[cisco-voip] Command for determining which dial-peer will be used on router?

Tony Fairhurst Tony.Fairhurst at kcom.com
Wed Feb 3 16:10:03 EST 2010


On an active call use 'show voice call status' this will show you which
dial peer was used.

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
cisco-voip-request at puck.nether.net
Sent: 03 February 2010 17:00
To: cisco-voip at puck.nether.net
Subject: cisco-voip Digest, Vol 76, Issue 3

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Today's Topics:

   1. Re: Command for determining which dial-peer will be used on
      router? (Robert Kulagowski)
   2. Re: Command for determining which dial-peer will be	usedon
      router? (UNCLASSIFIED) (Girard, Jeffrey COL MIL USA)
   3. Re: Command for determining which dial-peer will be used on
      router? (Lelio Fulgenzi)
   4. Cisco AXP-XMediusFAX? (Manoj Kalpage)
   5. Allocation of Speak Time to Users (mark baker)
   6. MOH Beeping (Kevin Dunn)
   7. Re: MOH Beeping (Peter Slow)
   8. Re: Command for determining which dial-peer will be	used on
      router? (Peter Slow)
   9. Re: Command for determining which dial-peer will be	used on
      router? (Peter Slow)
  10. Re: cisco 7941 switch port physical problem (Peter Slow)
  11. Re: Command for determining which dial-peer will be used on
      router? (Lelio Fulgenzi)
  12. Re: What is wrong with my destination pattern? (Peter Slow)
  13. Re: MOH Beeping (Kevin Dunn)
  14. Re: MOH Beeping (Kevin Dunn)
  15. Re: Allocation of Speak Time to Users (mark baker)
  16. Re: What is wrong with my destination pattern? (John Lange)
  17. Re: Unity 5.0 in the AD 2008 R2 (Ed Leatherman)
  18. Re: Allocation of Speak Time to Users (Mike Thompson)
  19. Re: Unity 5.0 in the AD 2008 R2 (Scott Kee)
  20. Re: What is wrong with my destination pattern? (Reynaldo Casta?o)
  21. Re: Allocation of Speak Time to Users (cips)
  22. Re: Allocation of Speak Time to Users (Ryan Ratliff)
  23. Re: Cisco AXP-XMediusFAX? (Nick Matthews)
  24. Re: MOH Beeping (Peter Slow)
  25. Question about CRS and using the Editor for queue	scripts
      (Terry Oakley)
  26. Re: Allocation of Speak Time to Users (Wes Sisk)
  27. Re: Allocation of Speak Time to Users (Wes Sisk)
  28. Re: MOH Beeping (Wes Sisk)
  29. Re: Question about CRS and using the Editor for queue	scripts
      (Ed Leatherman)
  30. Internaltional Calls Failing at the Bell PRI (John Lange)
  31. Re: MOH Beeping (Kevin Dunn)
  32. Re: Internaltional Calls Failing at the Bell PRI
      (Cristobal Priego)
  33. CFA CSS Activation Policy (Leslie Meade)
  34. OT/FYI: Using DBAN to wipe data on HP DL380s (Lelio Fulgenzi)
  35. Re: Internaltional Calls Failing at the Bell PRI (John Lange)
  36. Re: Internaltional Calls Failing at the Bell PRI
      (Cristobal Priego)
  37. UCM and UCCX directory synchronization (Matthew Linsemier)
  38. Re: Internaltional Calls Failing at the Bell PRI (Go0se)
  39. Re: MOH Beeping (Norton, Mike)
  40. Re: UCM and UCCX directory synchronization (Fuermann, Jason)
  41. Re: Internaltional Calls Failing at the Bell PRI (John Lange)
  42. Re: MOH Beeping (Kevin Dunn)
  43. Re: MOH Beeping (Wes Sisk)
  44. Re: Internaltional Calls Failing at the Bell PRI - SOLVED
      (John Lange)
  45. Re: Internaltional Calls Failing at the Bell PRI (Go0se)
  46. Re: cisco 7941 switch port physical problem (Lawrence E. Bakst)
  47. Re: Agent Status in UCCX (shary shary)
  48. Re: Agent Status in UCCX (Tanner Ezell)
  49. Re: cisco 7941 switch port physical problem (Abebe Amare)
  50. Re: cisco 7941 switch port physical problem (Abebe Amare)
  51. JTAPI user disappeared? (Ali El Moussaoui)
  52. Re: Command for determining which dial-peer will be	usedon
      router? (UNCLASSIFIED) (Dew Swen)
  53. Re: Command for determining which dial-peer will be	usedon
      router? (UNCLASSIFIED) (haroon rasheed)
  54. MGCP to PSTN - 0x80A9 - Temporary failure? (Stephen Greszczyszyn)
  55. DMA Validation Errors - DMA 7.1.3 (Aliberto, Nate)
  56. Re: Agent Status in UCCX (Tanner Ezell)
  57. Re: MGCP to PSTN - 0x80A9 - Temporary failure? (Ryan Ratliff)
  58. Re: MOH Beeping (Peter Slow)
  59. Re: DMA Validation Errors - DMA 7.1.3 (Ryan Ratliff)
  60. HuntPilot-Line Group - Phone Forwarded (Carter, Bill)
  61. Re: HuntPilot-Line Group - Phone Forwarded (Ryan Ratliff)


----------------------------------------------------------------------

Message: 1
Date: Tue, 02 Feb 2010 11:10:11 -0600
From: Robert Kulagowski <rkulagow at gmail.com>
To: Ryan Ratliff <rratliff at cisco.com>
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Command for determining which dial-peer will
	be used on router?
Message-ID: <4B685C73.8070601 at gmail.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed

Ryan Ratliff wrote:
> Try 'show dialplan number'.

That works, but it's not matching the "8T" dialpeers that I've got, only

those that have matchable digits, like 81312.......

But, the link that Wes sent has some additional information, and it
appears 
that "timeout" is the additional parameter I need.

Thanks!

BTW:  It sure would be nice if things were grouped together better;
you've 
got "show dialplan" on one hand and "test voice translation-pattern" on
the 
other.  Other than one group of people not talking to another group, is 
there a good reason why they're not both under either "show" or "test"?


------------------------------

Message: 2
Date: Tue, 2 Feb 2010 10:33:54 -0700
From: "Girard, Jeffrey COL MIL USA" <jeffrey.girard at us.army.mil>
To: "Robert Kulagowski" <rkulagow at gmail.com>,
	<cisco-voip at puck.nether.net>
Subject: Re: [cisco-voip] Command for determining which dial-peer will
	be	usedon router? (UNCLASSIFIED)
Message-ID:
	
<0B08C95FF256E9418A7229385AE0BA90D24BA2 at blis100be000101.nasw.ds.army.mil
>
	
Content-Type: text/plain;	charset="us-ascii"

Classification: UNCLASSIFIED
Caveats: FOUO

Show voice dialpeer

-----------------------------------------------------------
Jeffrey T. Girard ("Jeff")
COL, 53
Future Forces Integration Directorate (FFID), Deputy - Networks
office:  (915)568-1240  DSN 978
Mobile:  (915)727-4222
reply to:  jeffrey.girard at us.army.mil


-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Robert
Kulagowski
Sent: Tuesday, February 02, 2010 9:41 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Command for determining which dial-peer will be
usedon router?

I thought I saw in this list that there's a way to have the router show
you 
which dial-peer is going to be used based on a particular number
pattern. 
Sort of what you see when doing a "deb voip ccapi inout" when it shows
you 
the list of possible "outgoing dial peer=" values.

I'm not trying to test the voice translation-pattern stuff, but the
actual 
DP candidates.

Thanks.
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip
Classification: UNCLASSIFIED
Caveats: FOUO




------------------------------

Message: 3
Date: Tue, 2 Feb 2010 12:37:27 -0500 (EST)
From: Lelio Fulgenzi <lelio at uoguelph.ca>
To: Robert Kulagowski <rkulagow at gmail.com>
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Command for determining which dial-peer will
	be used on router?
Message-ID:
	
<1972444661.9132711265132246994.JavaMail.root at simcoe.cs.uoguelph.ca>
Content-Type: text/plain; charset="utf-8"

i like using the csim tool as well. 

--- 
Lelio Fulgenzi, B.A. 
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1 
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN) 
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
Cooking with unix is easy. You just sed it and forget it. 
- LFJ (with apologies to Mr. Popeil) 


----- Original Message ----- 
From: "Robert Kulagowski" <rkulagow at gmail.com> 
To: "Ryan Ratliff" <rratliff at cisco.com> 
Cc: cisco-voip at puck.nether.net 
Sent: Tuesday, February 2, 2010 12:10:11 PM GMT -05:00 US/Canada Eastern

Subject: Re: [cisco-voip] Command for determining which dial-peer will
be used on router? 

Ryan Ratliff wrote: 
> Try 'show dialplan number'. 

That works, but it's not matching the "8T" dialpeers that I've got, only

those that have matchable digits, like 81312....... 

But, the link that Wes sent has some additional information, and it
appears 
that "timeout" is the additional parameter I need. 

Thanks! 

BTW: It sure would be nice if things were grouped together better;
you've 
got "show dialplan" on one hand and "test voice translation-pattern" on
the 
other. Other than one group of people not talking to another group, is 
there a good reason why they're not both under either "show" or "test"? 
_______________________________________________ 
cisco-voip mailing list 
cisco-voip at puck.nether.net 
https://puck.nether.net/mailman/listinfo/cisco-voip 
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Message: 4
Date: Wed, 3 Feb 2010 02:42:48 +0900
From: Manoj Kalpage <manoj.kalpage at gmail.com>
To: Cisco Voip <cisco-voip at puck.nether.net>
Subject: [cisco-voip] Cisco AXP-XMediusFAX?
Message-ID:
	<bbe09c1d1002020942j544c95ffk87c5d1253a1970c0 at mail.gmail.com>
Content-Type: text/plain; charset="windows-1252"

Hi All,

We are using cisco T.38 IOS fax gateway for fax to e-mail solution.
There
are some issue like page truncated at the end. This happens specially
when
receive long fax. We are looking for alternative solution. I am
wondering how Cisco AXP-XMediusFAX? ?
Any comments would be appreciated.
http://www.cisco.com/en/US/prod/routers/ps9701/axp_promo.html

Thanks,
MK
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Message: 5
Date: Tue, 2 Feb 2010 17:21:07 +0000
From: mark baker <mb at c2ukltd.com>
To: "cisco-voip at puck.nether.net" <cisco-voip at puck.nether.net>
Subject: [cisco-voip] Allocation of Speak Time to Users
Message-ID: <C78E0F83.1C26%mb at c2ukltd.com>
Content-Type: text/plain; charset="us-ascii"

Hi Folks,

Is anybody aware of a mechanism in CCM to allow users to only dial
international numbers based on individual Pin Codes? Then on entry to
only
have access to the line for a predetermined amount of time set by the
CCM
admin?

Any advice would be greatly appreciated.

Kind Regards,

Mark Baker



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Message: 6
Date: Tue, 2 Feb 2010 11:53:17 -0600
From: Kevin Dunn <cheesevoice at gmail.com>
To: Cisco Voice <cisco-voip at puck.nether.net>
Subject: [cisco-voip] MOH Beeping
Message-ID:
	<84b74aec1002020953g5b7aa92cwa3f5459f3f97b99f at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"

Okay I have a TAC case open and I have tried changing configuration
settings, cables and hardware...

CUCM 7.0.2.2000-5
Fixed audio from XM radio (MOH-USB-AUDIO) card

when a caller is placed on hold and the speaker is activated there is an
audible (and quite annoying) beeping sound playing over the top of the
audio
file.

It is not audible on the handset or headset.

If I sniff the phone port and capture the audio file I can hear it.

If I record the audio file with my laptop (plugging mic cord into lappy
instead of MOH-USB) there is no beeping.
In my mind that eliminates the XM radio and cables, I have changed out
cables though, just in case.

I changes out MOH-USB cards and that also did nothing to eliminate the
issue.

I have upgraded firmware on the phones and that wasn't it either.
With the Sample Audio file (which is JAZZY) there is no beeping
regardless
of handset or speaker.

Any suggestions?
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Message: 7
Date: Tue, 2 Feb 2010 13:11:45 -0500
From: Peter Slow <peter.slow at gmail.com>
To: Kevin Dunn <cheesevoice at gmail.com>
Cc: Cisco Voice <cisco-voip at puck.nether.net>
Subject: Re: [cisco-voip] MOH Beeping
Message-ID:
	<53fc16d41002021011i577c0544wf0a1a40c2d883b18 at mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1

Kevin,
   Cool! Let's fix it =) A. what happens when you plug the USB audio
device into your laptop, instead of going directly into it via the mic
cord? ..and B. can i see the packet capture or hear the resulting
audio?

-Peter

On Tue, Feb 2, 2010 at 12:53 PM, Kevin Dunn <cheesevoice at gmail.com>
wrote:
> Okay I have a TAC case open and I have tried changing configuration
> settings, cables and hardware...
>
> CUCM 7.0.2.2000-5
> Fixed audio from XM radio (MOH-USB-AUDIO) card
>
> when a caller is placed on hold and the speaker is activated there is
an
> audible (and quite annoying) beeping sound playing over the top of the
audio
> file.
>
> It is not audible on the handset or headset.
>
> If I sniff the phone port and capture the audio file I can hear it.
>
> If I record the audio file with my laptop (plugging mic cord into
lappy
> instead of MOH-USB) there is no beeping.
> In my mind that eliminates the XM radio and cables, I have changed out
> cables though, just in case.
>
> I changes out MOH-USB cards and that also did nothing to eliminate the
> issue.
>
> I have upgraded firmware on the phones and that wasn't it either.
> With the Sample Audio file (which is JAZZY) there is no beeping
regardless
> of handset or speaker.
>
> Any suggestions?
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>


------------------------------

Message: 8
Date: Tue, 2 Feb 2010 13:16:04 -0500
From: Peter Slow <peter.slow at gmail.com>
To: Robert Kulagowski <rkulagow at gmail.com>
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Command for determining which dial-peer will
	be	used on router?
Message-ID:
	<53fc16d41002021016m6c8d6649s2c33fc1072675a73 at mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1

Robert,
  Show dialplan number, as Ryan  had said, is definitely the right
command. If it doesn't return your 8T dialpeer, that means it's
matching something else over it.
any destination pattern that matches more than one digit is
(basically) always going to get chosen over your 8T dialpeer.

-Peter

On Tue, Feb 2, 2010 at 12:10 PM, Robert Kulagowski <rkulagow at gmail.com>
wrote:
> Ryan Ratliff wrote:
>>
>> Try 'show dialplan number'.
>
> That works, but it's not matching the "8T" dialpeers that I've got,
only
> those that have matchable digits, like 81312.......
>
> But, the link that Wes sent has some additional information, and it
appears
> that "timeout" is the additional parameter I need.
>
> Thanks!
>
> BTW: ?It sure would be nice if things were grouped together better;
you've
> got "show dialplan" on one hand and "test voice translation-pattern"
on the
> other. ?Other than one group of people not talking to another group,
is
> there a good reason why they're not both under either "show" or
"test"?
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>


------------------------------

Message: 9
Date: Tue, 2 Feb 2010 13:17:38 -0500
From: Peter Slow <peter.slow at gmail.com>
To: Lelio Fulgenzi <lelio at uoguelph.ca>
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Command for determining which dial-peer will
	be	used on router?
Message-ID:
	<53fc16d41002021017jc7a8bc3l3b0d321b0cb58221 at mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1

csim IS awesome, but anyone using it in production should be aware
that it can occasionally crash your router and is totally unsupported.
be careful with it.

-Pete

On Tue, Feb 2, 2010 at 12:37 PM, Lelio Fulgenzi <lelio at uoguelph.ca>
wrote:
> i like using the csim tool as well.
>
> ---
> Lelio Fulgenzi, B.A.
> Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
> (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
> ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
> Cooking with unix is easy. You just sed it and forget it.
> ?? ? ? ? ? ? ? ? ? ? ? ? ? ? ?- LFJ (with apologies to Mr. Popeil)
>
>
> ----- Original Message -----
> From: "Robert Kulagowski" <rkulagow at gmail.com>
> To: "Ryan Ratliff" <rratliff at cisco.com>
> Cc: cisco-voip at puck.nether.net
> Sent: Tuesday, February 2, 2010 12:10:11 PM GMT -05:00 US/Canada
Eastern
> Subject: Re: [cisco-voip] Command for determining which dial-peer will
be
> used on router?
>
> Ryan Ratliff wrote:
>> Try 'show dialplan number'.
>
> That works, but it's not matching the "8T" dialpeers that I've got,
only
> those that have matchable digits, like 81312.......
>
> But, the link that Wes sent has some additional information, and it
appears
> that "timeout" is the additional parameter I need.
>
> Thanks!
>
> BTW: ?It sure would be nice if things were grouped together better;
you've
> got "show dialplan" on one hand and "test voice translation-pattern"
on the
> other. ?Other than one group of people not talking to another group,
is
> there a good reason why they're not both under either "show" or
"test"?
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>


------------------------------

Message: 10
Date: Tue, 2 Feb 2010 13:19:32 -0500
From: Peter Slow <peter.slow at gmail.com>
To: Abebe Amare <abucho at gmail.com>
Cc: cisco voip <cisco-voip at puck.nether.net>
Subject: Re: [cisco-voip] cisco 7941 switch port physical problem
Message-ID:
	<53fc16d41002021019h4e04f21et8a5b0a4afe7bedad at mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1

If you connect a different phone of the same type to the same port
with the same cable, does it turn on?

if yes, then RMA the first phone.

On Tue, Feb 2, 2010 at 9:43 AM, Abebe Amare <abucho at gmail.com> wrote:
> Hi,
>
> I am having problem with a new out of the box Cisco 7941 IP phone.
When I
> connect the switch port to a port on the PoE switch with a patch cord
it is
> not powering up. If I hold the cable tightly in place with my hand or
put
> something to hold the cable in place it will power up. I have changed
the
> cable several times but it won't work unless I hold it firmly. Does
this
> mean the switch port on the phone is damaged? do I need to process
RMA?
>
> Thanks in advance.
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>


------------------------------

Message: 11
Date: Tue, 2 Feb 2010 13:21:19 -0500 (EST)
From: Lelio Fulgenzi <lelio at uoguelph.ca>
To: Peter Slow <peter.slow at gmail.com>
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Command for determining which dial-peer will
	be used on router?
Message-ID:
	
<1066594105.9155381265134879639.JavaMail.root at simcoe.cs.uoguelph.ca>
Content-Type: text/plain; charset="utf-8"

ah! like the old "show ip mroute" command on the 7206VXR 12T train! 



lelio : hmmm, how do i find out my multicast routes? 
isp : do a "show ip mroute" 
lelio : ok. 
<pause> 
lelio : um, my router rebooted. 
isp : really? show commands shouldn't do that. it must have been
something else. try again. 
lelio : ok. 
<pause> 
lelio : um, my router rebooted again. 
isp : oh. ok, don't type that command in again. call Cisco TAC. 




----- Original Message ----- 
From: "Peter Slow" <peter.slow at gmail.com> 
To: "Lelio Fulgenzi" <lelio at uoguelph.ca> 
Cc: "Robert Kulagowski" <rkulagow at gmail.com>, cisco-voip at puck.nether.net

Sent: Tuesday, February 2, 2010 1:17:38 PM GMT -05:00 US/Canada Eastern 
Subject: Re: [cisco-voip] Command for determining which dial-peer will
be used on router? 

csim IS awesome, but anyone using it in production should be aware 
that it can occasionally crash your router and is totally unsupported. 
be careful with it. 

