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Today's Topics:

   1. VG 204 reliability issues? (Erick Wellnitz)
   2. CUCM 8.6 non-agent call recording? (Countryman, Edward)
   3. Re: CUCM 8.6 non-agent call recording? (Scott Voll)
   4. Re: CUCM 8.6 non-agent call recording? (Erick Wellnitz)
   5. Re: CUCM 8.6 non-agent call recording?
      (Jamie Gale -X (jamgale - Arc Solutions at Cisco))
   6. Re: Upgrade CUCM 7.0(3a) to 8.0(3) and move to UCS (Reto Gassmann)
   7. BLF CM 8.6 user defined? (Scott Voll)
   8. Re: Upgrade CUCM 7.0(3a) to 8.0(3) and move to UCS (Ryan Ratliff)
   9. Re: VG 204 reliability issues? (Carlo)
  10. Bootable 8.6.1.20000-1 (Nathan Reeves)
  11. TAPS with E.164 AppNote (Jason Aarons (AM))
  12. Re: TAPS with E.164 AppNote (Jason Aarons (AM))
  13. Re: TAPS with E.164 AppNote (Justin Steinberg)
  14. SIPC CIPC 8.6.3 not liking PLAR (Jason Aarons (AM))
  15. Re: TAPS with E.164 AppNote (Jason Aarons (AM))
  16. Re: SIPC CIPC 8.6.3 not liking PLAR (Matthew Loraditch)
  17. Re: SIPC CIPC 8.6.3 not liking PLAR (Jason Aarons (AM))
  18. Re: SIPC CIPC 8.6.3 not liking PLAR (Matthew Loraditch)
  19. Recommended SIP SBC (Heim, Dennis)
  20. Re: Recommended SIP SBC (Berry, Matthew)
  21. Re: CMBE upgrade 7.0 to 7.1.5 Attendant console issue (Wes Sisk)
  22. CCM 8.6 - SIP Register and authentication (Thierry De Vriendt)
  23. Re: CCM 8.6 - SIP Register and authentication (Jason Aarons (AM))
  24. Cisco 7925 / Aerohive (Jason Faraone)
  25. Re: VMWare support vis-a-vis Cisco (Wes Sisk)
  26. Re: Recommended SIP SBC (Erick Wellnitz)
  27. Re: CCM 8.6 - SIP Register and authentication (Matt Slaga (AM))


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

Message: 1
Date: Mon, 6 May 2013 11:42:07 -0500
From: Erick Wellnitz <[email protected]>
To: cisco-voip <[email protected]>
Subject: [cisco-voip] VG 204 reliability issues?
Message-ID:
        <CAK0wOsBBYQYau8AkoMzZc2qRx-1a1FPQbP3x==ng52w9rj6...@mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"

We have a few VG 204s that have issues once in a while with one or two of
the fxs ports stop functioning.

The VG is MGCP.  The problem is that traders use these lines all day and we
don't have time to monkey around with TAC to find the root cause.

We never see anything in logs or on the CUCM side that would indicate a
problem but a reboot if the VG always solves it.

Any ideas on how to either solve the issue or figure it out without impact
to uptime are greatly appreciated!

Thanks!
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Message: 2
Date: Mon, 6 May 2013 11:59:34 -0500
From: "Countryman, Edward" <[email protected]>
To: <[email protected]>
Subject: [cisco-voip] CUCM 8.6 non-agent call recording?
Message-ID:
        <C3B4BFD981268C48915DE5C216D800B92AED10@AEXMV13.phnet.phroot.local>
Content-Type: text/plain; charset="us-ascii"

Does anyone have direct experience or recommendations for telephone call
recording platforms and solutions that work well with CUCM 8.6?

 

I have the need to record calls to a handful of phones.  These are not
UCCX contact center agents

 

Any input would be welcomed!

 

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Message: 3
Date: Mon, 6 May 2013 10:21:44 -0700
From: Scott Voll <[email protected]>
To: "Countryman, Edward" <[email protected]>
Cc: "[email protected]" <[email protected]>
Subject: Re: [cisco-voip] CUCM 8.6 non-agent call recording?
Message-ID:
        <CAHgd+3_JdTEjzm9GQSdb2DxV-DG2iSbnRFxmBrz=qmb1n9h...@mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"

We are using Call Rex.

since our upgrade to CM 8.6 things have been working perfectly.

Shared lines are not supported, but Extension Mobiltiy is.  Works really
slick.

