[OSL | CCIE_Voice] UCCX Native codec g729

2013-06-16 Thread CISCO CCIE VOICE
Hi,

How can i verify that UCCX is using G729 codec native

Thnajs
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[OSL | CCIE_Voice] UCCX Native Codec G729

2013-06-16 Thread CISCO CCIE VOICE
Hi,

How can i verify that UCCX is using G729 codec native

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

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


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

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


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

 Hi,

 How can i verify that UCCX is using G729 codec native

 Thnajs



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Re: [OSL | CCIE_Voice] UCCX Native Codec G729

2013-06-16 Thread Somphol Boonjing
Hi,

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

--Somphol.


--Somphol


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

 Hi,

 How can i verify that UCCX is using G729 codec native

 Thanks


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Re: [OSL | CCIE_Voice] UCCX Native Codec G729

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


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

 Hi,

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

 --Somphol.


 --Somphol


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

 Hi,

 How can i verify that UCCX is using G729 codec native

 Thanks


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

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



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Re: [OSL | CCIE_Voice] Ringlist not updated

2013-06-16 Thread Bill
Doubt this is the issue in the lab but you never know.

If the phones are not registered, I doubt that anyone would expect the ring 
list to work


Sent from my iPad

On Jun 15, 2013, at 10:40 PM, nielsenj niels...@gmail.com wrote:

 or
 
 http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note09186a00806d774f.shtml
 
 It is a caching issue with the phone.
 
 Jason Nielsen
 
 On Sat, Jun 15, 2013 at 8:57 PM, Bill Lake whl...@gmail.com wrote:
 So how do you create the Ringlist.xml?  do you also ensure that you use the 
 exact format of the name (not ringlist.xml) and do you confirm that it is 
 correct on the CUCM-PUB-SUB?
 
 So my method for doing this would be to ssh into the CUCM
 
 Type: 
 
 admin:file dump tftp Ringlist.xml
 
 I then capture this, put it into word processor, edit it and save it as 
 Ringlist.xml
 
 I then upload it to both CUCM-PUB/SUB and restart TFTP services
 
 I then verify it is correct on CUCM by typing:
 
 admin:file dump tftp Ringlist.xml
 
 And verify on the phones after a reset.
 
 Bill
 
 
 On Sat, Jun 15, 2013 at 8:09 PM, Karen Johnson karen.johnson...@yahoo.ca 
 wrote:
 
 Yes, to both server and restart tftp. But the phones still show are u 
 there
 
 From: William Calvache williamcalva...@outlook.com; 
 To: Karen Johnson karen.johnson...@yahoo.ca; 
 ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com; 
 Subject: RE: [OSL | CCIE_Voice] Ringlist not updated 
 Sent: Sat, Jun 15, 2013 3:57:29 AM 
 
 Hi
 
 Did you upload the new Ringlist.xml file in both servers?
 
 William Fernando Calvache Gordillo
 Especialista en Comunicaciones Unificadas CISCO SYSTEMS
 CCNA-CCDA-CCVP
 Cel +(57) 318 598 2147
 Bogotá, Colombia
  
  
 Este mensaje y/o sus anexos son para uso exclusivo de su destinatario 
 intencional y puede contener información legalmente protegida por ser 
 privilegiada o confidencial. Si usted no es el destinatario intencional del 
 mensaje por favor infórmenos de inmediato y elimine el mensaje y sus anexos 
 de su computador y sistema de comunicaciones. Igualmente, le comunicamos 
 que cualquier retención, revisión no autorizada, distribución, divulgación, 
 reenvío, copia, impresión, reproducción, o uso indebido de este mensaje y/o 
 sus anexos, está estrictamente prohibida y sancionada legalmente
 
 
 Date: Thu, 13 Jun 2013 09:16:45 -0700
 From: karen.johnson...@yahoo.ca
 To: ccie_voice@onlinestudylist.com
 Subject: [OSL | CCIE_Voice] Ringlist not updated
 
 hi experts,
 
 in my last exam, i have removed the are you there ringlist from TFTP and 
 restart TFTP PUB/SUB. But still it show are you there
 
 - any idea why the TFTP not get updated ?  i check the replication was good 
 at that time.
 
 this never happen when i practice lab.
 
