying to register sccp resources to cucm?)

 You can run debug ccsip messages on cme to ensure you see the dtmf comes
 across the sip trunk from cucm.

 Dial peer 3600 for cue lists dtmf-relay sip-notify and just double check
 this is set the same inside cue.

 For an alternate test, when you place the same call can you leave a
 message ( 2 sec) and hang up without pressing pound?  Does the mwi come on
 and can the cme phone retrieve the voicemail after entering the pin?  If so
 use the same debug ccsip messages cmd to see the expected/normal debug
 output for the dtmf on this working scenario.

 Hope this helps...

 -Justin

 (Sent from my phone, please excuse and/or laugh at any typos.)
 On Jan 29, 2014 5:40 PM, Somphol Boonjing somp...@gmail.com wrote:


 On Thu, Jan 30, 2014 at 8:48 AM, Vignesh Sethuraman 
 sethuvign...@gmail.com wrote:

 Media Termination Point Required (Checked)
 MTP Preferred Originating CodecRequired Field: g711ulaw


 Hi Vignesh,

 I think if you can set these two to default settings which is MTP
 Required [uncheck], and MTP Prefered Codec: None, Leave the DTMF
 Signaling Method to No Preference.   Reset the SIP Trunk.

 You shouldn't need MTP for this operation.

 Then, if you really want to experiment with MTP insertion, I think you
 may find this article interesting -
 http://www.ucguerrilla.com/2012/12/ccie-v-i-shoulda-checked-that-tip-3-mtp.html
 .

 Regards,
 --Somphol.



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Re: [OSL | CCIE_Voice] DTMF Issue with SIP TRUNK (CUCM and CUE)

2014-01-30 Thread Vignesh Sethuraman
Hello All,

I have attached the debug ccsip messages output before and after using the
command. I do not have the answer why it resolved the dtmf-issue. If you
guys find something, please share it.

Thanks,
Viki





On Thu, Jan 30, 2014 at 4:16 PM, Moataz moataz_m...@yahoo.com wrote:

 no supplementary service affect only call forwarding and call transfer , i
 do not know how it solve DTMF

 Regards,
 Moataz Tolba


   On Thursday, 30 January 2014, 15:17, Mark Holloway m...@markholloway.com
 wrote:
  I understand how DTMF works on SIP Trunks, what I'm not clear on is how
 no supp services would have an impact on his DTMF issue. I'm trying to
 understand the logic of something changing with RFC2833 or SIP NOTIFY to
 the point where # is now recognized, yet without changing anything related
 to DTMF.  Wouldn't supp services only impact the signlaing behavior of the
 SIP 302 message itself?  But not DTMF?


 On Jan 30, 2014, at 8:00 AM, Bill Lake whl...@gmail.com wrote:

 Inbound SIP trunk from ITSP and CUE


 http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_configuration_example09186a00808f9666.shtml


 He would see the issue in the debugs




 On Thu, Jan 30, 2014 at 6:51 AM, Mark Holloway m...@markholloway.comwrote:

 Something doesn't seem to add up in my head. Supp Services shouldn't
 effect DTMF. Did you change anything related to the SIP Trunk on CUCM?  Or
 anything DTMF related on a dial-peer?

 On Jan 30, 2014, at 6:22 AM, Vignesh Sethuraman sethuvign...@gmail.com
 wrote:

 Hello Somphol/Justin,

 I have resolved the issue by adding the command no supplementary-service
 sip moved-temporarily.

 Thanks a lot Somphol for pointing the document to me.

 Thank you Justin for providing me the inputs.

 Regards,
 Viki









 On Thu, Jan 30, 2014 at 6:15 AM, Justin Carney 
 justin.s.car...@gmail.comwrote:

 I concur with Somphol's suggestion and that mtp shouldn't be required.
 You stated you can record the voicemail but I don't see the sdspfarm tag
 1 BR2-IOS-XCODE command under telephony-service.  Is your transcoder
 showing its registered with show sccp command?  I'm guessing that it is
 registered else you wouldn't be getting to cue using g729 that is coming
 over the wan (maybe the tag command just got lost on the copy/paste of the
 config to the email?).
 (Also for the sccp config you're missing the same tag command for the cfb
 and the conference hardware command.  You have the sccp ccm pointing to
 the cucm ip after cme, are you trying to register sccp resources to cucm?)
 You can run debug ccsip messages on cme to ensure you see the dtmf comes
 across the sip trunk from cucm.
 Dial peer 3600 for cue lists dtmf-relay sip-notify and just double check
 this is set the same inside cue.
 For an alternate test, when you place the same call can you leave a
 message ( 2 sec) and hang up without pressing pound?  Does the mwi come on
 and can the cme phone retrieve the voicemail after entering the pin?  If so
 use the same debug ccsip messages cmd to see the expected/normal debug
 output for the dtmf on this working scenario.
 Hope this helps...
 -Justin
 (Sent from my phone, please excuse and/or laugh at any typos.)
 On Jan 29, 2014 5:40 PM, Somphol Boonjing somp...@gmail.com wrote:


 On Thu, Jan 30, 2014 at 8:48 AM, Vignesh Sethuraman 
 sethuvign...@gmail.com wrote:

 Media Termination Point Required (Checked)
 MTP Preferred Originating CodecRequired Field: g711ulaw


 Hi Vignesh,

 I think if you can set these two to default settings which is MTP Required
 [uncheck], and MTP Prefered Codec: None, Leave the DTMF Signaling Method
 to No Preference.   Reset the SIP Trunk.

 You shouldn't need MTP for this operation.

 Then, if you really want to experiment with MTP insertion, I think you may
 find this article interesting -
 http://www.ucguerrilla.com/2012/12/ccie-v-i-shoulda-checked-that-tip-3-mtp.html
 .

 Regards,
 --Somphol.



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dtmf
Description: Binary data


dtmf
Description: Binary data
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[OSL | CCIE_Voice] DTMF Issue with SIP TRUNK (CUCM and CUE)

2014-01-29 Thread Vignesh Sethuraman
Hello All,

I am facing an issue with dtmf-relay. PhoneA registered to CUCM (SiteA)
calls PhoneD registered to CUCME (Site C). Between Regions G729 codec is
negotiated. PhoneD call-forward no answer to Voicemail. CUE integrated with
CME. After leaving the Voicemail from PhoneA to PhoneD, when I press # key
to send the Voicemail, it is not recognized.

Here is my scenario and the configuration.