Scott


On Mon, May 6, 2013 at 9:59 AM, Countryman, Edward <
[email protected]> wrote:

> Does anyone have direct experience or recommendations for telephone call
> recording platforms and solutions that work well with CUCM 8.6?****
>
> ** **
>
> I have the need to record calls to a handful of phones.  These are *not*UCCX 
> contact center agents
> ****
>
> ** **
>
> Any input would be welcomed!****
>
> ** **
>
> _______________________________________________
> cisco-voip mailing list
> [email protected]
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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Message: 4
Date: Mon, 6 May 2013 12:22:12 -0500
From: Erick Wellnitz <[email protected]>
To: "Countryman, Edward" <[email protected]>
Cc: cisco-voip <[email protected]>
Subject: Re: [cisco-voip] CUCM 8.6 non-agent call recording?
Message-ID:
        <cak0woscsg3vc7jkj404jhhw4vkn5xu4e6-koznmaz38rkhy...@mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"

We use RedBox Quantify.  I'm not quite sure I would recommend it.  It is
hard to get someone from suport to assist in anything and documentation is
sorely lacking.


On Mon, May 6, 2013 at 11:59 AM, Countryman, Edward <
[email protected]> wrote:

> Does anyone have direct experience or recommendations for telephone call
> recording platforms and solutions that work well with CUCM 8.6?****
>
> ** **
>
> I have the need to record calls to a handful of phones.  These are *not*UCCX 
> contact center agents
> ****
>
> ** **
>
> Any input would be welcomed!****
>
> ** **
>
> _______________________________________________
> cisco-voip mailing list
> [email protected]
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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Message: 5
Date: Mon, 6 May 2013 17:26:52 +0000
From: "Jamie Gale -X (jamgale - Arc Solutions at Cisco)"
        <[email protected]>
To: "Countryman, Edward" <[email protected]>
Cc: "<[email protected]>" <[email protected]>
Subject: Re: [cisco-voip] CUCM 8.6 non-agent call recording?
Message-ID:
        <[email protected]>
Content-Type: text/plain; charset="us-ascii"

Edward,

I would recommend you look at CallRex.

Kind Regards

Jamie


On May 6, 2013, at 12:59 PM, "Countryman, Edward" 
<[email protected]<mailto:[email protected]>>
 wrote:

Does anyone have direct experience or recommendations for telephone call 
recording platforms and solutions that work well with CUCM 8.6?

I have the need to record calls to a handful of phones.  These are not UCCX 
contact center agents

Any input would be welcomed!

_______________________________________________
cisco-voip mailing list
[email protected]<mailto:[email protected]>
https://puck.nether.net/mailman/listinfo/cisco-voip

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Message: 6
Date: Mon, 6 May 2013 20:17:54 +0200
From: Reto Gassmann <[email protected]>
To: Ryan Ratliff <[email protected]>
Cc: "[email protected]" <[email protected]>
Subject: Re: [cisco-voip] Upgrade CUCM 7.0(3a) to 8.0(3) and move to
        UCS
Message-ID:
        <cal4h0z5pr2wu867kynd4qyfjbeyyxhqxjd2vex-xhcyjnnh...@mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"

Hello Ryan

Is there a Problem with 8.0.3? We have to go there behause of the
compatibilty to the UCCE and we can not Update UCCE before the end of this
year. So we have to stick with 8.0.3 till then.

Regards Reto

Am Montag, 6. Mai 2013 schrieb Ryan Ratliff :

> In general you should be fine there.  Make sure you do your phone loads
> prior to the upgrade so you don't get stuck with all the phones upgrading
> when you are trying to come back into production.
>
> If you can do everything with the UCS cluster in a dead-net it will be
> less stressful watching that DRS restore go since you can leave the MCS
> servers up.
>
> I sincerely hope you aren't planning to stay on 8.0(3) for any length of
> time.  If you are a partner you should look into the migration scenarios
> for the Drive 2 Nine program (
> https://communities.cisco.com/community/partner/collaboration/migration)
> for more info and how 9.1(2) will make this upgrade process easier for you.
>  There is a customer portal with similar info in the works.
>
> -Ryan
>
> On May 4, 2013, at 2:41 PM, Reto Gassmann <[email protected]<javascript:_e({}, 
> 'cvml', '[email protected]');>>
> wrote:
>
> Hello group****
>
> We plan the upgrade from our CUCM 7.1(3a) Cluster (8 Servers) to 8.0(3).
> After the upgrade we also want to virtualize the cluster from MCS Servers
> to UCS. To reduce downtime we plan to do the upgrade with the following
> steps:****
> 1.       Setup a new CUCM 8.0(3) cluster on UCS with different IP
> addresses****
> 2.       Upgrade CUCM 7.1(3a) to 8.0(3) on MCS****
> 3.       Take backup (DRS) of upgraded CUCM 8.0(3) on MCS****
> 4.       Shutdown MCS Cluster****
> 5.       Change IP addresses to match old IP addresses on UCS cluster****
> 6.       Restore CUCM 8.0(3) on UCS servers****
>
> Can I do the upgrade as described or do I run into any issues?****
>
> If I want to upgrade the IP Phone (7960 / 7961) firmware a few days before
> the upgrade where do I have to install the firmware software package, just
> on the TFTP server or on all servers in the cluster?****
>
> Any comments are welcome.****
>
> Regards Reto****
>
>
> _______________________________________________
> cisco-voip mailing list
> [email protected] <javascript:_e({}, 'cvml',
> '[email protected]');>
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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Message: 7
Date: Mon, 6 May 2013 11:20:07 -0700
From: Scott Voll <[email protected]>
To: "[email protected]" <[email protected]>
Subject: [cisco-voip] BLF CM 8.6 user defined?
Message-ID:
        <cahgd+3-z39aqnwwa6lbv-4oygixt4+rkmkvvf72tgkdqxtp...@mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"

Back in the day end users could not define there own BLF / SDs.  is that
still the same case in the day of 8.6?

