[cisco-voip] RE:CallManager 4.1 on Un-approved Hardware
Brad Clarke
bclarke at hoodnet.net
Tue May 10 12:21:05 EDT 2005
And , FYI , the "Un-Approved" hardware we are testing is a HP DL320 G3, the
G2 are approved but end-of-life by HP. Hopefully Cisco will release an OS
Image update that supports the G3 version of this server because the old
G2's were the same thing as a MCS-7825.
-----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: Tuesday, May 10, 2005 12:00 PM
To: cisco-voip at puck.nether.net
Subject: cisco-voip Digest, Vol 27, Issue 32
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Today's Topics:
1. RE:CallManager 4.1 on Un-approved Hardware (Brad Clarke)
2. RE: CME, VPN, PIX IOS (Wes Sisk)
----------------------------------------------------------------------
Message: 1
Date: Tue, 10 May 2005 11:46:01 -0400
From: "Brad Clarke" <bclarke at hoodnet.net>
Subject: [cisco-voip] RE:CallManager 4.1 on Un-approved Hardware
To: <cisco-voip at puck.nether.net>
Message-ID: <200505101546.j4AFk1Tr021823 at puck.nether.net>
Content-Type: text/plain; charset="us-ascii"
Thanks for the tip, I did get it working on the test box by loading , 3.3
first, then upgrade to 4.0, then upgrade to 4.1, but that was a full day
wasted :)
I'm going to try the shutdown.exe on the next one!
-----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: Tuesday, May 10, 2005 11:23 AM
To: cisco-voip at puck.nether.net
Subject: cisco-voip Digest, Vol 27, Issue 31
Send cisco-voip mailing list submissions to
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When replying, please edit your Subject line so it is more specific than
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Today's Topics:
1. RE:CallManager 4.1 on Un-approved Hardware (Bradley Clarke)
(Nick Kassel)
2. Re: AS5350 SER and CLIP (Maciej Bylica)
3. Re: CME, VPN, PIX IOS (Candace Holman)
4. RE: CME, VPN, PIX IOS (Voll, Scott)
5. Re: AS5350 SER and CLIP (Yura Monchak)
----------------------------------------------------------------------
Message: 1
Date: Tue, 10 May 2005 15:19:52 +0100
From: "Nick Kassel" <Nick.Kassel at Charles-Stanley.co.uk>
Subject: [cisco-voip] RE:CallManager 4.1 on Un-approved Hardware
(Bradley Clarke)
To: <cisco-voip at puck.nether.net>
Message-ID:
<C0426FFA2483574C8256AF89257E4ABE0866D690 at CSEXCH01.charles-stanley.co.uk>
Content-Type: text/plain; charset="us-ascii"
I have done this recently, you need to copy the shutdown.exe from the utils
folder of a working server then run the call manager 4.1 install again as
there are files missing from the utils folder after you install the OS
upgrade.
-----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: Tuesday, May 10, 2005 2:56 PM
To: cisco-voip at puck.nether.net
Subject: cisco-voip Digest, Vol 27, Issue 29
Send cisco-voip mailing list submissions to
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When replying, please edit your Subject line so it is more specific than
"Re: Contents of cisco-voip digest..."
Today's Topics:
1. CallManager 4.1 on Un-approved Hardware (Bradley Clarke)
2. CME, VPN, PIX IOS (Jim McBurnett)
3. Re: NM-HDV2 and ISDN data backup (Arek Kuczorski)
4. How do i decode facility from isdn q931? (Ivan Ostres)
5. RE: SPAM-LOW: [cisco-voip] How do i decode facility from
isdn q931? (Ozcan YILDIZ)
6. web attendant (Erik Erasmus)
7. RE: web attendant (Wes Sisk)
----------------------------------------------------------------------
Message: 1
Date: Mon, 9 May 2005 17:14:03 -0400
From: "Bradley Clarke" <BClarke at cprgr.com>
Subject: [cisco-voip] CallManager 4.1 on Un-approved Hardware
To: <cisco-voip at puck.nether.net>
Message-ID: <200505092114.j49LE4Ql069223 at puck.nether.net>
Content-Type: text/plain; charset="us-ascii"
Has anyone built a 4.1 server for the lab with 4.1.