 - what possibilities i need to check ?
 
 k
 
 
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Re: [OSL | CCIE_Voice] UCCX Native codec g729

2013-06-16 Thread Bill
Establish a working call and use your show voice commands?

Sent from my iPad

On Jun 16, 2013, at 6:30 AM, CISCO CCIE VOICE ccievoic...@gmail.com wrote:

 Thanks  for your reply its fine that we have select G729 codec under System 
 Parametes but my question was is there a way  to test which shows UCCX is 
 using g729 codec native 
 
 
 On Sun, Jun 16, 2013 at 1:15 PM, OSL StudyList collaboration.c...@gmail.com 
 wrote:
 Go to CCX admin and look under System and select System Parameters
 —
 Sent from Mailbox for iPad
 
 
 On Sun, Jun 16, 2013 at 3:37 AM, CISCO CCIE VOICE ccievoic...@gmail.com 
 wrote:
 
 Hi,
 
 How can i verify that UCCX is using G729 codec native
 
 Thnajs
 
 ___
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 visit www.ipexpert.com
 
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 www.PlatinumPlacement.com
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Re: [OSL | CCIE_Voice] UCCX Native Codec G729

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


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

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

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

   *Bill Talley*
 UC Systems Consultant

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




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

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

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


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

 Hi,

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

  --Somphol.


 --Somphol


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

  Hi,

  How can i verify that UCCX is using G729 codec native

  Thanks


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 please visit www.ipexpert.com

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 www.PlatinumPlacement.com



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Re: [OSL | CCIE_Voice] UCCX Native Codec G729

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


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

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






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

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

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


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

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

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

   *Bill Talley*
 UC Systems Consultant

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




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

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

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


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

 Hi,

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

  --Somphol.


 --Somphol


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

  Hi,

  How can i verify that UCCX is using G729 codec native

  Thanks


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 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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 visit www.ipexpert.com

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Re: [OSL | CCIE_Voice] UCCX Native Codec G729

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


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

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


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

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

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


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

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






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

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

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


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

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

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

   *Bill Talley*
 UC Systems Consultant

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




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

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

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


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

 Hi,

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

  --Somphol.


 --Somphol


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

  Hi,

  How can i verify that UCCX is using G729 codec native

  Thanks


  ___
 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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 please visit www.ipexpert.com

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 Although this e-mail and any attachments are believed to be free of any
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Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 88, Issue 91

2013-06-16 Thread Justin McIntyre
Pressing the ? Twice will in fact give u the results your looking for.   The 
region setting on UCM will set the codec that u suoport during call setup on 
the phone and the system param settings for UCCX set the codec you support for 
call setup.   Thus as long as you have overlapping or at least one of the same 
codecs supported on both sides of connection the call will be able to setup the 
rtp connection between each.  Pressing ? twice onxe rtp session is established 
gives you the codec that was negotiated.  Keep in mond that the codec will 
change as different prompts or MOH are transgressed between the UCM and UCCX 
server.


Thanks,

Justin McIntyre


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[OSL | CCIE_Voice] Dialing *Extension to reach the person voicemail is not working on v9.1

2013-06-16 Thread Hesham Abdelkereem
Dear Experts,

I'd like to configure on CUCM when I dial *Extension then I can reach the
voicemail of the person directly and says Sorry Extension 1130 is not
available please record ur message

In V7 I just make a CTI Route Point with extension * on the internal
partition then forward all to voicemail then its absoultely working.

I have CUCM and Unity connection v9.1 and when I did that it just telling
enter your pin followed by pound like I am calling the normal voicemail
pilot number
I tried to tweak the forwarding routing rule and direct routing rule but no
chance unfortunately.

