[OSL | CCIE_Voice] selling IPX rack voucher..

2013-09-17 Thread Amit Sharma
guys..
i have around 17-20 ipx voice rack voucher..
if anyone want to buy please tell..


thanks
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[OSL | CCIE_Voice] cuc vm issue with hq user..

2013-08-26 Thread Amit Sharma
guys...when i working on ipx racks...
i am facing issue...
i can access cuc work on it...
when leave  vm from pstn or end user to nay cuc user...it goes...correctly..
but on ip phones..it shows no mwi and not vm notification..

 and also when i check vm..it says no vm for u...

what is issue and how can i fix it?

i tried 5-10 times with all config...but same results...
for hq, sb all end users...


please help em to fix it...

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[OSL | CCIE_Voice] lab qos for cue to cucm..

2013-08-17 Thread Amit Sharma
anyone tell me if this is right or need changes?
configure switch 1 with policies on the 3750 switch:

1: ensure cos value 5 is mapped to dscp ef /cue signal with cs3

2: in giga int 1/0/4, make sure all incoming cue signal traffic is amrked
with cs3  and guarantee to 7k bandwidth.
Anything in excess should be first amrked down to dsco value of 8 before
being transmitted.

3. use requirements listed in the cue section to deliver teh list of
protocols to be policed.


On HQ Switch side:
-
mls qos map policed-dscp  24 to 8
mls qos map cos-dscp 0 8 16 24 32 46 48 56
mls qos

class-map match-any SUB_TO_CUE
 match access-group name SUB_TO_CUE

policy-map SUB_TO_CUE
class SUB_TO_CUE
  set ip dscp cs3
  police 8000 8000 exceed-action policed-dscp-transmit
class class-default
  trust dscp

ip access-list extended SUB_TO_CUE
 permit tcp host 142.100.64.12 eq 2748 host 142.1.66.253
 permit tcp host 142.100.64.12 eq smtp host 142.1.66.253
 permit tcp host 142.100.64.12 eq 443 host 142.1.66.253
 permit tcp host 142.100.64.12 eq 8443 host 142.1.66.253
 permit tcp host 142.100.64.12 eq www host 142.1.66.253

interface GigabitEthernet1/0/1
description R1 Trunk
switchport trunk encapsulation dot1q
switchport trunk allowed vlan 1,100,102,202
switchport mode trunk
mls qos trust dscp
spanning-tree portfast trunk

interface GigabitEthernet1/0/3
description Publisher Port
switchport mode access
mls qos trust dscp
spanning-tree portfast


interface GigabitEthernet1/0/4
description Subscriber Port
service-policy input SUB_TO_CUE
switchport access vlan 100
switchport mode access
spanning-tree portfast

interface GigabitEthernet1/0/13
description *** IP Phones switchports
switchport access vlan 202
switchport mode access
switchport voice vlan 102
mls qos trust device cisco-phone
mls qos trust cos
spanning-tree portfast

interface GigabitEthernet1/0/14
description *** IP Phones switchports
switchport access vlan 202
switchport mode access
switchport voice vlan 102
mls qos trust device cisco-phone
mls qos trust cos
spanning-tree portfast

interface GigabitEthernet1/0/15
description *** IP Phones switchports
switchport access vlan 202
switchport mode access
switchport voice vlan 102
mls qos trust device cisco-phone
mls qos trust cos
spanning-tree portfast
!

=

On R3 router side:
--
-
!
class-map match-any cue_TO_sub
 match access-group name cue_TO_sub
!
policy-map cue_TO_sub
class cue_TO_sub
  set ip dscp cs3
  police 8000 8000 exceed-action policed-dscp-transmit
class class-default
  trust dscp
!
ip access-list extended cue_TO_sub
 permit tcp host 142.1.66.253 eq 2748 host 142.100.64.12
 permit tcp host 142.1.66.253 eq smtp host 142.100.64.12
 permit tcp host 142.1.66.253 eq 443 host 142.100.64.12
 permit tcp host 142.1.66.253 eq 8443 host 142.100.64.12
 permit tcp host 142.1.66.253 eq www host 142.100.64.12
!

interface serial 0/1/0:0.1 point-to-point
description serial port to HQ Router
service-policy output cue_TO_sub
!



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[OSL | CCIE_Voice] lab 5 sip + call

2013-08-09 Thread Amit Sharma
i am not able to connect call to pstn...
is it IOS issue or my config issue/

please help me...when trying without + call working...
but when aplly + called number call failes..
what could be issue?

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[OSL | CCIE_Voice] vm transfer with 2 key steps

2013-08-08 Thread Amit Sharma
how can we transfer call to vm..?
setup is like this...
when get call from pstn to 2001...
accept call and then press transfer key...then dial *2002 that is cti route
point..which forward all calls to vm..
ten i dial *2002 to transfer to vm...
it plays vm and going to vm...
but still call on 2001 phone...
i want to move call from 2001 and it should play vm prompt on pstn phone..

i enable transfer on hook but no affect...
anyone help me...!
what i missed...


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[OSL | CCIE_Voice] five hand lab 5 sip call

2013-08-08 Thread Amit Sharma
guys i am trying to call sip number on pstn side... but call not
connecting..
getting message check the number u dialed...

how can fix it/?