With everything being about presence, I would have thought Cisco would have
changed the policy on that.  Am I missing it?

TIA

Scott
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Message: 8
Date: Mon, 6 May 2013 14:47:08 -0400
From: Ryan Ratliff <[email protected]>
To: Reto Gassmann <[email protected]>
Cc: "[email protected]" <[email protected]>
Subject: Re: [cisco-voip] Upgrade CUCM 7.0(3a) to 8.0(3) and move to
        UCS
Message-ID: <[email protected]>
Content-Type: text/plain; charset="iso-8859-1"

CUCM 8.0 is no longer being maintained, so if you hit any bugs or want to 
introduce new devices you'll be looking at an 8.6 or later upgrade.  
That said, if you are being held back right now by UCCE then that's pretty darn 
good reason to hang back for a while.  

I would strongly recommend you read up on security by default (SBD) and be very 
comfortable with how it works before you do any more big changes to the 
cluster.  

-Ryan

On May 6, 2013, at 2:17 PM, Reto Gassmann <[email protected]> wrote:

Hello Ryan

Is there a Problem with 8.0.3? We have to go there behause of the compatibilty 
to the UCCE and we can not Update UCCE before the end of this year. So we have 
to stick with 8.0.3 till then.

Regards Reto

Am Montag, 6. Mai 2013 schrieb Ryan Ratliff :
In general you should be fine there.  Make sure you do your phone loads prior 
to the upgrade so you don't get stuck with all the phones upgrading when you 
are trying to come back into production.

If you can do everything with the UCS cluster in a dead-net it will be less 
stressful watching that DRS restore go since you can leave the MCS servers up.

I sincerely hope you aren't planning to stay on 8.0(3) for any length of time.  
If you are a partner you should look into the migration scenarios for the Drive 
2 Nine program 
(https://communities.cisco.com/community/partner/collaboration/migration) for 
more info and how 9.1(2) will make this upgrade process easier for you.  There 
is a customer portal with similar info in the works.

-Ryan

On May 4, 2013, at 2:41 PM, Reto Gassmann <[email protected]> wrote:

Hello group
 
We plan the upgrade from our CUCM 7.1(3a) Cluster (8 Servers) to 8.0(3). After 
the upgrade we also want to virtualize the cluster from MCS Servers to UCS. To 
reduce downtime we plan to do the upgrade with the following steps:
1.       Setup a new CUCM 8.0(3) cluster on UCS with different IP addresses
2.       Upgrade CUCM 7.1(3a) to 8.0(3) on MCS
3.       Take backup (DRS) of upgraded CUCM 8.0(3) on MCS
4.       Shutdown MCS Cluster
5.       Change IP addresses to match old IP addresses on UCS cluster
6.       Restore CUCM 8.0(3) on UCS servers
 
Can I do the upgrade as described or do I run into any issues?
 
If I want to upgrade the IP Phone (7960 / 7961) firmware a few days before the 
upgrade where do I have to install the firmware software package, just on the 
TFTP server or on all servers in the cluster?
 
Any comments are welcome.
 
Regards Reto
 
_______________________________________________
cisco-voip mailing list
[email protected]
https://puck.nether.net/mailman/listinfo/cisco-voip


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Message: 9
Date: Mon, 6 May 2013 11:58:29 -0700
From: Carlo <[email protected]>
To: cisco-voip <[email protected]>
Subject: Re: [cisco-voip] VG 204 reliability issues?
Message-ID: <[email protected]>
Content-Type: text/plain;       charset=us-ascii

I have had this problem before on 2800s. The mgcp on the gateway just stops 
talking to cucm. You don't need to reload, just stop and start mgcp. 

For a while, I had a program that went out and stopped and started mgcp once a 
week. 

Or you can move it to sip. Don't get me wrong, I love MGCP. It's fast and easy 
to maintain. But when I talk to Cisco on things. They are not doing much it 
keep mgcp going forward. And with 15.x sip is the answer, especially with fax.  