Any Ideas

Thank you very much in advance
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[OSL | CCIE_Voice] H323 Gateway for POTS with CUCM issue

2013-06-16 Thread Hesham Abdelkereem
Dear All.

I have configured H323 Gateway to use the 4 FXO ports on the router with
CUCM.
When I call in to the POTS line sometimes its working perfectly and
sometimes when I call in it give me like a faxtone and i hear no voice then
I drop the call and call again it works.
Also , There is a big delay to reach the PLAR number.

Do you have any ideas how to fix that?


Many Thanks,
Hesham
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[OSL | CCIE_Voice] Best Practice to block certain pattern

2013-06-16 Thread Hesham Abdelkereem
Dear All,


I'd like to block 91900 pattern efficient the CUCM.
What's the best and most efficent practice to do that?

Many Thanks,

Hesham
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[OSL | CCIE_Voice] SIP Timers fine tuning

2013-06-16 Thread Hesham Abdelkereem
Dear All,

I have a SIP Circuit to Verizon and when I call out I hear 3 rings first
before the call is actually routed to the PSTN.

Also , I have Automated Attendant and when I dial in to the AA the first 3
seconds are cut from the prompt?

Any Ideas what parameters should I change to fix that.


Thanks,
Hesham
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Re: [OSL | CCIE_Voice] Dialing *Extension to reach the person voicemail is not working on v9.1

2013-06-16 Thread Martin Sloan
Hi Hesham,

Do you have the VMbox mask set on the VM profile to strip the * ?  ie 
to send 1130 intead of *1130.


Marty


On Sun, Jun 16, 2013 at 11:28 AM, Hesham Abdelkereem 
heshamcentr...@gmail.com wrote:

 Dear Experts,

 I'd like to configure on CUCM when I dial *Extension then I can reach the
 voicemail of the person directly and says Sorry Extension 1130 is not
 available please record ur message

 In V7 I just make a CTI Route Point with extension * on the internal
 partition then forward all to voicemail then its absoultely working.

 I have CUCM and Unity connection v9.1 and when I did that it just telling
 enter your pin followed by pound like I am calling the normal voicemail
 pilot number
 I tried to tweak the forwarding routing rule and direct routing rule but
 no chance unfortunately.

 Any Ideas

 Thank you very much in advance

 ___
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 visit www.ipexpert.com

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 www.PlatinumPlacement.com

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Re: [OSL | CCIE_Voice] H323 Gateway for POTS with CUCM issue

2013-06-16 Thread Bill Lake
Do you think sharing your config might help?


On Sun, Jun 16, 2013 at 10:51 AM, Hesham Abdelkereem 
heshamcentr...@gmail.com wrote:

 Dear All.

 I have configured H323 Gateway to use the 4 FXO ports on the router with
 CUCM.
 When I call in to the POTS line sometimes its working perfectly and
 sometimes when I call in it give me like a faxtone and i hear no voice then
 I drop the call and call again it works.
 Also , There is a big delay to reach the PLAR number.

 Do you have any ideas how to fix that?


 Many Thanks,
 Hesham

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Re: [OSL | CCIE_Voice] H323 Gateway for POTS with CUCM issue

2013-06-16 Thread Hesham Abdelkereem
of course

voice service voip
 allow-connections h323 to h323
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
 h323
!
voice class codec 1
 codec preference 1 g711ulaw
 codec preference 2 g729r8
!
voice class h323 1
  h225 timeout tcp establish 3
  h225 timeout setup 3

interface Vlan100
 description ***Voice Vlan***
 ip address VLAN100IP 255.255.255.0
 ip pim dense-mode
 h323-gateway voip interface
 h323-gateway voip bind srcaddr VLAN100IP