(PhoneA) -- CUCM SIP TRUNK CUCME (PhoneD) --- CUE.

voice service voip
 ip address trusted list
  ipv4 0.0.0.0 0.0.0.0
 allow-connections h323 to h323
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
 fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
 h323
 sip
  bind control source-interface Loopback0
  bind media source-interface Loopback0
  registrar server expires max 600 min 60
!
voice class codec 1
 codec preference 1 g711ulaw
 codec preference 2 g729r8
!
telephony-service
 sdspfarm units 3
 sdspfarm transcode sessions 4
 max-ephones 5
 max-dn 10
 ip source-address 3.3.3.3 port 2000
 load 7945 term45.default.loads
 time-zone 28
 time-format 24
 date-format dd-mm-yy
 voicemail 3600
 max-conferences 8 gain -6
 call-forward pattern .T
 moh music-on-hold.au
 dn-webedit
 transfer-system full-consult
 transfer-pattern .T
 create cnf-files
!
sccp local Loopback0
sccp ccm 10.0.10.160 identifier 2 version 7.0
sccp ccm 3.3.3.3 identifier 1 version 7.0
sccp ip precedence 3
sccp
sccp ccm group 1
 associate ccm 1 priority 1
 associate ccm 2 priority 2
 associate profile 1 register BR2-IOS-XCODE
 associate profile 2 register BR2-IOS-CFB
 keepalive retries 5
 switchover method immediate
 switchback method immediate
 switchback interval 15
!
dspfarm profile 1 transcode
 codec g729r8
 codec g711ulaw
 codec g711alaw
 codec g729ar8
 codec g729abr8
 maximum sessions 4
 associate application SCCP
!
dspfarm profile 2 conference
 codec g711ulaw
 codec g711alaw
 codec g729ar8
 codec g729abr8
 codec g729r8
 codec g729br8
 maximum sessions 2
 associate application SCCP
!
dial-peer voice 1000 voip
 destination-pattern [15]...$
 session protocol sipv2
 session target ipv4:10.0.10.160
 incoming called-number .
 voice-class codec 1
 dtmf-relay sip-notify
 no vad
!
dial-peer voice 3600 voip
 destination-pattern 3[16]00$
 session protocol sipv2
 session target ipv4:10.10.202.100
 incoming called-number 399[89]
 dtmf-relay sip-notify
 codec g711ulaw
 no vad
!


*On the CUCM, I did the following,*Media Termination Point Required
(Checked)
MTP Preferred Originating CodecRequired Field: g711ulaw
DTMF Signaling MethodRequired Field: No preference
Non Secure SIP Trunk Profile: I am using TCP+UDP for INCOMING + Accept
Unsolicited Notification (Checked).

Please let me know what I am missing.

Thanks,
Viki
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[OSL | CCIE_Voice] SRST question

2014-01-27 Thread Vignesh Sethuraman
Dear All,

I have a basic question, when a site is in SRST and when I dial a PSTN
number or try to reach other site over PSTN, how come the GW at the SRST
site knows the called/calling party numbering plan and type. I do not have
any configuration related to that on the dial-peer or on the voice-port.

Are these information elements are negotiated and received from the PSTN
side?

For example, currently my GW is in SRST and when dial a PSTN number, it
correctly recognizes as local, national and international. Here is my
sample output after making a local and internation call to a PSTN number.

Calling Party Number i = 0x4180, '8631002'
Plan:ISDN, Type:Subscriber(local)
Called Party Number i = 0xC1, '8632683'

Calling Party Number i = 0x2180, '6178631002'
Plan:ISDN, Type:National
Called Party Number i = 0x91, '011916745738932'
Plan:ISDN, Type:International

Thanks,
Viki
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Re: [OSL | CCIE_Voice] IPMA- Assitant Device is not listed - Vol1 Task 9.4

2014-01-25 Thread Vignesh Sethuraman
Hello All,

I was using Cisco 7965 as manager Phone and SIP 9971 as Assistant phone is
9971.

Recently I came to know that only the following phones are supported for
Cisco Unified Communications Manager Assistant. Please see the below link.

https://supportforums.cisco.com/thread/2071497

• Cisco Unified IP Phone 7940G, 7941G, 7941G-GE, 7942G, 7945G, 7960G,
7961G, 7961G-GE, 7962G, 7965G, 7970G, 7971G-GE, or 7975G model supporting
Skinny Call Control Protocol (SCCP)

• Cisco Unified IP Phone 7941G, 7941G-GE, 7942G, 7945G, 7961G, 7961G-GE,
7962G, 7965G, 7970G, 7971G-GE, or 7975G model supporting SIP.

Thanks,

Viki




On Sun, Jan 12, 2014 at 2:08 PM, Vignesh Sethuraman
sethuvign...@gmail.comwrote:

 Dear All,

 During the IPMA configuration, when I tried to do the Assistant
 configuration for a user ID, the Device Name, Intercom Line  Primary Line
 are not listed.

 But the manager configuration is showing the relevant details in the
 details in the drop down.

 For the Assistant configuration, I have associated the phone to the end
 user (Assistant), made the extension 5002 as the primary extension for that
 user. Associated the user id (Assitant) as the Owner user ID in the
 Device page.

 I could not figure out what I am missing. I have followed the same steps
 as shown in the Vol 1 Solution Guide.

 Thanks,
 Viki



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[OSL | CCIE_Voice] CCIE_Voice Vol 1 Task 11.4

2014-01-25 Thread Vignesh Sethuraman
Hello All,

I could not get this question solved as per the solution mentioned in the
Proctor guide.

The SRST fallback incoming call at extn 1003 from PSTN hit the VM but I
don't get the Subscriber greeting but get a default message From a touch
tone telephone dial any extn.

I followed the OSL mail archive link  and added isdn outgoing ie
redirecting-number under the serial interface.

https://www.mail-archive.com/ccie_voice@onlinestudylist.com/msg13745.html

But still its not working.

Could you please someone tell me what I am missing.

Here is my config.

voice register pool  1
 id network 10.10.201.0 mask 255.255.255.0
 call-forward b2bua busy 5600
 call-forward b2bua noan 5600 timeout 8
!
voice translation-rule 1003
 rule 1 /^1002$/ /1003/
!
voice translation-profile REDIRECT
 translate redirect-called 1003
!
call-manager-fallback
 max-conferences 8 gain -6
 transfer-system full-consult
 ip source-address 2.2.2.2 port 2000
 max-ephones 10
 max-dn 20
 after-hours block pattern 1 91900 7-24
 voicemail 5600
 call-forward busy 5600
 call-forward noan 5600 timeout 8
 moh music-on-hold.au
 multicast moh 239.1.1.1 port 16384 route 2.2.2.2
 cor incoming CSS-LD default
 cor incoming CSS-LD 1 1002
!
dial-peer voice 5600 pots
 translation-profile outgoing REDIRECT
 destination-pattern 5600$
 no digit-strip
 port 0/0/0:23
 prefix 1212394
!
!
interface Serial0/0/0:23
 no ip address
 encapsulation hdlc
 isdn switch-type primary-ni
 isdn incoming-voice voice
 isdn outgoing ie redirecting-number
 no cdp enable
!

Thanks,

Viki
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Re: [OSL | CCIE_Voice] no DTMF Relay to UC via PSTN during HA (testing AAR)

2014-01-22 Thread Vignesh Sethuraman
Hello Mark,

yes, I do have *mgcp dtmf-relay voip codec all mode out-of-band.*

Thanks,
Viki




On Tue, Jan 21, 2014 at 8:57 PM, Mark Thrash (marthras)
marth...@cisco.comwrote:

   Do you have the command

  Mgcp dtmf codec all out

  In your mgcp config

   From: Vignesh Sethuraman sethuvign...@gmail.com
 Date: Tuesday, January 21, 2014 at 1:51 PM
 To: ccievoice ccie_voice@onlinestudylist.com
 Subject: [OSL | CCIE_Voice] no DTMF Relay to UC via PSTN during HA
 (testing AAR)

   Hello All,

 Unity Connection not recognizing the password (no DTMF) when the call
 is routed as following during a high availability situation.