-Pete 

On Tue, Feb 2, 2010 at 12:37 PM, Lelio Fulgenzi <lelio at uoguelph.ca>
wrote: 
> i like using the csim tool as well. 
> 
> --- 
> Lelio Fulgenzi, B.A. 
> Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1 
> (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN) 
> ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^

> Cooking with unix is easy. You just sed it and forget it. 
> - LFJ (with apologies to Mr. Popeil) 
> 
> 
> ----- Original Message ----- 
> From: "Robert Kulagowski" <rkulagow at gmail.com> 
> To: "Ryan Ratliff" <rratliff at cisco.com> 
> Cc: cisco-voip at puck.nether.net 
> Sent: Tuesday, February 2, 2010 12:10:11 PM GMT -05:00 US/Canada
Eastern 
> Subject: Re: [cisco-voip] Command for determining which dial-peer will
be 
> used on router? 
> 
> Ryan Ratliff wrote: 
>> Try 'show dialplan number'. 
> 
> That works, but it's not matching the "8T" dialpeers that I've got,
only 
> those that have matchable digits, like 81312....... 
> 
> But, the link that Wes sent has some additional information, and it
appears 
> that "timeout" is the additional parameter I need. 
> 
> Thanks! 
> 
> BTW: It sure would be nice if things were grouped together better;
you've 
> got "show dialplan" on one hand and "test voice translation-pattern"
on the 
> other. Other than one group of people not talking to another group, is

> there a good reason why they're not both under either "show" or
"test"? 
> _______________________________________________ 
> cisco-voip mailing list 
> cisco-voip at puck.nether.net 
> https://puck.nether.net/mailman/listinfo/cisco-voip 
> 
> _______________________________________________ 
> cisco-voip mailing list 
> cisco-voip at puck.nether.net 
> https://puck.nether.net/mailman/listinfo/cisco-voip 
> 
> 
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Message: 12
Date: Tue, 2 Feb 2010 13:24:42 -0500
From: Peter Slow <peter.slow at gmail.com>
To: Nick Matthews <matthnick at gmail.com>
Cc: Cisco VOIP <cisco-voip at puck.nether.net>
Subject: Re: [cisco-voip] What is wrong with my destination pattern?
Message-ID:
	<53fc16d41002021024k4576eceatf4ddfe89878d2fec at mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1

Nick was being nice. You need to set your dial peers up so that
they're more specific so that calls can only go where you want them
to. if all "9" calls should only go out the PRI, and you know how many
digits  you want sent to asterisk and what they should start with, you
should be using a pattern like [0-8]... to send calls to asterisk.
This wont fix your initial issue, the call failing out the PRI, but it
will save you a couple hours when you don't have to troubleshoot the
resulting call routing loops =)

-Peter

On Tue, Feb 2, 2010 at 11:30 AM, Nick Matthews <matthnick at gmail.com>
wrote:
> debug voip ccapi inout will give you some call routing details.
>
> i would suggest debug isdn q931. ?if it's going out isdn, isdn is
> rejecting it, it will try the next dial peer.
>
> you may also want to create more specific dial peers.
>
> 9(011).T
> 91..........
> 9........
>
> Like that, for US numbers at least.
>
> -nick
>
> On Tue, Feb 2, 2010 at 10:43 AM, John Lange <john at johnlange.ca> wrote:
>> We have a Cisco connected to a PRI acting as a VOIP gateway for an
>> Asterisk system. In our setup, we have every call arriving at the
>> gateway that begins with "9" routed to the PRI, and everything else
>> routed to the Asterisk server.
>>
>> In short, this means any number starting with "9" should be an
outbound
>> call (to the PRI), and everything else should be is an inbound call
(to
>> the Asterisk server).
>>
>> The problem is, any international call seems to be looping back to
the
>> Asterisk box. For example, if we dial '9011448712002000' it ends up
>> looping back to the Asterisk server as if the Cisco is ignoring the
9.
>>
>> Here are the dialpeers. Pretty straight forward. What could be wrong?
>>
>> What commands can I use to trace the progress of a call on the
console
>> to see why the Cisco is doing this?
>>
>> ---
>>
>> dial-peer voice 20 pots
>> ?destination-pattern 9
>> ?direct-inward-dial
>> ?port 0/3/0:23
>> ?forward-digits extra
>> !
>> dial-peer voice 40 voip
>> preference 1
>> destination-pattern .
>> session protocol sipv2
>> session target ipv4:192.168.134.9
>> session transport udp
>> dtmf-relay sip-notify rtp-nte
>> codec g711ulaw
>> fax rate 14400
>> fax protocol t38 ls-redundancy 2 hs-redundancy 1 fallback
pass-through
>> g711ulaw
>> no vad
>>
>> ---
>>
>>
>> --
>> John Lange
>> http://www.johnlange.ca
>>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>


------------------------------

Message: 13
Date: Tue, 2 Feb 2010 12:35:32 -0600
From: Kevin Dunn <cheesevoice at gmail.com>
To: Peter Slow <peter.slow at gmail.com>
Cc: Cisco Voice <cisco-voip at puck.nether.net>
Subject: Re: [cisco-voip] MOH Beeping
Message-ID:
	<84b74aec1002021035g6b8b620er9781b44025096da3 at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"

Peter

Here is the audio file.

I can plug the MOH-USB-AUDIO card into my laptop and I do not hear the
beeping.

Kevin

On Tue, Feb 2, 2010 at 12:11 PM, Peter Slow <peter.slow at gmail.com>
wrote:

> Kevin,
>   Cool! Let's fix it =) A. what happens when you plug the USB audio
> device into your laptop, instead of going directly into it via the mic
> cord? ..and B. can i see the packet capture or hear the resulting
> audio?
>
> -Peter
>
> On Tue, Feb 2, 2010 at 12:53 PM, Kevin Dunn <cheesevoice at gmail.com>
wrote:
> > Okay I have a TAC case open and I have tried changing configuration
> > settings, cables and hardware...
> >
> > CUCM 7.0.2.2000-5
> > Fixed audio from XM radio (MOH-USB-AUDIO) card
> >
> > when a caller is placed on hold and the speaker is activated there
is an
> > audible (and quite annoying) beeping sound playing over the top of
the
> audio
> > file.
> >
> > It is not audible on the handset or headset.
> >
> > If I sniff the phone port and capture the audio file I can hear it.
> >
> > If I record the audio file with my laptop (plugging mic cord into
lappy
> > instead of MOH-USB) there is no beeping.
> > In my mind that eliminates the XM radio and cables, I have changed
out
> > cables though, just in case.
> >
> > I changes out MOH-USB cards and that also did nothing to eliminate
the
> > issue.
> >
> > I have upgraded firmware on the phones and that wasn't it either.
> > With the Sample Audio file (which is JAZZY) there is no beeping
> regardless
> > of handset or speaker.
> >
> > Any suggestions?
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> >
>
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Message: 14
Date: Tue, 2 Feb 2010 12:37:05 -0600
From: Kevin Dunn <cheesevoice at gmail.com>
To: Peter Slow <peter.slow at gmail.com>
Cc: Cisco Voice <cisco-voip at puck.nether.net>
Subject: Re: [cisco-voip] MOH Beeping
Message-ID:
	<84b74aec1002021037r6bcac9f0gd28806c771c83dd8 at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"

grrr wrong file

This is the correct one

On Tue, Feb 2, 2010 at 12:35 PM, Kevin Dunn <cheesevoice at gmail.com>
wrote:

> Peter
>
> Here is the audio file.
>
> I can plug the MOH-USB-AUDIO card into my laptop and I do not hear the
> beeping.
>
> Kevin
>
>   On Tue, Feb 2, 2010 at 12:11 PM, Peter Slow
<peter.slow at gmail.com>wrote:
>
>> Kevin,
>>   Cool! Let's fix it =) A. what happens when you plug the USB audio
>> device into your laptop, instead of going directly into it via the
mic
>> cord? ..and B. can i see the packet capture or hear the resulting
>> audio?
>>
>> -Peter
>>
>> On Tue, Feb 2, 2010 at 12:53 PM, Kevin Dunn <cheesevoice at gmail.com>
>> wrote:
>> > Okay I have a TAC case open and I have tried changing configuration
>> > settings, cables and hardware...
>> >
>> > CUCM 7.0.2.2000-5
>> > Fixed audio from XM radio (MOH-USB-AUDIO) card
>> >
>> > when a caller is placed on hold and the speaker is activated there
is an
>> > audible (and quite annoying) beeping sound playing over the top of
the
>> audio
>> > file.
>> >
>> > It is not audible on the handset or headset.
>> >
>> > If I sniff the phone port and capture the audio file I can hear it.
>> >
>> > If I record the audio file with my laptop (plugging mic cord into
lappy
>> > instead of MOH-USB) there is no beeping.
>> > In my mind that eliminates the XM radio and cables, I have changed
out
>> > cables though, just in case.
>> >
>> > I changes out MOH-USB cards and that also did nothing to eliminate
the
>> > issue.
>> >
>> > I have upgraded firmware on the phones and that wasn't it either.
>> > With the Sample Audio file (which is JAZZY) there is no beeping
>> regardless
>> > of handset or speaker.
>> >
>> > Any suggestions?
>> > _______________________________________________
>> > cisco-voip mailing list
>> > cisco-voip at puck.nether.net
>> > https://puck.nether.net/mailman/listinfo/cisco-voip
>> >
>> >
>>
>
>
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Message: 15
Date: Tue, 2 Feb 2010 18:40:19 +0000
From: mark baker <mb at c2ukltd.com>
To: cips <cisco at cips.nl>
Cc: "cisco-voip at puck.nether.net" <cisco-voip at puck.nether.net>
Subject: Re: [cisco-voip] Allocation of Speak Time to Users
Message-ID: <34F5D7E4-D99C-4D9B-9173-4ABEBEC13395 at c2ukltd.com>
Content-Type: text/plain; charset="utf-8"

Many thanks for the prompt reply my friend - extremely useful. Do you
happen to know if there would be any way of injecting some sort of
audible warning rather than cut call. We are trying to create a 'pay as
you go' type service. Any advice would be extremely helpful????

Sent from my iPhone

On 2 Feb 2010, at 18:34, "cips" <cisco at cips.nl<mailto:cisco at cips.nl>>
wrote:

You could use ExtensionMobility for this. Create a device profile for
the use with international access. Associate this profile to the user.
The use needs to logon with personal userID and pincode and then the
device profile gets ?loaded? on the phone.

Set the enterprise param in de CCM to a specific amount of time they can
log in.
However if the user is logged in via EM the logout is not forced if
there is an active call.

Maybe this helps.

From:
cisco-voip-bounces at puck.nether.net<mailto:cisco-voip-bounces at puck.nether
.net> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of mark
baker
Sent: dinsdag 2 februari 2010 18:21
To: <mailto:cisco-voip at puck.nether.net>
cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>
Subject: [cisco-voip] Allocation of Speak Time to Users

Hi Folks,

Is anybody aware of a mechanism in CCM to allow users to only dial
international numbers based on individual Pin Codes? Then on entry to
only have access to the line for a predetermined amount of time set by
the CCM admin?

Any advice would be greatly appreciated.

Kind Regards,

Mark Baker

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------------------------------

Message: 16
Date: Tue, 02 Feb 2010 12:44:36 -0600
From: John Lange <john at johnlange.ca>
To: Peter Slow <peter.slow at gmail.com>
Cc: Cisco VOIP <cisco-voip at puck.nether.net>
Subject: Re: [cisco-voip] What is wrong with my destination pattern?
Message-ID: <1265136276.10432.145.camel at linux-k6vx.site>
Content-Type: text/plain; charset="UTF-8"

Thanks. For outbound we can't really make it any more specific than 9,
but for inbound we could tune it a lot since we always get calls
presented with 10 digits starting with our area code.

However, I'm confused about the "T" pattern. Unless i'm mistaken, that
is for "timeout" when collecting digits. Since these are 100% SIP calls,
the entire number is always sent in one block so that makes the "T"
irrelevant. Correct?

So what we've established is that the pattern matching is not the
underlying problem (it causes the looping when the PRI rejects the call
but it's not what's causing the call to fail in the first place).

So now we're on to debugging the PRI.

If anyone has any tips for configuring a Cisco to connect to a Bell PRI
please let me know. It's working fine but we just can't dial
international calls.

Regards,

-- 
John Lange
http://www.johnlange.ca


On Tue, 2010-02-02 at 13:24 -0500, Peter Slow wrote:
> Nick was being nice. You need to set your dial peers up so that
> they're more specific so that calls can only go where you want them
> to. if all "9" calls should only go out the PRI, and you know how many
> digits  you want sent to asterisk and what they should start with, you
> should be using a pattern like [0-8]... to send calls to asterisk.
> This wont fix your initial issue, the call failing out the PRI, but it
> will save you a couple hours when you don't have to troubleshoot the
> resulting call routing loops =)
> 
> -Peter
> 
> On Tue, Feb 2, 2010 at 11:30 AM, Nick Matthews <matthnick at gmail.com>
wrote:
> > debug voip ccapi inout will give you some call routing details.
> >
> > i would suggest debug isdn q931.  if it's going out isdn, isdn is
> > rejecting it, it will try the next dial peer.
> >
> > you may also want to create more specific dial peers.
> >
> > 9(011).T
> > 91..........
> > 9........
> >
> > Like that, for US numbers at least.
> >
> > -nick
> >
> > On Tue, Feb 2, 2010 at 10:43 AM, John Lange <john at johnlange.ca>
wrote:
> >> We have a Cisco connected to a PRI acting as a VOIP gateway for an
> >> Asterisk system. In our setup, we have every call arriving at the
> >> gateway that begins with "9" routed to the PRI, and everything else
> >> routed to the Asterisk server.
> >>
> >> In short, this means any number starting with "9" should be an
outbound
> >> call (to the PRI), and everything else should be is an inbound call
(to
> >> the Asterisk server).
> >>
> >> The problem is, any international call seems to be looping back to
the
> >> Asterisk box. For example, if we dial '9011448712002000' it ends up
> >> looping back to the Asterisk server as if the Cisco is ignoring the
9.
> >>
> >> Here are the dialpeers. Pretty straight forward. What could be
wrong?
> >>
> >> What commands can I use to trace the progress of a call on the
console
> >> to see why the Cisco is doing this?
> >>
> >> ---
> >>
> >> dial-peer voice 20 pots
> >>  destination-pattern 9
> >>  direct-inward-dial
> >>  port 0/3/0:23
> >>  forward-digits extra
> >> !
> >> dial-peer voice 40 voip
> >> preference 1
> >> destination-pattern .
> >> session protocol sipv2
> >> session target ipv4:192.168.134.9
> >> session transport udp
> >> dtmf-relay sip-notify rtp-nte
> >> codec g711ulaw
> >> fax rate 14400
> >> fax protocol t38 ls-redundancy 2 hs-redundancy 1 fallback
pass-through
> >> g711ulaw
> >> no vad
> >>
> >> ---
> >>
> >>
> >> --
> >> John Lange
> >> http://www.johnlange.ca
> >>
> >> _______________________________________________
> >> cisco-voip mailing list
> >> cisco-voip at puck.nether.net
> >> https://puck.nether.net/mailman/listinfo/cisco-voip
> >>
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >





------------------------------

Message: 17
Date: Tue, 2 Feb 2010 13:57:26 -0500
From: Ed Leatherman <ealeatherman at gmail.com>
To: Scott Kee <SKee at cmsstl.com>
Cc: "cisco-voip at puck.nether.net" <cisco-voip at puck.nether.net>
Subject: Re: [cisco-voip] Unity 5.0 in the AD 2008 R2
Message-ID:
	<94a1afde1002021057i58ced9e9l7ce922cf1f49dc1f at mail.gmail.com>
Content-Type: text/plain; charset="utf-8"

Unofficially, looks like this is what's causing the problem:

http://support.microsoft.com/default.aspx/kb/977180

Apparently applying the workaround in the KB article resolved the
problem,
however the fix is still unsupported by TAC and there isn't a likely
official supported solution to this from Cisco until after Unity 8 is
released.

We're going to put our DC rollout on hold for a few weeks and possibly
build
this scenario in a test environment ourselves and come to a decision
after
that. Perhaps in the meantime Unity 8 will be out and we will get a
supported fix soon after.

On Mon, Feb 1, 2010 at 3:43 PM, Scott Kee <SKee at cmsstl.com> wrote:

>  Ed,
>
> Our server guys are adding more Windows 2008 R2 DCs later this month.
I
> will keep you posted if I find any issues.. you do the same.
>
>
>
> Thanks,
>
>
>
>
>

-- 
Ed Leatherman
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Message: 18
Date: Tue, 2 Feb 2010 14:02:42 -0500
From: Mike Thompson <mthompson729 at gmail.com>
To: mark baker <mb at c2ukltd.com>
Cc: "cisco-voip at puck.nether.net" <cisco-voip at puck.nether.net>
Subject: Re: [cisco-voip] Allocation of Speak Time to Users
Message-ID: <28DD5780-66F9-4C44-8D98-68670EBBED8E at gmail.com>
Content-Type: text/plain; charset="utf-8"; Format="flowed";
	DelSp="yes"

I have also seen tcl scripts used in a similar fashion. An example was  
using a prepaid calling card.  Of course, that would require an h.323  
gateway.

Sent from my phone, apologies for any typos.

On Feb 2, 2010, at 1:40 PM, mark baker <mb at c2ukltd.com> wrote:

> Many thanks for the prompt reply my friend - extremely useful. Do  
> you happen to know if there would be any way of injecting some sort  
> of audible warning rather than cut call. We are trying to create a  
> 'pay as you go' type service. Any advice would be extremely  
> helpful????
>
> Sent from my iPhone
>
> On 2 Feb 2010, at 18:34, "cips" <cisco at cips.nl> wrote:
>
>> You could use ExtensionMobility for this. Create a device profile  
>> for the use with international access. Associate this profile to  
>> the user. The use needs to logon with personal userID and pincode  
>> and then the device profile gets ?loaded? on the phone.
>>
>>
>>
>> Set the enterprise param in de CCM to a specific amount of time  
>> they can log in.
>>
>> However if the user is logged in via EM the logout is not forced if  
>> there is an active call.
>>
>>
>>
>> Maybe this helps.
>>
>>
>>
>> From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip- 
>> bounces at puck.nether.net] On Behalf Of mark baker
>> Sent: dinsdag 2 februari 2010 18:21
>> To: cisco-voip at puck.nether.net
>> Subject: [cisco-voip] Allocation of Speak Time to Users
>>
>>
>>
>> Hi Folks,
>>
>> Is anybody aware of a mechanism in CCM to allow users to only dial  
>> international numbers based on individual Pin Codes? Then on entry  
>> to only have access to the line for a predetermined amount of time  
>> set by the CCM admin?
>>
>> Any advice would be greatly appreciated.
>>
>> Kind Regards,
>>
>> Mark Baker
>>
>>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
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Message: 19
Date: Tue, 2 Feb 2010 13:05:20 -0600
From: Scott Kee <SKee at cmsstl.com>
To: "'Ed Leatherman'" <ealeatherman at gmail.com>
Cc: "cisco-voip at puck.nether.net" <cisco-voip at puck.nether.net>
Subject: Re: [cisco-voip] Unity 5.0 in the AD 2008 R2
Message-ID:
	
<21712D91F95FB640822E2DFE19ADB7DB067FB17648 at CMSMAIL.Spectrumhealth.com>
	
Content-Type: text/plain; charset="utf-8"

Our Server guys are adding 2nd Windows 2008 R2 DC tonight.
I will let you know how it goes?

From: Ed Leatherman [mailto:ealeatherman at gmail.com]
Sent: Tuesday, February 02, 2010 12:57 PM
To: Scott Kee
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Unity 5.0 in the AD 2008 R2

Unofficially, looks like this is what's causing the problem:

http://support.microsoft.com/default.aspx/kb/977180

Apparently applying the workaround in the KB article resolved the
problem, however the fix is still unsupported by TAC and there isn't a
likely official supported solution to this from Cisco until after Unity
8 is released.

We're going to put our DC rollout on hold for a few weeks and possibly
build this scenario in a test environment ourselves and come to a
decision after that. Perhaps in the meantime Unity 8 will be out and we
will get a supported fix soon after.
On Mon, Feb 1, 2010 at 3:43 PM, Scott Kee
<SKee at cmsstl.com<mailto:SKee at cmsstl.com>> wrote:
Ed,
Our server guys are adding more Windows 2008 R2 DCs later this month.  I
will keep you posted if I find any issues.. you do the same.

Thanks,



--
Ed Leatherman
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------------------------------

Message: 20
Date: Tue, 2 Feb 2010 19:12:26 +0000
From: Reynaldo Casta?o <racu2000 at hotmail.com>
To: <john at johnlange.ca>
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] What is wrong with my destination pattern?
Message-ID: <SNT126-W276AAFB35F3F3136FDA631C8570 at phx.gbl>
Content-Type: text/plain; charset="iso-8859-1"


John,

 

you have to make a little change in your dial-peer configuration, as
follow:

 

dial-peer voice 20 pots
destination-pattern 9...............
direct-inward-dial
port 0/3/0:23
forward-digits extra


this way, you're telling the gateway, that all trafic beggining with 9,
must be redirected to port 0/3/0,

 

depending on the pattern, you can use something like this:

 

destination-pattern 9011............

 

hope this can help



 
> On Tue, Feb 2, 2010 at 10:43 AM, John Lange <john at johnlange.ca> wrote:
> > We have a Cisco connected to a PRI acting as a VOIP gateway for an
> > Asterisk system. In our setup, we have every call arriving at the
> > gateway that begins with "9" routed to the PRI, and everything else
> > routed to the Asterisk server.
> >
> > In short, this means any number starting with "9" should be an
outbound
> > call (to the PRI), and everything else should be is an inbound call
(to
> > the Asterisk server).
> >
> > The problem is, any international call seems to be looping back to
the
> > Asterisk box. For example, if we dial '9011448712002000' it ends up
> > looping back to the Asterisk server as if the Cisco is ignoring the
9.
> >
> > Here are the dialpeers. Pretty straight forward. What could be
wrong?
> >
> > What commands can I use to trace the progress of a call on the
console
> > to see why the Cisco is doing this?
> >
> > ---
> >
> > dial-peer voice 20 pots
> >  destination-pattern 9
> >  direct-inward-dial
> >  port 0/3/0:23
> >  forward-digits extra
> > !
> > dial-peer voice 40 voip
> > preference 1
> > destination-pattern .
> > session protocol sipv2
> > session target ipv4:192.168.134.9
> > session transport udp
> > dtmf-relay sip-notify rtp-nte
> > codec g711ulaw
> > fax rate 14400
> > fax protocol t38 ls-redundancy 2 hs-redundancy 1 fallback
pass-through
> > g711ulaw
> > no vad
> >
> > ---
> >
> >
> > --
> > John Lange
> > http://www.johnlange.ca
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
 		 	   		  
_________________________________________________________________
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Message: 21
Date: Tue, 2 Feb 2010 19:34:05 +0100
From: "cips" <cisco at cips.nl>
To: <cisco-voip at puck.nether.net>
Cc: 'mark baker' <mb at c2ukltd.com>
Subject: Re: [cisco-voip] Allocation of Speak Time to Users
Message-ID: <001e01caa436$4cf16c40$e6d444c0$@nl>
Content-Type: text/plain; charset="us-ascii"

You could use ExtensionMobility for this. Create a device profile for
the
use with international access. Associate this profile to the user. The
use
needs to logon with personal userID and pincode and then the device
profile
gets "loaded" on the phone.