Sent from my iPhone

On May 6, 2013, at 9:42 AM, Erick Wellnitz <[email protected]> wrote:

> We have a few VG 204s that have issues once in a while with one or two of the 
> fxs ports stop functioning. 
>  
> The VG is MGCP.  The problem is that traders use these lines all day and we 
> don't have time to monkey around with TAC to find the root cause. 
>  
> We never see anything in logs or on the CUCM side that would indicate a 
> problem but a reboot if the VG always solves it.
>  
> Any ideas on how to either solve the issue or figure it out without impact to 
> uptime are greatly appreciated!
>  
> Thanks!
> _______________________________________________
> cisco-voip mailing list
> [email protected]
> https://puck.nether.net/mailman/listinfo/cisco-voip



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

Message: 10
Date: Mon, 6 May 2013 23:27:55 +0800
From: Nathan Reeves <[email protected]>
To: "[email protected]" <[email protected]>
Subject: [cisco-voip] Bootable 8.6.1.20000-1
Message-ID:
        <cafi9pdrnuhj6+rekrqxajgne1kdr5gyn8jfixsy7kux0oku...@mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"

Anyone know where a bootable copy of cucm  8.6.1.20000-1 may be available
from.  Need it to migrate from hardware to VMware.  This is our current
running version and I want to complete a restore from backup for a sub move
to vm.

Thanks

Nathan.
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Message: 11
Date: Mon, 6 May 2013 17:06:54 -0400
From: "Jason Aarons (AM)" <[email protected]>
To: "cisco-voip ([email protected])"
        <[email protected]>
Subject: [cisco-voip] TAPS with E.164 AppNote
Message-ID:
        
<4e38db0a1959b04c8c83edcf069b53ed0d330ee...@usispclexdb01.na.didata.local>
        
Content-Type: text/plain; charset="windows-1252"

https://supportforums.cisco.com/docs/DOC-13144


I'm following these steps in editing the taps.aef file to have users enter a * 
then number to get TAPS to use E.164.

However  the last step I hit Ok and I get Error: Value Parsing Error, unable to 
parse expression

See attached screen shot;

[cid:[email protected]]


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Message: 12
Date: Mon, 6 May 2013 17:16:07 -0400
From: "Jason Aarons (AM)" <[email protected]>
To: "cisco-voip ([email protected])"
        <[email protected]>
Subject: Re: [cisco-voip] TAPS with E.164 AppNote
Message-ID:
        
<4e38db0a1959b04c8c83edcf069b53ed0d330ee...@usispclexdb01.na.didata.local>
        
Content-Type: text/plain; charset="windows-1252"

Reading the comments in the below URL I see my exact error and that a Mr. 
William E Welch has a updated .aef file and to click the URL to download, but I 
get an Access Denied from cisco.com ?


Here is the script location that William E Welch posted:

 https://communities.cisco.com/message/65893#65893

 I have used it on 9.0 Premium  but I have not used it on Standard or Enhanced. 
It is just a couple lines added to the current TAPS script. Good Work William.


From: Jason Aarons (AM)
Sent: Monday, May 06, 2013 5:07 PM
To: cisco-voip ([email protected])
Subject: TAPS with E.164 AppNote

https://supportforums.cisco.com/docs/DOC-13144


I'm following these steps in editing the taps.aef file to have users enter a * 
then number to get TAPS to use E.164.

However  the last step I hit Ok and I get Error: Value Parsing Error, unable to 
parse expression

See attached screen shot;

[cid:[email protected]]


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Message: 13
Date: Mon, 6 May 2013 17:56:31 -0400
From: Justin Steinberg <[email protected]>
To: "Jason Aarons (AM)" <[email protected]>
Cc: "cisco-voip \([email protected]\)"
        <[email protected]>
Subject: Re: [cisco-voip] TAPS with E.164 AppNote
Message-ID:
        <CACCAgha8=D5rxyO_WHRqwE=bzqb5w5hauro1f44jwus+gcr...@mail.gmail.com>
Content-Type: text/plain; charset="windows-1252"

look at the error messge, there are all those junk characters there that
cause the problem.  I don't know why they are inserted but it is easy
enough to remove them.

essentially look at the syntax of that step before you open the step to
edit it, then open the step and remove all the junk characters, then edit
according to the documentation.  that should fix it for you.


On Mon, May 6, 2013 at 5:16 PM, Jason Aarons (AM) <
[email protected]> wrote:

> Reading the comments in the below URL I see my exact error and that a Mr.
> William E Welch has a updated .aef file and to click the URL to download,
> but I get an Access Denied from cisco.com ?****
>
> ** **
>
> Here is the script location that William E Welch posted:****
>
>  https://communities.cisco.com/message/65893#65893****
>
>  I have used it on 9.0 Premium  but I have not used it on Standard or
> Enhanced. It is just a couple lines added to the current TAPS script. Good
> Work William.****
>
> ** **
>
> ** **
>
> *From:* Jason Aarons (AM)
> *Sent:* Monday, May 06, 2013 5:07 PM
> *To:* cisco-voip ([email protected])
> *Subject:* TAPS with E.164 AppNote****
>
> ** **
>
> https://supportforums.cisco.com/docs/DOC-13144****
>
> ** **
>
> ** **
>
> I?m following these steps in editing the taps.aef file to have users enter
> a * then number to get TAPS to use E.164.****
>
> ** **
>
> However  the last step I hit Ok and I get Error: Value Parsing Error,
> unable to parse expression****
>
> ** **
>
> See attached screen shot;****
>
> ** **
>
> ****
>
> ** **
>
> ** **
>
> ** **
>
> _______________________________________________
> cisco-voip mailing list
> [email protected]
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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Message: 14
Date: Mon, 6 May 2013 19:27:46 -0400
From: "Jason Aarons (AM)" <[email protected]>
To: "cisco-voip ([email protected])"
        <[email protected]>
Subject: [cisco-voip] SIPC CIPC 8.6.3 not liking PLAR
Message-ID:
        
