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Today's Topics:

   1. Re: SIP trunk failover question (Erick Wellnitz)
   2. Recommended SIP SBC (Robert Kulagowski)
   3. Re: SIP trunk failover question (Ryan Ratliff)
   4. Re: Branche CME with Head office CUCM (Ryan Ratliff)
   5. Re: Recommended SIP SBC (Matt Slaga (AM))
   6. Re: SIP trunk failover question (Robert Kulagowski)
   7. Re: SIP trunk failover question (Erick Wellnitz)
   8. TAPS and Intercom? Default Activated Device ? (Jason Aarons (AM))
   9. Re: TAPS and Intercom? Default Activated Device ? (Nate VanMaren)
  10. Extension Mobility Issue (Nick)
  11. Re: Extension Mobility Issue (Andy)
  12. Directory Lookups not working for MAC OS and Jabber       for Mac
      8.6 (Gregory Wenzel)
  13. Problem; Bulk Admin > Phones > Insert Phones then  error;
      Line number 4 is not configured for Intercom Line on the template
      (Jason Aarons (AM))
  14. Re: Directory Lookups not working for MAC OS and Jabber   for
      Mac 8.6 (Jason Aarons (AM))
  15. Re: Directory Lookups not working for MAC OS and Jabber   for
      Mac 8.6 (Gregory Wenzel)
  16. Re: Problem; Bulk Admin > Phones > Insert Phones then  error;
      Line number 4 is not configured for Intercom Line on the template
      (Jason Aarons (AM))
  17. Re: Extension Mobility Issue (Nick)
  18. Re: SIP trunk failover question (Ryan Ratliff)
  19. Shared Lines and Intercom Line? (Jason Aarons (AM))
  20. Re: Directory Lookups not working for MAC OS and Jabber   for
      Mac 8.6 (Jason Aarons (AM))
  21. Re: Shared Lines and Intercom Line? (Bill Talley)
  22. Re: Shared Lines and Intercom Line? (Bill Talley)
  23. Re: Shared Lines and Intercom Line? (Jason Aarons (AM))


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

Message: 1
Date: Tue, 14 May 2013 12:00:07 -0500
From: Erick Wellnitz <[email protected]>
To: Robert Kulagowski <[email protected]>
Cc: cisco-voip <[email protected]>
Subject: Re: [cisco-voip] SIP trunk failover question
Message-ID:
        <cak0wosclkk-citudr6nhaw0czb1klrzgkcznqvgeyysfkhv...@mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"

Looks like this isn't going to happen until our upgrade.  It's waited this
long already...


On Tue, May 14, 2013 at 9:54 AM, Erick Wellnitz <[email protected]>wrote:

> Great!  That's exactly what I was looking for but didn't know what it was
> called.
>
>
> On Tue, May 14, 2013 at 9:46 AM, Robert Kulagowski <[email protected]>wrote:
>
>> On Tue, May 14, 2013 at 9:27 AM, Erick Wellnitz <[email protected]>
>> wrote:
>> > I use top down distribution.  Are there any other settings I need to
>> make
>> > sure that the calls don't fail when the SIP trunk is unavailable for any
>> > reason?
>>
>> Depending on the version of CM that you're using, you may want to
>> check "SIP PING".
>>
>
>
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Message: 2
Date: Tue, 14 May 2013 11:33:40 -0500
From: Robert Kulagowski <[email protected]>
To: Cisco VOIP <[email protected]>
Subject: [cisco-voip] Recommended SIP SBC
Message-ID:
        <CAGe0w3RevfrW9h5aV=ezpaa0ytwwdwnpa6-gnslujnm_rsa...@mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1

[Sorry for breaking the thread, but original message is gone.]

Other than Acme Packet, has anyone else deployed something like a
Sonus? The merger with Oracle seems to be slowing things down over
there, and I may not be able to get equipment from the Demo Depot
until after they're done merging, and in the 3 weeks since our meeting
the AP sales guy still hasn't gotten me a copy of the "software only"
SBC to even start an eval even though I keep pestering him.

Something that can do 200 simultaneous sessions is what we're looking
for, G.711, no fancy transcoding required.

I found a SBC comparison report, but it's $3400, so a bit expensive,
but the executive summary put Sonus as their choice.

If you have experience with multiple SBCs and can go into why one and
not another then that's an added bonus.

Thanks.


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

Message: 3
Date: Tue, 14 May 2013 14:32:07 -0400
From: Ryan Ratliff <[email protected]>
To: Erick Wellnitz <[email protected]>
Cc: cisco-voip <[email protected]>
Subject: Re: [cisco-voip] SIP trunk failover question
Message-ID: <[email protected]>
Content-Type: text/plain; charset="iso-8859-1"

You may also want to look at the behavior available to you by having multiple 
route groups in a route list vs multiple devices in the RG. 
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/6_0_1/ccmsys/a03rp.html#wp1045311
 does a decent job of covering this.


-Ryan

On May 14, 2013, at 1:00 PM, Erick Wellnitz <[email protected]> wrote:

Looks like this isn't going to happen until our upgrade.  It's waited this long 
already...


On Tue, May 14, 2013 at 9:54 AM, Erick Wellnitz <[email protected]> wrote:
Great!  That's exactly what I was looking for but didn't know what it was 
called.


On Tue, May 14, 2013 at 9:46 AM, Robert Kulagowski <[email protected]> wrote:
On Tue, May 14, 2013 at 9:27 AM, Erick Wellnitz <[email protected]> wrote:
> I use top down distribution.  Are there any other settings I need to make
> sure that the calls don't fail when the SIP trunk is unavailable for any
> reason?

Depending on the version of CM that you're using, you may want to
check "SIP PING".