 

Set the enterprise param in de CCM to a specific amount of time they can
log
in.

However if the user is logged in via EM the logout is not forced if
there is
an active call.

 

Maybe this helps.

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of mark baker
Sent: dinsdag 2 februari 2010 18:21
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Allocation of Speak Time to Users

 

Hi Folks,

Is anybody aware of a mechanism in CCM to allow users to only dial
international numbers based on individual Pin Codes? Then on entry to
only
have access to the line for a predetermined amount of time set by the
CCM
admin?

Any advice would be greatly appreciated.

Kind Regards,

Mark Baker



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Message: 22
Date: Tue, 2 Feb 2010 14:19:24 -0500
From: Ryan Ratliff <rratliff at cisco.com>
To: mark baker <mb at c2ukltd.com>
Cc: "cisco-voip at puck.nether.net" <cisco-voip at puck.nether.net>
Subject: Re: [cisco-voip] Allocation of Speak Time to Users
Message-ID: <C6DF49B3-B7AA-4C58-AB31-6A6F5576B7F1 at cisco.com>
Content-Type: text/plain; charset="windows-1252"

In order to force the phone onhook you are going to need some
application that can control the phone.   You can try writing something
up in an IPCC script or could use something like CUAE to do it.
Sending audio to the phone while it is on a call may be tricky.  You
could use the whisper intercom feature for this or route every call
through a media port so your script can insert audio.

-Ryan

On Feb 2, 2010, at 1:40 PM, mark baker wrote:

Many thanks for the prompt reply my friend - extremely useful. Do you
happen to know if there would be any way of injecting some sort of
audible warning rather than cut call. We are trying to create a 'pay as
you go' type service. Any advice would be extremely helpful????

Sent from my iPhone

On 2 Feb 2010, at 18:34, "cips" <cisco at cips.nl> wrote:

> You could use ExtensionMobility for this. Create a device profile for
the use with international access. Associate this profile to the user.
The use needs to logon with personal userID and pincode and then the
device profile gets ?loaded? on the phone.
> 
>  
> 
> Set the enterprise param in de CCM to a specific amount of time they
can log in.
> 
> However if the user is logged in via EM the logout is not forced if
there is an active call.
> 
>  
> 
> Maybe this helps.
> 
>  
> 
> From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of mark baker
> Sent: dinsdag 2 februari 2010 18:21
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] Allocation of Speak Time to Users
> 
>  
> 
> Hi Folks,
> 
> Is anybody aware of a mechanism in CCM to allow users to only dial
international numbers based on individual Pin Codes? Then on entry to
only have access to the line for a predetermined amount of time set by
the CCM admin?
> 
> Any advice would be greatly appreciated.
> 
> Kind Regards,
> 
> Mark Baker
> 
> 
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

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Message: 23
Date: Tue, 2 Feb 2010 14:35:38 -0500
From: Nick Matthews <matthnick at gmail.com>
To: Manoj Kalpage <manoj.kalpage at gmail.com>
Cc: Cisco Voip <cisco-voip at puck.nether.net>
Subject: Re: [cisco-voip] Cisco AXP-XMediusFAX?
Message-ID:
	<56c3b48b1002021135r20e757e4k63297a92180b8912 at mail.gmail.com>
Content-Type: text/plain; charset=windows-1252

I believe fax to email is a feature that nearly every fax vendor is
going to provide, with differing minor features.  Admittedly, the
fax-to-email feature on routers is pretty fragile and partial pages on
long faxes is an ongoing problem.

-nick

On Tue, Feb 2, 2010 at 12:42 PM, Manoj Kalpage <manoj.kalpage at gmail.com>
wrote:
> Hi All,
>
> We are using cisco T.38 IOS fax gateway for fax to e-mail solution.
There
> are some?issue like page truncated at the end. This happens specially
when
> receive long fax. We are looking for alternative solution. I am
> wondering?how Cisco AXP-XMediusFAX? ?
> Any comments would be appreciated.
> http://www.cisco.com/en/US/prod/routers/ps9701/axp_promo.html
>
> Thanks,
> MK
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>


------------------------------

Message: 24
Date: Tue, 2 Feb 2010 14:39:24 -0500
From: Peter Slow <peter.slow at gmail.com>
To: Kevin Dunn <cheesevoice at gmail.com>
Cc: Cisco Voice <cisco-voip at puck.nether.net>
Subject: Re: [cisco-voip] MOH Beeping
Message-ID:
	<53fc16d41002021139l41e9a8c1j19afd8ca58b92161 at mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1

hrm, i was hoping to hear the beeping, but i listened to the first
minute or so of the good_au.au and didnt hear it... is it an actual
beep noise, or is there a better way to describe it? I'm imagining
(and listening for)  tone-on-hold type beeps.

-Peter

On Tue, Feb 2, 2010 at 1:37 PM, Kevin Dunn <cheesevoice at gmail.com>
wrote:
> grrr wrong file
>
> This is the correct one
>
> On Tue, Feb 2, 2010 at 12:35 PM, Kevin Dunn <cheesevoice at gmail.com>
wrote:
>>
>> Peter
>>
>> Here is the audio file.
>>
>> I can plug the MOH-USB-AUDIO card into my laptop and I do not hear
the
>> beeping.
>>
>> Kevin
>>
>> On Tue, Feb 2, 2010 at 12:11 PM, Peter Slow <peter.slow at gmail.com>
wrote:
>>>
>>> Kevin,
>>> ? Cool! Let's fix it =) A. what happens when you plug the USB audio
>>> device into your laptop, instead of going directly into it via the
mic
>>> cord? ..and B. can i see the packet capture or hear the resulting
>>> audio?
>>>
>>> -Peter
>>>
>>> On Tue, Feb 2, 2010 at 12:53 PM, Kevin Dunn <cheesevoice at gmail.com>
>>> wrote:
>>> > Okay I have a TAC case open and I have tried changing
configuration
>>> > settings, cables and hardware...
>>> >
>>> > CUCM 7.0.2.2000-5
>>> > Fixed audio from XM radio (MOH-USB-AUDIO) card
>>> >
>>> > when a caller is placed on hold and the speaker is activated there
is
>>> > an
>>> > audible (and quite annoying) beeping sound playing over the top of
the
>>> > audio
>>> > file.
>>> >
>>> > It is not audible on the handset or headset.
>>> >
>>> > If I sniff the phone port and capture the audio file I can hear
it.
>>> >
>>> > If I record the audio file with my laptop (plugging mic cord into
lappy
>>> > instead of MOH-USB) there is no beeping.
>>> > In my mind that eliminates the XM radio and cables, I have changed
out
>>> > cables though, just in case.
>>> >
>>> > I changes out MOH-USB cards and that also did nothing to eliminate
the
>>> > issue.
>>> >
>>> > I have upgraded firmware on the phones and that wasn't it either.
>>> > With the Sample Audio file (which is JAZZY) there is no beeping
>>> > regardless
>>> > of handset or speaker.
>>> >
>>> > Any suggestions?
>>> > _______________________________________________
>>> > cisco-voip mailing list
>>> > cisco-voip at puck.nether.net
>>> > https://puck.nether.net/mailman/listinfo/cisco-voip
>>> >
>>> >
>>
>
>


------------------------------

Message: 25
Date: Tue, 2 Feb 2010 12:03:06 -0700
From: Terry Oakley <Terry.Oakley at rdc.ab.ca>
To: "cisco-voip at puck.nether.net" <cisco-voip at puck.nether.net>
Subject: [cisco-voip] Question about CRS and using the Editor for
	queue	scripts
Message-ID:
	
<15F47B5DF14DB045A241B0B13672E6F0058646879A at RDCEXMAIL1.RDCSRVCS.ADS>
Content-Type: text/plain; charset="us-ascii"

Does anyone have a guide or site that explains the different prompts and
how to configure them in scripts, such as sample scripts.    The
Administrator guide is good but I seem to be missing the 'full'
understanding of what can and cannot be done.

Thanks

Terry

Terry Oakley
Telecommunication Coordinator, | Information Technology Services
100 College Blvd | Red Deer, AB T4N 5H5
Tel (403) 342-3521 |
Terry.Oakley at rdc.ab.ca<mailto:Terry.Oakley at rdc.ab.ca>
[cid:image001.jpg at 01CAA3FF.ADDF7010]<http://www.rdc.ab.ca/>

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Message: 26
Date: Tue, 02 Feb 2010 14:59:11 -0500
From: Wes Sisk <wsisk at cisco.com>
To: cips <cisco at cips.nl>
Cc: cisco-voip at puck.nether.net, 'mark baker' <mb at c2ukltd.com>
Subject: Re: [cisco-voip] Allocation of Speak Time to Users
Message-ID: <4B68840F.4080404 at cisco.com>
Content-Type: text/plain; charset="iso-8859-1"; Format="flowed"

Will em timed logout preemt an active call?

/Wes

On Tuesday, February 02, 2010 1:34:05 PM, cips <cisco at cips.nl> wrote:
>
> You could use ExtensionMobility for this. Create a device profile for 
> the use with international access. Associate this profile to the user.

> The use needs to logon with personal userID and pincode and then the 
> device profile gets "loaded" on the phone.
>
>  
>
> Set the enterprise param in de CCM to a specific amount of time they 
> can log in.
>
> However if the user is logged in via EM the logout is not forced if 
> there is an active call.
>
>  
>
> Maybe this helps.
>
>  
>
> *From:* cisco-voip-bounces at puck.nether.net 
> [mailto:cisco-voip-bounces at puck.nether.net] *On Behalf Of *mark baker
> *Sent:* dinsdag 2 februari 2010 18:21
> *To:* cisco-voip at puck.nether.net
> *Subject:* [cisco-voip] Allocation of Speak Time to Users
>
>  
>
> Hi Folks,
>
> Is anybody aware of a mechanism in CCM to allow users to only dial 
> international numbers based on individual Pin Codes? Then on entry to 
> only have access to the line for a predetermined amount of time set by

> the CCM admin?
>
> Any advice would be greatly appreciated.
>
> Kind Regards,
>
> Mark Baker
>
>
------------------------------------------------------------------------
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>   

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Message: 27
Date: Tue, 02 Feb 2010 15:00:59 -0500
From: Wes Sisk <wsisk at cisco.com>
To: mark baker <mb at c2ukltd.com>
Cc: "cisco-voip at puck.nether.net" <cisco-voip at puck.nether.net>
Subject: Re: [cisco-voip] Allocation of Speak Time to Users
Message-ID: <4B68847B.1090103 at cisco.com>
Content-Type: text/plain; charset="iso-8859-1"; Format="flowed"

FAC or CMC would work for PIN codes, however they will not limit call 
duration.

Only way to do both that I am aware of is via a script on CUAE, UCCX.  
It could be implemented similar to the calling card application in TCL 
on H.323 or SIP gateway.

/Wes

On Tuesday, February 02, 2010 12:21:07 PM, mark baker <mb at c2ukltd.com> 
wrote:
> Hi Folks,
>
> Is anybody aware of a mechanism in CCM to allow users to only dial 
> international numbers based on individual Pin Codes? Then on entry to 
> only have access to the line for a predetermined amount of time set by

> the CCM admin?
>
> Any advice would be greatly appreciated.
>
> Kind Regards,
>
> Mark Baker
>
>
>
------------------------------------------------------------------------
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>   

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Message: 28
Date: Tue, 02 Feb 2010 15:07:53 -0500
From: Wes Sisk <wsisk at cisco.com>
To: Kevin Dunn <cheesevoice at gmail.com>
Cc: Cisco Voice <cisco-voip at puck.nether.net>
Subject: Re: [cisco-voip] MOH Beeping
Message-ID: <4B688619.6010902 at cisco.com>
Content-Type: text/plain; charset="iso-8859-1"; Format="flowed"

In the very abstract this sounds like call waiting tone.  Can you 
clarify exactly what the user is doing when "caller is placed on hold 
and the speaker is activated"?  What buttons is the user pressing to 
accomplish this?

concurrent detailed CCM and SDL traces and a packet capture of all 
traffic to/from the phone could be used to isolate the source of this 
tone rather quickly.  You mention a TAC case, what is the number?

/Wes



On Tuesday, February 02, 2010 12:53:17 PM, Kevin Dunn 
<cheesevoice at gmail.com> wrote:
> Okay I have a TAC case open and I have tried changing configuration 
> settings, cables and hardware...
>  
> CUCM 7.0.2.2000-5
> Fixed audio from XM radio (MOH-USB-AUDIO) card
>  
> when a caller is placed on hold and the speaker is activated there is 
> an audible (and quite annoying) beeping sound playing over the top of 
> the audio file.
>  
> It is not audible on the handset or headset.
>  
> If I sniff the phone port and capture the audio file I can hear it.
>  
> If I record the audio file with my laptop (plugging mic cord into 
> lappy instead of MOH-USB) there is no beeping.
> In my mind that eliminates the XM radio and cables, I have changed out

> cables though, just in case.
>  
> I changes out MOH-USB cards and that also did nothing to eliminate the

> issue.
>  
> I have upgraded firmware on the phones and that wasn't it either.
> With the Sample Audio file (which is JAZZY) there is no beeping 
> regardless of handset or speaker.
>  
> Any suggestions?
>
------------------------------------------------------------------------
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>   

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Message: 29
Date: Tue, 2 Feb 2010 15:08:11 -0500
From: Ed Leatherman <ealeatherman at gmail.com>
To: Terry Oakley <Terry.Oakley at rdc.ab.ca>
Cc: "cisco-voip at puck.nether.net" <cisco-voip at puck.nether.net>
Subject: Re: [cisco-voip] Question about CRS and using the Editor for
	queue	scripts
Message-ID:
	<94a1afde1002021208p505626b3tf62f348d7e94c940 at mail.gmail.com>
Content-Type: text/plain; charset="utf-8"

Terry,

Have you looked at the CRS step reference?

There is also the script repository although it doesn't explicitly use
every
step in an example afaik.

On Tue, Feb 2, 2010 at 2:03 PM, Terry Oakley <Terry.Oakley at rdc.ab.ca>
wrote:

>  Does anyone have a guide or site that explains the different prompts
and
> how to configure them in scripts, such as sample scripts.    The
> Administrator guide is good but I seem to be missing the ?full?
> understanding of what can and cannot be done.
>
>
>
> Thanks
>
>
>
> Terry
>
>

-- 
Ed Leatherman
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Message: 30
Date: Tue, 02 Feb 2010 14:28:21 -0600
From: John Lange <john at johnlange.ca>
To: Cisco VOIP <cisco-voip at puck.nether.net>
Subject: [cisco-voip] Internaltional Calls Failing at the Bell PRI
Message-ID: <1265142501.10432.185.camel at linux-k6vx.site>
Content-Type: text/plain; charset="UTF-8"

Ok, now that I've figured out that it's not the dial pattern causing the
problem, I've turned on debugging and this is what I'm getting:

Feb  2 20:20:46.305: ISDN Se0/3/0:23 Q931: Applying typeplan for sw-type
0xD is 0x2 0x1, Calling num 2049757113
Feb  2 20:20:46.305: ISDN Se0/3/0:23 Q931: Applying typeplan for sw-type
0xD is 0x1 0x1, Called num 011448712002000
Feb  2 20:20:46.309: ISDN Se0/3/0:23 Q931: TX -> SETUP pd = 8  callref =
0x30FB 
	Bearer Capability i = 0x8090A2 
		Standard = CCITT 
		Transfer Capability = Speech  
		Transfer Mode = Circuit 
		Transfer Rate = 64 kbit/s 
	Channel ID i = 0xA98390 
		Exclusive, Channel 16 
	Calling Party Number i = 0x2180, '2049757113' 
		Plan:ISDN, Type:National 
	Called Party Number i = 0x91, '011448712002000' 
		Plan:ISDN, Type:International
Feb  2 20:20:46.405: ISDN Se0/3/0:23 Q931: RX <- RELEASE_COMP pd = 8
callref = 0xB0FB 
	Cause i = 0x809C - Invalid number format (incomplete number)
-----

I also tried numbering-type international, with the same result:

-----


Feb  2 20:24:43.921: ISDN Se0/3/0:23 Q931: Applying typeplan for sw-type
0xD is 0x2 0x1, Calling num 2049757113
Feb  2 20:24:43.921: ISDN Se0/3/0:23 Q931: Applying typeplan for sw-type
0xD is 0x1 0x1, Called num 011448712002000
Feb  2 20:24:43.921: ISDN Se0/3/0:23 Q931: TX -> SETUP pd = 8  callref =
0x3100 
	Bearer Capability i = 0x8090A2 
		Standard = CCITT 
		Transfer Capability = Speech  
		Transfer Mode = Circuit 
		Transfer Rate = 64 kbit/s 
	Channel ID i = 0xA98390 
		Exclusive, Channel 16 
	Calling Party Number i = 0x2180, '2049757113' 
		Plan:ISDN, Type:National 
	Called Party Number i = 0x91, '011448712002000' 
		Plan:ISDN, Type:International
Feb  2 20:24:44.025: ISDN Se0/3/0:23 Q931: RX <- RELEASE_COMP pd = 8
callref = 0xB100 
	Cause i = 0x809C - Invalid number format (incomplete number)

---

I tested this PRI on an Asterisk box and sending 011448712002000 works
so I'm at a loss as to why the Cisco is having problems.

Any suggestions?


-- 
John Lange
http://www.johnlange.ca



------------------------------

Message: 31
Date: Tue, 2 Feb 2010 14:30:21 -0600
From: Kevin Dunn <cheesevoice at gmail.com>
To: Wes Sisk <wsisk at cisco.com>
Cc: Cisco Voice <cisco-voip at puck.nether.net>
Subject: Re: [cisco-voip] MOH Beeping
Message-ID:
	<84b74aec1002021230l2cb4c986ke699410e7fc6bb83 at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"

Sev3 SR 613458017 Udhay has been pretty helpful so far checking the
configuration.
the sniff files are associated with that case,m you should be able to
see
them (otherwise I can add them in)

User experience:

I call my boss, he answers by pressing speaker button.
I place him on hold
He hears the "good.au" sound I enclosed for Peter to listen to

the sound is not unlike rapid repeated whistling...
I should note that when I use the handset I don't hear it, but when I
capture the packets it is still there, just not as loud

Kevin

On Tue, Feb 2, 2010 at 2:07 PM, Wes Sisk <wsisk at cisco.com> wrote:

> In the very abstract this sounds like call waiting tone.  Can you
clarify
> exactly what the user is doing when "caller is placed on hold and the
> speaker is activated"?  What buttons is the user pressing to
accomplish
> this?
>
> concurrent detailed CCM and SDL traces and a packet capture of all
traffic
> to/from the phone could be used to isolate the source of this tone
rather
> quickly.  You mention a TAC case, what is the number?
>
> /Wes
>
>
>
>
> On Tuesday, February 02, 2010 12:53:17 PM, Kevin Dunn
> <cheesevoice at gmail.com> <cheesevoice at gmail.com> wrote:
>
>   Okay I have a TAC case open and I have tried changing configuration
> settings, cables and hardware...
>
> CUCM 7.0.2.2000-5
> Fixed audio from XM radio (MOH-USB-AUDIO) card
>
> when a caller is placed on hold and the speaker is activated there is
an
> audible (and quite annoying) beeping sound playing over the top of the
audio
> file.
>
> It is not audible on the handset or headset.
>
> If I sniff the phone port and capture the audio file I can hear it.
>
> If I record the audio file with my laptop (plugging mic cord into
lappy
> instead of MOH-USB) there is no beeping.
> In my mind that eliminates the XM radio and cables, I have changed out
> cables though, just in case.
>
> I changes out MOH-USB cards and that also did nothing to eliminate the
> issue.
>
> I have upgraded firmware on the phones and that wasn't it either.
> With the Sample Audio file (which is JAZZY) there is no beeping
regardless
> of handset or speaker.
>
> Any suggestions?
>
> ------------------------------
>
> _______________________________________________
> cisco-voip mailing
listcisco-voip at puck.nether.nethttps://puck.nether.net/mailman/listinfo/c
isco-voip
>
>
>
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Message: 32
Date: Tue, 2 Feb 2010 12:34:26 -0800
From: Cristobal Priego <cristobalpriego at gmail.com>
To: John Lange <john at johnlange.ca>
Cc: Cisco VOIP <cisco-voip at puck.nether.net>
Subject: Re: [cisco-voip] Internaltional Calls Failing at the Bell PRI
Message-ID: <462C9150-3FBC-49A2-AE87-93CCA0769232 at gmail.com>
Content-Type: text/plain;	charset=us-ascii;	format=flowed;
delsp=yes

With the type international you don't need to send 011 to the pstn

Sent from my iPhone

On Feb 2, 2010, at 12:28 PM, John Lange <john at johnlange.ca> wrote:

> Ok, now that I've figured out that it's not the dial pattern causing  
> the
> problem, I've turned on debugging and this is what I'm getting:
>
> Feb  2 20:20:46.305: ISDN Se0/3/0:23 Q931: Applying typeplan for sw- 
> type
> 0xD is 0x2 0x1, Calling num 2049757113
> Feb  2 20:20:46.305: ISDN Se0/3/0:23 Q931: Applying typeplan for sw- 
> type
> 0xD is 0x1 0x1, Called num 011448712002000
> Feb  2 20:20:46.309: ISDN Se0/3/0:23 Q931: TX -> SETUP pd = 8   
> callref =
> 0x30FB
>    Bearer Capability i = 0x8090A2
>        Standard = CCITT
>        Transfer Capability = Speech
>        Transfer Mode = Circuit
>        Transfer Rate = 64 kbit/s
>    Channel ID i = 0xA98390
>        Exclusive, Channel 16
>    Calling Party Number i = 0x2180, '2049757113'
>        Plan:ISDN, Type:National
>    Called Party Number i = 0x91, '011448712002000'
>        Plan:ISDN, Type:International
> Feb  2 20:20:46.405: ISDN Se0/3/0:23 Q931: RX <- RELEASE_COMP pd = 8
> callref = 0xB0FB
>    Cause i = 0x809C - Invalid number format (incomplete number)
> -----
>
> I also tried numbering-type international, with the same result:
>
> -----
>
>
> Feb  2 20:24:43.921: ISDN Se0/3/0:23 Q931: Applying typeplan for sw- 
> type
> 0xD is 0x2 0x1, Calling num 2049757113
> Feb  2 20:24:43.921: ISDN Se0/3/0:23 Q931: Applying typeplan for sw- 
> type
> 0xD is 0x1 0x1, Called num 011448712002000
> Feb  2 20:24:43.921: ISDN Se0/3/0:23 Q931: TX -> SETUP pd = 8   
> callref =
> 0x3100
>    Bearer Capability i = 0x8090A2
>        Standard = CCITT
>        Transfer Capability = Speech
>        Transfer Mode = Circuit
>        Transfer Rate = 64 kbit/s
>    Channel ID i = 0xA98390
>        Exclusive, Channel 16
>    Calling Party Number i = 0x2180, '2049757113'
>        Plan:ISDN, Type:National
>    Called Party Number i = 0x91, '011448712002000'
>        Plan:ISDN, Type:International
> Feb  2 20:24:44.025: ISDN Se0/3/0:23 Q931: RX <- RELEASE_COMP pd = 8
> callref = 0xB100
>    Cause i = 0x809C - Invalid number format (incomplete number)
>
> ---
>
> I tested this PRI on an Asterisk box and sending 011448712002000 works
> so I'm at a loss as to why the Cisco is having problems.
>
> Any suggestions?
>
>
> -- 
> John Lange
> http://www.johnlange.ca
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip


------------------------------

Message: 33
Date: Tue, 2 Feb 2010 13:54:39 -0800
From: "Leslie Meade" <lmeade at signal.ca>
To: <cisco-voip at puck.nether.net>
Subject: [cisco-voip] CFA CSS Activation Policy
Message-ID:
	
<EAF4FF4AB2142D499F2F5A3B9D851529058AC4ED at exch-bo.MGVFS.McLeanNet>
Content-Type: text/plain; charset="us-ascii"

Changing the default from configured CSS to  activating device/line CSS,
does this need a CallManager service restart to take affect ?