I have done 3.3 and 4.0 without any trouble , but for some reason the
CallManager Install CDs (4.1.2) bomb out during the Setup with a File not
Found Error.
I also tried to run an upgrade from 3.3 to 4.1.2 using the CD with no luck.
Is there a file that needs to be copied to the server, or a special reg.
setting ?
I am going to try to bump the 3.3.x server to 4.x, then 4.1.x next.
Brad Clarke
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Message: 2
Date: Mon, 9 May 2005 20:57:06 -0400
From: "Jim McBurnett" <jim at tgasolutions.com>
Subject: [cisco-voip] CME, VPN, PIX IOS
To: <cisco-voip at puck.nether.net>
Message-ID:
<5432D045DAFD8040BCE549749263BD002B8DA4 at testsystem2.tga.local>
Content-Type: text/plain; charset="us-ascii"
Hello!
I have been testing a little something and I was wondering if anyone else
has seen this..
CME ver 3.2
PIX 6.3.3(132) on a 501
IP phone
|
PIX 501
|
Internet
|
2811 with CME/VPN
|
POTS lines
I have one way audio on the IP phone at the PIX location.
It does not matter how I try to call- To POTS or to other IP phones at the
2811 site, the other end can hear me, but not the other may around.
I have good ping times (sub 50ms)
Packet size has been adjusted up and down from 1400 to 1492
The VPN works fine for other DATA.. The Circuit was monitored on both ends,
neither was even approaching 30% capacity
I have check the VPN access-list and it all seems fine Traffic is flowing
for all the correct IP addresses.
Thanks,
Jim
------------------------------
Message: 3
Date: Tue, 10 May 2005 09:55:01 +0200
From: Arek Kuczorski <akuczorski at wp.pl>
Subject: Re: [cisco-voip] NM-HDV2 and ISDN data backup
To: mihi at ludens.elte.hu
Cc: 'cisco-voip' <cisco-voip at puck.nether.net>
Message-ID: <428068D5.8060903 at wp.pl>
Content-Type: text/plain; charset=us-ascii; format=flowed
Hi Marton,
Yeah but I'm little bit confusion because as stated here:
http://www.cisco.com/en/US/partner/tech/tk652/tk653/technologies_tech_no
te09186a0080387544.shtml
the NM-HDV2 has "enhanced time division multiplexing (TDM) capabilities"
which allow to switch
audio and data bit stream between ports on the same module. The ISDN
signalization is supported here. So I wonder since the NM-HDV2 can switch
ISDN data bitstrem maybe
it can terminate ISDN data too.
-Arek
MIHI wrote:
> Sure, NM-HDV and NM-HDV2 support data, but I think they support only
channel
> groups, see this doc:
>
http://cco.cisco.com/univercd/cc/td/doc/product/software/ios123/123newft
/123
> t/123t_7/hdd_vfnm.htm#wp1046592
> "Up to 32 HDLC channel groups with a total bandwidth of 2.048 Mbps"
> So they say "HDLC channel groups" not isdn PRI timeslots with data.
>
> Marton
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Arek
Kuczorski
> Sent: Monday, May 09, 2005 8:44 PM
> To: Walenta, Phil
> Cc: cisco-voip
> Subject: **SPAM** Re: [cisco-voip] NM-HDV2 and ISDN data backup
>
> As far as I know it can be used for data:
>
http://www.cisco.com/en/US/partner/tech/tk652/tk653/technologies_tech_no
te09
> 186a008039c333.shtml
>
http://www.cisco.com/en/US/partner/tech/tk652/tk653/technologies_tech_no
te09
> 186a0080387544.shtml
>
> Walenta, Phil wrote:
>
>>No. Because it's a voice-only card. Either of the options I gave you
>
> below should work, but NM-HDV's cannot be used for data.
>
>>-----Original Message-----
>>From: cisco-voip-bounces at puck.nether.net
>>[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Arek
>>Kuczorski
>>Sent: Monday, May 09, 2005 1:15 PM
>>Cc: cisco-voip
>>Subject: Re: [cisco-voip] NM-HDV2 and ISDN data backup
>>
>>Hello,
>>
>>thank You for Your answer, but my question was about using
NM-HDV2-2T1/E1
>
> module for data ISDN (backup).