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

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Message: 4
Date: Tue, 14 May 2013 14:33:35 -0400
From: Ryan Ratliff <[email protected]>
To: Adewale2 <[email protected]>
Cc: "[email protected]" <[email protected]>
Subject: Re: [cisco-voip] Branche CME with Head office CUCM
Message-ID: <[email protected]>
Content-Type: text/plain; charset="iso-8859-1"

The MGCP gateway will register with CUCM and so all calls that you want to go 
through the MGCP ports must route through CUCM.  

-Ryan

On May 14, 2013, at 11:41 AM, Adewale2 <[email protected]> wrote:


Hello All,
Is any way I can connect H.323(CME) with MGCP gateway on CUCM  for free call 
between Branch & head office please ?

Topology 
                                        [CUCM]
                                                |
CME------>        ?              MGCP_GW

Should i link it with CUCM or MGCP GW for TEHO ( Free WAN) calls?


Heaven is Real try your best to be there, and in all your getting get 
understanding
God's Own
Addy
_______________________________________________
cisco-voip mailing list
[email protected]
https://puck.nether.net/mailman/listinfo/cisco-voip

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Message: 5
Date: Tue, 14 May 2013 15:53:55 -0400
From: "Matt Slaga (AM)" <[email protected]>
To: Robert Kulagowski <[email protected]>, Cisco VOIP
        <[email protected]>
Subject: Re: [cisco-voip] Recommended SIP SBC
Message-ID:
        
<5fe225375f6e3f4493474b90bc1b94efc3f4dc8...@usispclexdb01.na.didata.local>
        
Content-Type: text/plain; charset="windows-1252"

A couple listed below.

Audiocodes
Ferrari (not the car)
Sangoma
Italtel
Ingate
TE Systems




From: cisco-voip [mailto:[email protected]] On Behalf Of 
Robert Kulagowski
Sent: Tuesday, May 14, 2013 12:34 PM
To: Cisco VOIP
Subject: [cisco-voip] Recommended SIP SBC



[Sorry for breaking the thread, but original message is gone.]

Other than Acme Packet, has anyone else deployed something like a
Sonus? The merger with Oracle seems to be slowing things down over
there, and I may not be able to get equipment from the Demo Depot
until after they're done merging, and in the 3 weeks since our meeting
the AP sales guy still hasn't gotten me a copy of the "software only"
SBC to even start an eval even though I keep pestering him.

Something that can do 200 simultaneous sessions is what we're looking
for, G.711, no fancy transcoding required.

I found a SBC comparison report, but it's $3400, so a bit expensive,
but the executive summary put Sonus as their choice.

If you have experience with multiple SBCs and can go into why one and
not another then that's an added bonus.

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


itevomcid
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Message: 6
Date: Tue, 14 May 2013 14:47:16 -0500
From: Robert Kulagowski <[email protected]>
To: cisco-voip <[email protected]>
Subject: Re: [cisco-voip] SIP trunk failover question
Message-ID:
        <CAGe0w3Rm0RTVpSMzCd38ch2kihscYGCicx1Lr4LntoaD-=f...@mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1

On Tue, May 14, 2013 at 1:32 PM, Ryan Ratliff <[email protected]> wrote:
> You may also want to look at the behavior available to you by having
> multiple route groups in a route list vs multiple devices in the RG.
> http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/6_0_1/ccmsys/a03rp.html#wp1045311
> does a decent job of covering this.

Can you dive a little deeper into this?

For my SIP testing, I've got a single route group, and in that route
group I have the SIP trunk first, then H.323 gateways after it. During
my testing the 2851 stopped responding; even with the ethernet port
shut down on the switch side (thereby removing any traffic if it was
due to a DOS), I still couldn't get a response on the console port,
and had to have remote hands reboot it. It was still "up" enough to
respond to pings (even though they were taking 2000ms instead of 2ms),
so CM was still sending it calls, but the caller was just hearing dead
air.

Since then we've been watching it like a hawk.

If the SIP trunk is moved to its own RG, and the RL is updated so that
it now has two RGs in it, would that have mitigated what happened?

Thanks.


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

Message: 7
Date: Tue, 14 May 2013 15:46:36 -0500
From: Erick Wellnitz <[email protected]>
To: Ryan Ratliff <[email protected]>
Cc: cisco-voip <[email protected]>
Subject: Re: [cisco-voip] SIP trunk failover question
Message-ID:
        <cak0wosczmazr8zwpxcxxzpjxncf75tch16cvzfkgwe7b1z+...@mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"

When I started my design I realized I had to use multiple RG for the RL and
that actually solves the problem on the 'global' side of the CUBE when
using the busy/unreachable service parameters.

SIP ping will solve the problem on the local side if routing gets botched
or some other issue disrupts connectivity to the CUBE.

Thanks Ryan and Robert.  I learned a thing or two about making my SIP
trunks a bit more resilient today.


