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Today's Topics:

   1. Re: 6961 Phone (David Zhars)
   2. Re: 6961 Phone (Nate VanMaren)
   3. Re: alert when PRI goes down (Erick B.)
   4. Re: problem with voicemail from PSTN (Wes Sisk)
   5. Call drop issue (Michele Russo (AM))


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

Message: 1
Date: Wed, 28 Nov 2012 12:29:24 -0500
From: David Zhars <[email protected]>
To: Steve Brickhouse <[email protected]>
Cc: "[email protected]" <[email protected]>
Subject: Re: [cisco-voip] 6961 Phone
Message-ID:
        <CADe=jTG6gQWpxFjEaQvrkYB=8luhc9e40ibgft2kspbg8gm...@mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"

Yup, have looked at that too, but it just looks clunky.
Was hoping kne of the newer phones might be a "sexier" solution.

On Wednesday, November 28, 2012, Steve Brickhouse <[email protected]>
wrote:
> What about just adding a sidecar?
>
> Sent from my iPhone
>
> On Nov 28, 2012, at 5:33 AM, David Zhars <[email protected]> wrote:
>
>> I have a situation where we place a (currently) 7961 phone in a common
room.  Each person is assigned one of those lines, and when they come in
they are expected to check the screen and see if they have voicemail.
>>
>> Of course, we are now getting a 7th person!!
>>
>> I was wondering if the 6961 could be a solution?  Does the button
assigned to a user change color designating voicemail?  Will this phone
work with UCM 8?  I saw the SPA501G (8 lines), but unsure if it works with
UCM or only Cisco's Small Business Pro Unified Communications 320??
>>
>> Thank you in advance!
>>
>> David
>> _______________________________________________
>> cisco-voip mailing list
>> [email protected]
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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Message: 2
Date: Wed, 28 Nov 2012 17:35:31 +0000
From: Nate VanMaren <[email protected]>
To: David Zhars <[email protected]>, Steve Brickhouse
        <[email protected]>
Cc: "[email protected]" <[email protected]>
Subject: Re: [cisco-voip] 6961 Phone
Message-ID:
        <[email protected]>
Content-Type: text/plain; charset="us-ascii"

9971+CKEM is pretty sexy.

From: [email protected] 
[mailto:[email protected]] On Behalf Of David Zhars
Sent: Wednesday, November 28, 2012 10:29 AM
To: Steve Brickhouse
Cc: [email protected]
Subject: Re: [cisco-voip] 6961 Phone

Yup, have looked at that too, but it just looks clunky.
Was hoping kne of the newer phones might be a "sexier" solution.

On Wednesday, November 28, 2012, Steve Brickhouse 
<[email protected]<mailto:[email protected]>> wrote:
> What about just adding a sidecar?
>
> Sent from my iPhone
>
> On Nov 28, 2012, at 5:33 AM, David Zhars 
> <[email protected]<mailto:[email protected]>> wrote:
>
>> I have a situation where we place a (currently) 7961 phone in a common room. 
>>  Each person is assigned one of those lines, and when they come in they are 
>> expected to check the screen and see if they have voicemail.
>>
>> Of course, we are now getting a 7th person!!
>>
>> I was wondering if the 6961 could be a solution?  Does the button assigned 
>> to a user change color designating voicemail?  Will this phone work with UCM 
>> 8?  I saw the SPA501G (8 lines), but unsure if it works with UCM or only 
>> Cisco's Small Business Pro Unified Communications 320??
>>
>> Thank you in advance!
>>
>> David
>> _______________________________________________
>> cisco-voip mailing list
>> [email protected]<mailto:[email protected]>
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>


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Message: 3
Date: Wed, 28 Nov 2012 12:26:37 -0600
From: "Erick B." <[email protected]>
To: "Abdul Salam ." <[email protected]>
Cc: Cisco VOIP <[email protected]>
Subject: Re: [cisco-voip] alert when PRI goes down
Message-ID:
        <cahsnbqzwqwwhk_-mvspaje4okfhqjxc6a9akbkj4dp-vvbg...@mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"

The RTMT MGCP alert will only be useful if your PRI is setup for MGCP.  If
it is H323 then you need to use syslogs, snmp traps. You could also put
together a EEM script that runs on the router to email you.