 SiteB PH2/PH3 ---  MGCP T1 Port of SiteB GW  My PSTN GW (use to switch
 call between all sites via pots dialpeers) - SiteA H323 GW - CUCM
 SUB  Unity Connection.

 *  The Unity Connection is playing Message -- Enter you PIN
 *  Unity Connection recognizes SiteB PH2 is a registered user's number , so
 asks for password
 *  When pressing password unity connection does not recognize that any key
 is pressed

 I am facing the same issue as mentioned in the below link but I am using 
 Skinny integration of CUC to CUCM.
 http://onlinestudylist.com/archives/ccie_voice/2013-August/085101.html

 Please let me know what I am missing.

 Thanks,
 Viki






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[OSL | CCIE_Voice] no DTMF Relay to UC via PSTN during HA (testing AAR)

2014-01-21 Thread Vignesh Sethuraman
Hello All,

Unity Connection not recognizing the password (no DTMF) when the call
is routed as following during a high availability situation.

SiteB PH2/PH3 ---  MGCP T1 Port of SiteB GW  My PSTN GW (use to switch
call between all sites via pots dialpeers) - SiteA H323 GW - CUCM
SUB  Unity Connection.

*  The Unity Connection is playing Message -- Enter you PIN
*  Unity Connection recognizes SiteB PH2 is a registered user's number , so
asks for password
*  When pressing password unity connection does not recognize that any key
is pressed

I am facing the same issue as mentioned in the below link but I am
using Skinny integration of CUC to CUCM.
http://onlinestudylist.com/archives/ccie_voice/2013-August/085101.html

Please let me know what I am missing.

Thanks,
Viki
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[OSL | CCIE_Voice] Changing Sampling rate on CUCM

2014-01-16 Thread Vignesh Sethuraman
Hello All,

Is there a possibility to change the sampling rate on CUCM. If so, please
let me know where can I find it.

Thanks,
Viki
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[OSL | CCIE_Voice] Extension Mobility - Vol1 Task 9.5

2014-01-13 Thread Vignesh Sethuraman
Hello All,

I have configured the following in the CUCME. On the Skinny phone, the
Login Softkey is greyed out so that I could not able to login to test the
EM feature.

I am not sure why the login softkey is greyed out and let me know how to
activate it.

telephony-service
 no auto-reg-ephone
 authentication credential cisco cisco
 max-ephones 5
 max-dn 10
 ip source-address 3.3.3.3 port 2000
 service phone webAccess 0
 url authentication http://3.3.3.3/CCMCIP/authenticate.asp cisco cisco
 time-zone 28
 time-format 24
 date-format dd-mm-yy
 max-conferences 8 gain -6
 moh music-on-hold.au
 dn-webedit
 transfer-system full-consult
 create cnf-files version-stamp 7960
!
voice logout-profile 1
number 3001 type normal
!
voice user-profile 1
 max-idle-time 10
 user br2phn3 password adgjm
 number 3102 type normal
 speed-dial 1 3005 label 3005
!
ephone  1
 device-security-mode none
 mac-address ..
 ephone-template 1
 max-calls-per-button 5
 busy-trigger-per-button 3
 type 7945
 logout-profile 1

Thanks
Viki
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Re: [OSL | CCIE_Voice] Extension Mobility - Vol1 Task 9.5

2014-01-13 Thread Vignesh Sethuraman
ip http server is also configured.


On Mon, Jan 13, 2014 at 8:51 PM, Vignesh Sethuraman
sethuvign...@gmail.comwrote:

 Hello All,

 I have configured the following in the CUCME. On the Skinny phone, the
 Login Softkey is greyed out so that I could not able to login to test the
 EM feature.

 I am not sure why the login softkey is greyed out and let me know how to
 activate it.

 telephony-service
  no auto-reg-ephone
  authentication credential cisco cisco
  max-ephones 5
  max-dn 10
  ip source-address 3.3.3.3 port 2000
  service phone webAccess 0
  url authentication http://3.3.3.3/CCMCIP/authenticate.asp cisco cisco
  time-zone 28
  time-format 24
  date-format dd-mm-yy
  max-conferences 8 gain -6
  moh music-on-hold.au
  dn-webedit
  transfer-system full-consult
  create cnf-files version-stamp 7960
 !
 voice logout-profile 1
 number 3001 type normal
 !
 voice user-profile 1
  max-idle-time 10
  user br2phn3 password adgjm
  number 3102 type normal
  speed-dial 1 3005 label 3005
 !
 ephone  1
  device-security-mode none
  mac-address ..
  ephone-template 1
  max-calls-per-button 5
  busy-trigger-per-button 3
  type 7945
  logout-profile 1

 Thanks
 Viki

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[OSL | CCIE_Voice] IPMA Phone Services URL

2014-01-12 Thread Vignesh Sethuraman
Hello All,

I am trying to find out the IP Phone services URL for IPMA from the DocCD
http://www.cisco.com/cisco/web/psa/default.html

Could you please someone point me to exact navigation.

Thanks,
Viki
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Re: [OSL | CCIE_Voice] IPMA Phone Services URL

2014-01-12 Thread Vignesh Sethuraman
Thanks Attila. I found IPMA URL in the Cisco Unified Communications
Manager Assistant with proxy line support topic in the feature and
services guide.


On Sun, Jan 12, 2014 at 12:03 PM, Attila Rumy rumy.att...@gmail.com wrote:

 Hi Viki,

 In Cisco Unified Communications Manager Features and Services Guide,
 Release 7.0(1) do a search for MAservice or find chapter called Cisco
 Unified Communications Manager Assistant With Proxy Line Support and do a
 search for http, this one is a bit easier to remember.

 Regards,
 Attila
 2014.01.12. 11:55 ezt írta (Vignesh Sethuraman sethuvign...@gmail.com
 ):

 Hello All,

 I am trying to find out the IP Phone services URL for IPMA from the DocCD
 http://www.cisco.com/cisco/web/psa/default.html

 Could you please someone point me to exact navigation.

 Thanks,
 Viki

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Re: [OSL | CCIE_Voice] B-ACD Fastbusy and Unallocated Number

2014-01-12 Thread Vignesh Sethuraman
Josh,

Thanks for your response. My BACD script is passing the calls to
hunt-groups as expected but the issue is am not hearing any welcome prompt.

When dial the pilot number, I dont hear any welcome prompt but the call is
passing through. when I press the dtmf digit 2, the call is passed on to
the huntgroup, at the sametime am hearing the music on hold.