On Tue, May 14, 2013 at 1:32 PM, Ryan Ratliff <[email protected]> wrote:

> You may also want to look at the behavior available to you by having
> multiple route groups in a route list vs multiple devices in the RG.
>
> http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/6_0_1/ccmsys/a03rp.html#wp1045311
>  does
> a decent job of covering this.
>
>
> -Ryan
>
> On May 14, 2013, at 1:00 PM, Erick Wellnitz <[email protected]>
> wrote:
>
> Looks like this isn't going to happen until our upgrade.  It's waited this
> long already...
>
>
> On Tue, May 14, 2013 at 9:54 AM, Erick Wellnitz 
> <[email protected]>wrote:
>
>> Great!  That's exactly what I was looking for but didn't know what it was
>> called.
>>
>>
>> On Tue, May 14, 2013 at 9:46 AM, Robert Kulagowski <[email protected]>wrote:
>>
>>> On Tue, May 14, 2013 at 9:27 AM, Erick Wellnitz <[email protected]>
>>> wrote:
>>> > I use top down distribution.  Are there any other settings I need to
>>> make
>>> > sure that the calls don't fail when the SIP trunk is unavailable for
>>> any
>>> > reason?
>>>
>>> Depending on the version of CM that you're using, you may want to
>>> check "SIP PING".
>>>
>>
>>
> _______________________________________________
> cisco-voip mailing list
> [email protected]
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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Message: 8
Date: Tue, 14 May 2013 21:33:58 -0400
From: "Jason Aarons (AM)" <[email protected]>
To: "cisco-voip ([email protected])"
        <[email protected]>
Subject: [cisco-voip] TAPS and Intercom? Default Activated Device ?
Message-ID:
        
<4e38db0a1959b04c8c83edcf069b53ed0d331c7...@usispclexdb01.na.didata.local>
        
Content-Type: text/plain; charset="windows-1252"

I have a large number of phones that I plan to TAP into CallManager 9.1.  These 
6945s all have the Intercom feature.  Can I TAP them in? Seems the Default 
Activated Device needs to be set.  Does TAPs set the Default Activated Device 
during the TAPs process?
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Message: 9
Date: Wed, 15 May 2013 02:50:40 +0000
From: Nate VanMaren <[email protected]>
To: "Jason Aarons (AM)" <[email protected]>, "cisco-voip
        ([email protected])" <[email protected]>
Subject: Re: [cisco-voip] TAPS and Intercom? Default Activated Device
        ?
Message-ID:
        <[email protected]>
Content-Type: text/plain; charset="us-ascii"

>From what I remember, no it does not.  But that was not 9.1, but I doubt it 
>changed.

From: cisco-voip [mailto:[email protected]] On Behalf Of Jason 
Aarons (AM)
Sent: Tuesday, May 14, 2013 7:34 PM
To: cisco-voip ([email protected])
Subject: [cisco-voip] TAPS and Intercom? Default Activated Device ?

I have a large number of phones that I plan to TAP into CallManager 9.1.  These 
6945s all have the Intercom feature.  Can I TAP them in? Seems the Default 
Activated Device needs to be set.  Does TAPs set the Default Activated Device 
during the TAPs process?


 NOTICE: This email message is for the sole use of the intended recipient(s) 
and may contain confidential and privileged information. Any unauthorized 
review, use, disclosure or distribution is prohibited. If you are not the 
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Message: 10
Date: Wed, 15 May 2013 11:18:44 +0100
From: Nick <[email protected]>
To: [email protected]
Subject: [cisco-voip] Extension Mobility Issue
Message-ID:
        <CAGJstULfRyYUhqELwmN7xYBdx52TnK_NaFop=ntlul7xog_...@mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"

Hi All



I have an issue with only one subscriber, which when phones are registered
to it and  you log in with extension mobility it does not display the
device profile info, the phone in CUCM admin shows the UDP as logged in and
when you go to extension mobility service, it shows the user as logged in.
I have tried reregistering the phone to the other 3 subscribers and the
phone registers fine and displays the profile. I have rebooted the affected
subscriber and during the reboot the phone registers with the backup
subscriber and display UDP fine, but when the phone registers with its
primary sub it the UDP will not display.


Anyone come across this before?

Nick
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Message: 11
Date: Wed, 15 May 2013 13:13:24 +0100
From: Andy <[email protected]>
To: [email protected]
Subject: Re: [cisco-voip] Extension Mobility Issue
Message-ID: <[email protected]>
Content-Type: text/plain; charset="iso-8859-1"; Format="flowed"

How is the replication on the cluster?

Regards

Andy

On 15/05/2013 11:18, Nick wrote:
>
> Hi All
>
> I have an issue with only one subscriber, which when phones are 
> registered to it and  you log in with extension mobility it does not 
> display the device profile info, the phone in CUCM admin shows the UDP 
> as logged in and when you go to extension mobility service, it shows 
> the user as logged in.  I have tried reregistering the phone to the 
> other 3 subscribers and the phone registers fine and displays the 
> profile. I have rebooted the affected subscriber and during the reboot 
> the phone registers with the backup subscriber and display UDP fine, 
> but when the phone registers with its primary sub it the UDP will not 
> display.
>
> Anyone come across this before?
> Nick
>
>
> _______________________________________________
> cisco-voip mailing list
> [email protected]
> https://puck.nether.net/mailman/listinfo/cisco-voip

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Message: 12
Date: Wed, 15 May 2013 12:48:48 +0000
From: Gregory Wenzel <[email protected]>
To: "[email protected]" <[email protected]>
Subject: [cisco-voip] Directory Lookups not working for MAC OS and
        Jabber  for Mac 8.6
Message-ID:
        <[email protected]>
Content-Type: text/plain; charset="us-ascii"

Hi all,

Has anyone run into an issue with Jabber directory lookup with a MAC that is 
not on a customer domain? I have a customer that I setup with SRST across a 
site-to-site VPN. The customer is claiming that the users on can't do lookups, 
yet all the other features of Jabber work. MAC OS 10.7, Jabber for Mac 8.6

TIA


Greg Wenzel




This message w/attachments (message) is solely for the use of the intended 
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and then please delete and destroy all copies and attachments, and be advised 
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Message: 13
Date: Wed, 15 May 2013 09:02:56 -0400
From: "Jason Aarons (AM)" <[email protected]>
To: "cisco-voip ([email protected])"
        <[email protected]>
Subject: [cisco-voip] Problem; Bulk Admin > Phones > Insert Phones
        then  error; Line number 4 is not configured for Intercom Line on the
        template
Message-ID:
        
<4e38db0a1959b04c8c83edcf069b53ed0d331c7...@usispclexdb01.na.didata.local>
        
Content-Type: text/plain; charset="windows-1252"

During Bulk Admin > Phones > Insert Phones > Specific Details I get the below 
error;

"Line number 4 is not configured for Intercom Line on the template"

.  I included a screen shot of the Phone Template showing Line 4 is an Intercom.