On Wed, Nov 28, 2012 at 1:20 AM, Abdul Salam . <[email protected]> wrote:

> see attached counter
>
>
>
> *---AS*
>
>
>
>
>
> On Wed, Nov 28, 2012 at 12:42 PM, Abdul Salam . <[email protected]> wrote:
>
>> I think you can look for RTMT alert D-channel oos
>>
>>
>>
>> *---AS*
>>
>>
>>
>>
>>
>> On Wed, Nov 28, 2012 at 11:14 AM, Lelio Fulgenzi <[email protected]>wrote:
>>
>>> Both the router and CUCM will send syslog messages to this affect. Once
>>> you configure syslog target, you can setup your syslog host with the
>>> appropriate software to send email. We use SEC in conjunction with a Linux
>>> based syslog daemon. There are other options like Solarwinds and Logzilla.
>>>
>>> I also believe RTMT can be configured to send email based on thresholds.
>>>
>>> Sent from my iPhone...
>>>
>>> "There's no place like 127.0.0.1"
>>>
>>> On Nov 28, 2012, at 12:33 AM, Shaihan Jaffrey <[email protected]>
>>> wrote:
>>>
>>> > Hi Team,
>>> > Is there any way to generate an alert on a specific email as soon  as
>>> the PRI goes down in cisco voice gateway.
>>> >
>>> > Regards,
>>> >
>>> > _______________________________________________
>>> > cisco-voip mailing list
>>> > [email protected]
>>> > https://puck.nether.net/mailman/listinfo/cisco-voip
>>>
>>> _______________________________________________
>>> cisco-voip mailing list
>>> [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: 4
Date: Wed, 28 Nov 2012 13:53:08 -0500
From: Wes Sisk <[email protected]>
To: Erick Wellnitz <[email protected]>
Cc: cisco-voip <[email protected]>
Subject: Re: [cisco-voip] problem with voicemail from PSTN
Message-ID: <[email protected]>
Content-Type: text/plain; charset=us-ascii

Most of the time AF is associated with codec negotiation failure or MTP 
allocation failure.  Take a look at the SDI/SDL traces to identify what 
capabilities are reported by each side and what codecs are available based on 
regions configuration and locations bandwidth.

/wes

On Nov 28, 2012, at 11:39 AM, Erick Wellnitz wrote:

Another weird issue.

Voicemail works fine internally.  When I call from outside through our h323 
gateway I get Cause i = 0x80AF - Resources unavailable, unspecified 

According to RTMT my MTP gets allocated.  

h323 Gateway ------g711------CUCM 8.6 -----------> SIP Trunk to Connection

Any guidance would be greatly appreciated
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------------------------------

Message: 5
Date: Wed, 28 Nov 2012 14:03:26 -0500
From: "Michele Russo (AM)" <[email protected]>
To: "[email protected]" <[email protected]>
Subject: [cisco-voip] Call drop issue
Message-ID:
        
<cb65a4d1d2e17640ba229cb811020f370f5ab7f...@usispclexdb01.na.didata.local>
        
Content-Type: text/plain; charset="windows-1252"

All,

I am working with a customer who is seeing sporadic call drops on their 
systems.  They following is the setup:

CME - H323 Trunk to Avaya - call routed out a LD PRI registered to the Avaya 
system.

The CME has 12 7937's and 4 biamp devices registered to it, the Biamps are SIP 
based phones.  The off-net calls drop sporadically and the seemingly relevant 
messages we see in the Wire Shark and Syslog  traces show:


10.146.128.50 (Avaya)    10.146.149.65     TCP        [TCP ACKed lost segment] 
h323hostcall > hfcs-manager [ACK] Seq=1 Ack=2 Win=8736 Len=0
TCP0: bad seg from 10.146.128.50 -- bad sequence number: port 35839 seq 
2919846836 ack 1534185920 rcvnxt 2919846837 rcvwnd 3146 len 0
TCP0: ACK timeout timer expired

With a disconnect Cause Value=38 (meaning network out of order).

I am attaching the router config, the syslog file and the Wireshark capture.

Note:  the CME and the Avaya re on the same 4507E switch and a review of both 
interfaces show literally zero CRC, Frame, Overrun or other L2 issues.  So it 
seems unlikely that there are any cabling issues, and the switch itself seems 
quite healthy.

Any thoughts/suggestions would be very helpful!  So far TAC has suggested we 
add the 'no vad' statement to the dial-peer 11 and allow-connections h323 to 
h323.  I am not sure either one of those configuration changes will fix this 
issue.

Thanks!

Michele Russo
Consultant
Dimension Data NA
202-460-3965 (cell)
571-203-4007 (desk)
[email protected]
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