Thanks,
Viki


On Sun, Jan 12, 2014 at 1:12 PM, Josh Petro josh.pe...@gmail.com wrote:

 You need to ensure the music on hold files referenced in your script are
 in flash with the correct, matching filename and in the correct directory.
 In your case, it looks like placing the files in the root if flash will
 work fine. If you dont have the files in the zip file you downloaded fron
 Cisco support, you'll need to download them for it to work. Without them,
 the gateway cannot run the script. Also, just to save you the headache most
 of us have had, don't be surprised if the timers don't react the way that
 they are configured. Please see other posts on the subject.
 Josh
 On Jan 10, 2014 3:26 PM, Vignesh Sethuraman sethuvign...@gmail.com
 wrote:

 Hi,

 I am getting a fastbusy tone and unallocated number messager when I tried
 to call the BACD pilot number from the PSTN and also from the CME
 registered phones. Here is my config.

 application
  service aa flash:app-b-acd-aa-3.0.0.2.tcl
   param number-of-hunt-grps 2
   paramspace english index 1
   param handoff-string aa
   paramspace english language en
   param max-time-vm-retry 2
   param aa-pilot 3500
   paramspace english location flash:
   param welcome-prompt _bacd_welcome.au
   param call-retry-timer 15
   param max-time-call-retry 90
   param voice-mail 3001
   param service-name queue
  !
  service queue flash:app-b-acd-3.0.0.2.tcl
   param queue-len 15
   param aa-hunt1 3210
   param queue-manager-debugs 1
   param aa-hunt2 3005
   param number-of-hunt-grps 2
 !
 dial-peer voice 3500 voip
  service aa
  destination-pattern 3500
  session target ipv4:3.3.3.3
  incoming called-number 3500
  dtmf-relay h245-alphanumeric
  codec g711ulaw
  no vad
 !
 voice service voip
 allow-connections h323 to sip
 !
 telephony-service
 moh music-on-hold.au
 !
 #sh flash:
 47   30421 Jan 9 2014 19:34:16 +01:00 app-b-acd-3.0.0.2.tcl
 48   55599 Jan 9 2014 19:34:16 +01:00 app-b-acd-aa-3.0.0.2.tcl
 50  496521 Jan 10 2014 20:48:16 +01:00 music-on-hold.au
 !
 voice hunt-group 1 parallel
  list 3001,3005
  pilot 3210

 Please let me know what I am missing.

 Thanks,
 Viki


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[OSL | CCIE_Voice] IPMA- Assitant Device is not listed - Vol1 Task 9.4

2014-01-12 Thread Vignesh Sethuraman
Dear All,

During the IPMA configuration, when I tried to do the Assistant
configuration for a user ID, the Device Name, Intercom Line  Primary Line
are not listed.

But the manager configuration is showing the relevant details in the
details in the drop down.

For the Assistant configuration, I have associated the phone to the end
user (Assistant), made the extension 5002 as the primary extension for that
user. Associated the user id (Assitant) as the Owner user ID in the
Device page.

I could not figure out what I am missing. I have followed the same steps as
shown in the Vol 1 Solution Guide.

Thanks,
Viki
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[OSL | CCIE_Voice] can anyone access ipexpert site?

2014-01-11 Thread Vignesh Sethuraman

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[OSL | CCIE_Voice] B-ACD Fastbusy and Unallocated Number

2014-01-10 Thread Vignesh Sethuraman
Hi,

I am getting a fastbusy tone and unallocated number messager when I tried
to call the BACD pilot number from the PSTN and also from the CME
registered phones. Here is my config.

application
 service aa flash:app-b-acd-aa-3.0.0.2.tcl
  param number-of-hunt-grps 2
  paramspace english index 1
  param handoff-string aa
  paramspace english language en
  param max-time-vm-retry 2
  param aa-pilot 3500
  paramspace english location flash:
  param welcome-prompt _bacd_welcome.au
  param call-retry-timer 15
  param max-time-call-retry 90
  param voice-mail 3001
  param service-name queue
 !
 service queue flash:app-b-acd-3.0.0.2.tcl
  param queue-len 15
  param aa-hunt1 3210
  param queue-manager-debugs 1
  param aa-hunt2 3005
  param number-of-hunt-grps 2
!
dial-peer voice 3500 voip
 service aa
 destination-pattern 3500
 session target ipv4:3.3.3.3
 incoming called-number 3500
 dtmf-relay h245-alphanumeric
 codec g711ulaw
 no vad
!
voice service voip
allow-connections h323 to sip
!
telephony-service
moh music-on-hold.au
!
#sh flash:
47   30421 Jan 9 2014 19:34:16 +01:00 app-b-acd-3.0.0.2.tcl
48   55599 Jan 9 2014 19:34:16 +01:00 app-b-acd-aa-3.0.0.2.tcl
50  496521 Jan 10 2014 20:48:16 +01:00 music-on-hold.au
!
voice hunt-group 1 parallel
 list 3001,3005
 pilot 3210

Please let me know what I am missing.

Thanks,
Viki
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[OSL | CCIE_Voice] Question on Media Resources

2014-01-04 Thread Vignesh Sethuraman
Hello All,

I have one PVDM3-16 on my BR1 Router. I can use it for IOS transcoder and
also for IOS CFB. In the ipexpert VoD, I heard Vik Mahli saying the DSP
resources cannot be shared between transcoder and Conference bridge but
when I tried in my Lab it is been shared. I hope PVDM3-16 has got 240 MIPS
which I can use for bearer channels, transcoder and Conference bridge. The
IOS based transcoder and IOS based Conference bridge are successfully
registered to CUCM. When I checked, the maximum session ?, I can see
transcoder can use 8 sessions and conference bridge can use 3 session. Out
of which I have used 4 sessions for transcoder and 1 session for Conference
bridge.

I am not sure what I have misunderstood in Vik's Lecture. Please correct me
if my understanding is wrong.

BR1-RTR#sh inventory | inc PVDM
NAME: PVDM3 DSP DIMM with 16 Channels on Slot 0 SubSlot 4, DESCR: PVDM3
DSP DIMM with 16 Channels
PID: PVDM3-16  , VID: V01 , SN: FOC15452LB4
!
controller T1 0/0/0
 pri-group timeslots 1-3,24 service mgcp
!
dspfarm profile 1 transcode
 codec g729r8
 codec g711ulaw
 codec g711alaw
 codec g729ar8
 codec g729abr8
 maximum sessions 4
 associate application SCCP
dspfarm profile 2 conference
 codec g711ulaw
 codec g711alaw
 codec g729ar8
 codec g729abr8
 codec g729r8
 codec g729br8
 maximum sessions 1
 associate application SCCP
!
sccp local Loopback0
sccp ccm X.X.X.X identifier 2 version 7.0
sccp ccm X.X.X.X identifier 1 version 7.0
sccp
sccp ccm group 1
 associate ccm 1 priority 1
 associate ccm 2 priority 2
 associate profile 2 register IOS-CFB-BR1
 associate profile 1 register IOS-XCODE-BR1
!
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[OSL | CCIE_Voice] Calling party Transformation pattern issue

2013-12-14 Thread Vignesh Sethuraman
Hello All,

I have 2 gateways one H323 GW (HQ) and one MGCP GW (BR1).

For 911 calls, I made the calling party transformation pattern as use
Device pool calling party transformation pattern CSS where I masked the
calling number as 7 Digits.

For Local calls, I created 2 RP one for BR1 and other for HQ. In this case
I used the Calling party transformation pattern on the RP level where I
masked the calling number for 10 digits.