Am I doing something wrong?



[cid:[email protected]]

Here was the data;
Device Name,Description,Device Pool,Phone Button Template,CSS,Location,Common 
Phone Profile,Owner User ID,Device Presence Group,Device Security 
Profile,Device Subscribe CSS,SIP Profile,MTP Preferred Originating Codec,Device 
Mobility Mode,Common Device Configuration,DND Option,Directory Number 1,Route 
Partition 1,Voice Mail Profile 1,Line CSS 1,Auto Answer 1,Forward All 
Destination 1,Forward All CSS 1,Forward Busy Internal Destination 1,Forward 
Busy Internal CSS 1,Forward Busy External Destination 1,Forward Busy External 
CSS 1,Forward No Answer Internal Destination 1,Forward No Answer Internal CSS 
1,Forward No Answer External Destination 1,Forward No Answer External CSS 
1,Forward No Coverage Internal Destination 1,Forward No Coverage Internal CSS 
1,Forward No Coverage External Destination 1,Forward No Coverage External CSS 
1,Forward No Answer Ring Duration 1,Call Pickup Group 1,Line Text Label 
1,External Phone Number Mask 1,Maximum Number of Calls 1,Busy Trigger 1,Visual 
Mess!
 age Waiting Indicator Policy 1,Ring setting (Phone Idle) 1,Ring Setting (Phone 
Active) 1,Line Description 1,Alerting Name 1,ASCII Alerting Name 1,Line 
Presence Group 1,Display 1,ASCII Display 1,ASCII Line Text Label 1,Hold 
Reversion Ring Duration 1,Hold Reversion Notification Interval 1,Audible 
Message Waiting Indicator Policy 1,Calling Search Space Activation Policy 1,Log 
Missed Calls 1,Directory Number 2,Route Partition 2,Voice Mail Profile 2,Line 
CSS 2,Forward All Destination 2,Forward All CSS 2,Forward Busy Internal 
Destination 2,Forward Busy Internal CSS 2,Forward Busy External Destination 
2,Forward Busy External CSS 2,Forward No Answer Internal Destination 2,Forward 
No Answer Internal CSS 2,Forward No Answer External Destination 2,Forward No 
Answer External CSS 2,Forward No Answer Ring Duration 2,Call Pickup Group 
2,Line Text Label 2,External Phone Number Mask 2,Maximum Number of Calls 2,Busy 
Trigger 2,Visual Message Waiting Indicator Policy 2,Ring setting (Phone Idle!
 ) 2,Ring Setting (Phone Active) 2,Line Description 2,Alerting Name 2,ASCII 
Alerting Name 2,Line Presence Group 2,Secondary CSS for Forward All 2,Forward 
on CTI Failure Destination 2,Forward on CTI Failure CSS 2,Display 2,ASCII 
Display 2,ASCII Line Text Label 2,Directory Number 3,Route Partition 3,Voice 
Mail Profile 3,Line CSS 3,Forward All Destination 3,Forward All CSS 3,Forward 
Busy Internal Destination 3,Forward Busy Internal CSS 3,Forward Busy External 
Destination 3,Forward Busy External CSS 3,Forward No Answer Internal 
Destination 3,Forward No Answer Internal CSS 3,Forward No Answer External 
Destination 3,Forward No Answer External CSS 3,Forward No Answer Ring Duration 
3,Call Pickup Group 3,Line Text Label 3,External Phone Number Mask 3,Maximum 
Number of Calls 3,Busy Trigger 3,Visual Message Waiting Indicator Policy 3,Ring 
setting (Phone Idle) 3,Ring Setting (Phone Active) 3,Line Description 
3,Alerting Name 3,ASCII Alerting Name 3,Line Presence Group 3,Secondary CSS for 
Forward All 3,Forward on CTI Failure Destination 3,Forward on CTI Fail!
 ure CSS 3,Display 3,ASCII Display 3,ASCII Line Text Label 3,Intercom Directory 
Number 1,Intercom Route Partition 1,Intercom Description 1,Intercom Alerting 
Name 1,Intercom ASCII Alerting Name 1,Intercom CSS 1,Intercom Presence Group 
1,Intercom Display 1,Intercom ASCII Display 1,Intercom Line Text Label 
1,Intercom ASCII Line Text Label 1,Intercom Speed Dial 1,Intercom External 
Phone Number Mask 1,Intercom Caller Name 1,Intercom Caller Number 1,Intercom 
Auto Answer 1,Intercom Default Activated Device 1
,0217 - Mary Betty Smith,DP_Newark_NC19,SIP 6945 
3L1I,CSS_Newark_831-Device,LOC_Newark,Standard Common Phone Profile,,Standard 
Presence group,Cisco 6945 - Standard SIP Non-Secure Profile,,Standard SIP 
Profile,711ulaw,Default,CDC_Newark,Ringer 
Off,\+19044910217,Z-MIGRATION-NewarkNorth,Newark_VM_Profile,CSS_NANP_International,Auto
 Answer Off,,,,,,,,,,,,,,,,,Mary Betty Smith - 0217,9044910217,2,2,Use System 
Policy,Use System Default,Use System Default,0217 - Mary Betty Smith,Mary Betty 
Smith,Mary Betty Smith,Standard Presence group,Mary Betty Smith,Mary Betty 
Smith,Mary Betty Smith - 0217,,,Default,Use System 
Default,t,\+19044916717,Z-MIGRATION-NewarkNorth,NoVoiceMail,International,,,,,,,,f,,,,,6717,9044916717,,,,Ring,,Mary
 Betty Smith,Mary Betty Smith,Mary Betty Smith,,,,,Mary Betty Smith,Mary Betty 
Smith,6717,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,0217,Intercom-PT,Mary Betty 
Smith,Mary Betty Smith,Mary Betty Smith,Intercom-CSS,,Mary Betty Smith,Mary 
Betty Smith,Intercom,Intercom,,,!
 ,,,None
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Message: 14
Date: Wed, 15 May 2013 09:04:25 -0400
From: "Jason Aarons (AM)" <[email protected]>
To: Gregory Wenzel <[email protected]>, "[email protected]"
        <[email protected]>
Subject: Re: [cisco-voip] Directory Lookups not working for MAC OS and
        Jabber  for Mac 8.6
Message-ID:
        