<4e38db0a1959b04c8c83edcf069b53ed0d330ee...@usispclexdb01.na.didata.local>
        
Content-Type: text/plain; charset="windows-1252"

I've setup Auto Registration Calling Search Space and did a  urgent Translation 
Pattern (blank) to 55-555 (PLAR)

But when I go offhook on CIPC(SIP) it doesn't do the PLAR.  I can manually dial 
55-5555.

Any ideas on why the CIPC SIP phone isn't using the Translation Pattern?
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Message: 15
Date: Mon, 6 May 2013 19:38:12 -0400
From: "Jason Aarons (AM)" <[email protected]>
To: Justin Steinberg <[email protected]>
Cc: "cisco-voip \([email protected]\)"
        <[email protected]>
Subject: Re: [cisco-voip] TAPS with E.164 AppNote
Message-ID:
        
<4e38db0a1959b04c8c83edcf069b53ed0d330ee...@usispclexdb01.na.didata.local>
        
Content-Type: text/plain; charset="windows-1252"

Well this is embarrassingly easy!!!  I just deleted everything to the left and 
it let me save it.  Now to test it.

Oddly I noticed I can't reactive debug the taps.aef script.  It never fires off 
the debug.  I wanted to remove the Press 1 for English option.

From: Justin Steinberg [mailto:[email protected]]
Sent: Monday, May 06, 2013 5:57 PM
To: Jason Aarons (AM)
Cc: cisco-voip ([email protected])
Subject: Re: [cisco-voip] TAPS with E.164 AppNote


look at the error messge, there are all those junk characters there that cause 
the problem.  I don't know why they are inserted but it is easy enough to 
remove them.

essentially look at the syntax of that step before you open the step to edit 
it, then open the step and remove all the junk characters, then edit according 
to the documentation.  that should fix it for you.

On Mon, May 6, 2013 at 5:16 PM, Jason Aarons (AM) 
<[email protected]<mailto:[email protected]>> wrote:
Reading the comments in the below URL I see my exact error and that a Mr. 
William E Welch has a updated .aef file and to click the URL to download, but I 
get an Access Denied from cisco.com<http://cisco.com> ?


Here is the script location that William E Welch posted:

 https://communities.cisco.com/message/65893#65893

 I have used it on 9.0 Premium  but I have not used it on Standard or Enhanced. 
It is just a couple lines added to the current TAPS script. Good Work William.


From: Jason Aarons (AM)
Sent: Monday, May 06, 2013 5:07 PM
To: cisco-voip ([email protected]<mailto:[email protected]>)
Subject: TAPS with E.164 AppNote

https://supportforums.cisco.com/docs/DOC-13144


I'm following these steps in editing the taps.aef file to have users enter a * 
then number to get TAPS to use E.164.

However  the last step I hit Ok and I get Error: Value Parsing Error, unable to 
parse expression

See attached screen shot;

[cid:[email protected]]




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Message: 16
Date: Tue, 7 May 2013 00:47:34 +0000
From: Matthew Loraditch <[email protected]>
To: "Jason Aarons (AM)" <[email protected]>
Cc: "cisco-voip \([email protected]\)"
        <[email protected]>
Subject: Re: [cisco-voip] SIPC CIPC 8.6.3 not liking PLAR
Message-ID:
        <[email protected]>
Content-Type: text/plain; charset="windows-1252"

For PLAR on sip dialing rules must be setup. I just encountered this a few 
weeks ago myself. The newer version docs address this. Sorry I don't have it in 
front of me right now.

Sent from my iPad

On May 6, 2013, at 7:30 PM, "Jason Aarons (AM)" 
<[email protected]<mailto:[email protected]>> wrote:

I?ve setup Auto Registration Calling Search Space and did a  urgent Translation 
Pattern (blank) to 55-555 (PLAR)

But when I go offhook on CIPC(SIP) it doesn?t do the PLAR.  I can manually dial 
55-5555.

Any ideas on why the CIPC SIP phone isn?t using the Translation Pattern?
_______________________________________________
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[email protected]<mailto:[email protected]>
https://puck.nether.net/mailman/listinfo/cisco-voip
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Message: 17
Date: Mon, 6 May 2013 20:53:38 -0400
From: "Jason Aarons (AM)" <[email protected]>
To: Matthew Loraditch <[email protected]>
Cc: "cisco-voip \([email protected]\)"
        <[email protected]>
Subject: Re: [cisco-voip] SIPC CIPC 8.6.3 not liking PLAR
Message-ID:
        
<4e38db0a1959b04c8c83edcf069b53ed0d330ee...@usispclexdb01.na.didata.local>
        
Content-Type: text/plain; charset="windows-1252"

Based on some docs is SIP plar only supported on;

"Only Cisco SIP IP Phone model 7940/41, 7960/61, and 7970/71 support PLAR."