Issue here is, the calls passing over H323 GW is showing 7 digits as
calling number for 911 and 10 digits for local calls.

But the calls passing over MGCP GW is showing 7 digits for 911 and 7 digits
for local calls. The digit manipulation done at Route pattern level for BR1
location is not taken into effect. I did no mgcp and mgcp but still no luck.

Please let me know what I am missing.

Thanks,
Viki
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[OSL | CCIE_Voice] Conferencing and Transcoding guide for 15M

2013-12-07 Thread Vignesh Sethuraman
Hello,

I am trying to find out the document Configuring Conferencing and
Transcoding for Voice Gateway Routers' on the IOS 15MT using the
products/Technology page but could not see it.

I am not able to find it out on any of the config guides in the below URL.
http://www.cisco.com/en/US/products/ps12745/products_installation_and_configuration_guides_list.html#anchor18

I am aware it is available on 12.4T but could not find the same on 15M.

Can you please point me the exact path on the Cisco documentation page.

Thanks,
Viki
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[OSL | CCIE_Voice] MGCP GW Configuration Guide

2013-11-21 Thread Vignesh Sethuraman
Dear All,

Is this document available for the Voice candidates in the Lab.

MGCP and Related Protocols Configuration Guide
http://www.cisco.com/en/US/docs/ios-xml/ios/voice/mgcp/configuration/12-4t/vm-12-4t-book.html

If not, please let me know the on the SRND, on which topic the MGCP GW
configuration is listed.

Thanks,
Viki
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[OSL | CCIE_Voice] VG 202 integration CME (H323)

2013-11-18 Thread Vignesh Sethuraman
Hello All,

I have integrated VG202 with CME using H323. The integration is through IP
connectivity.

I can make calls from IPphone registered to CME to the Analog phone
connected to VG202.

I could not hear any dial-tone when the Analog phone goes off-hook nor I
can dial any Cisco IP phone connected to CME.  I tried to connect to
different port on VG202, used different Analog phone, reloaded the VG202.
No luck.

Please find below the config of  VG202.

voice service voip
 ip address trusted list
  ipv4 0.0.0.0 0.0.0.0
 allow-connections h323 to h323
 supplementary-service h450.12
 fax protocol pass-through g711alaw
 h323
 modem passthrough nse codec g711alaw
!
voice class h323 1
  h225 timeout tcp establish 3
!
voice-card 0
!
interface FastEthernet0/0
 ip address X.X.X.X 255.255.255.0
 duplex auto
 speed auto
 h323-gateway voip interface
 h323-gateway voip bind srcaddr X.X.X.X
!
interface FastEthernet0/1
 no ip address
 duplex auto
 speed auto
!
ip forward-protocol nd
!
no ip http server
ip route 0.0.0.0 0.0.0.0 X.X.X.X
!
control-plane
!
!
voice-port 0/0
 no supervisory disconnect lcfo
 ring frequency 20
 ring cadence pattern11
 disconnect-ack
 compand-type a-law
 cptone MY
 timeouts initial 60
 timeouts interdigit 60
 timeouts ringing infinity
 caller-id enable
!
voice-port 0/1
 no supervisory disconnect lcfo
 disconnect-ack
 compand-type a-law
 cptone MY
 timeouts initial 60
 timeouts interdigit 60
 timeouts ringing infinity
 timeouts power-denial 500
 caller-id enable
!
no ccm-manager fax protocol cisco
!
mgcp profile default
!
!
dial-peer voice 805 voip
 destination-pattern 0T
 modem passthrough nse codec g711alaw
 session target ipv4:X.X.X.X (CME IP)
 voice-class h323 1
 codec g711alaw
 fax protocol none
 no vad
!
dial-peer voice 4805 voip
 modem passthrough nse codec g711alaw
 incoming called-number .
 voice-class h323 1
 codec g711alaw
 fax protocol none
 no vad
!
dial-peer voice 1 pots
 destination-pattern 4805
 port 0/0
 forward-digits all
!
!
!
line con 0
 login local
 no modem enable
line aux 0
line vty 0 4
 login local
 transport input all
!
end

Thanks  Regards,
Vignesh S
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Re: [OSL | CCIE_Voice] Phone View Unified FX

2013-11-01 Thread Vignesh Sethuraman
Dear All,

Regarding the Phone view software for controlling the phones, I have few
questions to other study mates who are using Phone View for their
preparation

Can you confirm you if the Application User (Admin User)  a End User
(Phone User) account in Call Manager for PhoneView to use is pre configured
for us on the proctor labs CUCM and those do have the following
permissions? is the phone I am trying to control is already associated with
the Phone User account? Is the phone I am trying to control already have
web access enabled?
Admin User
• Standard CCM Server Monitoring
• Standard EM Authentication Proxy Rights
• Standard Tab Sync User
Phone User
• Standard CTI Enabled
• Standard CTI Allow Control of Phones supporting Connected Xfer and conf
Or do I need to do all the above mentioned setup during my session?
Thanks,
Viki





On Thu, Oct 31, 2013 at 2:44 PM, Vignesh Sethuraman
sethuvign...@gmail.comwrote:

 Hello Experts,

 I am trying to use Phone view (lab edition) software to control my Lab
 phones. I can see registered phones on the Phone view but when dial any
 extension from any of the registered phones, for example 5001, I see
 message at the bottom on the activity log stating Command (Key:KeyPad5)
 sent to device (SEPMAC of my Phone) using thread (3) with response failure
 .

 I tried to contact the Unified support team on this issue but I have not
 received any response from the support team for last 1 month.

 If any of you have experienced this issue, please guide me to resolve this.

 Thanks,
 Vignesh



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[OSL | CCIE_Voice] Phone View Unified FX

2013-10-31 Thread Vignesh Sethuraman
Hello Experts,

I am trying to use Phone view (lab edition) software to control my Lab
phones. I can see registered phones on the Phone view but when dial any
extension from any of the registered phones, for example 5001, I see
message at the bottom on the activity log stating Command (Key:KeyPad5)
sent to device (SEPMAC of my Phone) using thread (3) with response failure
.

I tried to contact the Unified support team on this issue but I have not
received any response from the support team for last 1 month.

If any of you have experienced this issue, please guide me to resolve this.

Thanks,
Vignesh
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[OSL | CCIE_Voice] Volume 1 Task 11.3

2013-10-21 Thread Vignesh Sethuraman
Dear All,

In the CCIE Voice, IPX volume 1 task 11.3, I am unable to understand what
would be testing result if I press 3 as the caller input.

For caller input 3, the question says, option 3 should allow callers to
modify and enable any greeting for the call handler (including Alternate
Greetings) providing that the caller is the subscriber HQ Phone2 or BR1
Phone2.

I tried to call 2123945000 from PSTN and pressed 3 as the caller input. It
reaches the system callhander I created with extension as 5000. Since I
have chosen for input 3, the conversation as Greetings Administrator,
greeting administrator prompt is asking me to dial the call handler
extension number. when I dial the HQ Phone2 or BR1 phone 2 extn number it
is saying wrong call handler extension. It accepts only when I dial 5000,
so how it relates to HQ phone 2 or BR1 Phone 2 as the call handler owner?
Do i need to dial 5000 from HQ phone 2 or BR1 Phone 2 or am I missing basic
understanding?

