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

2013-08-01 Thread Bill
Could be hitting this

CSCsd14203

Symptom : BACD AA script crashes while trying to play Music on Hold (MoH) 
Conditions : This happens with 12.4 (4) XC and 12.4 (4)T images.

Workaround : Configure Live MoH. Even if you don’t have a livefeed source, the 
script will failover to the MoH file in flash.

The sample below has ephone-dn config for “dummy” MoH livefeed:

ephone-dn 20 number dummy-number moh out-call BCDA! dial-peer voice 7 pots 
destination-pattern BCDA!

Also enable debug ephone moh. Having dummy MoH Livefeed allows the MoH server 
to get started.



Try configuring the live source




Sent from my iPad

On Aug 1, 2013, at 12:17 AM, Amit Sharma aryan231...@gmail.com wrote:

 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
 !
 
 
 -- 
 Thanks  Regard's
 Amit Sharma
 
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Re: [OSL | CCIE_Voice] CAC RSVP Not enough bandwidth tip

2013-08-01 Thread Amdo Ngawa
Suresh Bhandari's link is helpful

Check if this helps:
http://sureshbhandari.com.np/rsvp-configuration-and-errors/

Amdo


On Tue, Jul 30, 2013 at 12:31 AM, ie ravindra ieravin...@gmail.com wrote:

 Hi Amdo,

 I have faced the same thing when I am in the real lab all local site calls
 went failed. But every thing correct. I can't understand the logic behind
 of that. Let me explain some workaround I did.

 I changed the location of Site C to Hub none in the lab in my lab day.
 then local calls goes out without issue. so try like this. create different
 location for Site A , Site B , and Site C. other than hub_none and Assign
 it to respective sites and assign required bandwidth. Then let me know what
 happens.  I guess the problem is with hub none location. But let me know
 the results.

 Thanks,
 Ravi,


 On Mon, Jul 29, 2013 at 7:13 PM, Amdo Ngawa datapack...@gmail.com wrote:

 Hi Folks:

 I had been having trouble with CAC RSVP lately. Once I configured CAC
 RSVP between two sites, the calls between the two sides failed with error
 *Not enough bandwidth'*

 Checked my configuration thoroughly. Locations CAC configured; mrg and
 mrgls proper, mtp profiles fine; sccp and dspfarm created and registered
 with CUCM; all good to go.
 Found suggestions online, but most talked about proper configuration
 which I had already done. However, there was one quirky suggestion that I
 tried:
 Assuming CAC RSVP is between HQ and other site, I removed the hq-rsvp-mtp
 from the mrg-hq and inserted a default sub or pub (mtp_2,mtp_1) and saved.
 Thereafter I removed the default mtp from mrg-hq and reinserted the
 hq-mtp-rsvp and saved the changes and reset. For good measure, I clicked
 the resync-bandwidth for for both sites in the Location section
 (SettingsLocation)
 The calls started to work. show ip rsvp reservation and sh ip rsvp int
 showed output that I wanted.

 Thanks,

 Amdo

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

2013-08-01 Thread Drake J
Hello All,

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

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

2013-08-01 Thread Amdo Ngawa
I am having similar issue. No resolution so far? Will read and google.

Thanks


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

 HI experts,

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

 Thnks






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

2013-08-01 Thread Ramcharan Arya
Hi,

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

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



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

 Hello All,

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

 -Drake

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

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

___
For more information regarding industry leading CCIE Lab training, please visit 
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Re: [OSL | CCIE_Voice] NO Extension in CME-SRST

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

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



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

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

 Thanks


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

 HI experts,

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

 Thnks






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

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



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

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

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

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