 

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Message: 34
Date: Tue, 2 Feb 2010 17:04:56 -0500 (EST)
From: Lelio Fulgenzi <lelio at uoguelph.ca>
To: cisco-voip voyp list <cisco-voip at puck.nether.net>
Subject: [cisco-voip] OT/FYI: Using DBAN to wipe data on HP DL380s
Message-ID:
	
<347963835.9281591265148296045.JavaMail.root at simcoe.cs.uoguelph.ca>
Content-Type: text/plain; charset="utf-8"

For those of you who have requirements to wipe data off of disks before
decommissioning them, this may be useful (if you didn't already know
about it). Apparently, the latest version of DBAN does not recognize the
SmartArray controllers and you need to use an old version in order to
make things work. 



reference: 

http://forums13.itrc.hp.com/service/forums/questionanswer.do?admit=10944
7627+1265139762899+28353475&threadId=1390222 



Hi, 

Recent versions of DBAN don't include the cciss driver needed to
recognise the SmartArray controller in the ProLiants and doesn't
officially support any RAID cards, if memory serves. 

You could try hooking the drives up to a SCSI interface instead, or try
using one of the older versions from here: 

http://sourceforge.net/projects/dban/files/ 

which did have the cciss driver. 

I seem to recall using 1.0.6 on a ProLiant in the dim and distant past,
although you may want to try some of the others... 


--- 
Lelio Fulgenzi, B.A. 
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1 
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN) 
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
Cooking with unix is easy. You just sed it and forget it. 
- LFJ (with apologies to Mr. Popeil) 

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Message: 35
Date: Tue, 02 Feb 2010 16:27:08 -0600
From: John Lange <john at johnlange.ca>
To: Go0se <me at go0se.com>
Cc: 'Cisco VOIP' <cisco-voip at puck.nether.net>
Subject: Re: [cisco-voip] Internaltional Calls Failing at the Bell PRI
Message-ID: <1265149628.10432.241.camel at linux-k6vx.site>
Content-Type: text/plain; charset="UTF-8"

With the following config I get a message from Bell "Call cannot be
completed as dialed". How do I see what number, plan & type is actually
being transmitted down the PRI to Bell?

It seems like the translation isn't working but I'm not sure.

--
voice translation-rule 1
 rule 1 /^9/ // type any unknown plan any unknown

dial-peer voice 25 pots
 numbering-type unknown
 destination-pattern 9011
 translate-outgoing called 1
 direct-inward-dial
 port 0/3/0:23
 forward-digits all
!

----- debug -----


Feb  2 22:21:56.305: //-1/34F30C39B653/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=9011448712002000, Peer Info
Type=DIALPEER_INFO_SPEECH
Feb  2 22:21:56.305: //-1/34F30C39B653/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=9011448712002000
Feb  2 22:21:56.305: //-1/34F30C39B653/DPM/dpMatchCore:
   Dial String=9011448712002000, Expanded String=9011448712002000,
Calling Number=
   Timeout=TRUE, Is Incoming=FALSE, Peer Info Type=DIALPEER_INFO_SPEECH
Feb  2 22:21:56.305: //-1/34F30C39B653/DPM/MatchNextPeer:
   Result=Success(0); Outgoing Dial-peer=20 Is Matched
Feb  2 22:21:56.305: //-1/34F30C39B653/DPM/MatchNextPeer:
   Result=Success(0); Outgoing Dial-peer=25 Is Matched
Feb  2 22:21:56.305: //-1/34F30C39B653/DPM/dpMatchPeersCore:
   Result=Success(0) after DP_MATCH_DEST
Feb  2 22:21:56.305: //-1/34F30C39B653/DPM/dpMatchPeersMoreArg:
   Result=SUCCESS(0) 
   List of Matched Outgoing Dial-peer(s): 
     1: Dial-peer Tag=25 
     2: Dial-peer Tag=20
Feb  2 16:21:59: %VOIPAAA-5-VOIP_CALL_HISTORY: CallLegType 2,
ConnectionId 34F30C39 F8011DF B6539757 A2E40164, SetupTime 16:21:56.297
CST Tue Feb 2 2010, PeerAddress 2049757113, PeerSubAddress ,
DisconnectCause 10  , DisconnectText normal call clearing (16),
ConnectTime 16:21:59.437 CST Tue Feb 2 2010, DisconnectTime 16:21:59.437
CST Tue Feb 2 2010, CallOrigin 2, ChargedUnits 0, InfoType 2,
TransmitPackets 149, TransmitBytes 23840, ReceivePackets 144,
ReceiveBytes 23040
Feb  2 16:21:59: %VOIPAAA-5-VOIP_CALL_HISTORY: CallLegType 1,
ConnectionId 34F30C39 F8011DF B6539757 A2E40164, SetupTime 16:21:56.433
CST Tue Feb 2 2010, PeerAddress 9011448712002000, PeerSubAddress ,
DisconnectCause 10  , DisconnectText normal call clearing (16),
ConnectTime 16:21:59.493 CST Tue Feb 2 2010, DisconnectTime 16:21:59.493
CST Tue Feb 2 2010, CallOrigin 1, ChargedUnits 0, InfoType 2,
TransmitPackets 144, TransmitBytes 24192, ReceivePackets 149,
ReceiveBytes 23840t  


-- 
John Lange
http://www.johnlange.ca



------------------------------

Message: 36
Date: Tue, 2 Feb 2010 14:48:15 -0800
From: Cristobal Priego <cristobalpriego at gmail.com>
To: John Lange <john at johnlange.ca>
Cc: Cisco VOIP <cisco-voip at puck.nether.net>
Subject: Re: [cisco-voip] Internaltional Calls Failing at the Bell PRI
Message-ID: <E31FF601-8D76-447E-B897-910318423252 at gmail.com>
Content-Type: text/plain;	charset=us-ascii;	format=flowed;
delsp=yes

Enable debug usenet q931 debugs and you will see what you're sending


Sent from my iPhone

On Feb 2, 2010, at 2:27 PM, John Lange <john at johnlange.ca> wrote:

> With the following config I get a message from Bell "Call cannot be
> completed as dialed". How do I see what number, plan & type is  
> actually
> being transmitted down the PRI to Bell?
>
> It seems like the translation isn't working but I'm not sure.
>
> --
> voice translation-rule 1
> rule 1 /^9/ // type any unknown plan any unknown
>
> dial-peer voice 25 pots
> numbering-type unknown
> destination-pattern 9011
> translate-outgoing called 1
> direct-inward-dial
> port 0/3/0:23
> forward-digits all
> !
>
> ----- debug -----
>
>
> Feb  2 22:21:56.305: //-1/34F30C39B653/DPM/dpMatchPeersCore:
>   Calling Number=, Called Number=9011448712002000, Peer Info
> Type=DIALPEER_INFO_SPEECH
> Feb  2 22:21:56.305: //-1/34F30C39B653/DPM/dpMatchPeersCore:
>   Match Rule=DP_MATCH_DEST; Called Number=9011448712002000
> Feb  2 22:21:56.305: //-1/34F30C39B653/DPM/dpMatchCore:
>   Dial String=9011448712002000, Expanded String=9011448712002000,
> Calling Number=
>   Timeout=TRUE, Is Incoming=FALSE, Peer Info Type=DIALPEER_INFO_SPEECH
> Feb  2 22:21:56.305: //-1/34F30C39B653/DPM/MatchNextPeer:
>   Result=Success(0); Outgoing Dial-peer=20 Is Matched
> Feb  2 22:21:56.305: //-1/34F30C39B653/DPM/MatchNextPeer:
>   Result=Success(0); Outgoing Dial-peer=25 Is Matched
> Feb  2 22:21:56.305: //-1/34F30C39B653/DPM/dpMatchPeersCore:
>   Result=Success(0) after DP_MATCH_DEST
> Feb  2 22:21:56.305: //-1/34F30C39B653/DPM/dpMatchPeersMoreArg:
>   Result=SUCCESS(0)
>   List of Matched Outgoing Dial-peer(s):
>     1: Dial-peer Tag=25
>     2: Dial-peer Tag=20
> Feb  2 16:21:59: %VOIPAAA-5-VOIP_CALL_HISTORY: CallLegType 2,
> ConnectionId 34F30C39 F8011DF B6539757 A2E40164, SetupTime  
> 16:21:56.297
> CST Tue Feb 2 2010, PeerAddress 2049757113, PeerSubAddress ,
> DisconnectCause 10  , DisconnectText normal call clearing (16),
> ConnectTime 16:21:59.437 CST Tue Feb 2 2010, DisconnectTime  
> 16:21:59.437
> CST Tue Feb 2 2010, CallOrigin 2, ChargedUnits 0, InfoType 2,
> TransmitPackets 149, TransmitBytes 23840, ReceivePackets 144,
> ReceiveBytes 23040
> Feb  2 16:21:59: %VOIPAAA-5-VOIP_CALL_HISTORY: CallLegType 1,
> ConnectionId 34F30C39 F8011DF B6539757 A2E40164, SetupTime  
> 16:21:56.433
> CST Tue Feb 2 2010, PeerAddress 9011448712002000, PeerSubAddress ,
> DisconnectCause 10  , DisconnectText normal call clearing (16),
> ConnectTime 16:21:59.493 CST Tue Feb 2 2010, DisconnectTime  
> 16:21:59.493
> CST Tue Feb 2 2010, CallOrigin 1, ChargedUnits 0, InfoType 2,
> TransmitPackets 144, TransmitBytes 24192, ReceivePackets 149,
> ReceiveBytes 23840t
>
>
> -- 
> John Lange
> http://www.johnlange.ca
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip


------------------------------

Message: 37
Date: Tue, 02 Feb 2010 17:00:37 -0500
From: Matthew Linsemier <mlinsemier at apassurance.com>
To: Cisco VoIPoE List <cisco-voip at puck.nether.net>
Subject: [cisco-voip] UCM and UCCX directory synchronization
Message-ID: <C78E0AB5.30CE%mlinsemier at apassurance.com>
Content-Type: text/plain; charset="iso-8859-1"

All,

Recently we had an employee leave the company who worked in our CAC.
They
were removed from Windows AD but they also were an agent within UCCX.
In
UCM their ID is gone (LDAP Syncing is on), however within UCCX I can
still
see them assigned to a phone and queue with an extension that I have
reassigned to another employee as of today.

Is there a way to see these orphaned ID?s in UCM and delete them?  I
tried
to do a manual synchronization in the SideA web portal, but I received
?Error, call Technical Support?.  I plan on rebooting the server during
the
next maintenance window to see if this would resolve the issues.  Any
input
would be greatly appreciated.

Thanks!

Matt
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Message: 38
Date: Tue, 2 Feb 2010 15:43:54 -0600
From: "Go0se" <me at go0se.com>
To: "'John Lange'" <john at johnlange.ca>,	"'Cisco VOIP'"
	<cisco-voip at puck.nether.net>
Subject: Re: [cisco-voip] Internaltional Calls Failing at the Bell PRI
Message-ID: <00f601caa450$d2c2c840$784858c0$@com>
Content-Type: text/plain;	charset="us-ascii"

If it's connecting to Bell, try setting the called type and plan both to
UNKNOWN.

-Go0se

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of John Lange
Sent: Tuesday, February 02, 2010 2:28 PM
To: Cisco VOIP
Subject: [cisco-voip] Internaltional Calls Failing at the Bell PRI

Ok, now that I've figured out that it's not the dial pattern causing the
problem, I've turned on debugging and this is what I'm getting:

Feb  2 20:20:46.305: ISDN Se0/3/0:23 Q931: Applying typeplan for sw-type
0xD is 0x2 0x1, Calling num 2049757113
Feb  2 20:20:46.305: ISDN Se0/3/0:23 Q931: Applying typeplan for sw-type
0xD is 0x1 0x1, Called num 011448712002000
Feb  2 20:20:46.309: ISDN Se0/3/0:23 Q931: TX -> SETUP pd = 8  callref =
0x30FB 
	Bearer Capability i = 0x8090A2 
		Standard = CCITT 
		Transfer Capability = Speech  
		Transfer Mode = Circuit 
		Transfer Rate = 64 kbit/s 
	Channel ID i = 0xA98390 
		Exclusive, Channel 16 
	Calling Party Number i = 0x2180, '2049757113' 
		Plan:ISDN, Type:National 
	Called Party Number i = 0x91, '011448712002000' 
		Plan:ISDN, Type:International
Feb  2 20:20:46.405: ISDN Se0/3/0:23 Q931: RX <- RELEASE_COMP pd = 8
callref = 0xB0FB 
	Cause i = 0x809C - Invalid number format (incomplete number)
-----

I also tried numbering-type international, with the same result:

-----


Feb  2 20:24:43.921: ISDN Se0/3/0:23 Q931: Applying typeplan for sw-type
0xD is 0x2 0x1, Calling num 2049757113
Feb  2 20:24:43.921: ISDN Se0/3/0:23 Q931: Applying typeplan for sw-type
0xD is 0x1 0x1, Called num 011448712002000
Feb  2 20:24:43.921: ISDN Se0/3/0:23 Q931: TX -> SETUP pd = 8  callref =
0x3100 
	Bearer Capability i = 0x8090A2 
		Standard = CCITT 
		Transfer Capability = Speech  
		Transfer Mode = Circuit 
		Transfer Rate = 64 kbit/s 
	Channel ID i = 0xA98390 
		Exclusive, Channel 16 
	Calling Party Number i = 0x2180, '2049757113' 
		Plan:ISDN, Type:National 
	Called Party Number i = 0x91, '011448712002000' 
		Plan:ISDN, Type:International
Feb  2 20:24:44.025: ISDN Se0/3/0:23 Q931: RX <- RELEASE_COMP pd = 8
callref = 0xB100 
	Cause i = 0x809C - Invalid number format (incomplete number)

---

I tested this PRI on an Asterisk box and sending 011448712002000 works
so I'm at a loss as to why the Cisco is having problems.

Any suggestions?


-- 
John Lange
http://www.johnlange.ca

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip



------------------------------

Message: 39
Date: Tue, 2 Feb 2010 15:17:53 -0700
From: "Norton, Mike" <mikenorton at pwsd76.ab.ca>
To: Kevin Dunn <cheesevoice at gmail.com>
Cc: Cisco Voice <cisco-voip at puck.nether.net>
Subject: Re: [cisco-voip] MOH Beeping
Message-ID:
	
<096D50507635B645A061351B1308C90B04DAE67E9F at pwsdexchange03.pwsb33.ab.ca>
	
Content-Type: text/plain; charset="us-ascii"

Listening to the capture, it sounds like ambient electrical/RF noise is
getting into the analog audio signal. I'm surprised this isn't a more
common problem. Server racks are probably one of the least ideal places
to try to run unbalanced analog audio signals.

--
Mike Norton
I.T. Support
Peace Wapiti School Division No. 76
Helpdesk: 780-831-3080
Direct: 780-831-3076


From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Kevin Dunn
Sent: February-02-10 1:30 PM
To: Wes Sisk
Cc: Cisco Voice
Subject: Re: [cisco-voip] MOH Beeping

Sev3 SR 613458017 Udhay has been pretty helpful so far checking the
configuration.
the sniff files are associated with that case,m you should be able to
see them (otherwise I can add them in)

User experience:

I call my boss, he answers by pressing speaker button.
I place him on hold
He hears the "good.au" sound I enclosed for Peter to listen to

the sound is not unlike rapid repeated whistling...
I should note that when I use the handset I don't hear it, but when I
capture the packets it is still there, just not as loud

Kevin
On Tue, Feb 2, 2010 at 2:07 PM, Wes Sisk
<wsisk at cisco.com<mailto:wsisk at cisco.com>> wrote:
In the very abstract this sounds like call waiting tone.  Can you
clarify exactly what the user is doing when "caller is placed on hold
and the speaker is activated"?  What buttons is the user pressing to
accomplish this?

concurrent detailed CCM and SDL traces and a packet capture of all
traffic to/from the phone could be used to isolate the source of this
tone rather quickly.  You mention a TAC case, what is the number?

/Wes




On Tuesday, February 02, 2010 12:53:17 PM, Kevin Dunn
<cheesevoice at gmail.com><mailto:cheesevoice at gmail.com> wrote:
Okay I have a TAC case open and I have tried changing configuration
settings, cables and hardware...

CUCM 7.0.2.2000-5
Fixed audio from XM radio (MOH-USB-AUDIO) card

when a caller is placed on hold and the speaker is activated there is an
audible (and quite annoying) beeping sound playing over the top of the
audio file.

It is not audible on the handset or headset.

If I sniff the phone port and capture the audio file I can hear it.

If I record the audio file with my laptop (plugging mic cord into lappy
instead of MOH-USB) there is no beeping.
In my mind that eliminates the XM radio and cables, I have changed out
cables though, just in case.

I changes out MOH-USB cards and that also did nothing to eliminate the
issue.

I have upgraded firmware on the phones and that wasn't it either.
With the Sample Audio file (which is JAZZY) there is no beeping
regardless of handset or speaker.

Any suggestions?




________________________________






_______________________________________________

cisco-voip mailing list

cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>

https://puck.nether.net/mailman/listinfo/cisco-voip




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Message: 40
Date: Tue, 2 Feb 2010 17:18:31 -0600
From: "Fuermann, Jason" <JBF005 at shsu.edu>
To: "'Matthew Linsemier'" <mlinsemier at apassurance.com>, Cisco VoIPoE
	List	<cisco-voip at puck.nether.net>
Subject: Re: [cisco-voip] UCM and UCCX directory synchronization
Message-ID:
	<8FAC1E47484E43469AA28DBF35C955E4A867901B10 at EXMBX.SHSU.EDU>
Content-Type: text/plain; charset="us-ascii"

You didn't say what version you were running, but to v7, admin guide
7-11 Implications of Deleting Agents in Unified CM
"When Unified CCX detects that the agent no longer exists in Unified CM,
it does not automatically delete that agent from the Unified CCX
database. Instead, the Unified CCX Resources page displays a new link
called Inactive Agents. When you click this link, Unified CCX displays a
list of agents deleted from Unified CM but still existing in the Unified
CCX database. In this case, select the agents to delete from Unified CCX
by checking the check box next to the required agent (or select all
agents for deletion by clicking Check All). Then click Delete to remove
the selected agents from the Unified CCX database. Unless you follow
this procedure, agents deleted in Unified CM will continue to appear in
the agents list in the Unified CCX Resources page, but they will not be
able to log in as the Unified CM authentication will not be successful."