voice-port 0/1/0
 no battery-reversal
 no comfort-noise
 connection plar opx 
 caller-id enable
!
voice-port 0/1/1
 no battery-reversal
 no comfort-noise
 connection plar opx 
 caller-id enable
!
voice-port 0/1/2
 no battery-reversal
 no comfort-noise
 connection plar opx 
 caller-id enable
!
voice-port 0/1/3
 no battery-reversal
 no comfort-noise
 connection plar opx 
caller-id enable
!
dial-peer voice 1 pots
 description ** FXO pots dial-peer **
 incoming called-number .
 port 0/1/0
!
dial-peer voice 2 pots
 description ** FXO pots dial-peer **
 incoming called-number .
 port 0/1/1
!
dial-peer voice 3 pots
 description ** FXO pots dial-peer **
 incoming called-number .
 port 0/1/2
!
dial-peer voice 4 pots
 description ** FXO pots dial-peer **
 incoming called-number .
 port 0/1/3
!
dial-peer voice 100 voip
 preference 2
 destination-pattern .T
 session target ipv4:172.30.55.11
 voice-class codec 1
 voice-class h323 1
 dtmf-relay h245-signal h245-alphanumeric
 no vad
!
dial-peer voice 5 pots
 translation-profile outgoing STRIP9
 preference 1
 destination-pattern .T
 port 0/1/0
!
dial-peer voice 6 pots
 translation-profile outgoing STRIP9
 preference 1
 destination-pattern .T
 port 0/1/1
!
dial-peer voice 7 pots
 translation-profile outgoing STRIP9
 preference 1
 destination-pattern .T
 port 0/1/2
!
dial-peer voice 8 pots
 translation-profile outgoing STRIP9
 preference 1
 destination-pattern .T
 port 0/1/3
!
!
!




Thank you very much


On 16 June 2013 09:14, Bill Lake whl...@gmail.com wrote:

 Do you think sharing your config might help?


 On Sun, Jun 16, 2013 at 10:51 AM, Hesham Abdelkereem 
 heshamcentr...@gmail.com wrote:

 Dear All.

 I have configured H323 Gateway to use the 4 FXO ports on the router with
 CUCM.
 When I call in to the POTS line sometimes its working perfectly and
 sometimes when I call in it give me like a faxtone and i hear no voice then
 I drop the call and call again it works.
 Also , There is a big delay to reach the PLAR number.

 Do you have any ideas how to fix that?


 Many Thanks,
 Hesham

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Re: [OSL | CCIE_Voice] Ringlist not updated

2013-06-16 Thread Karen Johnson
hi all,
 
yes i upload to both server and same method as Bill mention here.
and when I file view tftp ringlist.xml i did not see the 'are you there 
anymore.
but in my phones , even after few TFTP and phone restart , i still see the are 
you there
 
 



From: Bill Lake whl...@gmail.com
To: Karen Johnson karen.johnson...@yahoo.ca 
Cc: William Calvache williamcalva...@outlook.com; CCIE Study 
ccie_voice@onlinestudylist.com 
Sent: Saturday, June 15, 2013 7:57:29 PM
Subject: Re: [OSL | CCIE_Voice] Ringlist not updated

So how do you create the Ringlist.xml?  do you also ensure that you use the 
exact format of the name (not ringlist.xml) and do you confirm that it is 
correct on the CUCM-PUB-SUB?

So my method for doing this would be to ssh into the CUCM

Type: admin:file dump tftp Ringlist.xml
I then capture this, put it into word processor, edit it and save it as 
Ringlist.xml

I then upload it to both CUCM-PUB/SUB and restart TFTP services

I then verify it is correct on CUCM by typing:admin:file dump tftp Ringlist.xml
And verify on the phones after a reset.