Could you please someone explain, how the Greetings Administrator works. I
could not find the testing or the verification in the solution guide.

Thanks,
Viki
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Re: [OSL | CCIE_Voice] Volume 1 Task 11.3

2013-10-21 Thread Vignesh Sethuraman
Hello Martin and Bill,

I have already assigned HQph2 and BR1ph2 as call handler owners, is this
you mean as assigning the role or something else?

Thanks,
Viki

On Monday, October 21, 2013, Martin Sloan wrote:

 Also make sure to assign the 'Greetings Administrator' role to the
 subscriber/end user account.


 On Mon, Oct 21, 2013 at 3:15 PM, Bill Hatcher 
 wchatc...@gmail.comjavascript:_e({}, 'cvml', 'wchatc...@gmail.com');
  wrote:

 First you will need to configure some users in unity, and assign them as
 administrators on Call Handlers.

 When you hit the greeting administrator you will be prompted to enter
 your user ID and password, example 5002 and a vm password of 12345.  Once
 you have been authenticated it will ask you to enter the number of the call
 handler you wish to change followed by #.  After that just follow the
 prompts.


 On Mon, Oct 21, 2013 at 12:21 PM, Vignesh Sethuraman 
 sethuvign...@gmail.com javascript:_e({}, 'cvml',
 'sethuvign...@gmail.com'); wrote:

 Dear All,

 In the CCIE Voice, IPX volume 1 task 11.3, I am unable to understand
 what would be testing result if I press 3 as the caller input.

 For caller input 3, the question says, option 3 should allow callers to
 modify and enable any greeting for the call handler (including Alternate
 Greetings) providing that the caller is the subscriber HQ Phone2 or BR1
 Phone2.

 I tried to call 2123945000 from PSTN and pressed 3 as the caller input.
 It reaches the system callhander I created with extension as 5000. Since I
 have chosen for input 3, the conversation as Greetings Administrator,
 greeting administrator prompt is asking me to dial the call handler
 extension number. when I dial the HQ Phone2 or BR1 phone 2 extn number it
 is saying wrong call handler extension. It accepts only when I dial 5000,
 so how it relates to HQ phone 2 or BR1 Phone 2 as the call handler owner?
 Do i need to dial 5000 from HQ phone 2 or BR1 Phone 2 or am I missing basic
 understanding?

 Could you please someone explain, how the Greetings Administrator works.
 I could not find the testing or the verification in the solution guide.

 Thanks,
 Viki

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Re: [OSL | CCIE_Voice] Outbound PSTN call drops

2013-10-21 Thread Vignesh Sethuraman
Hi Samson,

Have you hard coded the isdn channel to ascending or descending. If so try
to remove that and check.

Did you try isdn bchan-negotiate?

Did you see any errors on the output of show controllers t1/e1, and show
isdn status

Thanks,
Viki

On Monday, October 21, 2013, Samson Kareem wrote:

 Hi All,

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


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


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

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

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

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

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

 does anyone have any pointers?

 thanks
 Samson




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Re: [OSL | CCIE_Voice] Volume 1 Task 11.3

2013-10-21 Thread Vignesh Sethuraman
I meant the owner of HQph2 and BR1ph2 as the call handler owner.

On Monday, October 21, 2013, Vignesh Sethuraman wrote:

 Hello Martin and Bill,

 I have already assigned HQph2 and BR1ph2 as call handler owners, is this
 you mean as assigning the role or something else?

 Thanks,
 Viki

 On Monday, October 21, 2013, Martin Sloan wrote:

 Also make sure to assign the 'Greetings Administrator' role to the
 subscriber/end user account.


 On Mon, Oct 21, 2013 at 3:15 PM, Bill Hatcher wchatc...@gmail.comwrote:

 First you will need to configure some users in unity, and assign them as
 administrators on Call Handlers.

 When you hit the greeting administrator you will be prompted to enter
 your user ID and password, example 5002 and a vm password of 12345.  Once
 you have been authenticated it will ask you to enter the number of the call
 handler you wish to change followed by #.  After that just follow the
 prompts.


 On Mon, Oct 21, 2013 at 12:21 PM, Vignesh Sethuraman 
 sethuvign...@gmail.com wrote:

 Dear All,

 In the CCIE Voice, IPX volume 1 task 11.3, I am unable to understand
 what would be testing result if I press 3 as the caller input.

 For caller input 3, the question says, option 3 should allow callers
 to modify and enable any greeting for the call handler (including Alternate
 Greetings) providing that the caller is the subscriber HQ Phone2 or BR1
 Phone2.

 I tried to call 2123945000 from PSTN and pressed 3 as the caller
 input. It reaches the system callhander I created with extension as 5000.
 Since I have chosen for input 3, the conversation as Greetings
 Administrator, greeting administrator prompt is asking me to dial the call
 handler extension number. when I dial the HQ Phone2 or BR1 phone 2 extn
 number it is saying wrong call handler extension. It accepts only when I
 dial 5000, so how it relates to HQ phone 2 or BR1 Phone 2 as the call
 handler owner? Do i need to dial 5000 from HQ phone 2 or BR1 Phone 2 or am
 I missing basic understanding?

 Could you please someone explain, how the Greetings Administrator
 works. I could not find the testing or the verification in the solution
 guide.

 Thanks,
 Viki

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 please visit www.ipexpert.com

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 www.PlatinumPlacement.com



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

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




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

2013-10-05 Thread Vignesh Sethuraman
Dear All,

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

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

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

Thanks,
Viki
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[OSL | CCIE_Voice] Softkey Template on SIP 9971 registered to CME

2013-08-21 Thread Vignesh Sethuraman
Hello Experts,

I have registered my hardware 9971 SIP phone to CME. I would like to know
how to change the softkey template of 9971 SIP Phone to have the ad-hoc
conference facility. Moreover, do I need to do anything specific to make
9971 as dual-line as like it is did in Skinny Phones.

I tried to check few documentation, documentation say we need to change
feature policy template. I am not sure if this is applicable to phones
registered to CME. Is this the way to change the softkey template of 9971
SIP phone registered to the CME.

Thanks,
Viki
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[OSL | CCIE_Voice] Extension Mobility on CUCME

2013-08-09 Thread Vignesh Sethuraman
Hello,



I am trying to setup the Extension Mobility on CME,  but when I press the
Mobility key, it shows key is not active

here is my config

*telephony-service*

* no auto-reg-ephone*

* authentication credential username password*

* em keep-history*

* max-ephones 1*

* max-dn 2 no-reg both*

* ip source-address 10.0.38.254 port 2000*

* service phone webAccess 0*

* system message ITASSISTANT*

* url authentication http://10.0.38.254/CCMCIP/authenticate.asp username
password*

* load 7945 flash0:term45.default.loads*

* time-format 24*

* date-format dd-mm-yy*

* max-conferences 8 gain -6*

* dn-webedit *

* transfer-system full-consult*

* create cnf-files*

*!*

*voice logout-profile 400*

* pin 2400*

* user 2400 password cisco*

* number 250032400 type normal*

*!*

*voice user-profile 2400*

* pin 2400*

* user 250032400 password 2400*

* number 250032400 type normal*

*!*

*ephone  1*

* device-security-mode none*

* mac-address 0021.55D6.05AE*

* ephone-template 1*

* type 7945*

* no auto-line*

* logout-profile 400*



Please let me know what I am missing.