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[OSL | CCIE_Voice] cue register issue with ccm

2013-08-08 Thread Amit Sharma
guys.,,
anyone help me as i am using ipx remote racks...and always issue to
register cue with cucm...


i am doing all config but it is not registering with cucm...

please tell me how can fix it?

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[OSL | CCIE_Voice] tfr to vm....

2013-08-04 Thread Amit Sharma
guys,

when i accept a call from pstn phone and then transfer to vm by press shift
and speed dail that is configure for VMit still showing to press
transfer again

but as i apply same process in SRST case...for 4001it works and call
moves goes away from 40001 and it shows on pstn phone to play vm...



what could be issue due to that it is not working with cucm but working
with SiteC srst case?


anyone help would be more appreciated!

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[OSL | CCIE_Voice] SRST TCL not reaching vm...

2013-07-31 Thread Amit Sharma
can anyone tell me what is issue with this log and config?

ul 31 23:43:14.446: //142//TCL :/tcl_PutsObjCmd: TCL AA:  Destination
Huntgroup = 4888 
Jul 31 23:43:14.458: //35//TCL :/tcl_PutsObjCmd: TCL B-ACD:   THE QUEUE
IS {TclModule_498EB2F0_0_3103288 {CALL_PROGRESS} {1375314194}} 
SiteC-RTR#
Jul 31 23:43:15.042: %ISDN-6-CONNECT: Interface Serial0/0/0:0 is now
connected to 8522535 N/A
SiteC-RTR#
Jul 31 23:43:24.434: //35//TCL :/tcl_PutsObjCmd: TCL B-ACD:   THE QUEUE
IS {TclModule_498EB2F0_0_3103288 {CALL_PROGRESS} {1375314204}} 
Jul 31 23:43:24.434: //35//TCL :/tcl_PutsObjCmd: TCL B-ACD:  Received
HELLO from TclModule_498EB2F0_0_3103288 
SiteC-RTR#
Jul 31 23:43:27.130: //35//TCL :/tcl_PutsObjCmd: TCL B-ACD:   B-ACD
Service Running 
Jul 31 23:43:27.130: //35//TCL :/tcl_PutsObjCmd: TCL B-ACD: 
{TclModule_498EB2F0_0_3103288 {CALL_PROGRESS} {1375314204}} 
SiteC-RTR#
Jul 31 23:43:34.434: //35//TCL :/tcl_PutsObjCmd: TCL B-ACD:   THE QUEUE
IS {TclModule_498EB2F0_0_3103288 {CALL_PROGRESS} {1375314214}} 
Jul 31 23:43:34.434: //35//TCL :/tcl_PutsObjCmd: TCL B-ACD:  Received
HELLO from TclModule_498EB2F0_0_3103288 
Jul 31 23:43:34.502: //142//TCL :/tcl_PutsObjCmd:
 proc act_CallSetupDone
Jul 31 23:43:34.502:
Jul 31 23:43:34.502: //142//TCL :/tcl_PutsObjCmd:
 IP IVR Leg Setup Status = ls_002
Jul 31 23:43:34.502:
Jul 31 23:43:34.506: //142//TCL :/tcl_PutsObjCmd: TCL AA:  Agent did not
answer; Received Call NOAN 
Jul 31 23:43:34.510: //35//TCL :/tcl_PutsObjCmd:
SiteC-RTR#TCL B-ACD:   THE QUEUE IS {TclModule_498EB2F0_0_3103288
{CALL_NOAN} {1375314214}} 
SiteC-RTR#
Jul 31 23:43:43.963: //142//TCL :/tcl_PutsObjCmd: TCL AA: -- MEDIA STATUS =
ms_000 --
Jul 31 23:43:43.983: //142//AFW_:/AFW_FSM_Drive: Tcl_Eval to drive FSM
inside Tcl modulespace. code=1 code=ERROR
Jul 31 23:43:43.983: TCL script failure
Result:
 Play Failed
Jul 31 23:43:43.983:TCL script failure errorInfo:
Play Failed
while executing
media play leg_incoming moh:livemoh
(procedure act_EvMediaDone line 19)
invoked from within
act_EvMediaDone
SiteC-RTR#



application
 service app-b-acd-aa
  param voice-mail *4001
  paramspace english index 0
  param max-time-call-retry 55
  param service-name app-b-acd
  param number-of-hunt-grps 1
  param drop-through-option 1
  paramspace english language en
  param handoff-string app-b-acd-aa
  param max-time-vm-retry 1
  paramspace english location flash:
  param aa-pilot 4000
  param second-greeting-time 30
  param drop-through-prompt _bacd_welcome.au
  param call-retry-timer 10
 !
 service app-b-acd
  param queue-len 2
  param aa-hunt1 4888
  param queue-manager-debugs 1
  param number-of-hunt-grps 1
!
dial-peer voice 4000 voip
 service app-b-acd-aa
 destination-pattern 4000
 session target ipv4:142.102.66.254
 incoming called-number 4000
 dtmf-relay h245-alphanumeric
 codec g711ulaw
!
dial-peer voice 4001 pots
 service app-b-acd-aa
 incoming called-number 4000
!
telephony-service
moh en_bacd_music-on-hold.au
!
ccm-manager music-on-hold
!
ephone-hunt 1 longest-idle
 pilot 4888
 list 4101, 4102
 timeout 10, 10
 auto logout 2 dynamic
!