<4e38db0a1959b04c8c83edcf069b53ed0d331c7...@usispclexdb01.na.didata.local>
        
Content-Type: text/plain; charset="windows-1252"

Does it work in the office?   If not part of a domain how is the directory 
lookup occurring?

The Jabber Administration guide has some good details on this.

From: cisco-voip [mailto:[email protected]] On Behalf Of 
Gregory Wenzel
Sent: Wednesday, May 15, 2013 8:49 AM
To: [email protected]
Subject: [cisco-voip] Directory Lookups not working for MAC OS and Jabber for 
Mac 8.6


Hi all,

Has anyone run into an issue with Jabber directory lookup with a MAC that is 
not on a customer domain? I have a customer that I setup with SRST across a 
site-to-site VPN. The customer is claiming that the users on can't do lookups, 
yet all the other features of Jabber work. MAC OS 10.7, Jabber for Mac 8.6

TIA


Greg Wenzel




This message w/attachments (message) is solely for the use of the intended 
recipient(s) and may contain information that is privileged, confidential or 
proprietary. If you are not an intended recipient, please notify the sender, 
and then please delete and destroy all copies and attachments, and be advised 
that any review or dissemination of, or the taking of any action in reliance 
on, the information contained in or attached to this message is prohibited. 
Unless specifically indicated, this message is not an offer to sell or a 
solicitation of any products.   

itevomcid
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Message: 15
Date: Wed, 15 May 2013 13:09:16 +0000
From: Gregory Wenzel <[email protected]>
To: "Jason Aarons (AM)" <[email protected]>,
        "[email protected]" <[email protected]>
Subject: Re: [cisco-voip] Directory Lookups not working for MAC OS and
        Jabber  for Mac 8.6
Message-ID:
        <[email protected]>
Content-Type: text/plain; charset="us-ascii"

Good point. I will look in that direction. Thanks Jason :)

From: Jason Aarons (AM) [mailto:[email protected]]
Sent: Wednesday, May 15, 2013 9:04 AM
To: Gregory Wenzel; [email protected]
Subject: RE: [cisco-voip] Directory Lookups not working for MAC OS and Jabber 
for Mac 8.6

Does it work in the office?   If not part of a domain how is the directory 
lookup occurring?

The Jabber Administration guide has some good details on this.

From: cisco-voip [mailto:[email protected]] On Behalf Of 
Gregory Wenzel
Sent: Wednesday, May 15, 2013 8:49 AM
To: [email protected]<mailto:[email protected]>
Subject: [cisco-voip] Directory Lookups not working for MAC OS and Jabber for 
Mac 8.6


Hi all,

Has anyone run into an issue with Jabber directory lookup with a MAC that is 
not on a customer domain? I have a customer that I setup with SRST across a 
site-to-site VPN. The customer is claiming that the users on can't do lookups, 
yet all the other features of Jabber work. MAC OS 10.7, Jabber for Mac 8.6

TIA


Greg Wenzel




This message w/attachments (message) is solely for the use of the intended 
recipient(s) and may contain information that is privileged, confidential or 
proprietary. If you are not an intended recipient, please notify the sender, 
and then please delete and destroy all copies and attachments, and be advised 
that any review or dissemination of, or the taking of any action in reliance 
on, the information contained in or attached to this message is prohibited. 
Unless specifically indicated, this message is not an offer to sell or a 
solicitation of any products.   

itevomcid

This message w/attachments (message) is solely for the use of the intended 
recipient(s) and may contain information that is privileged, confidential or 
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Message: 16
Date: Wed, 15 May 2013 09:31:56 -0400
From: "Jason Aarons (AM)" <[email protected]>
To: "cisco-voip ([email protected])"
        <[email protected]>
Subject: Re: [cisco-voip] Problem; Bulk Admin > Phones > Insert Phones
        then  error; Line number 4 is not configured for Intercom Line on the
        template
Message-ID:
        
<4e38db0a1959b04c8c83edcf069b53ed0d331c7...@usispclexdb01.na.didata.local>
        
Content-Type: text/plain; charset="windows-1252"

Only after I spent an hour, composed a lengthy email, I then did an Export All 
Details and see the Intercom Directory Number 4 should be Intercom Directory 
Number 1.  BAT columns sometimes require a little Prozac.

From: Jason Aarons (AM)
Sent: Wednesday, May 15, 2013 9:03 AM
To: cisco-voip ([email protected])
Subject: Problem; Bulk Admin > Phones > Insert Phones then error; Line number 4 
is not configured for Intercom Line on the template

During Bulk Admin > Phones > Insert Phones > Specific Details I get the below 
error;

"Line number 4 is not configured for Intercom Line on the template"

.  I included a screen shot of the Phone Template showing Line 4 is an Intercom.