Bill

On Sat, Jun 15, 2013 at 8:09 PM, Karen Johnson karen.johnson...@yahoo.ca 
wrote: 
Yes, to both server and restart tftp. But the phones still show are u there 


From: William Calvache williamcalva...@outlook.com; To: Karen Johnson 
karen.johnson...@yahoo.ca; ccie_voice@onlinestudylist.com 
ccie_voice@onlinestudylist.com; Subject: RE: [OSL | CCIE_Voice] Ringlist not 
updated Sent: Sat, Jun 15, 2013 3:57:29 AM 
HiDid you upload the new Ringlist.xml file in both servers? 
William Fernando Calvache Gordillo
Especialista en Comunicaciones Unificadas CISCO SYSTEMS
CCNA-CCDA-CCVP
Cel +(57) 318 598 2147

Bogotá, Colombia
 
 
Este mensaje y/o sus anexos son para uso exclusivo de su destinatario 
intencional y puede contener información legalmente protegida por ser 
privilegiada o confidencial. Si usted no es el destinatario intencional del 
mensaje por favor infórmenos de inmediato y elimine el mensaje y sus anexos de 
su computador y sistema de comunicaciones. Igualmente, le comunicamos que 
cualquier retención, revisión no autorizada, distribución, divulgación, 
reenvío, copia, impresión, reproducción, o uso indebido de este mensaje y/o 
sus anexos, está estrictamente prohibida y sancionada legalmente


Date: Thu, 13 Jun 2013 09:16:45 -0700From: karen.johnson...@yahoo.cato: 
ccie_voice@onlinestudylist.comSubject: [OSL | CCIE_Voice] Ringlist not updated 
hi experts,
in my last exam, i have removed the are you there ringlist from TFTP and 
restart TFTP PUB/SUB. But still it show are you there
- any idea why the TFTP not get updated ?  i check the replication was good at 
that time.
this never happen when i practice lab.
- what possibilities i need to check ?

k
___ For more information regarding 
industry leading CCIE Lab training, please visit http://www.ipexpert.com/ Are 
you a CCNP or CCIE and looking for a job? Check out 
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___
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visit http://www.ipexpert.com/

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Re: [OSL | CCIE_Voice] Ringlist not updated

2013-06-16 Thread Karen Johnson
thanks i will try this if replicate the same in my home lab




From: nielsenj niels...@gmail.com
To: Bill Lake whl...@gmail.com 
Cc: Karen Johnson karen.johnson...@yahoo.ca; CCIE Study 
ccie_voice@onlinestudylist.com; William Calvache 
williamcalva...@outlook.com 
Sent: Saturday, June 15, 2013 9:40:38 PM
Subject: Re: [OSL | CCIE_Voice] Ringlist not updated

orhttp://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note09186a00806d774f.shtmlIt
 is a caching issue with the phone.Jason Nielsen 
On Sat, Jun 15, 2013 at 8:57 PM, Bill Lake whl...@gmail.com wrote: 
So how do you create the Ringlist.xml?  do you also ensure that you use the 
exact format of the name (not ringlist.xml) and do you confirm that it is 
correct on the CUCM-PUB-SUB?

So my method for doing this would be to ssh into the CUCM

Type: admin:file dump tftp Ringlist.xml
I then capture this, put it into word processor, edit it and save it as 
Ringlist.xml

I then upload it to both CUCM-PUB/SUB and restart TFTP services

I then verify it is correct on CUCM by typing:admin:file dump tftp Ringlist.xml
And verify on the phones after a reset.

Bill

On Sat, Jun 15, 2013 at 8:09 PM, Karen Johnson karen.johnson...@yahoo.ca 
wrote: 
Yes, to both server and restart tftp. But the phones still show are u there 


From: William Calvache williamcalva...@outlook.com; To: Karen Johnson 
karen.johnson...@yahoo.ca; ccie_voice@onlinestudylist.com 
ccie_voice@onlinestudylist.com; Subject: RE: [OSL | CCIE_Voice] Ringlist 
not updated Sent: Sat, Jun 15, 2013 3:57:29 AM 
HiDid you upload the new Ringlist.xml file in both servers? 
William Fernando Calvache Gordillo
Especialista en Comunicaciones Unificadas CISCO SYSTEMS
CCNA-CCDA-CCVP
Cel +(57) 318 598 2147