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Amit Sharma
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[OSL | CCIE_Voice] srst of sitec cme

2013-07-27 Thread Amit Sharma
when go in srst...as i apply speed dail it removes ephone extension from
phone

but when go back to cucm from srst it comes automaticlaly...

why is it happening?

any solution!


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Amit Sharma
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[OSL | CCIE_Voice] SiteC SRST

2013-07-27 Thread Amit Sharma
Guys,

when go in sitec srst all working ok...

when come backto cucm phones register but vm is not working...


i reload cue and cucm service restart but no affect...


what could be issue  and how can fix it?

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Amit Sharma
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[OSL | CCIE_Voice] issue with sip trunk with MTP

2013-07-20 Thread Amit Sharma
guys, i have made this setupa nd capture call logsin pstn phone
when i see it is not accepting called party prefix + that i applied in
roulte list...
Can anyone help me why it is not taking it?
my setup is like this:

cucm...sip trunk with ip of PSTN GW...
sip trunk amd cucm default mtp under MRG and MRG is under MRGL...
then i have SIP RGthat covers SIP TRUNK..
then RLthat covers SIP RG
then under RLi selected SIP TRUNk and applied called prefix + and
calling number on...that is on phones..

then create route pattern :
9011.322!

but when i call it shows only 322number not accepting + prefix for
called number..
?



PSTN-WAN#
Jul 19 23:33:35.711: //25//CCAPI/cc_api_caps_ind:
   Call Entry Is Not Found
Jul 19 23:33:35.715: //-1/7753160E804A/CCAPI/cc_api_display_ie_subfields:
   cc_api_call_setup_ind_common:
   cisco-username=+14082022001
   - ccCallInfo IE subfields -
   cisco-ani=14082022001
   cisco-anitype=1
   cisco-aniplan=1
   cisco-anipi=0
   cisco-anisi=1
   dest=3225251234
   cisco-desttype=1
   cisco-destplan=1
   cisco-rdie=
   cisco-rdn=
   cisco-rdntype=0
   cisco-rdnplan=0
   cisco-rdnpi=-1
   cisco-rdnsi=-1
   cisco-redirectreason=-1   fwd_final_type =0
   final_redirectNumber =
   hunt_group_timeout =0

Jul 19 23:33:35.715: //-1/7753160E804A/CCAPI/cc_api_call_setup_ind_common:
   Interface=0x4854FBAC, Call Info(
   Calling Number=14082022001,(Calling Name=)(TON=International, NPI=ISDN,
Scree
ning=User, Passed, Presentation=Allowed),
   Called Number=3225251234(TON=International, NPI=ISDN),
   Calling Translated=FALSE, Subscriber Type S
PSTN-WAN#tr=Unknown, FinalDestinationFlag=TRUE,
   Incoming Dial-peer=0, Progress Indication=NULL(0), Calling IE
Present=TRUE,
   Source Trkgrp Route Label=, Target Trkgrp Route Label=, CLID
Transparent=FALS
E), Call Id=25
Jul 19 23:33:35.715: //-1//CCAPI/cc_get_feature_vsa:

Jul 19 23:33:35.715: :cc_get_feature_vsa malloc success
Jul 19 23:33:35.715: //-1//CCAPI/cc_get_feature_vsa:

Jul 19 23:33:35.715:  cc_get_feature_vsa count is 1
Jul 19 23:33:35.715: //-1//CCAPI/cc_get_feature_vsa:

Jul 19 23:33:35.715: :FEATURE_VSA attributes are:
feature_name:0,feature_time:11
90726336,feature_id:25
Jul 19 23:33:35.715: //25/7753160E804A/CCAPI/cc_api_call_setup_ind_common:
   Set Up Event Sent;
   Call Info(Calling Number=14082022001(TON=International, NPI=ISDN,
Screening=U
ser, Passed, Presentation=Allowed),
   Called Number=3225251234(TON=International, NPI=ISDN))
Jul 19 23:33:35.719: //25/7753160E804A/CCAPI/cc_process_call_setup_ind:
   Event=0x48963510
Jul 19 23:33:35.719: //25/7753160E804A/CCAPI/ccCallSetContext:
   Context=0x493EF1D8
Jul 19 23:33:35.719: //25/7753160E804A/CCAPI/cc_process_call_setup_ind:
   CCAPI handed cid 25 with tag 0 to app _ManagedAppProcess_Default
Jul 19 23:33:35.719: //25/7753160E804A/CCAPI/ccCallProceeding:
   Progress Indication=NULL(0)
Jul 19 23:33:35.723: //25/7753160E804A/CCAPI/ccCallDisconnect:
   Cause Value=1, Tag=0x0, Call Entry(Previous Disconnect Cause=0,
Disconnect
Ca
use=0)
Jul 19 23:33:35.723: //25/7753160E804A/CCAPI/ccCallDisconnect:
   Cause Value=1, Call Entry(Responsed=TRUE, Cause Value=1)
Jul 19 23:33:35.735: //25/7753160E804A/CCAPI/cc_api_call_disconnect_done:
   Disposition=0, Interface=0x4854FBAC, Tag=0x0, Call Id=25,
   Call Entry(Disconnect Cause=1, Voice Class Cause Code=0, Retry Count=0)
Jul 19 23:33:35.735: //25/7753160E804A/CCAPI/cc_api_call_disconnect_done:
   Call Disconnect Event Sent
Jul 19 23:33:35.735: //-1//CCAPI/cc_free_feature_vsa:

Jul 19 23:33:35.735: :cc_free_feature_vsa freeing 46F90AB8
Jul 19 23:33:35.735: //-1//CCAPI/cc_free_feature_vsa:

Jul 19 23:33:35.735:  vsacount in free is 0
PSTN-WAN#

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[OSL | CCIE_Voice] qos for cue that coming to switch 1/4 port

2013-07-16 Thread Amit Sharma
guys..i have this config...

mla qos
mls qos mal policed-dscp 24 to 8
mls qos map cos-dscp 0 8  16 24 32 46 48 56

ip access-list extended signal
permit tcp host 10.10.210.10 eq 2748 host 10.10.115.2
permit tcp host 10.10.210.11 eq 2748 host 10.10.115.2



class-map match any signal
match access-group name signal


policy-map cue
class signal
set dscp cs3
police 8000 8000 exceed action transmit dscp policed
class class-default
fair queue


int giga1/0/4
service policy input cue

exit



is this config correct or need something else for complete it?

we will apply mls qos trust dscp on all trunk and server ports./..
ip phone pot will sue...

mls qos trust cos
mls qos trust device ciscp phone//


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[OSL | CCIE_Voice] srst call issue

2013-07-10 Thread Amit Sharma
guys i applied siteb-srst...but when trying call outside and inside not
working..

how can fix it?



i also face issue when apply one button login service on phones...
when try to access service...showing requesting...?

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SiteB-RTR#sh isdn status 
Global ISDN Switchtype = primary-ni
ISDN Serial0/0/0:23 interface
dsl 0, interface ISDN Switchtype = primary-ni
Layer 1 Status:
ACTIVE
Layer 2 Status:
TEI = 0, Ces = 1, SAPI = 0, State = MULTIPLE_FRAME_ESTABLISHED
Layer 3 Status:
0 Active Layer 3 Call(s)
Active dsl 0 CCBs = 0
The Free Channel Mask:  0x800F
Number of L2 Discards = 0, L2 Session ID = 0
Total Allocated ISDN CCBs = 0
SiteB-RTR#
SiteB-RTR#

SiteB-RTR#sh ephone registered 


ephone-1[0] Mac:000F.8FC7.BB42 TCP socket:[2] activeLine:0 REGISTERED in SCCP 
ver 11/9
mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 caps:8 
privacy:1
IP:10.10.201.40 49882 Telecaster 7960  keepalive 7 max_line 6
button 1: dn 1  number 3001  CM Fallback CH1   IDLE CH2   IDLE 
CH3   IDLE CH4   IDLE CH5   IDLE CH6   IDLE CH7 
  IDLE CH8   IDLE 
Preferred Codec: g711ulaw 