Am I doing something wrong?



[cid:[email protected]]

Here was the data;
Device Name,Description,Device Pool,Phone Button Template,CSS,Location,Common 
Phone Profile,Owner User ID,Device Presence Group,Device Security 
Profile,Device Subscribe CSS,SIP Profile,MTP Preferred Originating Codec,Device 
Mobility Mode,Common Device Configuration,DND Option,Directory Number 1,Route 
Partition 1,Voice Mail Profile 1,Line CSS 1,Auto Answer 1,Forward All 
Destination 1,Forward All CSS 1,Forward Busy Internal Destination 1,Forward 
Busy Internal CSS 1,Forward Busy External Destination 1,Forward Busy External 
CSS 1,Forward No Answer Internal Destination 1,Forward No Answer Internal CSS 
1,Forward No Answer External Destination 1,Forward No Answer External CSS 
1,Forward No Coverage Internal Destination 1,Forward No Coverage Internal CSS 
1,Forward No Coverage External Destination 1,Forward No Coverage External CSS 
1,Forward No Answer Ring Duration 1,Call Pickup Group 1,Line Text Label 
1,External Phone Number Mask 1,Maximum Number of Calls 1,Busy Trigger 1,Visual 
Mess!
 age Waiting Indicator Policy 1,Ring setting (Phone Idle) 1,Ring Setting (Phone 
Active) 1,Line Description 1,Alerting Name 1,ASCII Alerting Name 1,Line 
Presence Group 1,Display 1,ASCII Display 1,ASCII Line Text Label 1,Hold 
Reversion Ring Duration 1,Hold Reversion Notification Interval 1,Audible 
Message Waiting Indicator Policy 1,Calling Search Space Activation Policy 1,Log 
Missed Calls 1,Directory Number 2,Route Partition 2,Voice Mail Profile 2,Line 
CSS 2,Forward All Destination 2,Forward All CSS 2,Forward Busy Internal 
Destination 2,Forward Busy Internal CSS 2,Forward Busy External Destination 
2,Forward Busy External CSS 2,Forward No Answer Internal Destination 2,Forward 
No Answer Internal CSS 2,Forward No Answer External Destination 2,Forward No 
Answer External CSS 2,Forward No Answer Ring Duration 2,Call Pickup Group 
2,Line Text Label 2,External Phone Number Mask 2,Maximum Number of Calls 2,Busy 
Trigger 2,Visual Message Waiting Indicator Policy 2,Ring setting (Phone Idle!
 ) 2,Ring Setting (Phone Active) 2,Line Description 2,Alerting Name 2,ASCII 
Alerting Name 2,Line Presence Group 2,Secondary CSS for Forward All 2,Forward 
on CTI Failure Destination 2,Forward on CTI Failure CSS 2,Display 2,ASCII 
Display 2,ASCII Line Text Label 2,Directory Number 3,Route Partition 3,Voice 
Mail Profile 3,Line CSS 3,Forward All Destination 3,Forward All CSS 3,Forward 
Busy Internal Destination 3,Forward Busy Internal CSS 3,Forward Busy External 
Destination 3,Forward Busy External CSS 3,Forward No Answer Internal 
Destination 3,Forward No Answer Internal CSS 3,Forward No Answer External 
Destination 3,Forward No Answer External CSS 3,Forward No Answer Ring Duration 
3,Call Pickup Group 3,Line Text Label 3,External Phone Number Mask 3,Maximum 
Number of Calls 3,Busy Trigger 3,Visual Message Waiting Indicator Policy 3,Ring 
setting (Phone Idle) 3,Ring Setting (Phone Active) 3,Line Description 
3,Alerting Name 3,ASCII Alerting Name 3,Line Presence Group 3,Secondary CSS for 
Forward All 3,Forward on CTI Failure Destination 3,Forward on CTI Fail!
 ure CSS 3,Display 3,ASCII Display 3,ASCII Line Text Label 3,Intercom Directory 
Number 1,Intercom Route Partition 1,Intercom Description 1,Intercom Alerting 
Name 1,Intercom ASCII Alerting Name 1,Intercom CSS 1,Intercom Presence Group 
1,Intercom Display 1,Intercom ASCII Display 1,Intercom Line Text Label 
1,Intercom ASCII Line Text Label 1,Intercom Speed Dial 1,Intercom External 
Phone Number Mask 1,Intercom Caller Name 1,Intercom Caller Number 1,Intercom 
Auto Answer 1,Intercom Default Activated Device 1
,0217 - Mary Betty Smith,DP_Newark_NC19,SIP 6945 
3L1I,CSS_Newark_831-Device,LOC_Newark,Standard Common Phone Profile,,Standard 
Presence group,Cisco 6945 - Standard SIP Non-Secure Profile,,Standard SIP 
Profile,711ulaw,Default,CDC_Newark,Ringer 
Off,\+19044910217,Z-MIGRATION-NewarkNorth,Newark_VM_Profile,CSS_NANP_International,Auto
 Answer Off,,,,,,,,,,,,,,,,,Mary Betty Smith - 0217,9044910217,2,2,Use System 
Policy,Use System Default,Use System Default,0217 - Mary Betty Smith,Mary Betty 
Smith,Mary Betty Smith,Standard Presence group,Mary Betty Smith,Mary Betty 
Smith,Mary Betty Smith - 0217,,,Default,Use System 
Default,t,\+19044916717,Z-MIGRATION-NewarkNorth,NoVoiceMail,International,,,,,,,,f,,,,,6717,9044916717,,,,Ring,,Mary
 Betty Smith,Mary Betty Smith,Mary Betty Smith,,,,,Mary Betty Smith,Mary Betty 
Smith,6717,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,0217,Intercom-PT,Mary Betty 
Smith,Mary Betty Smith,Mary Betty Smith,Intercom-CSS,,Mary Betty Smith,Mary 
Betty Smith,Intercom,Intercom,,,!
 ,,,None
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Message: 17
Date: Wed, 15 May 2013 14:45:12 +0100
From: Nick <[email protected]>
To: Andy <[email protected]>
Cc: [email protected]
Subject: Re: [cisco-voip] Extension Mobility Issue
Message-ID:
        <CAGJstU+SyXSNV7L9x=EWMMqcfa-aiJ425SnT=y+i1hqk6fx...@mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"