Thanks,

Viki
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[OSL | CCIE_Voice] Fwd: IPX Vol1 Lab 5C Task 5.8

2013-05-07 Thread Vignesh Sethuraman
Hi,

I am working on IPX Vol1 Lab 5C Task 5.8, I am not able to get the calls
working. When I checked the output of debug gatekeeper main 10, I could see
the following logs on the PSTN router (Remote Gatekeeper).

PSTNRouter#
Apr 27 09:55:21.071: ////GK/gk_rassrv_lrq:
(916745738932) Tech-prefix match failed.
Apr 27 09:55:21.071: ////GK/gk_rassrv_lrq:
(916745738932) Matched zone-prefix 91
Apr 27 09:55:21.071: ////GK/gk_rassrv_lrq: checking
the source of the LRQ. source_endptp=0x0
Apr 27 09:55:21.071: ////GK/gk_rassrv_lrq: srcvia
found gkname of source zone. looking up HQ-RTR in zone list
Apr 27 09:55:21.071: ////GK/gk_rassrv_lrq: about to
check the source side, src_zonep=0x33CB1380
Apr 27 09:55:21.071: ////GK/gk_rassrv_lrq: matched
zone is HQ-RTR
Apr 27 09:55:21.071: ////GK/gk_rassrv_lrq  and
z_invianamelen=0
Apr 27 09:55:21.071: ////GK/gk_rassrv_lrq: about to
check the destination side, zonep=0x33CB10C0
Apr 27 09:55:21.071: ////GK/gk_rassrv_lrq:matched
zone is PSTN-WAN
PSTNRouter#
Apr 27 09:55:21.071: ////GK/gk_rassrv_lrq  and
z_outvianamelen=0
Apr 27 09:55:21.071: ////GK/gk_rassrv_lrq:
(916745738932) Default-tech gateway selection failed for zone PSTN-WAN
PSTNRouter#
PSTNRouter#
Apr 27 09:55:35.971: ////GK/gk_process: got a TIMER
event

Apr 27 09:55:35.971: ////GK/gk_handle_timers

Apr 27 09:55:35.971: ////GK/gk_handle_timers:
managed timer expired 0x2C5C9EB0

Here are my local GK and remote GK configuration.

HQ_CO#sh run | s gatekeeper
gatekeeper
 zone local SPAIN X 10.0.10.254
 zone local HQ-RTR ipexpert.com
 zone remote PSTN-WAN ipexpert.com 10.10.100.2 1719
 zone prefix PSTN-WAN 91*
gw-type-prefix 1#* default-technology
 no shutdown


PSTNRouter#sh run | s gatekeeper
gatekeeper
 zone local PSTN-WAN ipexpert.com
 zone remote HQ-RTR ipexpert.com 10.0.10.254 1719
 zone remote US ipexpert.com 10.0.48.254 1719
 zone prefix PSTN-WAN 34*
 zone prefix PSTN-WAN 91*
 gw-type-prefix 1#* default-technology
 no shutdown
PSTNRouter#

can you please let me know what I am missing.

Thanks,
Viki
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[OSL | CCIE_Voice] IPX Vol1 Lab 5C Task 5.8

2013-04-27 Thread Vignesh Sethuraman
Hi,

I am working on IPX Vol1 Lab 5C Task 5.8, I am not able to get the calls
working. When I checked the output of debug gatekeeper main 10, I could see
the following logs on the PSTN router (Remote Gatekeeper).

PSTNRouter#
Apr 27 09:55:21.071: ////GK/gk_rassrv_lrq:
(916745738932) Tech-prefix match failed.
Apr 27 09:55:21.071: ////GK/gk_rassrv_lrq:
(916745738932) Matched zone-prefix 91
Apr 27 09:55:21.071: ////GK/gk_rassrv_lrq: checking
the source of the LRQ. source_endptp=0x0
Apr 27 09:55:21.071: ////GK/gk_rassrv_lrq: srcvia
found gkname of source zone. looking up HQ-RTR in zone list
Apr 27 09:55:21.071: ////GK/gk_rassrv_lrq: about to
check the source side, src_zonep=0x33CB1380
Apr 27 09:55:21.071: ////GK/gk_rassrv_lrq: matched
zone is HQ-RTR
Apr 27 09:55:21.071: ////GK/gk_rassrv_lrq  and
z_invianamelen=0
Apr 27 09:55:21.071: ////GK/gk_rassrv_lrq: about to
check the destination side, zonep=0x33CB10C0
Apr 27 09:55:21.071: ////GK/gk_rassrv_lrq:matched
zone is PSTN-WAN
PSTNRouter#
Apr 27 09:55:21.071: ////GK/gk_rassrv_lrq  and
z_outvianamelen=0
Apr 27 09:55:21.071: ////GK/gk_rassrv_lrq:
(916745738932) Default-tech gateway selection failed for zone PSTN-WAN
PSTNRouter#
PSTNRouter#
Apr 27 09:55:35.971: ////GK/gk_process: got a TIMER
event

Apr 27 09:55:35.971: ////GK/gk_handle_timers

Apr 27 09:55:35.971: ////GK/gk_handle_timers:
managed timer expired 0x2C5C9EB0

Here are my local GK and remote GK configuration.

HQ_CO#sh run | s gatekeeper
gatekeeper
 zone local SPAIN domain-name 10.0.10.254
 zone local HQ-RTR domain-name
 zone remote PSTN-WAN ipexpert.com 10.10.100.2 1719
 zone prefix PSTN-WAN 91*
gw-type-prefix 1#* default-technology
 no shutdown


PSTNRouter#sh run | s gatekeeper
gatekeeper
 zone local PSTN-WAN ipexpert.com
 zone remote HQ-RTR ipexpert.com 10.0.10.254 1719
 zone remote US ipexpert.com 10.0.48.254 1719
 zone prefix PSTN-WAN 34*
 zone prefix PSTN-WAN 91*
 gw-type-prefix 1#* default-technology
 no shutdown
PSTNRouter#

can you please let me know what I am missing.

Thanks,
Viki
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[OSL | CCIE_Voice] CCIE Voice Vol1 Task 5.7

2013-03-16 Thread vignesh sethuraman
Hello Experts,

I am working on Task 5.7 from Vol1. Question is to block the 91900? numbers. I 
have configured a Route pattern to block this number but this Route pattern is 
overridden by a another Route pattern 9.1[2-9]XX[2-9]XX which I have 
created for Task 5.6.I understand the longest match wins but I need to make it 
work as said in the question to play the error message the precedence used is 
not authorized for your line. 

Could you please let me know is there a way to make this work as expected in 
the question or am I missing something.