SiteB-RTR#
SiteB-RTR#
SiteB-RTR#
SiteB-RTR#
SiteB-RTR#
SiteB-RTR#
SiteB-RTR#
SiteB-RTR#
SiteB-RTR#
Jul 10 03:22:23.517: ISDN Se0/0/0:23 Q931: RX - SETUP pd = 8  callref = 0x0085 
Bearer Capability i = 0x8090A2 
Standard = CCITT 
Transfer Capability = Speech  
Transfer Mode = Circuit 
Transfer Rate = 64 kbit/s 
Channel ID i = 0xA98381 
Exclusive, Channel 1 
Progress Ind i = 0x8583 - Origination address is non-ISDN  
Display i = 'PSTN-SITEB-PHON' 
Calling Party Number i = 0x4180, '525' 
Plan:ISDN, Type:Subscriber(local) 
Called Party Number i = 0xC1, '3033001' 
Plan:ISDN, Type:Subscriber(local)
Jul 10 03:22:23.517: ISDN Se0/0/0:23 Q931: Received SETUP  callref = 0x8085 
callID = 0x0005 switch = primary-ni interface = User 
SiteB-RTR#
Jul 10 03:22:23.537: ISDN Se0/0/0:23 Q931: TX - CALL_PROC pd = 8  callref = 
0x8085 
Channel ID i = 0xA98381 
Exclusive, Channel 1
SiteB-RTR#
Jul 10 03:22:26.569: ISDN Se0/0/0:23 Q931: TX - DISCONNECT pd = 8  callref = 
0x8085 
Cause i = 0x80A6 - Network out of order
Jul 10 03:22:26.577: ISDN Se0/0/0:23 Q931: RX - RELEASE pd = 8  callref = 
0x0085 
Cause i = 0x82E4 - Invalid information element contents
Jul 10 03:22:26.581: ISDN Se0/0/0:23 Q931: TX - RELEASE_COMP pd = 8  callref = 
0x8085
SiteB-RTR#
Jul 10 03:22:37.169: ISDN Se0/0/0:23 Q931: RX - SETUP pd = 8  callref = 0x0086 
Bearer Capability i = 0x8090A2 
Standard = CCITT 
Transfer Capability = Speech  
Transfer Mode = Circuit 
Transfer Rate = 64 kbit/s 
Channel ID i = 0xA98381 
Exclusive, Channel 1 
Progress Ind i = 0x8583 - Origination address is non-ISDN  
Display i = 'PSTN-SITEB-PHON' 
Calling Party Number i = 0x4180, '525' 
Plan:ISDN, Type:Subscriber(local) 
Called Party Number i = 0x80, '97230330001' 
Plan:Unknown, Type:Unknown
Jul 10 03:22:37.169: ISDN Se0/0/0:23 Q931: Received SETUP  callref = 0x8086 
callID = 0x0006 switch = primary-ni interface = User 
Jul 10 03:22:37.189: ISDN Se0/0/0:23 Q931: Applying typeplan for sw-type 0xD is 
0x4 0x1, Calling num 525
Jul 10 03:22:37.193: ISDN Se0/0/0:23 Q931: Sending SETUP  callref = 0x0082 
callID = 0x8003 switch = primary-ni interface = User 
Jul 10 03:22:37.193: ISDN Se0/0/0:23 Q931: TX - CALL_PROC pd = 8  callref = 
0x8086 
Ch
SiteB-RTR#annel ID i = 0xA98381 
Exclusive, Channel 1
Jul 10 03:22:37.193: ISDN Se0/0/0:23 Q931: TX - SETUP pd = 8  callref = 0x0082 
Bearer Capability i = 0x8090A2 
Standard = CCITT 
Transfer Capability = Speech  
Transfer Mode = Circuit 
Transfer Rate = 64 kbit/s 
Channel ID i = 0xA98384 
Exclusive, Channel 4 
Progress Ind i = 0x8583 - Origination address is non-ISDN  
Calling Party Number i = 0x4180, '525' 
Plan:ISDN, Type:Subscriber(local) 
Called Party Number i = 0xC1, '7230330001' 
Plan:ISDN, Type:Subscriber(local)
Jul 10 03:22:37.225: ISDN Se0/0/0:23 Q931: RX - CALL_PROC pd = 8  callref = 
0x8082 
Channel ID i = 0xA98384 
Exclusive, Channel 4
Jul 10 03:22:37.233: ISDN Se0/0/0:23 Q931: RX - DISCONNECT pd = 8  callref = 
0x8082 
Cause i = 0x8281 - Unallocated/unassigned number
Jul 10 03:22:37.233: ISDN Se0/0/0:23 Q931: TX - RELEASE pd = 8  callref = 
0x0082
Jul 10 03:22:37.245: ISDN Se0/0/0:23 Q931: RX - RELEASE_COMP pd = 8  callref = 
0x8082
Jul

[OSL | CCIE_Voice] will phoneview will show srst ip phones for site-b and site-c

2013-07-07 Thread Amit Sharma
Dear All,

I am using proctor labs for my study...
as my all connected all phones for hq, site-b and site-c showing in
phoneview
but when goes in srst for site-b and c i cant see phone working in
phoneview...


how can i fix it?
what is solution to see phone in phoneview in srst case?

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[OSL | CCIE_Voice] how can see phones in SRST mode when using phoneview

2013-07-06 Thread Amit Sharma
Guys,

as i am using phoneview for remote access of ip phones...
when i put site-b  c router in SRST modei cant see working mode these
site phones...


how can i access these phonesand also verify call from these site
phones...SRST mode...?


i can see phone register with SRST router...

but i want to see them in phoneview so that check by call from them...!

can any one tell me if they did on proctor labs?


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[OSL | CCIE_Voice] phoneview in SRST

2013-07-06 Thread Amit Sharma
did anyone use phone view with lab phones to access remotely in case of
SRST?

if anyone sue tell me how can use it?


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[OSL | CCIE_Voice] how use phoneview in srst case?

2013-07-05 Thread Amit Sharma
guys,

can anyone tell me if using proctor labs for ccie voice study?
how can use phoneview to access remote phones in srst case that we apply on
site-b and site c router/

as i applied srst i lost access with these site phones...

if u guys did tell me how can use those phones for call test in case of
srst.../
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[OSL | CCIE_Voice] TCL applied but when calling 4000 call disconnected...

2013-06-30 Thread Amit Sharma
 services dspfarm
!
!
application
  service app-b-acd-aa
  param voice-mail 4200
  paramspace english index 0
  param max-time-call-retry 700
  param service-name callq
  param number-of-hunt-grps 1
  param number-of-hunt-grps 1
  param queue-len 10
  param drop-through-option 1
  paramspace english language en
  param handoff-string aa
  param max-time-vm-retry 2
  param aa-hunt1 4500
  paramspace english location flash:
  param aa-pilot 4000
  param second-greeting-time 60
  param drop-through-prompt _bacd_welcome.au
  param queue-manager-debugs 1
  param call-retry-timer 15
  !
!
!
!
!


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[OSL | CCIE_Voice] IPCC extension not showing in CUCM a nd also not showing rmcm user

2013-06-30 Thread Amit Sharma
guys,

anyone using proctor lab and use this issue?

iPCC extension not showing...

rmcm not showing under cucm?


how can fix it?