Or will CIPC and other models support PLAR ?  What model phone did you get it 
working on ?

From: Matthew Loraditch [mailto:[email protected]]
Sent: Monday, May 06, 2013 8:48 PM
To: Jason Aarons (AM)
Cc: cisco-voip ([email protected])
Subject: Re: [cisco-voip] SIPC CIPC 8.6.3 not liking PLAR


For PLAR on sip dialing rules must be setup. I just encountered this a few 
weeks ago myself. The newer version docs address this. Sorry I don't have it in 
front of me right now.

Sent from my iPad

On May 6, 2013, at 7:30 PM, "Jason Aarons (AM)" 
<[email protected]<mailto:[email protected]>> wrote:
I've setup Auto Registration Calling Search Space and did a  urgent Translation 
Pattern (blank) to 55-555 (PLAR)

But when I go offhook on CIPC(SIP) it doesn't do the PLAR.  I can manually dial 
55-5555.

Any ideas on why the CIPC SIP phone isn't using the Translation Pattern?
_______________________________________________
cisco-voip mailing list
[email protected]<mailto:[email protected]>
https://puck.nether.net/mailman/listinfo/cisco-voip


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Message: 18
Date: Tue, 7 May 2013 01:33:12 +0000
From: Matthew Loraditch <[email protected]>
To: "Jason Aarons (AM)" <[email protected]>
Cc: "cisco-voip \([email protected]\)"
        <[email protected]>
Subject: Re: [cisco-voip] SIPC CIPC 8.6.3 not liking PLAR
Message-ID:
        <[email protected]>
Content-Type: text/plain; charset="us-ascii"

It was actually an ATA187.


Matthew G. Loraditch - CCNP-Voice, CCNA, CCDA

1965 Greenspring Drive
Timonium, MD 21093

voice. 410.252.8830
fax.  410.252.9284

Twitter<http://twitter.com/heliontech>  |  
Facebook<http://www.facebook.com/#!/pages/Helion/252157915296>  | 
Website<http://www.heliontechnologies.com/>  |  Email 
Support<mailto:[email protected]?subject=Technical%20Support%20Request>


From: Jason Aarons (AM) [mailto:[email protected]]
Sent: Monday, May 06, 2013 8:54 PM
To: Matthew Loraditch
Cc: cisco-voip ([email protected])
Subject: RE: [cisco-voip] SIPC CIPC 8.6.3 not liking PLAR

Based on some docs is SIP plar only supported on;

"Only Cisco SIP IP Phone model 7940/41, 7960/61, and 7970/71 support PLAR."

Or will CIPC and other models support PLAR ?  What model phone did you get it 
working on ?

From: Matthew Loraditch [mailto:[email protected]]
Sent: Monday, May 06, 2013 8:48 PM
To: Jason Aarons (AM)
Cc: cisco-voip ([email protected]<mailto:[email protected]>)
Subject: Re: [cisco-voip] SIPC CIPC 8.6.3 not liking PLAR


For PLAR on sip dialing rules must be setup. I just encountered this a few 
weeks ago myself. The newer version docs address this. Sorry I don't have it in 
front of me right now.

Sent from my iPad

On May 6, 2013, at 7:30 PM, "Jason Aarons (AM)" 
<[email protected]<mailto:[email protected]>> wrote:
I've setup Auto Registration Calling Search Space and did a  urgent Translation 
Pattern (blank) to 55-555 (PLAR)

But when I go offhook on CIPC(SIP) it doesn't do the PLAR.  I can manually dial 
55-5555.

Any ideas on why the CIPC SIP phone isn't using the Translation Pattern?
_______________________________________________
cisco-voip mailing list
[email protected]<mailto:[email protected]>
https://puck.nether.net/mailman/listinfo/cisco-voip


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Message: 19
Date: Tue, 7 May 2013 07:47:16 -0500
From: "Heim, Dennis" <[email protected]>
To: "[email protected]" <[email protected]>
Subject: [cisco-voip] Recommended SIP SBC
Message-ID:
        <[email protected]>
Content-Type: text/plain; charset="us-ascii"

Any recommendations for a non-Cisco SIP SBC product? I need  one with a gui and 
prefer some multi-tenant options.

Dennis Heim | Sr. Unified Collaboration Team Lead & Solution Architect
World Wide Technology | 314.212.1814 | 
[email protected]<mailto:[email protected]>
"Creating Impact, Ignition & Scalability"

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Message: 20
Date: Tue, 7 May 2013 08:45:07 -0500
From: "Berry, Matthew" <[email protected]>
To: "Heim, Dennis" <[email protected]>, "[email protected]"
        <[email protected]>
Subject: Re: [cisco-voip] Recommended SIP SBC
Message-ID:
        <[email protected]>
Content-Type: text/plain; charset="us-ascii"

My vote would be Acme Packet's Session Director.