Bogotá, Colombia
 
 
Este mensaje y/o sus anexos son para uso exclusivo de su destinatario 
intencional y puede contener información legalmente protegida por ser 
privilegiada o confidencial. Si usted no es el destinatario intencional del 
mensaje por favor infórmenos de inmediato y elimine el mensaje y sus anexos 
de su computador y sistema de comunicaciones. Igualmente, le comunicamos que 
cualquier retención, revisión no autorizada, distribución, divulgación, 
reenvío, copia, impresión, reproducción, o uso indebido de este mensaje y/o 
sus anexos, está estrictamente prohibida y sancionada legalmente


Date: Thu, 13 Jun 2013 09:16:45 -0700From: karen.johnson...@yahoo.cato: 
ccie_voice@onlinestudylist.comSubject: [OSL | CCIE_Voice] Ringlist not 
updated 
hi experts,
in my last exam, i have removed the are you there ringlist from TFTP and 
restart TFTP PUB/SUB. But still it show are you there
- any idea why the TFTP not get updated ?  i check the replication was good 
at that time.
this never happen when i practice lab.
- what possibilities i need to check ?

k
___ For more information 
regarding industry leading CCIE Lab training, please visit 
http://www.ipexpert.com/ Are you a CCNP or CCIE and looking for a job? Check 
out http://www.platinumplacement.com/ 
___
For more information regarding industry leading CCIE Lab training, please 
visit http://www.ipexpert.com/

Are you a CCNP or CCIE and looking for a job? Check out 
http://www.platinumplacement.com/

___
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visit http://www.ipexpert.com/

Are you a CCNP or CCIE and looking for a job? Check out 
http://www.platinumplacement.com/
___
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[OSL | CCIE_Voice] Change B-Channel Maintenance Status

2013-06-16 Thread Karen Johnson
when do MGCP, i come across service parameter Change B-Channel Maintenance 
Status and they only have 5 box.
 
- does that mean in one cluster we can have 5 GW that use partial PRI with MGCP 
?
- if i plan to use partial PRI in MGCP,   do i need to also busy out  above 
paramater or 
  pri-group  timeslots 1-12,16 service mgcp  command in GW is enough?
 
- based on UCM help page, we only need use busy out if (Inhibit re-starts at 
PRI initialization is checked)
  I just try to make sense of this statement
 
tks
K.
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Re: [OSL | CCIE_Voice] Change B-Channel Maintenance Status

2013-06-16 Thread Bill Lake
I would do the config manually on the gateway and not worry about this in the 
real world

Sent from my iPhone

On Jun 16, 2013, at 4:25 PM, Karen Johnson karen.johnson...@yahoo.ca wrote:

 when do MGCP, i come across service parameter Change B-Channel Maintenance 
 Status and they only have 5 box.
  
 - does that mean in one cluster we can have 5 GW that use partial PRI with 
 MGCP ?
 - if i plan to use partial PRI in MGCP,   do i need to also busy out  above 
 paramater or
   pri-group  timeslots 1-12,16 service mgcp  command in GW is enough?
  
 - based on UCM help page, we only need use busy out if (Inhibit re-starts 
 at PRI initialization is checked)
   I just try to make sense of this statement
  
 tks
 K.
  