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[OSL | CCIE_Voice] SRST issue

2013-06-28 Thread Amit Sharma
Dear guys,


When i applied SRST on site-b and site-c router
i cant see any ephone under router

how can verify it on both routers?


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[OSL | CCIE_Voice] LLQ issue between HQ and BR1

2013-06-28 Thread Amit Sharma
guys see below config...that showing after apply config inter link goes
down...


how can fix it and what is issue?





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Re: [OSL | CCIE_Voice] LLQ issue between HQ and BR1

2013-06-28 Thread Amit Sharma
uys see below config...that showing after apply config inter link goes
down...


how can fix it and what is issue?


SiteA-RTR(config-fr-dlci)#au
SiteA-RTR(config-fr-dlci)#auto qo
SiteA-RTR(config-fr-dlci)#auto qos voi
SiteA-RTR(config-fr-dlci)#auto qos voip tr
SiteA-RTR(config-fr-dlci)#auto qos voip trust
Jun 28 19:26:25.103: %OSPF-5-ADJCHG: Process 1, Nbr 10.10.101.1 on
Serial0/0/1:0.1 from FULL to DOWN, Neighbor Down: Interface down or detached
SiteA-RTR(config-fr-dlci)#auto qos voip trust
%Creating new map-class.
SiteA-RTR(config-fr-dlci)#
SiteA-RTR(config-fr-dlci)#
SiteA-RTR(config-fr-dlci)#
SiteA-RTR(config-fr-dlci)#
SiteA-RTR(config-fr-dlci)#
Jun 28 19:28:05.241: %RMON-5-FALLINGTRAP: Falling trap is generated because
the value of cbQosCMDropBitRate.210.9467521 has fallen below the
falling-threshold value 0
SiteA-RTR(config-fr-dlci


==
teB-RTR(config-subif)#
SiteB-RTR(config-subif)#
SiteB-RTR(config-subif)#fra
SiteB-RTR(config-subif)#frame-relay in
SiteB-RTR(config-subif)#frame-relay int
SiteB-RTR(config-subif)#frame-relay interface-dlci 101
SiteB-RTR(config-fr-dlci)#
Jun 28 19:26:28.739: %OSPF-5-ADJCHG: Process 1, Nbr 10.10.100.1 on
Serial0/0/1:0.1 from FULL to DOWN, Neighbor Down: Dead timer expired
SiteB-RTR(config-fr-dlci)#do sh run int
SiteB-RTR(config-fr-dlci)#do sh run int ser
SiteB-RTR(config-fr-dlci)#do sh run int ser
SiteB-RTR(config-fr-dlci)#do sh run int ser0/0/0:0.1 po
SiteB-RTR(config-fr-dlci)#do sh run int ser0/0/0:0.1 po
sh run int ser0/0/0:0.1 po


SiteA-RTR(config-fr-dlci)#au
SiteA-RTR(config-fr-dlci)#auto qo
SiteA-RTR(config-fr-dlci)#auto qos voi
SiteA-RTR(config-fr-dlci)#auto qos voip tr
SiteA-RTR(config-fr-dlci)#auto qos voip trust
Jun 28 19:26:25.103: %OSPF-5-ADJCHG: Process 1, Nbr 10.10.101.1 on
Serial0/0/1:0.1 from FULL to DOWN, Neighbor Down: Interface down or detached
SiteA-RTR(config-fr-dlci)#auto qos voip trust
%Creating new map-class.
SiteA-RTR(config-fr-dlci)#
SiteA-RTR(config-fr-dlci)#
SiteA-RTR(config-fr-dlci)#
SiteA-RTR(config-fr-dlci)#
SiteA-RTR(config-fr-dlci)#
Jun 28 19:28:05.241: %RMON-5-FALLINGTRAP: Falling trap is generated because
the value of cbQosCMDropBitRate.210.9467521 has fallen below the
falling-threshold value 0
SiteA-RTR(config-fr-dlci


On Fri, Jun 28, 2013 at 6:31 PM, Amit Sharma aryan231...@gmail.com wrote:

 guys see below config...that showing after apply config inter link goes
 down...


 how can fix it and what is issue?





 --
 Thanks  Regard's
 Amit Sharma




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[OSL | CCIE_Voice] guys need help for ipcc extension..

2013-06-27 Thread Amit Sharma
guys...

really hard time...

i applied config...to enable ipcc extension in cucm..\\\

http://www.cisco.com/en/US/products/sw/custcosw/ps1846/products_tech_note09186a008088cff0.shtml

Cisco CallManager 5.x, ICD Extension is known as IPCC Express Extension.
You can update the IPCC Express Extension on Cisco CallManager 5.x when you
run the

 *run sql update processconfig set paramvalue=T *

*
*

*where paramname like ‘IAQInstalledFlag’* command in CLI mode on the
publisher server.

after that i restart cucm servers...when comes back server...i can ssh but
not able to login cucm admin...


but after that getting db error...when login on cucm...

i used this link to fix it...but not working...successful..

https://supportforums.cisco.com/docs/DOC-27232

can someone tell me how fix it?