Their Net-Net Enterprise Session Director can run on a VM and support 25-250 
sessions.

Maybe my good friend and former CCIE Voice study partner, Mark Holloway, can 
comment on a good Acme solution.

Mark?

Matthew

From: cisco-voip [mailto:[email protected]] On Behalf Of Heim, 
Dennis
Sent: Tuesday, May 07, 2013 7:47 AM
To: [email protected]
Subject: [cisco-voip] Recommended SIP SBC

Any recommendations for a non-Cisco SIP SBC product? I need  one with a gui and 
prefer some multi-tenant options.

Dennis Heim | Sr. Unified Collaboration Team Lead & Solution Architect
World Wide Technology | 314.212.1814 | 
[email protected]<mailto:[email protected]>
"Creating Impact, Ignition & Scalability"

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Message: 21
Date: Tue, 7 May 2013 10:30:38 -0400
From: Wes Sisk <[email protected]>
To: Mike <[email protected]>
Cc: [email protected]
Subject: Re: [cisco-voip] CMBE upgrade 7.0 to 7.1.5 Attendant console
        issue
Message-ID: <[email protected]>
Content-Type: text/plain; charset="us-ascii"

CTI just translates message, the actual registration is processed by the ccm 
process. verify the config and then bounce the ccm process.

On May 6, 2013, at 8:47 AM, Mike <[email protected]> wrote:

Hey all I upgraded to 7.1.5 with no issue everything works except the attendant 
console CTI route point has a status of rejected. I restarted the service and 
CTI manager with no luck.
 
I created another attendant console route point and it registers no problems.
 
Anyone ever seen this?
 
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Message: 22
Date: Tue, 7 May 2013 15:49:37 +0200
From: "Thierry De Vriendt" <[email protected]>
To: <[email protected]>
Subject: [cisco-voip] CCM 8.6 - SIP Register and authentication
Message-ID: <[email protected]>
Content-Type: text/plain; charset="us-ascii"

Dear All,

 

We would like to configure on our Cisco Call Manager 8.6 different SIP
trunks to the same Asterisk server (same destination IP). The utility to
have different SIP trunks is for billing purposes inside the Asterisk server
(1 SIP trunk per customer). The username/password sends over the SIP
messages will make the difference.  

 

Is it possible that the Call Manager sends SIP register messages to the
Asterisk server (for incoming calls from the Asterisk)?

Is it possible that the Call Manager sends login/password within the SIP
messages when sending a call to the Asterisk server?

 

I don't really see in the configuration how to associate username/password
with a SIP trunk.

 

Thanks for your help.

 

Thierry D.

 

 

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Message: 23
Date: Tue, 7 May 2013 10:42:06 -0400
From: "Jason Aarons (AM)" <[email protected]>
To: Thierry De Vriendt <[email protected]>,
        "[email protected]" <[email protected]>
Subject: Re: [cisco-voip] CCM 8.6 - SIP Register and authentication
Message-ID:
        
<4e38db0a1959b04c8c83edcf069b53ed0d330ee...@usispclexdb01.na.didata.local>
        
Content-Type: text/plain; charset="windows-1252"

I'm suspecting the answer is you need a CUBE then CUBE to Asterisk

From: cisco-voip [mailto:[email protected]] On Behalf Of 
Thierry De Vriendt
Sent: Tuesday, May 07, 2013 9:50 AM
To: [email protected]
Subject: [cisco-voip] CCM 8.6 - SIP Register and authentication


Dear All,

We would like to configure on our Cisco Call Manager 8.6 different SIP trunks 
to the same Asterisk server (same destination IP). The utility to have 
different SIP trunks is for billing purposes inside the Asterisk server (1 SIP 
trunk per customer). The username/password sends over the SIP messages will 
make the difference.

Is it possible that the Call Manager sends SIP register messages to the 
Asterisk server (for incoming calls from the Asterisk)?
Is it possible that the Call Manager sends login/password within the SIP 
messages when sending a call to the Asterisk server?

I don't really see in the configuration how to associate username/password with 
a SIP trunk.

Thanks for your help.

Thierry D.




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Message: 24
Date: Tue, 7 May 2013 14:51:00 +0000
From: Jason Faraone <[email protected]>
To: "'[email protected]'" <[email protected]>
Subject: [cisco-voip] Cisco 7925 / Aerohive
Message-ID:
        <fb0beb07ca11e54a83b331ad1c263ed665bc0...@chq-mdf-mbx-01.paulo.com>
Content-Type: text/plain; charset="us-ascii"

I'm currently evaluating different wireless networking solutions, and 
Aerohive's offering is very compelling. The only reservation I have is that 
Cisco wireless (7921 / 7925) is critical in our environment, and buying Cisco 
wireless would be a safe bet.