Hi All

dbreplication was being reported as good (all 2's) via RTMT and the Unfied
CM Database Status report was shoing as no errors and all fine. In the end
I did a utils dbreplication reset for the affected subscriber and
eventually it came back and was working fine. Its not good that the
replication was reporting as fine when there clearly was an issue with it
though,  this is 9.1.1a

Cheers

Nick

On 15 May 2013 13:13, Andy <[email protected]> wrote:

>  How is the replication on the cluster?
>
> Regards
>
> Andy
>
> On 15/05/2013 11:18, Nick wrote:
>
>  Hi All
>
>
>
> I have an issue with only one subscriber, which when phones are registered
> to it and  you log in with extension mobility it does not display the
> device profile info, the phone in CUCM admin shows the UDP as logged in and
> when you go to extension mobility service, it shows the user as logged in.
> I have tried reregistering the phone to the other 3 subscribers and the
> phone registers fine and displays the profile. I have rebooted the affected
> subscriber and during the reboot the phone registers with the backup
> subscriber and display UDP fine, but when the phone registers with its
> primary sub it the UDP will not display.
>
>
>  Anyone come across this before?
>
> Nick
>
>
> _______________________________________________
> cisco-voip mailing 
> [email protected]https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
> _______________________________________________
> cisco-voip mailing list
> [email protected]
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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Message: 18
Date: Wed, 15 May 2013 09:50:50 -0400
From: Ryan Ratliff <[email protected]>
To: Robert Kulagowski <[email protected]>
Cc: cisco-voip <[email protected]>
Subject: Re: [cisco-voip] SIP trunk failover question
Message-ID: <[email protected]>
Content-Type: text/plain; charset="us-ascii"

Actually when I started the email I was thinking that there were big 
differences in how RLs hunt between members of a RG and between RGs, but that 
document actually helped correct my misunderstanding.   The biggest reason I 
can think now to use separate RGs if if you need to do digit manipulation when 
sending the call to one gw vs another and wish to do it at the RL level.


-Ryan

On May 14, 2013, at 3:47 PM, Robert Kulagowski <[email protected]> wrote:

On Tue, May 14, 2013 at 1:32 PM, Ryan Ratliff <[email protected]> wrote:
> You may also want to look at the behavior available to you by having
> multiple route groups in a route list vs multiple devices in the RG.
> http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/6_0_1/ccmsys/a03rp.html#wp1045311
> does a decent job of covering this.

Can you dive a little deeper into this?

For my SIP testing, I've got a single route group, and in that route
group I have the SIP trunk first, then H.323 gateways after it. During
my testing the 2851 stopped responding; even with the ethernet port
shut down on the switch side (thereby removing any traffic if it was
due to a DOS), I still couldn't get a response on the console port,
and had to have remote hands reboot it. It was still "up" enough to
respond to pings (even though they were taking 2000ms instead of 2ms),
so CM was still sending it calls, but the caller was just hearing dead
air.

Since then we've been watching it like a hawk.

If the SIP trunk is moved to its own RG, and the RL is updated so that
it now has two RGs in it, would that have mitigated what happened?

Thanks.
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Message: 19
Date: Wed, 15 May 2013 09:57:44 -0400
From: "Jason Aarons (AM)" <[email protected]>
To: "cisco-voip ([email protected])"
        <[email protected]>
Subject: [cisco-voip] Shared Lines and Intercom Line?
Message-ID:
        
<4e38db0a1959b04c8c83edcf069b53ed0d331c7...@usispclexdb01.na.didata.local>
        
Content-Type: text/plain; charset="windows-1252"

Any brilliant idea on how to handle Intercom feature on CallManager 9.1 when 
you have shared lines?

I was going to make Line 1 4 digit number.
I was going to make Line 4 - Intercom DN with same 4 digit number.

If I have a Shared Line between two phones Intercom won't work (right?).  Wish 
CallManager would just multicast to the two phones (isnt' that how CME works?)
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Message: 20
Date: Wed, 15 May 2013 09:59:38 -0400
From: "Jason Aarons (AM)" <[email protected]>
To: Gregory Wenzel <[email protected]>, "[email protected]"
        <[email protected]>
Subject: Re: [cisco-voip] Directory Lookups not working for MAC OS and
        Jabber  for Mac 8.6
Message-ID:
        
<4e38db0a1959b04c8c83edcf069b53ed0d331c7...@usispclexdb01.na.didata.local>
        
Content-Type: text/plain; charset="windows-1252"

I think your solution will involve moving everyone to a xml file you have to 
write (both macs and pcs)  which I don't like. It creates a lot of work.