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[OSL | CCIE_Voice] cucm end user not showing ipcc extension and also not able to find rmcm in cucm

2013-06-26 Thread Amit Sharma
hello guys,

already done cucm and uccx integration but still not able to see ipcc
extension under cucm end user

also not able to see rmcm user in application user

how can fix it and what is issue?


anyone using ipexpert solution with proctor labs please tell me how u
working for this option...?

how u using one button login without these options?



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[OSL | CCIE_Voice] need help guys..

2013-06-26 Thread Amit Sharma
I am having proctor labs...'
when i am working for cucm and uccx integration task..
it is already done by default for us in lab...
but when check in cucm not able to see rmcm user that used in uccx ...

and when i try to add ipcc extension in cucm end user...it is not having
any option to add it...


is it any config issue or i missed something that need to do for fix it?



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Re: [OSL | CCIE_Voice] not able to login cupc

2013-06-23 Thread Amit Sharma
thanks Bill!
i will check and update if it works...

for cue i see that loopback 0 is haivng /32 ip so that service module not
accepting ip
if we use /24 ip then it will worki will test it and confirm you...


thanks for all your support!



On Sun, Jun 23, 2013 at 3:04 PM, Bill Lake whl...@gmail.com wrote:

 Run the CUPS system troubleshooter and it can be very helpful

 If that shows all good try restarting CUPS server

 Third issue is find the fix on Cisco's website at (might want to learn to
 find this page)


 http://www.cisco.com/en/US/products/sw/custcosw/ps1846/products_tech_note09186a008088cff0.shtml

 Change the parm to T so it will show up

 No routing to CUE, not configured correctly, bad password, reboot of cue,
 cue being a pain, OK you get the idea, you have to eliminate each and every
 reason it might not work, also sometimes just restarting CUE when you know
 it is right will help.

 Post your config but usually it would look something like this (this is
 from memory so please if a command is slightly off fix it)

 interface loopback 0
 ip ospf network point-to-point

 int service-engine 0/0
 ip unnumbered loopback 0
 service-module ip address ip 255.255.255.255
 service-module ip default-gateway ip of loopback
 no shut

 ip route ip of CUE 255.255.255.255 service-engine 0/0

 Your IP's will very as will the module type.

 Good luck



 On Sat, Jun 22, 2013 at 10:31 PM, Amit Sharma aryan231...@gmail.comwrote:

 guys.
 i am trying to login cupcbut not successful..
 showing login issue?
 how can fix it?



 second i am not be able to see cucm end users in cupc for desktop
 control...
 what could be the issue?

 third issue is that in ccm end user i dont have option to add ipcc
 extnesion...
 how can see that option to add ipcc extension...for one button login...?



 fourth point is that i configure cue with loopback 1 ip but after
 config...not seen register cti port and cti route point why?

 if using lo 0 it is not accepting under service modulehow can use lo0
 ipr range for service module?


 --
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[OSL | CCIE_Voice] not able to login cupc

2013-06-22 Thread Amit Sharma
guys.
i am trying to login cupcbut not successful..
showing login issue?
how can fix it?



second i am not be able to see cucm end users in cupc for desktop control...
what could be the issue?

third issue is that in ccm end user i dont have option to add ipcc
extnesion...
how can see that option to add ipcc extension...for one button login...?



fourth point is that i configure cue with loopback 1 ip but after
config...not seen register cti port and cti route point why?

if using lo 0 it is not accepting under service modulehow can use lo0
ipr range for service module?


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[OSL | CCIE_Voice] uccx not working

2013-06-19 Thread Amit Sharma
Guys,

when i connect to proactor labs...with initial and final configuration...
most cases, uccx is not coming up...

if it comes up...then try to login..and get login error...


sometimes...seeing uccx account in cucm application user some
time...not...seen uccx in cucm...
is this pod issue or some config issue?

can someone help me how can fix this issue?

if i have no uccx account in cucm,..how can login in uccx?

if connect with podbut cant ping uccxhow can able to wokr on it?
i cant see uccx in cdp of switchthen what can i do to bring it back?



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[OSL | CCIE_Voice] cti route point isnot registering

2013-06-17 Thread Amit Sharma
dear guys
i create the cti route point,..but not registering...
how can register it?

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[OSL | CCIE_Voice] uccx integration with cucm

2013-06-17 Thread Amit Sharma
dear guys,

what required configuration for integrate cucm with uccx..?

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[OSL | CCIE_Voice] ip pblue multilab version..

2013-06-14 Thread Amit Sharma
Hello guys..
i have multi lab version and try to sue with cucm
I want to know how can register multiple ip blue..
I have 4 loop back adapter but all showing same mac address so how can i
will use for register separate phone for this ip blue?

can some one help me?

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[OSL | CCIE_Voice] how can add multiple ipblue from my pc to remote labs

2013-06-09 Thread Amit Sharma
Hello guys...
i tried to use loop back adapter to register ipblue with remote lab but it
is working for 1 phone in 1 time...
can someone tell me how can i register 3 phones with remote lab?
i am using phoneview...but in that i am not be able to call 1 phoen to
another...
can someone help me to fix it?



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Re: [OSL | CCIE_Voice] Correct order for Directory services

2013-06-07 Thread Amit Sharma
when i try to use these steps my cipc showing error..when going to reset/
why is this happening and how can fix it?