Hope this helps

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Matthew
Linsemier
Sent: Tuesday, February 02, 2010 4:01 PM
To: Cisco VoIPoE List
Subject: [cisco-voip] UCM and UCCX directory synchronization

All,

Recently we had an employee leave the company who worked in our CAC.
They were removed from Windows AD but they also were an agent within
UCCX.  In UCM their ID is gone (LDAP Syncing is on), however within UCCX
I can still see them assigned to a phone and queue with an extension
that I have reassigned to another employee as of today.

Is there a way to see these orphaned ID's in UCM and delete them?  I
tried to do a manual synchronization in the SideA web portal, but I
received "Error, call Technical Support".  I plan on rebooting the
server during the next maintenance window to see if this would resolve
the issues.  Any input would be greatly appreciated.

Thanks!

Matt
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------------------------------

Message: 41
Date: Tue, 02 Feb 2010 17:45:09 -0600
From: John Lange <john at johnlange.ca>
To: Go0se <me at go0se.com>
Cc: 'Cisco VOIP' <cisco-voip at puck.nether.net>
Subject: Re: [cisco-voip] Internaltional Calls Failing at the Bell PRI
Message-ID: <1265154309.10432.267.camel at linux-k6vx.site>
Content-Type: text/plain; charset="UTF-8"

On Tue, 2010-02-02 at 15:43 -0600, Go0se wrote:
> If it's connecting to Bell, try setting the called type and plan both
to
> UNKNOWN.

I did speak with Bell tech and though they were not able to take a look
at the D channel, he said pretty much the same thing but how do you do
that?

Set both the called _and_ calling party, type & plan to unknown. So I
tried:

dial-peer voice 25 pots
 numbering-type unknown
 destination-pattern 9011
 direct-inward-dial
 port 0/3/0:23
 forward-digits 0
 prefix 011

But, so far as I can tell, "numbering-type" is for matching peers, not
for changing the numbering type and I can't see any way to change the
numbering type or plan in the dial-peer except with a translation rule.

But I can not get my translation rule to work...

---

voice translation-rule 1
 rule 1 /^9/ /9/ type any unknown plan any unknown

dial-peer voice 25 pots
 destination-pattern 9011
 prefix 011
 direct-inward-dial
 port 0/3/0:23
 forward-digits extra
 translate-outgoing called 1
!

-- debug --

The last few lines of the debug tell the tail..

Called Party Number i = 0x91, '011448712002000' 
 Plan:ISDN, Type:International

If I could just get the translation to work so I could set the plan &
type, I think the problem would be solved...

--- and sorry for the long post but here is the full debug isdn &
dialpeer ----


Feb  2 23:34:34.414: //-1/5A96E357B8A8/DPM/dpAssociateIncomingPeerCore:
   Calling Number=2049757113, Called Number=9011448712002000,
Voice-Interface=0x0,
   Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search
Type=PEER_TYPE_VOICE,
   Peer Info Type=DIALPEER_INFO_SPEECH
Feb  2 23:34:34.414: //-1/5A96E357B8A8/DPM/dpAssociateIncomingPeerCore:
   Match Rule=DP_MATCH_REQUEST_URI;
URI=sip:9011448712002000 at 192.168.134.20
Feb  2 23:34:34.414: //-1/5A96E357B8A8/DPM/dpMatchPeertype:
   Is Incoming=TRUE, Number Expansion=FALSE
Feb  2 23:34:34.414: //-1/5A96E357B8A8/DPM/dpMatchCore:
   Dial String=, Expanded String=, Calling Number=
   Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
Feb  2 23:34:34.414: //-1/5A96E357B8A8/DPM/dpMatchCore:
   Result=-1
Feb  2 23:34:34.414: //-1/5A96E357B8A8/DPM/dpMatchPeertype:exit at 5404 
Feb  2 23:34:34.414: //-1/5A96E357B8A8/DPM/dpAssociateIncomingPeerCore:
   Match Rule=DP_MATCH_TO_URI; URI=sip:9011448712002000 at 192.168.134.20
Feb  2 23:34:34.414: //-1/5A96E357B8A8/DPM/dpMatchPeertype:
   Is Incoming=TRUE, Number Expansion=FALSE
Feb  2 23:34:34.414: //-1/5A96E357B8A8/DPM/dpMatchCore:
   Dial String=, Expanded String=, Calling Number=
   Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
Feb  2 23:34:34.414: //-1/5A96E357B8A8/DPM/dpMatchCore:
   Result=-1
Feb  2 23:34:34.414: //-1/5A96E357B8A8/DPM/dpMatchPeertype:exit at 5404 
Feb  2 23:34:34.414: //-1/5A96E357B8A8/DPM/dpAssociateIncomingPeerCore:
   Match Rule=DP_MATCH_FROM_URI; URI=sip:2049757113 at 192.168.134.9
Feb  2 23:34:34.414: //-1/5A96E357B8A8/DPM/dpMatchPeertype:
   Is Incoming=TRUE, Number Expansion=FALSE
Feb  2 23:34:34.414: //-1/5A96E357B8A8/DPM/dpMatchCore:
   Dial String=, Expanded String=, Calling Number=
   Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
Feb  2 23:34:34.414: //-1/5A96E357B8A8/DPM/dpMatchCore:
   Result=-1
Feb  2 23:34:34.414: //-1/5A96E357B8A8/DPM/dpMatchPeertype:exit at 5404 
Feb  2 23:34:34.414: //-1/5A96E357B8A8/DPM/dpAssociateIncomingPeerCore:
   Match Rule=DP_MATCH_INCOMING_DNIS; Called Number=9011448712002000
Feb  2 23:34:34.414: //-1/5A96E357B8A8/DPM/dpMatchPeertype:
   Is Incoming=TRUE, Number Expansion=FALSE
Feb  2 23:34:34.414: //-1/5A96E357B8A8/DPM/dpMatchCore:
   Dial String=9011448712002000, Expanded String=9011448712002000,
Calling Number=
   Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
Feb  2 23:34:34.414: //-1/5A96E357B8A8/DPM/dpMatchCore:
   Result=-1
Feb  2 23:34:34.414: //-1/5A96E357B8A8/DPM/dpMatchPeertype:exit at 5404 
Feb  2 23:34:34.414: //-1/5A96E357B8A8/DPM/dpAssociateIncomingPeerCore:
   Match Rule=DP_MATCH_ANSWER; Calling Number=2049757113
Feb  2 23:34:34.414: //-1/5A96E357B8A8/DPM/dpMatchPeertype:
   Is Incoming=TRUE, Number Expansion=FALSE
Feb  2 23:34:34.414: //-1/5A96E357B8A8/DPM/dpMatchCore:
   Dial String=, Expanded String=, Calling Number=2049757113T
   Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
Feb  2 23:34:34.414: //-1/5A96E357B8A8/DPM/dpMatchCore:
   Result=-1
Feb  2 23:34:34.414: //-1/5A96E357B8A8/DPM/dpMatchPeertype:exit at 5404 
Feb  2 23:34:34.414: //-1/5A96E357B8A8/DPM/dpAssociateIncomingPeerCore:
   Match Rule=DP_MATCH_ORIGINATE; Calling Number=2049757113
Feb  2 23:34:34.414: //-1/5A96E357B8A8/DPM/dpMatchPeertype:
   Is Incoming=TRUE, Number Expansion=FALSE
Feb  2 23:34:34.414: //-1/5A96E357B8A8/DPM/dpMatchCore:
   Dial String=, Expanded String=, Calling Number=2049757113T
   Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
Feb  2 23:34:34.418: //-1/5A96E357B8A8/DPM/MatchNextPeer:
   Result=Success(0); Incoming Dial-peer=40 Is Matched
Feb  2 23:34:34.418: //-1/5A96E357B8A8/DPM/MatchNextPeer:
   Result=Success(0); Incoming Dial-peer=45 Is Matched
Feb  2 23:34:34.418: //-1/5A96E357B8A8/DPM/dpMatchPeertype:exit at 5404 
Feb  2 23:34:34.418: //-1/5A96E357B8A8/DPM/dpAssociateIncomingPeerCore:
   Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=40
Feb  2 23:34:34.418:
//-1/5A96E357B8A8/DPM/dpAssociateIncomingPeerSPI:exit at 5938 
Feb  2 23:34:34.422: //-1/5A96E357B8A8/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=9011448712002000, Peer Info
Type=DIALPEER_INFO_SPEECH
Feb  2 23:34:34.422: //-1/5A96E357B8A8/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=9011448712002000
Feb  2 23:34:34.422: //-1/5A96E357B8A8/DPM/dpMatchCore:
   Dial String=9011448712002000, Expanded String=9011448712002000,
Calling Number=
   Timeout=TRUE, Is Incoming=FALSE, Peer Info Type=DIALPEER_INFO_SPEECH
Feb  2 23:34:34.422: //-1/5A96E357B8A8/DPM/MatchNextPeer:
   Result=Success(0); Outgoing Dial-peer=20 Is Matched
Feb  2 23:34:34.422: //-1/5A96E357B8A8/DPM/MatchNextPeer:
   Result=Success(0); Outgoing Dial-peer=25 Is Matched
Feb  2 23:34:34.422: //-1/5A96E357B8A8/DPM/dpMatchPeersCore:
   Result=Success(0) after DP_MATCH_DEST
Feb  2 23:34:34.422: //-1/5A96E357B8A8/DPM/dpMatchPeersMoreArg:
   Result=SUCCESS(0) 
   List of Matched Outgoing Dial-peer(s): 
     1: Dial-peer Tag=25 
     2: Dial-peer Tag=20
Feb  2 23:34:34.422: ISDN Se0/3/0:23 Q931: Applying typeplan for sw-type
0xD is 0x2 0x1, Calling num 2049757113
Feb  2 23:34:34.426: ISDN Se0/3/0:23 Q931: Applying typeplan for sw-type
0xD is 0x1 0x1, Called num 011448712002000
Feb  2 23:34:34.426: ISDN Se0/3/0:23 Q931: TX -> SETUP pd = 8  callref =
0x3244 
	Bearer Capability i = 0x8090A2 
		Standard = CCITT 
		Transfer Capability = Speech  
		Transfer Mode = Circuit 
		Transfer Rate = 64 kbit/s 
	Channel ID i = 0xA98395 
		Exclusive, Channel 21 
	Calling Party Number i = 0x2180, '2049757113' 
		Plan:ISDN, Type:National 
	Called Party Number i = 0x91, '011448712002000' 
		Plan:ISDN, Type:International
Feb  2 23:34:34.526: ISDN Se0/3/0:23 Q931: RX <- RELEASE_COMP pd = 8
callref = 0xB244 
	Cause i = 0x809C - Invalid number format (incomplete number)
Feb  2 17:34:34: %VOIPAAA-5-VOIP_CALL_HISTORY: CallLegType 1,
ConnectionId 5A96E357 F8A11DF B8A89757 A2E40164, SetupTime 17:34:34.530
CST Tue Feb 2 2010, PeerAddress 9011448712002000, PeerSubAddress ,
DisconnectCause 1C  , DisconnectText invalid number (28), ConnectTime
17:34:34.530 CST Tue Feb 2 2010, DisconnectTime 17:34:34.530 CST Tue Feb
2 2010, CallOrigin 1, ChargedUnits 0, InfoType 2, TransmitPackets 0,
TransmitBytes 0, ReceivePackets 0, ReceiveBytes 0
Feb  2 23:34:34.534: ISDN Se0/3/0:23 Q931: Applying typeplan for sw-type
0xD is 0x2 0x1, Calling num 2049757113
Feb  2 23:34:34.534: ISDN Se0/3/0:23 Q931: Applying typeplan for sw-type
0xD is 0x1 0x1, Called num 011448712002000
Feb  2 23:34:34.534: ISDN Se0/3/0:23 Q931: TX -> SETUP pd = 8  callref =
0x3245 
	Bearer Capability i = 0x8090A2 
		Standard = CCITT 
		Transfer Capability = Speech  
		Transfer Mode = Circuit 
		Transfer Rate = 64 kbit/s 
	Channel ID i = 0xA98395 
		Exclusive, Channel 21 
	Calling Party Number i = 0x2180, '2049757113' 
		Plan:ISDN, Type:National 
	Called Party Number i = 0x91, '011448712002000' 
		Plan:ISDN, Type:International
Feb  2 23:34:34.666: ISDN Se0/3/0:23 Q931: RX <- RELEASE_COMP pd = 8
callref = 0xB245 
	Cause i = 0x809C - Invalid number format (incomplete number)
Feb  2 17:34:34: %VOIPAAA-5-VOIP_CALL_HISTORY: CallLegType 1,
ConnectionId 5A96E357 F8A11DF B8A89757 A2E40164, SetupTime 17:34:34.670
CST Tue Feb 2 2010, PeerAddress 9011448712002000, PeerSubAddress ,
DisconnectCause 1C  , DisconnectText invalid number (28), ConnectTime
17:34:34.670 CST Tue Feb 2 2010, DisconnectTime 17:34:34.670 CST Tue Feb
2 2010, CallOrigin 1, ChargedUnits 0, InfoType 2, TransmitPackets 0,
TransmitBytes 0, ReceivePackets 0, ReceiveBytes 0
Feb  2 17:34:34: %VOIPAAA-5-VOIP_CALL_HISTORY: CallLegType 2,
ConnectionId 5A96E357 F8A11DF B8A89757 A2E40164, SetupTime 17:34:34.422
CST Tue Feb 2 2010, PeerAddress 2049757113, PeerSubAddress ,
DisconnectCause 1C  , DisconnectText invalid number (28), ConnectTime
17:34:34.682 CST Tue Feb 2 2010, DisconnectTime 17:34:34.682 CST Tue Feb
2 2010, CallOrigin 2, ChargedUnits 0, InfoType 2, TransmitPackets 0,
TransmitBytes 0, ReceivePackets 0, ReceiveBytes 0


-- 
John Lange
http://www.johnlange.ca



------------------------------

Message: 42
Date: Tue, 2 Feb 2010 17:54:13 -0600
From: Kevin Dunn <cheesevoice at gmail.com>
To: Wes Sisk <wsisk at cisco.com>
Cc: "Norton, Mike" <mikenorton at pwsd76.ab.ca>,	Cisco Voice
	<cisco-voip at puck.nether.net>
Subject: Re: [cisco-voip] MOH Beeping
Message-ID:
	<84b74aec1002021554ma535a55pa83f682c04c8f305 at mail.gmail.com>
Content-Type: text/plain; charset="windows-1252"

I will give it a shot

Thanks Les

On Tue, Feb 2, 2010 at 5:37 PM, Wes Sisk <wsisk at cisco.com> wrote:

> Thanks Kevin,
>
> So:
> noise is not present recording from USB source to your laptop.
> noise is present even in the packets transmitted over the wire to the
phone
> when USB device is connected to server.
>
> Based on the nature of sound I agree with Mike that it sounds like
> electrical interference.  Are the server and rack grounded?  When you
> captured "good.au" what source of power and and ground was your laptop
> using?
>
> Do you have another analog source such as an iPod or mp3 player that
would
> use a different, or possibly floating ground, to test with?
>
> You have eliminated most of the devices in the audio path. The only
way to
> isolate further inside the server is to use a root account to dump
audio
> directly off the linux DSP device before the IPVMSApp kernel driver
attempts
> to packetize it for transmission over the network.  If noise is
present in
> that sample then it is either introduced by electrical interface
between the
> source and USB device or introduced in the USB device itself.
>
> /Wes
>
>
>
> On Tuesday, February 02, 2010 5:17:53 PM, Norton, Mike
> <mikenorton at pwsd76.ab.ca> <mikenorton at pwsd76.ab.ca> wrote:
>
>  Listening to the capture, it sounds like ambient electrical/RF noise
is
> getting into the analog audio signal. I?m surprised this isn?t a more
common
> problem. Server racks are probably one of the least ideal places to
try to
> run unbalanced analog audio signals.
>
>
>
> --
>
> Mike Norton
>
> I.T. Support
>
> Peace Wapiti School Division No. 76
>
> Helpdesk: 780-831-3080
>
> Direct: 780-831-3076
>
>
>
>
>
> *From:* cisco-voip-bounces at puck.nether.net [
>
mailto:cisco-voip-bounces at puck.nether.net<cisco-voip-bounces at puck.nether
.net>]
> *On Behalf Of *Kevin Dunn
> *Sent:* February-02-10 1:30 PM
> *To:* Wes Sisk
> *Cc:* Cisco Voice
> *Subject:* Re: [cisco-voip] MOH Beeping
>
>
>
> Sev3 SR 613458017 Udhay has been pretty helpful so far checking the
> configuration.
>
> the sniff files are associated with that case,m you should be able to
see
> them (otherwise I can add them in)
>
>
>
> User experience:
>
>
>
> I call my boss, he answers by pressing speaker button.
>
> I place him on hold
>
> He hears the "good.au" sound I enclosed for Peter to listen to
>
>
>
> the sound is not unlike rapid repeated whistling...
>
> I should note that when I use the handset I don't hear it, but when I
> capture the packets it is still there, just not as loud
>
>
>
> Kevin
>
> On Tue, Feb 2, 2010 at 2:07 PM, Wes Sisk <wsisk at cisco.com> wrote:
>
> In the very abstract this sounds like call waiting tone.  Can you
clarify
> exactly what the user is doing when "caller is placed on hold and the
> speaker is activated"?  What buttons is the user pressing to
accomplish
> this?
>
> concurrent detailed CCM and SDL traces and a packet capture of all
traffic
> to/from the phone could be used to isolate the source of this tone
rather
> quickly.  You mention a TAC case, what is the number?
>
> /Wes
>
>
>
>
>
> On Tuesday, February 02, 2010 12:53:17 PM, Kevin Dunn
> <cheesevoice at gmail.com> <cheesevoice at gmail.com> wrote:
>
>   Okay I have a TAC case open and I have tried changing configuration
> settings, cables and hardware...
>
>
>
> CUCM 7.0.2.2000-5
>
> Fixed audio from XM radio (MOH-USB-AUDIO) card
>
>
>
> when a caller is placed on hold and the speaker is activated there is
an
> audible (and quite annoying) beeping sound playing over the top of the
audio
> file.
>
>
>
> It is not audible on the handset or headset.
>
>
>
> If I sniff the phone port and capture the audio file I can hear it.
>
>
>
> If I record the audio file with my laptop (plugging mic cord into
lappy
> instead of MOH-USB) there is no beeping.
>
> In my mind that eliminates the XM radio and cables, I have changed out
> cables though, just in case.
>
>
>
> I changes out MOH-USB cards and that also did nothing to eliminate the
> issue.
>
>
>
> I have upgraded firmware on the phones and that wasn't it either.
>
> With the Sample Audio file (which is JAZZY) there is no beeping
regardless
> of handset or speaker.
>
>
>
> Any suggestions?
>
> ------------------------------
>
>
>
>
> _______________________________________________
>
> cisco-voip mailing list
>
> cisco-voip at puck.nether.net
>
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
>
>
>
>
> ------------------------------
>
> _______________________________________________
> cisco-voip mailing
listcisco-voip at puck.nether.nethttps://puck.nether.net/mailman/listinfo/c
isco-voip
>
>
>
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------------------------------

Message: 43
Date: Tue, 02 Feb 2010 18:37:14 -0500
From: Wes Sisk <wsisk at cisco.com>
To: Kevin Dunn <cheesevoice at gmail.com>
Cc: "Norton, Mike" <mikenorton at pwsd76.ab.ca>,	Cisco Voice
	<cisco-voip at puck.nether.net>
Subject: Re: [cisco-voip] MOH Beeping
Message-ID: <4B68B72A.7000605 at cisco.com>
Content-Type: text/plain; charset="iso-8859-1"; Format="flowed"

Thanks Kevin,

So:
noise is not present recording from USB source to your laptop.
noise is present even in the packets transmitted over the wire to the 
phone when USB device is connected to server.

Based on the nature of sound I agree with Mike that it sounds like 
electrical interference.  Are the server and rack grounded?  When you 
captured "good.au" what source of power and and ground was your laptop 
using?

Do you have another analog source such as an iPod or mp3 player that 
would use a different, or possibly floating ground, to test with?

You have eliminated most of the devices in the audio path. The only way 
to isolate further inside the server is to use a root account to dump 
audio directly off the linux DSP device before the IPVMSApp kernel 
driver attempts to packetize it for transmission over the network.  If 
noise is present in that sample then it is either introduced by 
electrical interface between the source and USB device or introduced in 
the USB device itself.