 ___
 For more information regarding industry leading CCIE Lab training, please 
 visit www.ipexpert.com
 
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 www.PlatinumPlacement.com
___
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Re: [OSL | CCIE_Voice] Ringlist not updated

2013-06-16 Thread Rrcrumm
Do a debug tftp server and see if you r getting the file to the phone

Ex

On Jun 16, 2013, at 2:10 PM, Karen Johnson karen.johnson...@yahoo.ca wrote:

 hi all,
  
 yes i upload to both server and same method as Bill mention here.
 and when I file view tftp ringlist.xml i did not see the 'are you there 
 anymore.
 but in my phones , even after few TFTP and phone restart , i still see the 
 are you there
  
  
 
 From: Bill Lake whl...@gmail.com
 To: Karen Johnson karen.johnson...@yahoo.ca 
 Cc: William Calvache williamcalva...@outlook.com; CCIE Study 
 ccie_voice@onlinestudylist.com 
 Sent: Saturday, June 15, 2013 7:57:29 PM
 Subject: Re: [OSL | CCIE_Voice] Ringlist not updated
 So how do you create the Ringlist.xml?  do you also ensure that you use the 
 exact format of the name (not ringlist.xml) and do you confirm that it is 
 correct on the CUCM-PUB-SUB?
 So my method for doing this would be to ssh into the CUCM
 Type: admin:file dump tftp Ringlist.xml
 I then capture this, put it into word processor, edit it and save it as 
 Ringlist.xml
 I then upload it to both CUCM-PUB/SUB and restart TFTP services
 I then verify it is correct on CUCM by typing:admin:file dump tftp 
 Ringlist.xml
 And verify on the phones after a reset.
 Bill
 On Sat, Jun 15, 2013 at 8:09 PM, Karen Johnson karen.johnson...@yahoo.ca 
 wrote:
 Yes, to both server and restart tftp. But the phones still show are u there
 From: William Calvache williamcalva...@outlook.com; To: Karen Johnson 
 karen.johnson...@yahoo.ca; ccie_voice@onlinestudylist.com 
 ccie_voice@onlinestudylist.com; Subject: RE: [OSL | CCIE_Voice] Ringlist 
 not updated Sent: Sat, Jun 15, 2013 3:57:29 AM
 HiDid you upload the new Ringlist.xml file in both servers?
 William Fernando Calvache Gordillo
 Especialista en Comunicaciones Unificadas CISCO SYSTEMS
 CCNA-CCDA-CCVP
 Cel +(57) 318 598 2147
 Bogotá, Colombia
  
  
 Este mensaje y/o sus anexos son para uso exclusivo de su destinatario 
 intencional y puede contener información legalmente protegida por ser 
 privilegiada o confidencial. Si usted no es el destinatario intencional del 
 mensaje por favor infórmenos de inmediato y elimine el mensaje y sus anexos 
 de su computador y sistema de comunicaciones. Igualmente, le comunicamos que 
 cualquier retención, revisión no autorizada, distribución, divulgación, 
 reenvío, copia, impresión, reproducción, o uso indebido de este mensaje y/o 
 sus anexos, está estrictamente prohibida y sancionada legalmente
 Date: Thu, 13 Jun 2013 09:16:45 -0700From: karen.johnson...@yahoo.cato: 
 ccie_voice@onlinestudylist.comSubject: [OSL | CCIE_Voice] Ringlist not updated
 hi experts,
 in my last exam, i have removed the are you there ringlist from TFTP and 
 restart TFTP PUB/SUB. But still it show are you there
 - any idea why the TFTP not get updated ?  i check the replication was good 
 at that time.
 this never happen when i practice lab.
 - what possibilities i need to check ?
 k
 ___ For more information 
 regarding industry leading CCIE Lab training, please visit 
 http://www.ipexpert.com/ Are you a CCNP or CCIE and looking for a job? Check 
 out http://www.platinumplacement.com/
 
 ___
 For more information regarding industry leading CCIE Lab training, please 
 visit http://www.ipexpert.com/
 
 Are you a CCNP or CCIE and looking for a job? Check out 
 http://www.platinumplacement.com/
 ___
 For more information regarding industry leading CCIE Lab training, please 
 visit www.ipexpert.com
 
 Are you a CCNP or CCIE and looking for a job? Check out 
 www.PlatinumPlacement.com
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

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

Re: [OSL | CCIE_Voice] Change B-Channel Maintenance Status

2013-06-16 Thread Robert Thomas
Karen,

Those parameters are old story. The were used back in the 6608 days, have
no use on the current MGCP GW. If you configure your PRI as fractional with
less timeslots, CUCM would audit the port and the GW report back an error
for that endpoint, CUCM will then mark it out of service automatically.