On Fri, Jun 7, 2013 at 1:01 PM, Randall Saborio ill2...@gmail.com wrote:

 I mean, here:
 [image: Inline image 1]


 On Fri, Jun 7, 2013 at 3:48 AM, Randall Saborio ill2...@gmail.com wrote:

 Btw, here is screenshot of my phone:
 [image: http://192.168.11.14/CGI/Screenshot]


 On Fri, Jun 7, 2013 at 3:47 AM, Randall Saborio ill2...@gmail.comwrote:

 Hi folks (and Bill),

 Just want to check up with you, because I have tried a few times to do
 the procedure of deleting the Directory Services and then adding them
 again, but when I do this, the services never come up in the original order.

 Check my procedure:
 - Delete the directory services from Phone Services page.
 - Check the Release Notes for CUCM 7.0 and take down the queries for
 enabling back the services:
 Eg:
 run sql insert into telecasterservice
 (pkid,Name,NameASCII,Description,URLTemplate,tkPhoneService,EnterpriseSubscription,Priority)
 values('7eca2cf1-0c8d-4df4-a807-124b18fe89a4','Corporate
 Directory','Corporate Directory','Corporate
 Directory','Application:Cisco/CorporateDirectory',1,'t',6)

 run sql insert into telecasterservice
 (pkid,Name,NameASCII,Description,URLTemplate,tkPhoneService,EnterpriseSubscription,Priority)
 values('d0059763-cdcc-4be7-a2a8-bbd4aac73f63','Missed Calls','Missed
 Calls','Missed Calls','Application:Cisco/MissedCalls',1,'t',1)

 etc

 - Then I modify all the 't' for 'f' and paste into CLI.

 - Result is the phone that should miss some or all of the services shows
 up correctly with missing or No services configured.

 - However the good phone will always show the services in alphabetical
 order, no matter what the priority is set to.

 - Followed the steps on ucguerrilla to no avail:

 http://www.ucguerrilla.com/2012/07/using-sql-to-fix-cucm-enterprise.html#more

 Any suggestions on what could I be doing wrong? Any validation that this
 or other procedure should get the services in proper order?

 Thanks!



 --
 Randall da ill Saborio
 CCIE Voice Wannabe #10054675811




 --
 Randall da ill Saborio
 CCIE Voice Wannabe #10054675811




 --
 Randall da ill Saborio
 CCIE Voice Wannabe #10054675811

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[OSL | CCIE_Voice] uccx login showing unauthorized...

2013-05-31 Thread Amit Sharma
please help how can fix it?
when i login uccx with any browser it giving unauthorized message...


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[OSL | CCIE_Voice] can i install many phones from pc?

2013-04-30 Thread Amit Sharma
AS i am connected for lab over vpn and trying to register soft clients with
cucm...and cme...can i use multiple phones to register?

would be work if i have multiple loopback ports of network in system?
i created 2 loopback of ethernet in system...each for each soft phone..
can vpn adaptor sue with cipc?

please help me if anyone have idea

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[OSL | CCIE_Voice] lab-2a--workbook-1

2013-04-28 Thread Amit Sharma
Hello experts,

i have some confusion...
how can i register my system, cipc, x-lite and blue phone when using all
labs for tests features...as doing now lab-2 then 3...4..5.till end...


what phone would be pstn phone?
please response as i am stuck here...



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[OSL | CCIE_Voice] lab-2of workbook-1

2013-04-28 Thread Amit Sharma
dear experts,

which phone will work for PSTN phone?

How can register x-lite, blue and cipc from my system to lab devices as per
lab question needs?



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[OSL | CCIE_Voice] which phone be PSTN phone

2013-04-27 Thread Amit Sharma
dear guys,

I am working on vol-1, lab-4 and 5...
which one would be PSTN phone..?

how can i select PSTN phone for calling internal phones?



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[OSL | CCIE_Voice] how fix br2 router with PSTN router

2013-04-16 Thread Amit Sharma
i configure PSTN router ip as ntp server in BR2 router but still not
showing fixed

it is not showing synchronized...
how can fixed it?



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Re: [OSL | CCIE_Voice] lab--6 new question some i found here!

2011-11-26 Thread Amit Sharma
HI,
Is there any change in the IPCC script in lab 6

Regards,




 From: andrew margis andrewvoice2...@gmail.com
To: ccie_voice@onlinestudylist.com 
Sent: Friday, 25 November 2011 8:13 PM
Subject: [OSL | CCIE_Voice] lab--6 new question some i found here!
 

Hi,
 
http://www. certknowledge .com/forum/index.php?topic=39.0  
(It's certknowledge(.)com/forum)
 
I m working on early offer solution for sip.
 
Regards 
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Re: [OSL | CCIE_Voice] phone need password

2011-06-11 Thread Amit Sharma
yes dear,,,
this is always come for sip phones

please use cisco always and it is default for sip.

Amit

On Sat, Jun 11, 2011 at 6:16 AM, donny f f.faraday...@gmail.com wrote:

 hi all,

 does any one know why , when I change PHONE setting, by press **# , it ask
 for password ?
 and username is Administrator , cause I never setup one, and this is SIP
 phone

 tks

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