Has anyone deployed Cisco wireless voice over Aerohive access points, and if 
so, how successful was it?
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Message: 25
Date: Tue, 7 May 2013 11:02:35 -0400
From: Wes Sisk <[email protected]>
To: Ed Leatherman <[email protected]>
Cc: Cisco VOIP <[email protected]>
Subject: Re: [cisco-voip] VMWare support vis-a-vis Cisco
Message-ID: <[email protected]>
Content-Type: text/plain; charset=us-ascii

ISV1 should get sync'd over to VMWare. I've forwarded your report to that team.

Otherwise ISV1 allows Cisco TAC to engage VMWare support when necessary.

-Wes

On May 2, 2013, at 7:57 AM, Ed Leatherman <[email protected]> wrote:

Question here for anyone that's dealt with this before.

I have some UCS hardware, which was purchased with VMWare licenses all from 
Cisco. Have had it for a few years now, and we've been buying support for the 
VMWare from Cisco under an ISV1 "smartnet" contract. 

When I go into my license page on VMWare.com it always shows these are expired 
support-wise, and no one at cisco that i've tried so far seems to know why. I 
decided to just get a support contract directly from VMWare this year for that 
reason, and they don't show us as having ever paid for support, so the quote is 
basically huge with multiple reinstatement fees.

Is the ISV1 support I purchased from Cisco only Cisco support, does VMWare not 
recognized it? How do other vendors do this? Our server team renews their 
support via Dell and never has this problem.

Thanks!

-- 
Ed Leatherman
_______________________________________________
cisco-voip mailing list
[email protected]
https://puck.nether.net/mailman/listinfo/cisco-voip




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

Message: 26
Date: Tue, 7 May 2013 10:17:05 -0500
From: Erick Wellnitz <[email protected]>
To: "Berry, Matthew" <[email protected]>
Cc: "[email protected]" <[email protected]>
Subject: Re: [cisco-voip] Recommended SIP SBC
Message-ID:
        <cak0wosdhnv1ohegvnmhdzsovhkox7bce0cr9s2xtaccodgp...@mail.gmail.com>
Content-Type: text/plain; charset="windows-1252"

I would second the ACME solutions.


On Tue, May 7, 2013 at 8:45 AM, Berry, Matthew <[email protected]>wrote:

> My vote would be Acme Packet?s Session Director.****
>
> ** **
>
> Their Net-Net Enterprise Session Director can run on a VM and support
> 25-250 sessions.****
>
> ** **
>
> Maybe my good friend and former CCIE Voice study partner, Mark Holloway,
> can comment on a good Acme solution.****
>
> ** **
>
> Mark?****
>
> * *
>
> *Matthew *
>
> ** **
>
> *From:* cisco-voip [mailto:[email protected]] *On Behalf
> Of *Heim, Dennis
> *Sent:* Tuesday, May 07, 2013 7:47 AM
> *To:* [email protected]
> *Subject:* [cisco-voip] Recommended SIP SBC****
>
> ** **
>
> Any recommendations for a non-Cisco SIP SBC product? I need  one with a
> gui and prefer some multi-tenant options.****
>
> ** **
>
> *Dennis Heim | **Sr. Unified Collaboration Team Lead & Solution Architect
> *****
>
> World Wide Technology | 314.212.1814 | [email protected]**
>
> ?Creating Impact, Ignition & Scalability?****
>
> ** **
>
> _______________________________________________
> cisco-voip mailing list
> [email protected]
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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Message: 27
Date: Tue, 7 May 2013 11:49:10 -0400
From: "Matt Slaga (AM)" <[email protected]>
To: Thierry De Vriendt <[email protected]>,
        "[email protected]" <[email protected]>
Subject: Re: [cisco-voip] CCM 8.6 - SIP Register and authentication
Message-ID:
        
<5fe225375f6e3f4493474b90bc1b94efc3f4dc8...@usispclexdb01.na.didata.local>
        
Content-Type: text/plain; charset="windows-1252"

I would suggest multiple SIP trunks using different port combinations rather 
than user/pass to distinguish.



From: cisco-voip [mailto:[email protected]] On Behalf Of 
Thierry De Vriendt
Sent: Tuesday, May 07, 2013 9:50 AM
To: [email protected]
Subject: [cisco-voip] CCM 8.6 - SIP Register and authentication


Dear All,

We would like to configure on our Cisco Call Manager 8.6 different SIP trunks 
to the same Asterisk server (same destination IP). The utility to have 
different SIP trunks is for billing purposes inside the Asterisk server (1 SIP 
trunk per customer). The username/password sends over the SIP messages will 
make the difference.

Is it possible that the Call Manager sends SIP register messages to the 
Asterisk server (for incoming calls from the Asterisk)?
Is it possible that the Call Manager sends login/password within the SIP 
messages when sending a call to the Asterisk server?

I don't really see in the configuration how to associate username/password with 
a SIP trunk.

Thanks for your help.

Thierry D.




itevomcid
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Subject: Digest Footer

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