On Sun, Jun 16, 2013 at 2:25 PM, Karen Johnson karen.johnson...@yahoo.cawrote:

 when do MGCP, i come across service parameter Change B-Channel
 Maintenance Status and they only have 5 box.

 - does that mean in one cluster we can have 5 GW that use partial PRI with
 MGCP ?
 - if i plan to use partial PRI in MGCP,   do i need to also busy out 
 above paramater or
   pri-group  timeslots 1-12,16 service mgcp  command in GW is enough?

 - based on UCM help page, we only need use busy out if (Inhibit
 re-starts at PRI initialization is checked)
   I just try to make sense of this statement

 tks
 K.


 ___
 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




-- 
Robert Thomas Zamora
tho...@gmail.com +50689389544
http://cr.linkedin.com/pub/robert-thomas-zamora/29/913/8a8
CCNP, CCNP Voice
___
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Re: [OSL | CCIE_Voice] SIP Timers fine tuning

2013-06-16 Thread Robert Thomas
You should look into Early offer and Early media. Perhaps you might need
PRACK enabled, to cut throught the audio before the call connects. Usually
your Telco can give the specific requirements you need.


On Sun, Jun 16, 2013 at 8:53 AM, Hesham Abdelkereem 
heshamcentr...@gmail.com wrote:

 Dear All,

 I have a SIP Circuit to Verizon and when I call out I hear 3 rings first
 before the call is actually routed to the PSTN.

 Also , I have Automated Attendant and when I dial in to the AA the first 3
 seconds are cut from the prompt?

 Any Ideas what parameters should I change to fix that.


 Thanks,
 Hesham

 ___
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 visit www.ipexpert.com

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 www.PlatinumPlacement.com




-- 
Robert Thomas Zamora
tho...@gmail.com +50689389544
http://cr.linkedin.com/pub/robert-thomas-zamora/29/913/8a8
CCNP, CCNP Voice
___
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Re: [OSL | CCIE_Voice] Best Practice to block certain pattern

2013-06-16 Thread Robert Thomas
Specific translation pattern with Block this Pattern.


On Sun, Jun 16, 2013 at 8:51 AM, Hesham Abdelkereem 
heshamcentr...@gmail.com wrote:

 Dear All,


 I'd like to block 91900 pattern efficient the CUCM.
 What's the best and most efficent practice to do that?

 Many Thanks,

 Hesham

 ___
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 visit www.ipexpert.com

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 www.PlatinumPlacement.com




-- 
Robert Thomas Zamora
tho...@gmail.com +50689389544
http://cr.linkedin.com/pub/robert-thomas-zamora/29/913/8a8
CCNP, CCNP Voice
___
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Re: [OSL | CCIE_Voice] Correct order for Directory services

2013-06-16 Thread Brian Meade
 
  Hope this helps
 
  Bill
 
  On Fri, Jun 7, 2013 at 5:01 AM, 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.com
 wrote:
 
  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
 
  ___
  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
 
 
 
 
 
  --
  Randall da ill Saborio
  CCIE Voice Wannabe #10054675811
 
  ___
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  please visit www.ipexpert.com
 
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  www.PlatinumPlacement.com
 
 
 
 
  ___
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  visit www.ipexpert.com
 
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  www.PlatinumPlacement.com
 



 --
 Robert Thomas Zamora
 tho...@gmail.com +50689389544
 http://cr.linkedin.com/pub/robert-thomas-zamora/29/913/8a8
 CCNP, CCNP Voice
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 End of CCIE_Voice Digest, Vol 88, Issue 98
 **

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