From: Gregory Wenzel [mailto:[email protected]]
Sent: Wednesday, May 15, 2013 9:09 AM
To: Jason Aarons (AM); [email protected]
Subject: RE: [cisco-voip] Directory Lookups not working for MAC OS and Jabber 
for Mac 8.6

Good point. I will look in that direction. Thanks Jason :)

From: Jason Aarons (AM) [mailto:[email protected]]
Sent: Wednesday, May 15, 2013 9:04 AM
To: Gregory Wenzel; 
[email protected]<mailto:[email protected]>
Subject: RE: [cisco-voip] Directory Lookups not working for MAC OS and Jabber 
for Mac 8.6

Does it work in the office?   If not part of a domain how is the directory 
lookup occurring?

The Jabber Administration guide has some good details on this.

From: cisco-voip [mailto:[email protected]] On Behalf Of 
Gregory Wenzel
Sent: Wednesday, May 15, 2013 8:49 AM
To: [email protected]<mailto:[email protected]>
Subject: [cisco-voip] Directory Lookups not working for MAC OS and Jabber for 
Mac 8.6


Hi all,

Has anyone run into an issue with Jabber directory lookup with a MAC that is 
not on a customer domain? I have a customer that I setup with SRST across a 
site-to-site VPN. The customer is claiming that the users on can't do lookups, 
yet all the other features of Jabber work. MAC OS 10.7, Jabber for Mac 8.6

TIA


Greg Wenzel




This message w/attachments (message) is solely for the use of the intended 
recipient(s) and may contain information that is privileged, confidential or 
proprietary. If you are not an intended recipient, please notify the sender, 
and then please delete and destroy all copies and attachments, and be advised 
that any review or dissemination of, or the taking of any action in reliance 
on, the information contained in or attached to this message is prohibited. 
Unless specifically indicated, this message is not an offer to sell or a 
solicitation of any products.   

itevomcid
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Message: 21
Date: Wed, 15 May 2013 09:07:03 -0500
From: Bill Talley <[email protected]>
To: "Jason Aarons (AM)" <[email protected]>
Cc: "cisco-voip \([email protected]\)"
        <[email protected]>
Subject: Re: [cisco-voip] Shared Lines and Intercom Line?
Message-ID: <8088554404929858283@unknownmsgid>
Content-Type: text/plain; charset="windows-1252"

Sorry, clarification, not EXACTLY what you want to do...  The intercom
extension on the other phones with the shared lines will have to be
different than the primary extension.

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

On May 15, 2013, at 8:59 AM, "Jason Aarons (AM)" <
[email protected]> wrote:

Any brilliant idea on how to handle Intercom feature on CallManager 9.1
when you have shared lines?



I was going to make Line 1 4 digit number.

I was going to make Line 4 ? Intercom DN with same 4 digit number.



If I have a Shared Line between two phones Intercom won?t work (right?).
Wish CallManager would just multicast to the two phones (isnt? that how CME
works?)

_______________________________________________
cisco-voip mailing list
[email protected]
https://puck.nether.net/mailman/listinfo/cisco-voip
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Message: 22
Date: Wed, 15 May 2013 09:04:29 -0500
From: Bill Talley <[email protected]>
To: "Jason Aarons (AM)" <[email protected]>
Cc: "cisco-voip \([email protected]\)"
        <[email protected]>
Subject: Re: [cisco-voip] Shared Lines and Intercom Line?
Message-ID: <-5226693633808422936@unknownmsgid>
Content-Type: text/plain; charset="windows-1252"

What you're wanting to do will work as long as the 4-digit intercom
extension is only assigned to a single phone.

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

On May 15, 2013, at 8:59 AM, "Jason Aarons (AM)" <
[email protected]> wrote:

Any brilliant idea on how to handle Intercom feature on CallManager 9.1
when you have shared lines?



I was going to make Line 1 4 digit number.

I was going to make Line 4 ? Intercom DN with same 4 digit number.



If I have a Shared Line between two phones Intercom won?t work (right?).
Wish CallManager would just multicast to the two phones (isnt? that how CME
works?)

_______________________________________________
cisco-voip mailing list
[email protected]
https://puck.nether.net/mailman/listinfo/cisco-voip
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Message: 23
Date: Wed, 15 May 2013 10:36:30 -0400
From: "Jason Aarons (AM)" <[email protected]>
To: Bill Talley <[email protected]>
Cc: "cisco-voip \([email protected]\)"
        <[email protected]>
Subject: Re: [cisco-voip] Shared Lines and Intercom Line?
Message-ID:
        
<4e38db0a1959b04c8c83edcf069b53ed0d331c7...@usispclexdb01.na.didata.local>
        
Content-Type: text/plain; charset="windows-1252"

Yeah I got 5,000 phone in a excel spreadsheet. I'll have to see how I can 
programmatically search for duplicates (this might be a case for using MS 
Access to de-duplicate)

1 to many would be a nice feature (eg Intercom-DN being shared and use 
multicast to send).

From: Bill Talley [mailto:[email protected]]
Sent: Wednesday, May 15, 2013 10:07 AM
To: Jason Aarons (AM)
Cc: cisco-voip ([email protected])
Subject: Re: [cisco-voip] Shared Lines and Intercom Line?


Sorry, clarification, not EXACTLY what you want to do...  The intercom 
extension on the other phones with the shared lines will have to be different 
than the primary extension.

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

On May 15, 2013, at 8:59 AM, "Jason Aarons (AM)" 
<[email protected]<mailto:[email protected]>> wrote:
Any brilliant idea on how to handle Intercom feature on CallManager 9.1 when 
you have shared lines?

I was going to make Line 1 4 digit number.
I was going to make Line 4 - Intercom DN with same 4 digit number.

If I have a Shared Line between two phones Intercom won't work (right?).  Wish 
CallManager would just multicast to the two phones (isnt' that how CME works?)
_______________________________________________
cisco-voip mailing list
[email protected]<mailto:[email protected]>
https://puck.nether.net/mailman/listinfo/cisco-voip


itevomcid
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