/Wes


On Tuesday, February 02, 2010 5:17:53 PM, Norton, Mike 
<mikenorton at pwsd76.ab.ca> wrote:
>
> Listening to the capture, it sounds like ambient electrical/RF noise 
> is getting into the analog audio signal. I'm surprised this isn't a 
> more common problem. Server racks are probably one of the least ideal 
> places to try to run unbalanced analog audio signals.
>
>  
>
> -- 
>
> Mike Norton
>
> I.T. Support
>
> Peace Wapiti School Division No. 76
>
> Helpdesk: 780-831-3080
>
> Direct: 780-831-3076
>
>  
>
>  
>
> *From:* cisco-voip-bounces at puck.nether.net 
> [mailto:cisco-voip-bounces at puck.nether.net] *On Behalf Of *Kevin Dunn
> *Sent:* February-02-10 1:30 PM
> *To:* Wes Sisk
> *Cc:* Cisco Voice
> *Subject:* Re: [cisco-voip] MOH Beeping
>
>  
>
> Sev3 SR 613458017 Udhay has been pretty helpful so far checking the 
> configuration.
>
> the sniff files are associated with that case,m you should be able to 
> see them (otherwise I can add them in)
>
>  
>
> User experience:
>
>  
>
> I call my boss, he answers by pressing speaker button.
>
> I place him on hold
>
> He hears the "good.au" sound I enclosed for Peter to listen to
>
>  
>
> the sound is not unlike rapid repeated whistling...
>
> I should note that when I use the handset I don't hear it, but when I 
> capture the packets it is still there, just not as loud
>
>  
>
> Kevin
>
> On Tue, Feb 2, 2010 at 2:07 PM, Wes Sisk <wsisk at cisco.com 
> <mailto:wsisk at cisco.com>> wrote:
>
> In the very abstract this sounds like call waiting tone.  Can you 
> clarify exactly what the user is doing when "caller is placed on hold 
> and the speaker is activated"?  What buttons is the user pressing to 
> accomplish this?
>
> concurrent detailed CCM and SDL traces and a packet capture of all 
> traffic to/from the phone could be used to isolate the source of this 
> tone rather quickly.  You mention a TAC case, what is the number?
>
> /Wes
>
>
>
>
>
> On Tuesday, February 02, 2010 12:53:17 PM, Kevin Dunn 
> <cheesevoice at gmail.com> <mailto:cheesevoice at gmail.com> wrote:
>
>     Okay I have a TAC case open and I have tried changing
>     configuration settings, cables and hardware...
>
>      
>
>     CUCM 7.0.2.2000-5
>
>     Fixed audio from XM radio (MOH-USB-AUDIO) card
>
>      
>
>     when a caller is placed on hold and the speaker is activated there
>     is an audible (and quite annoying) beeping sound playing over the
>     top of the audio file.
>
>      
>
>     It is not audible on the handset or headset.
>
>      
>
>     If I sniff the phone port and capture the audio file I can hear
it.
>
>      
>
>     If I record the audio file with my laptop (plugging mic cord into
>     lappy instead of MOH-USB) there is no beeping.
>
>     In my mind that eliminates the XM radio and cables, I have changed
>     out cables though, just in case.
>
>      
>
>     I changes out MOH-USB cards and that also did nothing to eliminate
>     the issue.
>
>      
>
>     I have upgraded firmware on the phones and that wasn't it either.
>
>     With the Sample Audio file (which is JAZZY) there is no beeping
>     regardless of handset or speaker.
>
>      
>
>     Any suggestions?
>
>
------------------------------------------------------------------------
>
>
>         
>
>      
>
>     _______________________________________________
>
>     cisco-voip mailing list
>
>     cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>
>
>     https://puck.nether.net/mailman/listinfo/cisco-voip
>
>       
>
>  
>
>  
>
>
------------------------------------------------------------------------
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>   

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------------------------------

Message: 44
Date: Tue, 02 Feb 2010 20:27:56 -0600
From: John Lange <john at johnlange.ca>
To: "'Cisco VOIP'" <cisco-voip at puck.nether.net>
Subject: Re: [cisco-voip] Internaltional Calls Failing at the Bell PRI
	- SOLVED
Message-ID: <1265164076.10432.308.camel at linux-k6vx.site>
Content-Type: text/plain; charset="UTF-8"

Just to finish off my own thread; after a lot of work, I was finally
able to get the international dialing working on our Cisco VOIP Gateway.

Specifically I implemented a new dial peer just for international
dailing which forces the dialing plan and type to 'unknown' instead of
the cisco default which is 'international' (which doesn't work with
Bell).

Here are the relevant parts:

voice translation-rule 1
rule 1 // // type any unknown plan any unknown
!
voice translation-profile isdn_map
translate called 1
!
dial-peer voice 25 pots
translation-profile outgoing isdn_map
destination-pattern 9011T
direct-inward-dial
port 0/3/0:23
prefix 011
!         

---
As a side note I have no idea why it wouldn't work when my rule said:

rule 1/^9/ //

But anyhow, the above is now working.

Thanks to everyone who made suggestions.

-- 
John Lange
http://www.johnlange.ca



------------------------------

Message: 45
Date: Tue, 2 Feb 2010 20:37:46 -0600
From: "Go0se" <me at go0se.com>
To: "'John Lange'" <john at johnlange.ca>
Cc: 'Cisco VOIP' <cisco-voip at puck.nether.net>
Subject: Re: [cisco-voip] Internaltional Calls Failing at the Bell PRI
Message-ID: <001101caa479$dfff9eb0$9ffedc10$@com>
Content-Type: text/plain;	charset="UTF-8"

You do it on the serial interface with an "isdn map address" command.
The below config is off of a working CME:

!
interface Serial0/0/0:23
 no ip address
 encapsulation hdlc
 isdn switch-type primary-ni
 isdn incoming-voice voice
 isdn map address ^011* plan unknown type unknown
 no cdp enable
!

-Go0se

-----Original Message-----
From: John Lange [mailto:john at johnlange.ca] 
Sent: Tuesday, February 02, 2010 5:45 PM
To: Go0se
Cc: 'Cisco VOIP'
Subject: RE: [cisco-voip] Internaltional Calls Failing at the Bell PRI

On Tue, 2010-02-02 at 15:43 -0600, Go0se wrote:
> If it's connecting to Bell, try setting the called type and plan both
to
> UNKNOWN.

I did speak with Bell tech and though they were not able to take a look
at the D channel, he said pretty much the same thing but how do you do
that?

Set both the called _and_ calling party, type & plan to unknown. So I
tried:

dial-peer voice 25 pots
 numbering-type unknown
 destination-pattern 9011
 direct-inward-dial
 port 0/3/0:23
 forward-digits 0
 prefix 011

But, so far as I can tell, "numbering-type" is for matching peers, not
for changing the numbering type and I can't see any way to change the
numbering type or plan in the dial-peer except with a translation rule.

But I can not get my translation rule to work...

---

voice translation-rule 1
 rule 1 /^9/ /9/ type any unknown plan any unknown

dial-peer voice 25 pots
 destination-pattern 9011
 prefix 011
 direct-inward-dial
 port 0/3/0:23
 forward-digits extra
 translate-outgoing called 1
!

-- debug --

The last few lines of the debug tell the tail..

Called Party Number i = 0x91, '011448712002000' 
 Plan:ISDN, Type:International

If I could just get the translation to work so I could set the plan &
type, I think the problem would be solved...

--- and sorry for the long post but here is the full debug isdn &
dialpeer ----


Feb  2 23:34:34.414: //-1/5A96E357B8A8/DPM/dpAssociateIncomingPeerCore:
   Calling Number=2049757113, Called Number=9011448712002000,
Voice-Interface=0x0,
   Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search
Type=PEER_TYPE_VOICE,
   Peer Info Type=DIALPEER_INFO_SPEECH
Feb  2 23:34:34.414: //-1/5A96E357B8A8/DPM/dpAssociateIncomingPeerCore:
   Match Rule=DP_MATCH_REQUEST_URI;
URI=sip:9011448712002000 at 192.168.134.20
Feb  2 23:34:34.414: //-1/5A96E357B8A8/DPM/dpMatchPeertype:
   Is Incoming=TRUE, Number Expansion=FALSE
Feb  2 23:34:34.414: //-1/5A96E357B8A8/DPM/dpMatchCore:
   Dial String=, Expanded String=, Calling Number=
   Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
Feb  2 23:34:34.414: //-1/5A96E357B8A8/DPM/dpMatchCore:
   Result=-1
Feb  2 23:34:34.414: //-1/5A96E357B8A8/DPM/dpMatchPeertype:exit at 5404 
Feb  2 23:34:34.414: //-1/5A96E357B8A8/DPM/dpAssociateIncomingPeerCore:
   Match Rule=DP_MATCH_TO_URI; URI=sip:9011448712002000 at 192.168.134.20
Feb  2 23:34:34.414: //-1/5A96E357B8A8/DPM/dpMatchPeertype:
   Is Incoming=TRUE, Number Expansion=FALSE
Feb  2 23:34:34.414: //-1/5A96E357B8A8/DPM/dpMatchCore:
   Dial String=, Expanded String=, Calling Number=
   Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
Feb  2 23:34:34.414: //-1/5A96E357B8A8/DPM/dpMatchCore:
   Result=-1
Feb  2 23:34:34.414: //-1/5A96E357B8A8/DPM/dpMatchPeertype:exit at 5404 
Feb  2 23:34:34.414: //-1/5A96E357B8A8/DPM/dpAssociateIncomingPeerCore:
   Match Rule=DP_MATCH_FROM_URI; URI=sip:2049757113 at 192.168.134.9
Feb  2 23:34:34.414: //-1/5A96E357B8A8/DPM/dpMatchPeertype:
   Is Incoming=TRUE, Number Expansion=FALSE
Feb  2 23:34:34.414: //-1/5A96E357B8A8/DPM/dpMatchCore:
   Dial String=, Expanded String=, Calling Number=
   Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
Feb  2 23:34:34.414: //-1/5A96E357B8A8/DPM/dpMatchCore:
   Result=-1
Feb  2 23:34:34.414: //-1/5A96E357B8A8/DPM/dpMatchPeertype:exit at 5404 
Feb  2 23:34:34.414: //-1/5A96E357B8A8/DPM/dpAssociateIncomingPeerCore:
   Match Rule=DP_MATCH_INCOMING_DNIS; Called Number=9011448712002000
Feb  2 23:34:34.414: //-1/5A96E357B8A8/DPM/dpMatchPeertype:
   Is Incoming=TRUE, Number Expansion=FALSE
Feb  2 23:34:34.414: //-1/5A96E357B8A8/DPM/dpMatchCore:
   Dial String=9011448712002000, Expanded String=9011448712002000,
Calling Number=
   Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
Feb  2 23:34:34.414: //-1/5A96E357B8A8/DPM/dpMatchCore:
   Result=-1
Feb  2 23:34:34.414: //-1/5A96E357B8A8/DPM/dpMatchPeertype:exit at 5404 
Feb  2 23:34:34.414: //-1/5A96E357B8A8/DPM/dpAssociateIncomingPeerCore:
   Match Rule=DP_MATCH_ANSWER; Calling Number=2049757113
Feb  2 23:34:34.414: //-1/5A96E357B8A8/DPM/dpMatchPeertype:
   Is Incoming=TRUE, Number Expansion=FALSE
Feb  2 23:34:34.414: //-1/5A96E357B8A8/DPM/dpMatchCore:
   Dial String=, Expanded String=, Calling Number=2049757113T
   Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
Feb  2 23:34:34.414: //-1/5A96E357B8A8/DPM/dpMatchCore:
   Result=-1
Feb  2 23:34:34.414: //-1/5A96E357B8A8/DPM/dpMatchPeertype:exit at 5404 
Feb  2 23:34:34.414: //-1/5A96E357B8A8/DPM/dpAssociateIncomingPeerCore:
   Match Rule=DP_MATCH_ORIGINATE; Calling Number=2049757113
Feb  2 23:34:34.414: //-1/5A96E357B8A8/DPM/dpMatchPeertype:
   Is Incoming=TRUE, Number Expansion=FALSE
Feb  2 23:34:34.414: //-1/5A96E357B8A8/DPM/dpMatchCore:
   Dial String=, Expanded String=, Calling Number=2049757113T
   Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
Feb  2 23:34:34.418: //-1/5A96E357B8A8/DPM/MatchNextPeer:
   Result=Success(0); Incoming Dial-peer=40 Is Matched
Feb  2 23:34:34.418: //-1/5A96E357B8A8/DPM/MatchNextPeer:
   Result=Success(0); Incoming Dial-peer=45 Is Matched
Feb  2 23:34:34.418: //-1/5A96E357B8A8/DPM/dpMatchPeertype:exit at 5404 
Feb  2 23:34:34.418: //-1/5A96E357B8A8/DPM/dpAssociateIncomingPeerCore:
   Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=40
Feb  2 23:34:34.418:
//-1/5A96E357B8A8/DPM/dpAssociateIncomingPeerSPI:exit at 5938 
Feb  2 23:34:34.422: //-1/5A96E357B8A8/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=9011448712002000, Peer Info
Type=DIALPEER_INFO_SPEECH
Feb  2 23:34:34.422: //-1/5A96E357B8A8/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=9011448712002000
Feb  2 23:34:34.422: //-1/5A96E357B8A8/DPM/dpMatchCore:
   Dial String=9011448712002000, Expanded String=9011448712002000,
Calling Number=
   Timeout=TRUE, Is Incoming=FALSE, Peer Info Type=DIALPEER_INFO_SPEECH
Feb  2 23:34:34.422: //-1/5A96E357B8A8/DPM/MatchNextPeer:
   Result=Success(0); Outgoing Dial-peer=20 Is Matched
Feb  2 23:34:34.422: //-1/5A96E357B8A8/DPM/MatchNextPeer:
   Result=Success(0); Outgoing Dial-peer=25 Is Matched
Feb  2 23:34:34.422: //-1/5A96E357B8A8/DPM/dpMatchPeersCore:
   Result=Success(0) after DP_MATCH_DEST
Feb  2 23:34:34.422: //-1/5A96E357B8A8/DPM/dpMatchPeersMoreArg:
   Result=SUCCESS(0) 
   List of Matched Outgoing Dial-peer(s): 
     1: Dial-peer Tag=25 
     2: Dial-peer Tag=20
Feb  2 23:34:34.422: ISDN Se0/3/0:23 Q931: Applying typeplan for sw-type
0xD is 0x2 0x1, Calling num 2049757113
Feb  2 23:34:34.426: ISDN Se0/3/0:23 Q931: Applying typeplan for sw-type
0xD is 0x1 0x1, Called num 011448712002000
Feb  2 23:34:34.426: ISDN Se0/3/0:23 Q931: TX -> SETUP pd = 8  callref =
0x3244 
	Bearer Capability i = 0x8090A2 
		Standard = CCITT 
		Transfer Capability = Speech  
		Transfer Mode = Circuit 
		Transfer Rate = 64 kbit/s 
	Channel ID i = 0xA98395 
		Exclusive, Channel 21 
	Calling Party Number i = 0x2180, '2049757113' 
		Plan:ISDN, Type:National 
	Called Party Number i = 0x91, '011448712002000' 
		Plan:ISDN, Type:International
Feb  2 23:34:34.526: ISDN Se0/3/0:23 Q931: RX <- RELEASE_COMP pd = 8
callref = 0xB244 
	Cause i = 0x809C - Invalid number format (incomplete number)
Feb  2 17:34:34: %VOIPAAA-5-VOIP_CALL_HISTORY: CallLegType 1,
ConnectionId 5A96E357 F8A11DF B8A89757 A2E40164, SetupTime 17:34:34.530
CST Tue Feb 2 2010, PeerAddress 9011448712002000, PeerSubAddress ,
DisconnectCause 1C  , DisconnectText invalid number (28), ConnectTime
17:34:34.530 CST Tue Feb 2 2010, DisconnectTime 17:34:34.530 CST Tue Feb
2 2010, CallOrigin 1, ChargedUnits 0, InfoType 2, TransmitPackets 0,
TransmitBytes 0, ReceivePackets 0, ReceiveBytes 0
Feb  2 23:34:34.534: ISDN Se0/3/0:23 Q931: Applying typeplan for sw-type
0xD is 0x2 0x1, Calling num 2049757113
Feb  2 23:34:34.534: ISDN Se0/3/0:23 Q931: Applying typeplan for sw-type
0xD is 0x1 0x1, Called num 011448712002000
Feb  2 23:34:34.534: ISDN Se0/3/0:23 Q931: TX -> SETUP pd = 8  callref =
0x3245 
	Bearer Capability i = 0x8090A2 
		Standard = CCITT 
		Transfer Capability = Speech  
		Transfer Mode = Circuit 
		Transfer Rate = 64 kbit/s 
	Channel ID i = 0xA98395 
		Exclusive, Channel 21 
	Calling Party Number i = 0x2180, '2049757113' 
		Plan:ISDN, Type:National 
	Called Party Number i = 0x91, '011448712002000' 
		Plan:ISDN, Type:International
Feb  2 23:34:34.666: ISDN Se0/3/0:23 Q931: RX <- RELEASE_COMP pd = 8
callref = 0xB245 
	Cause i = 0x809C - Invalid number format (incomplete number)
Feb  2 17:34:34: %VOIPAAA-5-VOIP_CALL_HISTORY: CallLegType 1,
ConnectionId 5A96E357 F8A11DF B8A89757 A2E40164, SetupTime 17:34:34.670
CST Tue Feb 2 2010, PeerAddress 9011448712002000, PeerSubAddress ,
DisconnectCause 1C  , DisconnectText invalid number (28), ConnectTime
17:34:34.670 CST Tue Feb 2 2010, DisconnectTime 17:34:34.670 CST Tue Feb
2 2010, CallOrigin 1, ChargedUnits 0, InfoType 2, TransmitPackets 0,
TransmitBytes 0, ReceivePackets 0, ReceiveBytes 0
Feb  2 17:34:34: %VOIPAAA-5-VOIP_CALL_HISTORY: CallLegType 2,
ConnectionId 5A96E357 F8A11DF B8A89757 A2E40164, SetupTime 17:34:34.422
CST Tue Feb 2 2010, PeerAddress 2049757113, PeerSubAddress ,
DisconnectCause 1C  , DisconnectText invalid number (28), ConnectTime
17:34:34.682 CST Tue Feb 2 2010, DisconnectTime 17:34:34.682 CST Tue Feb
2 2010, CallOrigin 2, ChargedUnits 0, InfoType 2, TransmitPackets 0,
TransmitBytes 0, ReceivePackets 0, ReceiveBytes 0


-- 
John Lange
http://www.johnlange.ca



------------------------------

Message: 46
Date: Tue, 2 Feb 2010 23:16:34 -0500
From: "Lawrence E. Bakst" <ml at iridescent.org>
To: Peter Slow <peter.slow at gmail.com>, Abebe Amare <abucho at gmail.com>
Cc: cisco voip <cisco-voip at puck.nether.net>
Subject: Re: [cisco-voip] cisco 7941 switch port physical problem
Message-ID: <p06240810c78ea8aa98f2@[192.168.0.7]>
Content-Type: text/plain; charset="us-ascii"

Does the plastic tab on the RJ-45 connector click into the socket? Some
cables with anti-snag connectors are a very tight fit. The ones I got at
Fry's you really have to push it very hard before it clicks in.


At 1:19 PM -0500 2/2/10, Peter Slow wrote:
>If you connect a different phone of the same type to the same port
>with the same cable, does it turn on?
>
>if yes, then RMA the first phone.
>
>On Tue, Feb 2, 2010 at 9:43 AM, Abebe Amare <abucho at gmail.com> wrote:
>> Hi,
>>
>> I am having problem with a new out of the box Cisco 7941 IP phone.
When I
>> connect the switch port to a port on the PoE switch with a patch cord
it is
>> not powering up. If I hold the cable tightly in place with my hand or
put
>> something to hold the cable in place it will power up. I have changed
the
>> cable several times but it won't work unless I hold it firmly. Does
this
>> mean the switch port on the phone is damaged? do I need to process
RMA?
>>
>> Thanks in advance.
>>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>>
>_______________________________________________
>cisco-voip mailing list
>cisco-voip at puck.nether.net
>https://puck.nether.net/mailman/listinfo/cisco-voip


-- 
leb at iridescent.org



------------------------------

Message: 47
Date: Wed, 3 Feb 2010 05:30:32 +0000
From: shary shary <shaary1 at hotmail.com>
To: <jbf005 at shsu.edu>, <tanner.ezell at gmail.com>
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Agent Status in UCCX
Message-ID: <SNT127-W156CB4BA0A436B5C7AE30AEE560 at phx.gbl>
Content-Type: text/plain; charset="iso-8859-1"



yes Tanner you got my point. I want this but how could i do this i did
it once the call gets queued but i want to take decision before queuing
call whether the agent is log in or not if not then transfer the call to
another queue.


 
> From: JBF005 at shsu.edu
> To: tanner.ezell at gmail.com; shaary1 at hotmail.com
> CC: cisco-voip at puck.nether.net
> Date: Tue, 2 Feb 2010 08:38:45 -0600
> Subject: RE: [cisco-voip] Agent Status in UCCX
> 
> Report statistic can accomplish this, but I think what you want to do
is queue the call twice. Within your first queue just queue it again to
the second queue and whoever becomes available first will get the call.
 		 	   		  
_________________________________________________________________
Got a cool Hotmail story? Tell us now
http://clk.atdmt.com/UKM/go/195013117/direct/01/
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Message: 48
Date: Wed, 3 Feb 2010 00:33:55 -0500
From: Tanner Ezell <tanner.ezell at gmail.com>
To: shary shary <shaary1 at hotmail.com>
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Agent Status in UCCX
Message-ID:
	<9c4f122d1002022133p3ff8fe3fl89d15740acda624d at mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1

Use the Get Reporting Statistics Step, first box should be CSQ
Resources or something on that order, then "logged in resources" then
supply the CSQ name, and a return variable, then you can test against
that.