Thanks,
Vignesh
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[OSL | CCIE_Voice] Nortel to Cisco Unified Communication Migration

2013-03-13 Thread vignesh sethuraman
Hello Experts,

I am not sure if this the right forum to post my question but giving a try.

I am in the initial stage of a project for Migrating Nortel to Cisco UC. I need 
a Generic Design document and the implementation plan which I can use as a 
reference to start with the project and will customize as per the customer 
requirement. If someone has any case study or any document related to this, 
could you please send me.


Thanks,
Vignesh___
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Re: [OSL | CCIE_Voice] all incoming calls to HQ phones failing

2013-03-09 Thread vignesh sethuraman
Hi Farooq,

Did you see calls coming into the mgcp gw, use debug isdn q931 to check.

Try no mgcp  mgcp on the Hq gw.

Thanks,
Viki
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Re: [OSL | CCIE_Voice] VM in SRST

2013-03-07 Thread vignesh sethuraman
Hi, 

I am working on CCIE Voice IPX Vol1 task 5.1, as mentioned in the question, I 
removed the SIP and tried to configure the HQ Router as H.323 GW. The issue is 
it is affecting the task that I did in Lab 4A (4.6 and 4.7).

Basic question,can I have a router acting as H.323 GW and also as the 
Gatekeeper.

I suspect the port 1720 is being used by the H.323 GW for the task 5.1 and the 
same cannot be used for the Gatekeeper funtion so it is affecting the task 4.6 
 4.7.

One thing I have not done as per the workbook is I have not loaded the new 
configs for this Lab, instead am configuring it on top of Lab 4A. 
 
Can you please help me in this regard.

Thanks,
Vignesh___
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[OSL | CCIE_Voice] CCIE Voice Vol1 Task 4.2 Workbook Video Solution

2013-02-12 Thread vignesh sethuraman
Hello All,


I was listening to Vol1 workbook video solution, in Task 4.2,  question was to 
add BR2 as the H323 GW but the video solution is about adding HQ RTR as MGCP GW 
and configuring Route Groups etc.

Am I missing something or my understanding of Task is wrong?

Thanks,
Vignesh___
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Re: [OSL | CCIE_Voice] CCIE Voice - IPX Vol1 - Lab 4A - Task 4.7

2013-02-06 Thread vignesh sethuraman
Hello All,

The solution for my issue was both the H323 trunk and Gatekeeper controlled 
trunk should have the significant digits as 4 for the inbound calls.

And the Gatekeeper controlled trunk should have Enable Inbound Faststart. 


Thanks Ryan for the hint, the only change I did was on the Gatekeeper 
controlled trunk instead of H323GW. When I did that on H323GW the issue was 
there so just tried it on the Gatekeeper controlled trunk and removed it on the 
H323GW, it started working.


Thanks,
Viki




 From: Ryan Maxam ryan.ma...@gmail.com
To: vignesh sethuraman sethuvign...@yahoo.co.in 
Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com 
Sent: Tuesday, 5 February 2013 8:25 PM
Subject: Re: [OSL | CCIE_Voice] CCIE Voice - IPX Vol1 - Lab 4A - Task 4.7
 

I ran into this same problem.  if you check Enable Inbound FastStart on you 
H323 gateway that should correct the problem.  Its caused because you are 
communicating between a SIP trunk and an H323 trunk.  

On Tuesday, February 5, 2013, vignesh sethuraman  wrote:

Hello,


In the task 4.7 of IPX Vol1 Lab 4A, am able to call and answer the calls to 
BR2 from HQ and BR1. 


But when I dial HQ phones or BR1 phones from BR2, am getting a ring back tone 
but I could not answer the calls. Even after picking the handset, I could hear 
the ringback on the BR2 phones.


I have kept the significant digits as 4 on both the CME (H.323GW) trunk and 
Gatekeeper controlled trunk.


Could you please someone help to resolve this.


Thanks,
Viki___
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Re: [OSL | CCIE_Voice] CCIE Voice - IPX Vol1 - Lab 4A - Task 4.7

2013-02-06 Thread vignesh sethuraman
Missed to mention, this issue was faced only when I originate a call from SIP 
Phone registeredto CME towards SCCP/SIP Phones registered to UCM. If I 
originate a call from SCCP Phone registered to CME towards SCCP/SIP Phones 
registered to UCM everything is working fine without Enabling Inbound 
Faststart on the Gatekeeper controlled Trunk.


In either of the case, the significant digits on the inbound calls on both the 
H323GW trunk and Gatekeeper controlled trunk should be 4.

Just sending this as it might be useful to someone facing the same issue.


Thanks,
Viki




 From: vignesh sethuraman sethuvign...@yahoo.co.in
To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com 
Sent: Wednesday, 6 February 2013 10:51 AM
Subject: Re: [OSL | CCIE_Voice] CCIE Voice - IPX Vol1 - Lab 4A - Task 4.7
 

Hello All,

The solution for my issue was both the H323 trunk and Gatekeeper controlled 
trunk should have the significant digits as 4 for the inbound calls.

And the Gatekeeper controlled trunk should have Enable Inbound Faststart. 


Thanks Ryan for the hint, the only change I did was on the Gatekeeper 
controlled trunk instead of H323GW. When I did that on H323GW the issue was 
there so just tried it on the Gatekeeper controlled trunk and removed it on the 
H323GW, it started working.


Thanks,
Viki




 From: Ryan Maxam ryan.ma...@gmail.com
To: vignesh sethuraman sethuvign...@yahoo.co.in 
Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com 
Sent: Tuesday, 5 February 2013 8:25 PM
Subject: Re: [OSL | CCIE_Voice] CCIE Voice - IPX Vol1 - Lab 4A - Task 4.7
 

I ran into this same problem.  if you check Enable Inbound FastStart on you 
H323 gateway that should correct the problem.  Its caused because you are 
communicating between a SIP trunk and an H323 trunk.  

On Tuesday, February 5, 2013, vignesh sethuraman  wrote

Hello,


In the task 4.7 of IPX Vol1 Lab 4A, am able to call and answer the calls to 
BR2 from HQ and BR1. 


But when I dial HQ phones or BR1 phones from BR2, am getting a ring back tone 
but I could not answer the calls. Even after picking the handset, I could hear 
the ringback on the BR2 phones.


I have kept the significant digits as 4 on both the CME (H.323GW) trunk and 
Gatekeeper controlled trunk.


Could you please someone help to resolve this.


Thanks,
Viki


___
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www.ipexpert.com

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

Are you a CCNP or CCIE and looking for a job? Check out 
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[OSL | CCIE_Voice] CCIE Voice - IPX Vol1 - Lab 4A - Task 4.7

2013-02-05 Thread vignesh sethuraman
Hello,

In the task 4.7 of IPX Vol1 Lab 4A, am able to call and answer the calls to BR2 
from HQ and BR1. 

But when I dial HQ phones or BR1 phones from BR2, am getting a ring back tone 
but I could not answer the calls. Even after picking the handset, I could hear 
the ringback on the BR2 phones.

I have kept the significant digits as 4 on both the CME (H.323GW) trunk and 
Gatekeeper controlled trunk.

Could you please someone help to resolve this.

Thanks,
Viki___
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www.ipexpert.com

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