On Wed, Feb 3, 2010 at 12:30 AM, shary shary <shaary1 at hotmail.com>
wrote:
>
> yes Tanner you?got my point. I want this but how could i do this i did
it
> once the?call gets queued but i want?to take decision before queuing
call
> whether the agent is log in or not if not then transfer the call to
another
> queue.
>
>
>
>> From: JBF005 at shsu.edu
>> To: tanner.ezell at gmail.com; shaary1 at hotmail.com
>> CC: cisco-voip at puck.nether.net
>> Date: Tue, 2 Feb 2010 08:38:45 -0600
>> Subject: RE: [cisco-voip] Agent Status in UCCX
>>
>> Report statistic can accomplish this, but I think what you want to do
is
>> queue the call twice. Within your first queue just queue it again to
the
>> second queue and whoever becomes available first will get the call.
>
> ________________________________
> Got a cool Hotmail story? Tell us now



-- 
Regards,
Tanner Ezell


------------------------------

Message: 49
Date: Wed, 3 Feb 2010 09:29:14 +0300
From: Abebe Amare <abucho at gmail.com>
To: Peter Slow <peter.slow at gmail.com>
Cc: cisco voip <cisco-voip at puck.nether.net>
Subject: Re: [cisco-voip] cisco 7941 switch port physical problem
Message-ID:
	<bdd7c6271002022229m237d6fb4o9940618eed9f4b81 at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"

Hi Peter,

I have tried with another same model phone with the same cable on the
same
switch port and it works fine. I guess it has to be replaced. Thanks for
your help.

best regards,

Abebe Amare
Network Engineer - VivaCell

On Tue, Feb 2, 2010 at 9:19 PM, Peter Slow <peter.slow at gmail.com> wrote:

> If you connect a different phone of the same type to the same port
> with the same cable, does it turn on?
>
> if yes, then RMA the first phone.
>
> On Tue, Feb 2, 2010 at 9:43 AM, Abebe Amare <abucho at gmail.com> wrote:
> > Hi,
> >
> > I am having problem with a new out of the box Cisco 7941 IP phone.
When I
> > connect the switch port to a port on the PoE switch with a patch
cord it
> is
> > not powering up. If I hold the cable tightly in place with my hand
or put
> > something to hold the cable in place it will power up. I have
changed the
> > cable several times but it won't work unless I hold it firmly. Does
this
> > mean the switch port on the phone is damaged? do I need to process
RMA?
> >
> > Thanks in advance.
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> >
>
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Message: 50
Date: Wed, 3 Feb 2010 09:31:30 +0300
From: Abebe Amare <abucho at gmail.com>
To: "Lawrence E. Bakst" <ml at iridescent.org>
Cc: cisco voip <cisco-voip at puck.nether.net>
Subject: Re: [cisco-voip] cisco 7941 switch port physical problem
Message-ID:
	<bdd7c6271002022231n39d73009gb3473c0e20879055 at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"

Hi ,

Yes the plastic tab clicks on the socket but somehow it feels different
than
on other phones.

best regards,

Abebe Amare
Network Engineer - VivaCell

On Wed, Feb 3, 2010 at 7:16 AM, Lawrence E. Bakst <ml at iridescent.org>
wrote:

> Does the plastic tab on the RJ-45 connector click into the socket?
Some
> cables with anti-snag connectors are a very tight fit. The ones I got
at
> Fry's you really have to push it very hard before it clicks in.
>
>
> At 1:19 PM -0500 2/2/10, Peter Slow wrote:
> >If you connect a different phone of the same type to the same port
> >with the same cable, does it turn on?
> >
> >if yes, then RMA the first phone.
> >
> >On Tue, Feb 2, 2010 at 9:43 AM, Abebe Amare <abucho at gmail.com> wrote:
> >> Hi,
> >>
> >> I am having problem with a new out of the box Cisco 7941 IP phone.
When
> I
> >> connect the switch port to a port on the PoE switch with a patch
cord it
> is
> >> not powering up. If I hold the cable tightly in place with my hand
or
> put
> >> something to hold the cable in place it will power up. I have
changed
> the
> >> cable several times but it won't work unless I hold it firmly. Does
this
> >> mean the switch port on the phone is damaged? do I need to process
RMA?
> >>
> >> Thanks in advance.
> >>
> >> _______________________________________________
> >> cisco-voip mailing list
> >> cisco-voip at puck.nether.net
> >> https://puck.nether.net/mailman/listinfo/cisco-voip
> >>
> >>
> >_______________________________________________
> >cisco-voip mailing list
> >cisco-voip at puck.nether.net
> >https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
> --
> leb at iridescent.org
>
>
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Message: 51
Date: Wed, 3 Feb 2010 09:17:19 +0200
From: Ali El Moussaoui <mousawi.ali at gmail.com>
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] JTAPI user disappeared?
Message-ID:
	<5e5306941002022317i6c0155a0h8408f14f2635a757 at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"

Hello,

I have weired behavior on my system .. its IPCCX 4.0 with CM 4.2 . When
i do
Check resync under the JTAPI system it tells me that the jtapi user does
not
exist and the ports are not associated.. I click on synch it says jtapi
user
created and ports associated...

I checked users in call manager but its not there ... re check on IPCCX
same
thing again !!! I tried to creat the JTAPI user and error occured with
no
explanation or any details!!

I am not IPCCX expert any feed back on how to troubleshoot is highly
appreciated.

Note that everyting is wokring fine but i can not add any new control
group.

Regards,
Ali
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Message: 52
Date: Wed, 3 Feb 2010 10:57:14 +0200
From: Dew Swen <dew.swen at gmail.com>
To: "Girard, Jeffrey COL MIL USA" <jeffrey.girard at us.army.mil>
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Command for determining which dial-peer will
	be	usedon router? (UNCLASSIFIED)
Message-ID:
	<ae5778961002030057u7147cd39wbc077ed4c20e35df at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"

sh voice call status
sh call active voice brief

-
Dew Swen


On Tue, Feb 2, 2010 at 7:33 PM, Girard, Jeffrey COL MIL USA <
jeffrey.girard at us.army.mil> wrote:

> Classification: UNCLASSIFIED
> Caveats: FOUO
>
> Show voice dialpeer
>
> -----------------------------------------------------------
> Jeffrey T. Girard ("Jeff")
> COL, 53
> Future Forces Integration Directorate (FFID), Deputy - Networks
> office:  (915)568-1240  DSN 978
> Mobile:  (915)727-4222
> reply to:  jeffrey.girard at us.army.mil
>
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Robert
> Kulagowski
> Sent: Tuesday, February 02, 2010 9:41 AM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] Command for determining which dial-peer will be
> usedon router?
>
> I thought I saw in this list that there's a way to have the router
show
> you
> which dial-peer is going to be used based on a particular number
> pattern.
> Sort of what you see when doing a "deb voip ccapi inout" when it shows
> you
> the list of possible "outgoing dial peer=" values.
>
> I'm not trying to test the voice translation-pattern stuff, but the
> actual
> DP candidates.
>
> Thanks.
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> Classification: UNCLASSIFIED
> Caveats: FOUO
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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Message: 53
Date: Wed, 3 Feb 2010 13:13:33 +0400
From: haroon rasheed <hrasheed at istnetworks.com>
To: Dew Swen <dew.swen at gmail.com>
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Command for determining which dial-peer will
	be	usedon router? (UNCLASSIFIED)
Message-ID:
	<23423c7a1002030113pe07abe8wadd8d2ce328723a5 at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"

try

debug voice dialpeer inout




On Wed, Feb 3, 2010 at 12:57 PM, Dew Swen <dew.swen at gmail.com> wrote:

> sh voice call status
> sh call active voice brief
>
> -
> Dew Swen
>
>
> On Tue, Feb 2, 2010 at 7:33 PM, Girard, Jeffrey COL MIL USA <
> jeffrey.girard at us.army.mil> wrote:
>
>> Classification: UNCLASSIFIED
>> Caveats: FOUO
>>
>> Show voice dialpeer
>>
>> -----------------------------------------------------------
>> Jeffrey T. Girard ("Jeff")
>> COL, 53
>> Future Forces Integration Directorate (FFID), Deputy - Networks
>> office:  (915)568-1240  DSN 978
>> Mobile:  (915)727-4222
>> reply to:  jeffrey.girard at us.army.mil
>>
>>
>> -----Original Message-----
>> From: cisco-voip-bounces at puck.nether.net
>> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Robert
>> Kulagowski
>> Sent: Tuesday, February 02, 2010 9:41 AM
>> To: cisco-voip at puck.nether.net
>> Subject: [cisco-voip] Command for determining which dial-peer will be
>> usedon router?
>>
>> I thought I saw in this list that there's a way to have the router
show
>> you
>> which dial-peer is going to be used based on a particular number
>> pattern.
>> Sort of what you see when doing a "deb voip ccapi inout" when it
shows
>> you
>> the list of possible "outgoing dial peer=" values.
>>
>> I'm not trying to test the voice translation-pattern stuff, but the
>> actual
>> DP candidates.
>>
>> Thanks.
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>> Classification: UNCLASSIFIED
>> Caveats: FOUO
>>
>>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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Message: 54
Date: Wed, 3 Feb 2010 13:29:12 +0000
From: Stephen Greszczyszyn <sgreszcz at gmail.com>
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] MGCP to PSTN - 0x80A9 - Temporary failure?
Message-ID:
	<71601cd61002030529u6577c842rd38159cc8ac6bb9b at mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1

Hi there,

I'm working on a lab scenario where I'm making calls from CUCM7 via an
IOS MGCP gateway out to another router simulating the PSTN.  The call
connects OK, but after about 1 minute - 2 minutes, the call drops.  I
have tried the same test through another H.323 gateway and the call
stays up fine.  Running 12.4(24)T2 on the MGCP gateway.  Here is the
information:

MGCP gateway:
Feb  3 11:53:33.851: ISDN Se0/1/0:15 Q931: TX -> DISCONNECT pd = 8
callref = 0x0007
       Cause i = 0x80A9 - Temporary failure

PSTN:
*Feb  3 11:59:57.011: ISDN Se0/3/0:15 Q931: RX <- DISCONNECT pd = 8
callref = 0x0007
       Cause i = 0x80A9 - Temporary failure
*Feb  3 11:59:57.015: %ISDN-6-DISCONNECT: Interface Serial0/3/0:0
disconnected from +16178631002 , call lasted 72 seconds

I did some searching on this list and there were some references to
faulty DSPs.  Mine seem to look OK?

MGCP:
br1#sh voice dsp group all
DSP groups on slot 0:
dsp 1:
 State: UP, firmware: 24.3.2
 Max signal/voice channel: 16/16
 Max credits: 240
 num_of_sig_chnls_allocated: 8
 Transcoding channels allocated: 0
 Group: FLEX_GROUP_VOICE, complexity: FLEX
   Shared credits: 225, reserved credits: 0
   Signaling channels allocated: 8
   Voice channels allocated: 1
   Credits used: 15
   Voice channels:
     Ch01: voice port: 0/1/0:15.1, codec: g711ulaw, credits allocated:
15

PSTN:
WAN_PSTN#sh voice dsp group all
DSP groups on slot 0:
dsp 1:
 State: UP, firmware: 24.3.2
 Max signal/voice channel: 16/16
 Max credits: 240
 num_of_sig_chnls_allocated: 16
 Transcoding channels allocated: 0
 Group: FLEX_GROUP_VOICE, complexity: FLEX
   Shared credits: 225, reserved credits: 0
   Signaling channels allocated: 16
   Voice channels allocated: 1
   Credits used: 15
   Voice channels:
     Ch01: voice port: 0/3/0:15.1, codec: g711ulaw, credits allocated:
15

I also get some weird stuff around the time of call failure when I
enable MGCP debugging:

debug mgcp errors
debug mgcp packet

It seems as though my MGCP gateway tells CUCM to drop the call and
also sends the disconnect to the PSTN:

MGCP:
Feb  3 13:09:00.557: ISDN Se0/1/0:15 Q921: User TX -> INFO sapi=0
tei=0, ns=14 nr=15
Feb  3 13:09:00.557: ISDN Se0/1/0:15 Q931: DISCONNECT pd = 8  callref =
0x0003
       Cause i = 0x80A9 - Temporary failure

PSTN:
*Feb  3 13:15:23.583: ISDN Se0/3/0:15 Q921: Net RX <- INFO sapi=0
tei=0, ns=14 nr=15
*Feb  3 13:15:23.583: ISDN Se0/3/0:15 Q931: DISCONNECT pd = 8  callref =
0x0003
       Cause i = 0x80A9 - Temporary failure

(Previous call traces - MGCP debug)

Feb  3 12:52:45.713: Following traceback is for INFO ONLY. -Traceback=
0x40F3126Cz 0x40F5BA6Cz 0x40F4296Cz 0x40F3ABC4z 0x40F333D4z
0x40EEED44z 0x40EF43F0z 0x435D69D8z 0x435D69BCz
Feb  3 12:52:45.717: MGCP Packet sent to 10.10.210.11:2427--->
DLCX 205358175 S0/SU1/DS1-0/1 at br1.proctorlabs.com MGCP 0.1
C: D000000002a512c9000000F500000002
I: 3
P: PS=5000, OS=800000, PR=4978, OR=796480, PL=22, JI=7, LA=0
E: 502
<---

Feb  3 12:52:45.725: MGCP Packet received from 10.10.210.11:2427--->
200 205358175
<---

Feb  3 12:52:45.725:
//-1/xxxxxxxxxxxx/MGCP/mgcp_mp_get_not_entity(830):[lvl=2]Invalid
parameter (pkt 0x4A428FE0 pkt->mgcp_parm_lines 0x00000000)
Feb  3 12:52:45.733: ISDN Se0/1/0:15 Q931: TX -> DISCONNECT pd = 8
callref = 0x0002
       Cause i = 0x80A9 - Temporary failure
Feb  3 12:52:45.745: ISDN Se0/1/0:15 Q931: RX <- RELEASE pd = 8
callref = 0x8002
Feb  3 12:52:45.757: ISDN Se0/1/0:15 Q931: TX -> RELEASE_COMP pd = 8
callref = 0x0002
Feb  3 12:53:04.977: MGCP Packet sent to 10.10.210.11:2427--->
NTFY 205358176 *@br1.proctorlabs.com MGCP 0.1
X: 0
O:
<---

In the CUCM traces, it isn't clear to me why the call is being dumped:

02/03/2010 12:52:45.690 CCM|MGCPHandler received msg from: 10.10.110.2
DLCX 205358175 S0/SU1/DS1-0/1 at br1.proctorlabs.com MGCP 0.1
C: D000000002a512c9000000F500000002
I: 3
P: PS=5000, OS=800000, PR=4978, OR=796480, PL=22, JI=7, LA=0
E: 502
|<CLID::StandAloneCluster><NID::10.10.210.11><CT::2,100,132,1.837><IP::1
0.10.110.2><DEV::><LVL::Significant><MASK::2000>
02/03/2010 12:52:45.690
CCM|<CLID::StandAloneCluster><NID::10.10.210.11><CT::2,100,132,1.837><MN
::MGCPEndPoint><MV::S0/SU1/DS1-0/1 at br1.proctorlabs.com><DEV::><LVL::All>
<MASK::ffff>
02/03/2010 12:52:45.691 CCM|MGCPHandler send msg SUCCESSFULLY to:
10.10.110.2
200 205358175
|<CLID::StandAloneCluster><NID::10.10.210.11><CT::2,100,132,1.837><IP::1
0.10.110.2><DEV::S0/SU1 at br1.proctorlabs.com><LVL::Significant><MASK::200
0>
02/03/2010 12:52:45.692 CCM|ConnectionManager -
wait_AuDisconnectRequest(44372680,44372681),disconnectType(1),
IFHandling(0,0)|<CLID::StandAloneCluster><NID::10.10.210.11><CT::2,100,1
32,1.837><IP::10.10.110.2><DEV::S0/SU1 at br1.proctorlabs.com><LVL::Arbitra
ry><MASK::0800>
02/03/2010 12:52:45.692 CCM|ConnectionManager -
storeMediaInfo(44372680): EXISTING ENTRY DISCOVERED,
size=2|<CLID::StandAloneCluster><NID::10.10.210.11><CT::2,100,132,1.837>
<IP::10.10.110.2><DEV::S0/SU1 at br1.proctorlabs.com><LVL::Arbitrary><MASK:
:0800>
02/03/2010 12:52:45.692 CCM|ConnectionManager -
storeMediaInfo(44372681): EXISTING ENTRY DISCOVERED,
size=2|<CLID::StandAloneCluster><NID::10.10.210.11><CT::2,100,132,1.837>
<IP::10.10.110.2><DEV::S0/SU1 at br1.proctorlabs.com><LVL::Arbitrary><MASK:
:0800>
02/03/2010 12:52:45.692 CCM|MediaCoordinator -
wait_AuDisconnectRequest,CI(44372680,44372681),IFCreated(1,1)|<CLID::Sta
ndAloneCluster><NID::10.10.210.11><CT::2,100,132,1.837><IP::10.10.110.2>
<DEV::S0/SU1 at br1.proctorlabs.com><LVL::Significant><MASK::0800>
02/03/2010 12:52:45.692 CCM|MediaCoordinator -
wait_AuDisconnectRequest - sending disconnect to
MediaManager(48)|<CLID::StandAloneCluster><NID::10.10.210.11><CT::2,100,
132,1.837><IP::10.10.110.2><DEV::S0/SU1 at br1.proctorlabs.com><LVL::Signif
icant><MASK::0800>

...

02/03/2010 12:52:45.694 CCM|MGCPpn9d - Dump portInfo table:
 portInfo[00] endpoint=S0/SU1/DS1-0/1 at br1.proctorlabs.com, ci=44372681
 portInfo[01] endpoint=S0/SU1/DS1-0/2 at br1.proctorlabs.com, ci=0
 portInfo[02] endpoint=S0/SU1/DS1-0/3 at br1.proctorlabs.com, ci=0

...

02/03/2010 12:52:45.698 CCM|Out Message -- PriDisconnectMsg --
Protocol=
PriEuroProtocol|<CLID::StandAloneCluster><NID::10.10.210.11><LVL::Signif
icant><MASK::0040>
02/03/2010 12:52:45.698 CCM|Ie - Q931CauseIe IEData= 08 02 80 A9
|<CLID::StandAloneCluster><NID::10.10.210.11><LVL::State
Transition><MASK::0040>

 Thanks for any suggestions!


------------------------------

Message: 55
Date: Wed, 3 Feb 2010 08:18:54 -0500
From: "Aliberto, Nate" <Nate.Aliberto at COREBTS.com>
To: "cisco-voip at puck.nether.net" <cisco-voip at puck.nether.net>
Subject: [cisco-voip] DMA Validation Errors - DMA 7.1.3
Message-ID:
	
<8932AD4A92EF974AB9D814B448798F3B2247DB816E at CDC001024.COREBTS.local>
Content-Type: text/plain; charset="us-ascii"


All,

Performing an upgrade from CCM version 4.1.3es100 running on OS version
4.4aSR4 to a new platform with CUCM 7.1.3b.  When running the DMA
(7.1.3) export on the currently deployed publisher, all data is exported
sucessfully but fails during validation with the following error.



Output from DMA trace:

***> DATA PROCESSING TESTS REVEALED AN ISSUE WHICH GENERATED THE
FOLLOWING EVENT:
             Sorry.  Unable to parse and identify error number.  Raw
Message follows:
             02/02/2010 20:37:51.895 installdb|   installFull *ERROR*
Prior Cancel or Error Processing installSql
(c:\tmp\db\sql\dbfuncs_win.sql)|


***> DATA PROCESSING TESTS REVEALED AN ISSUE WHICH GENERATED THE
FOLLOWING EVENT:
             Sorry.  Unable to parse and identify error number.  Raw
Message follows:
             02/02/2010 20:37:52.504 installdb|   installFull *ERROR*
Prior Cancel or Error Processing installSql (c:\tmp\db\sql\makedb.sql)|


***> DATA PROCESSING TESTS REVEALED AN ISSUE WHICH GENERATED THE
FOLLOWING EVENT:
             The error number is:  [-27002]
             The error message is:  [-27002   No connections are allowed
in quiescent mode.
]
             Sorry.  Unable to parse and format record info from log
message.  Raw Message follows:
                           [02/02/2010 20:37:54.567 installdb|
validateDB *ERROR* SQL Exception [0xffff9686] [Failed to connect to
datasource: [Informix][Informix ODBC Driver][Informix]Unspecified System
Error =  -27002.]|
]


I've got a TAC case open but was wondering if anyone had seen this in
their travels.
In any event I'll update with a resolution when one is found.

Thanks
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------------------------------

Message: 56
Date: Wed, 3 Feb 2010 09:02:12 -0500
From: Tanner Ezell <tanner.ezell at gmail.com>
To: shary shary <shaary1 at hotmail.com>
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Agent Status in UCCX
Message-ID:
	<9c4f122d1002030602safa29c4i1f99907b55b8954c at mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1

The Get Reporting Statistic Step will return -1 if the CSQ (or
whatever target) is not found.

On Wed, Feb 3, 2010 at 7:32 AM, shary shary <shaary1 at hotmail.com> wrote:
> Tanner,
>
> i tried this option and playing the value stored in Return Variable as
an
> Integer and but for every action whether the agent is ready , not
ready ,
> talking or logged out? for every it is palying "negative 1"
>
>
> Regards,
> Sheeraz Saeed lodhi.
>
>
>
>
>> Date: Wed, 3 Feb 2010 00:33:55 -0500
>> Subject: Re: [cisco-voip] Agent Status in UCCX
>> From: tanner.ezell at gmail.com
>> To: shaary1 at hotmail.com
>> CC: jbf005 at shsu.edu; cisco-voip at puck.nether.net
>>
>> Use the Get Reporting Statistics Step, first box should be CSQ
>> Resources or something on that order, then "logged in resources" then
>> supply the CSQ name, and a return variable, then you can test against
>> that.
>>
>> On Wed, Feb 3, 2010 at 12:30 AM, shary shary <shaary1 at hotmail.com>
wrote:
>> >
>> > yes Tanner you?got my point. I want this but how could i do this i
did
>> > it
>> > once the?call gets queued but i want?to take decision before
queuing
>> > call
>> > whether the agent is log in or not if not then transfer the call to
>> > another
>> > queue.
>> >
>> >
>> >
>> >> From: JBF005 at shsu.edu
>> >> To: tanner.ezell at gmail.com; shaary1 at hotmail.com
>> >> CC: cisco-voip at puck.nether.net
>> >> Date: Tue, 2 Feb 2010 08:38:45 -0600
>> >> Subject: RE: [cisco-voip] Agent Status in UCCX
>> >>
>> >> Report statistic can accomplish this, but I think what you want to
do
>> >> is
>> >> queue the call twice. Within your first queue just queue it again
to
>> >> the
>> >> second queue and whoever becomes available first will get the
call.
>> >
>> > ________________________________
>> > Got a cool Hotmail story? Tell us now
>>
>>
>>
>> --
>> Regards,
>> Tanner Ezell
>
> ________________________________
> Got a cool Hotmail story? Tell us now



-- 
Regards,
Tanner Ezell


------------------------------

Message: 57
Date: Wed, 3 Feb 2010 09:51:51 -0500
From: Ryan Ratliff <rratliff at cisco.com>
To: Stephen Greszczyszyn <sgreszcz at gmail.com>
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] MGCP to PSTN - 0x80A9 - Temporary failure?
Message-ID: <ACAB09AF-1C2C-4A99-BD79-8DF9E6AAE621 at cisco.com>
Content-Type: text/plain; charset=us-ascii

Normally a temp failure coming from CUCM means a device involved in the
call unregistered.  With MGCP gateways this is most common when you have
MGCP messages getting lost between CUCM and the router.  CUCM will
retransmit a message 3 times before unregistering the gateway.

In this case you have a traceback followed by the gateway sending a DLCX
(delete connection) message to the CUCM, which ends up with CUCM tearing
down the call.  

I'd recommend putting your traceback (and the other required info) into
the output interpreter at
https://www.cisco.com/cgi-bin/Support/OutputInterpreter/home.pl.  

-Ryan

On Feb 3, 2010, at 8:29 AM, Stephen Greszczyszyn wrote:

Hi there,

I'm working on a lab scenario where I'm making calls from CUCM7 via an
IOS MGCP gateway out to another router simulating the PSTN.  The call
connects OK, but after about 1 minute - 2 minutes, the call drops.  I
have tried the same test through another H.323 gateway and the call
stays up fine.  Running 12.4(24)T2 on the MGCP gateway.  Here is the
information:

MGCP gateway:
Feb  3 11:53:33.851: ISDN Se0/1/0:15 Q931: TX -> DISCONNECT pd = 8
callref = 0x0007
      Cause i = 0x80A9 - Temporary failure

PSTN:
*Feb  3 11:59:57.011: ISDN Se0/3/0:15 Q931: RX <- DISCONNECT pd = 8
callref = 0x0007
      Cause i = 0x80A9 - Temporary failure
*Feb  3 11:59:57.015: %ISDN-6-DISCONNECT: Interface Serial0/3/0:0
disconnected from +16178631002 , call lasted 72 seconds

I did some searching on this list and there were some references to
faulty DSPs.  Mine seem to look OK?

MGCP:
br1#sh voice dsp group all
DSP groups on slot 0:
dsp 1:
State: UP, firmware: 24.3.2
Max signal/voice channel: 16/16
Max credits: 240
num_of_sig_chnls_allocated: 8
Transcoding channels allocated: 0
Group: FLEX_GROUP_VOICE, complexity: FLEX
  Shared credits: 225, reserved credits: 0
  Signaling channels allocated: 8
  Voice channels allocated: 1
  Credits used: 15
  Voice channels:
    Ch01: voice port: 0/1/0:15.1, codec: g711ulaw, credits allocated: 15

PSTN:
WAN_PSTN#sh voice dsp group all
DSP groups on slot 0:
dsp 1:
State: UP, firmware: 24.3.2
Max signal/voice channel: 16/16
Max credits: 240
num_of_sig_chnls_allocated: 16
Transcoding channels allocated: 0
Group: FLEX_GROUP_VOICE, complexity: FLEX
  Shared credits: 225, reserved credits: 0
  Signaling channels allocated: 16
  Voice channels allocated: 1
  Credits used: 15
  Voice channels:
    Ch01: voice port: 0/3/0:15.1, codec: g711ulaw, credits allocated: 15

I also get some weird stuff around the time of call failure when I
enable MGCP debugging:

debug mgcp errors
debug mgcp packet

It seems as though my MGCP gateway tells CUCM to drop the call and
also sends the disconnect to the PSTN:

MGCP:
Feb  3 13:09:00.557: ISDN Se0/1/0:15 Q921: User TX -> INFO sapi=0
tei=0, ns=14 nr=15
Feb  3 13:09:00.557: ISDN Se0/1/0:15 Q931: DISCONNECT pd = 8  callref =
0x0003
      Cause i = 0x80A9 - Temporary failure

PSTN:
*Feb  3 13:15:23.583: ISDN Se0/3/0:15 Q921: Net RX <- INFO sapi=0
tei=0, ns=14 nr=15
*Feb  3 13:15:23.583: ISDN Se0/3/0:15 Q931: DISCONNECT pd = 8  callref =
0x0003
      Cause i = 0x80A9 - Temporary failure

(Previous call traces - MGCP debug)

Feb  3 12:52:45.713: Following traceback is for INFO ONLY. -Traceback=
0x40F3126Cz 0x40F5BA6Cz 0x40F4296Cz 0x40F3ABC4z 0x40F333D4z
0x40EEED44z 0x40EF43F0z 0x435D69D8z 0x435D69BCz
Feb  3 12:52:45.717: MGCP Packet sent to 10.10.210.11:2427--->
DLCX 205358175 S0/SU1/DS1-0/1 at br1.proctorlabs.com MGCP 0.1
C: D000000002a512c9000000F500000002
I: 3
P: PS=5000, OS=800000, PR=4978, OR=796480, PL=22, JI=7, LA=0
E: 502
<---

Feb  3 12:52:45.725: MGCP Packet received from 10.10.210.11:2427--->
200 205358175
<---

Feb  3 12:52:45.725:
//-1/xxxxxxxxxxxx/MGCP/mgcp_mp_get_not_entity(830):[lvl=2]Invalid
parameter (pkt 0x4A428FE0 pkt->mgcp_parm_lines 0x00000000)
Feb  3 12:52:45.733: ISDN Se0/1/0:15 Q931: TX -> DISCONNECT pd = 8
callref = 0x0002
      Cause i = 0x80A9 - Temporary failure
Feb  3 12:52:45.745: ISDN Se0/1/0:15 Q931: RX <- RELEASE pd = 8
callref = 0x8002
Feb  3 12:52:45.757: ISDN Se0/1/0:15 Q931: TX -> RELEASE_COMP pd = 8
callref = 0x0002
Feb  3 12:53:04.977: MGCP Packet sent to 10.10.210.11:2427--->
NTFY 205358176 *@br1.proctorlabs.com MGCP 0.1
X: 0
O:
<---

In the CUCM traces, it isn't clear to me why the call is being dumped:

02/03/2010 12:52:45.690 CCM|MGCPHandler received msg from: 10.10.110.2
DLCX 205358175 S0/SU1/DS1-0/1 at br1.proctorlabs.com MGCP 0.1
C: D000000002a512c9000000F500000002
I: 3
P: PS=5000, OS=800000, PR=4978, OR=796480, PL=22, JI=7, LA=0
E: 502
|<CLID::StandAloneCluster><NID::10.10.210.11><CT::2,100,132,1.837><IP::1
0.10.110.2><DEV::><LVL::Significant><MASK::2000>
02/03/2010 12:52:45.690
CCM|<CLID::StandAloneCluster><NID::10.10.210.11><CT::2,100,132,1.837><MN
::MGCPEndPoint><MV::S0/SU1/DS1-0/1 at br1.proctorlabs.com><DEV::><LVL::All>
<MASK::ffff>
02/03/2010 12:52:45.691 CCM|MGCPHandler send msg SUCCESSFULLY to:
10.10.110.2
200 205358175
|<CLID::StandAloneCluster><NID::10.10.210.11><CT::2,100,132,1.837><IP::1
0.10.110.2><DEV::S0/SU1 at br1.proctorlabs.com><LVL::Significant><MASK::200
0>
02/03/2010 12:52:45.692 CCM|ConnectionManager -
wait_AuDisconnectRequest(44372680,44372681),disconnectType(1),
IFHandling(0,0)|<CLID::StandAloneCluster><NID::10.10.210.11><CT::2,100,1
32,1.837><IP::10.10.110.2><DEV::S0/SU1 at br1.proctorlabs.com><LVL::Arbitra
ry><MASK::0800>
02/03/2010 12:52:45.692 CCM|ConnectionManager -
storeMediaInfo(44372680): EXISTING ENTRY DISCOVERED,
size=2|<CLID::StandAloneCluster><NID::10.10.210.11><CT::2,100,132,1.837>
<IP::10.10.110.2><DEV::S0/SU1 at br1.proctorlabs.com><LVL::Arbitrary><MASK:
:0800>
02/03/2010 12:52:45.692 CCM|ConnectionManager -
storeMediaInfo(44372681): EXISTING ENTRY DISCOVERED,
size=2|<CLID::StandAloneCluster><NID::10.10.210.11><CT::2,100,132,1.837>
<IP::10.10.110.2><DEV::S0/SU1 at br1.proctorlabs.com><LVL::Arbitrary><MASK:
:0800>
02/03/2010 12:52:45.692 CCM|MediaCoordinator -
wait_AuDisconnectRequest,CI(44372680,44372681),IFCreated(1,1)|<CLID::Sta
ndAloneCluster><NID::10.10.210.11><CT::2,100,132,1.837><IP::10.10.110.2>
<DEV::S0/SU1 at br1.proctorlabs.com><LVL::Significant><MASK::0800>
02/03/2010 12:52:45.692 CCM|MediaCoordinator -
wait_AuDisconnectRequest - sending disconnect to
MediaManager(48)|<CLID::StandAloneCluster><NID::10.10.210.11><CT::2,100,
132,1.837><IP::10.10.110.2><DEV::S0/SU1 at br1.proctorlabs.com><LVL::Signif
icant><MASK::0800>

...

02/03/2010 12:52:45.694 CCM|MGCPpn9d - Dump portInfo table:
portInfo[00] endpoint=S0/SU1/DS1-0/1 at br1.proctorlabs.com, ci=44372681
portInfo[01] endpoint=S0/SU1/DS1-0/2 at br1.proctorlabs.com, ci=0
portInfo[02] endpoint=S0/SU1/DS1-0/3 at br1.proctorlabs.com, ci=0

...

02/03/2010 12:52:45.698 CCM|Out Message -- PriDisconnectMsg --
Protocol=
PriEuroProtocol|<CLID::StandAloneCluster><NID::10.10.210.11><LVL::Signif
icant><MASK::0040>
02/03/2010 12:52:45.698 CCM|Ie - Q931CauseIe IEData= 08 02 80 A9
|<CLID::StandAloneCluster><NID::10.10.210.11><LVL::State
Transition><MASK::0040>

Thanks for any suggestions!
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------------------------------

Message: 58
Date: Wed, 3 Feb 2010 10:40:48 -0500
From: Peter Slow <peter.slow at gmail.com>
To: Kevin Dunn <cheesevoice at gmail.com>
Cc: "Norton, Mike" <mikenorton at pwsd76.ab.ca>,	Cisco Voice
	<cisco-voip at puck.nether.net>
Subject: Re: [cisco-voip] MOH Beeping
Message-ID:
	<53fc16d41002030740u5c033ba3k7b7e4acb73ae1486 at mail.gmail.com>
Content-Type: text/plain; charset=windows-1252

Touching on what Mike and Wes said, to test the interference idea...
I'm guessing you tried plugging the USB audio device into your laptop
at your desk or some place other than the rack. Can we see if you hear
the beeping if we put your laptop on top of the MCS server in the rack
and try it there? if it's RF interference it'll have to do with your
proximity to the device generating the RF noise.

-Peter

On Tue, Feb 2, 2010 at 6:54 PM, Kevin Dunn <cheesevoice at gmail.com>
wrote:
> I will give it a shot
>
> Thanks Les
>
> On Tue, Feb 2, 2010 at 5:37 PM, Wes Sisk <wsisk at cisco.com> wrote:
>>
>> Thanks Kevin,
>>
>> So:
>> noise is not present recording from USB source to your laptop.
>> noise is present even in the packets transmitted over the wire to the
>> phone when USB device is connected to server.
>>
>> Based on the nature of sound I agree with Mike that it sounds like
>> electrical interference.? Are the server and rack grounded?? When you
>> captured "good.au" what source of power and and ground was your
laptop
>> using?
>>
>> Do you have another analog source such as an iPod or mp3 player that
would
>> use a different, or possibly floating ground, to test with?
>>
>> You have eliminated most of the devices in the audio path. The only
way to
>> isolate further inside the server is to use a root account to dump
audio
>> directly off the linux DSP device before the IPVMSApp kernel driver
attempts
>> to packetize it for transmission over the network.? If noise is
present in
>> that sample then it is either introduced by electrical interface
between the
>> source and USB device or introduced in the USB device itself.
>>
>> /Wes
>>
>>
>> On Tuesday, February 02, 2010 5:17:53 PM, Norton, Mike
>> <mikenorton at pwsd76.ab.ca> wrote:
>>
>> Listening to the capture, it sounds like ambient electrical/RF noise
is
>> getting into the analog audio signal. I?m surprised this isn?t a more
common
>> problem. Server racks are probably one of the least ideal places to
try to
>> run unbalanced analog audio signals.
>>
>>
>>
>> --
>>
>> Mike Norton
>>
>> I.T. Support
>>
>> Peace Wapiti School Division No. 76
>>
>> Helpdesk: 780-831-3080
>>
>> Direct: 780-831-3076
>>
>>
>>
>>
>>
>> From: cisco-voip-bounces at puck.nether.net
>> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Kevin Dunn
>> Sent: February-02-10 1:30 PM
>> To: Wes Sisk
>> Cc: Cisco Voice
>> Subject: Re: [cisco-voip] MOH Beeping
>>
>>
>>
>> Sev3 SR 613458017 Udhay has been pretty helpful so far checking the
>> configuration.
>>
>> the sniff files are associated with that case,m you should be able to
see
>> them (otherwise I can add them in)
>>
>>
>>
>> User experience:
>>
>>
>>
>> I call my boss, he answers by pressing speaker button.
>>
>> I place him on hold
>>
>> He hears the "good.au" sound I enclosed for Peter to listen to
>>
>>
>>
>> the sound is not unlike rapid repeated whistling...
>>
>> I should note that when I use the handset I don't hear it, but when I
>> capture the packets it is still there, just not as loud
>>
>>
>>
>> Kevin
>>
>> On Tue, Feb 2, 2010 at 2:07 PM, Wes Sisk <wsisk at cisco.com> wrote:
>>
>> In the very abstract this sounds like call waiting tone.? Can you
clarify
>> exactly what the user is doing when "caller is placed on hold and the
>> speaker is activated"?? What buttons is the user pressing to
accomplish
>> this?
>>
>> concurrent detailed CCM and SDL traces and a packet capture of all
traffic
>> to/from the phone could be used to isolate the source of this tone
rather
>> quickly.? You mention a TAC case, what is the number?
>>
>> /Wes
>>
>>
>>
>> On Tuesday, February 02, 2010 12:53:17 PM, Kevin Dunn
>> <cheesevoice at gmail.com> wrote:
>>
>> Okay I have a TAC case open and I have tried changing configuration
>> settings, cables and hardware...
>>
>>
>>
>> CUCM 7.0.2.2000-5
>>
>> Fixed audio from XM radio (MOH-USB-AUDIO) card
>>
>>
>>
>> when a caller is placed on hold and the speaker is activated there is
an
>> audible (and quite annoying) beeping sound playing over the top of
the audio
>> file.
>>
>>
>>
>> It is not audible on the handset or headset.
>>
>>
>>
>> If I sniff the phone port and capture the audio file I can hear it.
>>
>>
>>
>> If I record the audio file with my laptop (plugging mic cord into
lappy
>> instead of MOH-USB) there is no beeping.
>>
>> In my mind that eliminates the XM radio and cables, I have changed
out
>> cables though, just in case.
>>
>>
>>
>> I changes out MOH-USB cards and that also did nothing to eliminate
the
>> issue.
>>
>>
>>
>> I have upgraded firmware on the phones and that wasn't it either.
>>
>> With the Sample Audio file (which is JAZZY) there is no beeping
regardless
>> of handset or speaker.
>>
>>
>>
>> Any suggestions?
>>
>> ________________________________
>>
>>
>>
>>
>>
>> _______________________________________________
>>
>> cisco-voip mailing list
>>
>> cisco-voip at puck.nether.net
>>
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>>
>>
>>
>>
>>
>>
>> ________________________________
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>


------------------------------

Message: 59
Date: Wed, 3 Feb 2010 10:46:19 -0500
From: Ryan Ratliff <rratliff at cisco.com>
To: "Aliberto, Nate" <Nate.Aliberto at corebts.com>
Cc: "cisco-voip at puck.nether.net" <cisco-voip at puck.nether.net>
Subject: Re: [cisco-voip] DMA Validation Errors - DMA 7.1.3
Message-ID: <9D94C2AC-F979-47F8-8F6B-AC9A458F4412 at cisco.com>
Content-Type: text/plain; charset="us-ascii"

The DMA process basically exports your SQL database to an informix db on
the same server, then runs the validation against the informix db.  In
your case it's not able to connect to the informix db it looks like.

Do you have Prognosis or any other monitoring software installed?  If
so, stop it and see if that fixes DMA.

-Ryan

On Feb 3, 2010, at 8:18 AM, Aliberto, Nate wrote:

 
All,
 
Performing an upgrade from CCM version 4.1.3es100 running on OS version
4.4aSR4 to a new platform with CUCM 7.1.3b.  When running the DMA
(7.1.3) export on the currently deployed publisher, all data is exported
sucessfully but fails during validation with the following error.
 
 
 
Output from DMA trace:
 
***> DATA PROCESSING TESTS REVEALED AN ISSUE WHICH GENERATED THE
FOLLOWING EVENT: 
             Sorry.  Unable to parse and identify error number.  Raw
Message follows:
             02/02/2010 20:37:51.895 installdb|   installFull *ERROR*
Prior Cancel or Error Processing installSql
(c:\tmp\db\sql\dbfuncs_win.sql)|
 
 
***> DATA PROCESSING TESTS REVEALED AN ISSUE WHICH GENERATED THE
FOLLOWING EVENT: 
             Sorry.  Unable to parse and identify error number.  Raw
Message follows:
             02/02/2010 20:37:52.504 installdb|   installFull *ERROR*
Prior Cancel or Error Processing installSql (c:\tmp\db\sql\makedb.sql)|
 
 
***> DATA PROCESSING TESTS REVEALED AN ISSUE WHICH GENERATED THE
FOLLOWING EVENT: 
             The error number is:  [-27002]
             The error message is:  [-27002   No connections are allowed
in quiescent mode.
]
             Sorry.  Unable to parse and format record info from log
message.  Raw Message follows:
                           [02/02/2010 20:37:54.567 installdb|
validateDB *ERROR* SQL Exception [0xffff9686] [Failed to connect to
datasource: [Informix][Informix ODBC Driver][Informix]Unspecified System
Error =  -27002.]|
]
 
 
I've got a TAC case open but was wondering if anyone had seen this in
their travels.
In any event I'll update with a resolution when one is found.
 
Thanks
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Message: 60
Date: Wed, 3 Feb 2010 10:11:45 -0600
From: "Carter, Bill" <bcarter at sentinel.com>
To: "Cisco Voip" <cisco-voip at puck.nether.net>
Subject: [cisco-voip] HuntPilot-Line Group - Phone Forwarded
Message-ID:
	<C0B4574561D1E04DBB500BA062BAF22601894D23 at Mail1.sentinel.com>
Content-Type: text/plain; charset="us-ascii"

I have a Hunt Pilot -> HuntList -> Line group with members DN-1111 and
DN-2222. The Hunt List is configured as Broadcast. 

 

DN-1111 is set to cfwdall 5555. When a call is placed to the Hunt Pilot,
will DN-1111 ring or will the call not be presented to it?

 

 

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------------------------------

Message: 61
Date: Wed, 3 Feb 2010 11:24:13 -0500
From: Ryan Ratliff <rratliff at cisco.com>
To: "Carter, Bill" <bcarter at sentinel.com>
Cc: Cisco Voip <cisco-voip at puck.nether.net>
Subject: Re: [cisco-voip] HuntPilot-Line Group - Phone Forwarded
Message-ID: <56D1258C-267E-494A-B2A3-D9235FD78C4E at cisco.com>
Content-Type: text/plain; charset="us-ascii"

Intercepts (forwards) are ignored for calls routing through a hunt
pilot.   The phone should ring.

-Ryan

On Feb 3, 2010, at 11:11 AM, Carter, Bill wrote:

I have a Hunt Pilot -> HuntList -> Line group with members DN-1111 and
DN-2222. The Hunt List is configured as Broadcast.
 
DN-1111 is set to cfwdall 5555. When a call is placed to the Hunt Pilot,
will DN-1111 ring or will the call not be presented to it?
 
 
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