>
>>It's unnecessary to use MFT VICs since this module has two T1/E1 ports
>
> onboard. I'd like to use one of them to voice ISDN and second to
terminate
> ISDN data backup. Is it possible ?
>
>>Any advice will be appreciated
>>Arek
>>
>>
>>Walenta, Phil wrote:
>>
>>
>>
>>>The documentation is correct, but it really should state data isn't
>
> supported. Any MFT-T1 or MFT-E1 card that based in an NM-HDV slot
cannot be
> used for data because the carrier card is voice only.
>
>>>However, if you used a 28XX router, or 38XX router, you could put the
>
> MFT-2T1/E1 card in a hwic slot, have internal DSP's, and I believe you
could
> then split the duty of the card. The only potential issue would be
the line
> clocking.
>
>>>Alternatively, I believe you could use a 26XX/36XX/37XX router with
an AIM
>
> card for voice.
>
>>>-----Original Message-----
>>>From: cisco-voip-bounces at puck.nether.net
>>>[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Arkadiusz
>>>Kuczorski
>>>Sent: Monday, May 09, 2005 8:26 AM
>>>To: cisco-voip
>>>Subject: [cisco-voip] NM-HDV2 and ISDN data backup
>>>
>>>Hello All,
>>>
>>>I need some advice regards NM-HDV2-2T1/E1. Does this module can be
used to
>
> perform ISDN dial backup ?
>
>>>Q&A
>
>
(http://www.cisco.com/warp/public/cc/pd/nemnsw/callmn/prodlit/sopra_qa.p
df)
> is stated that it doesn't "...support integrated voice and serial WAN
data
> access over PRI" but I wonder about "INTEGRATED voice and serial WAN".
Is
> that mean voice and data couldn't be used on single interface, or at
all ?
> Can I configure E1 PRI (voice) on one port and E1 PRI (Data) on the
second
> and perform ISDN dial-in backup from other routers ?
>
>>>TIA
>>>Arek
>
>
>
>
------------------------------
Message: 4
Date: Tue, 10 May 2005 10:29:05 +0200
From: "Ivan Ostres" <ivan.ostres at verso.hr>
Subject: [cisco-voip] How do i decode facility from isdn q931?
To: <cisco-voip at puck.nether.net>
Message-ID: <8899FEC181C5CC488CAE4D5A0670E7B00461C7 at pismo.verso.hr>
Content-Type: text/plain; charset="iso-8859-1"
Hi group,
how can I decode Facility info from "debug isdn q931" received form qsig
trunk? The message is like this:
*Mar 4 21:11:42.492: ISDN Se0/0:15 Q931: RX <- ALERTING pd = 8 callref =
0x8004
Facility i =
0x91AA068001008201008B0100A11402010C06042B0C09018009564552534F20313032
Shift to Codeset 5
Codeset 5 IE 0x32 i = 0x81
I would like to know does it carry dial reason (it should be Fwd
(NoReply)) and original number? Any tools or documentation available on this
subject?
Thanks,
Ivan
------------------------------
Message: 5
Date: Tue, 10 May 2005 12:22:50 +0300
From: "Ozcan YILDIZ" <ozcanyildiz at academytech.com>
Subject: RE: SPAM-LOW: [cisco-voip] How do i decode facility from
isdn q931?
To: "'Ivan Ostres'" <ivan.ostres at verso.hr>,
<cisco-voip at puck.nether.net>
Message-ID: <200505100924.j4A9O0Ne052130 at puck.nether.net>
Content-Type: text/plain; charset="us-ascii"
Dear Ivan,
Yes it carries dial reason.. you can view the documentation below. But i
amnot sure if it works for your case.
http://www.cisco.com/univercd/cc/td/doc/product/software/ios120/120newft
/120
limit/120xk/1207xk/e1_vo_xk.htm
If you want send us your case in a detailed form maybe list owners and
members canhelp you .
Best Regards
Ozcan YILDIZ
General Manager
AcademyTech Training and Consultancy Services
Tel: + 90 (216) 349 23 11 (pbx)
Fax: + 90 (216) 349 24 39
E-mail : ozcanyildiz at academytech.com
URL: http://www.academytech.com
-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ivan Ostres
Sent: Tuesday, May 10, 2005 11:29 AM
To: cisco-voip at puck.nether.net
Subject: SPAM-LOW: [cisco-voip] How do i decode facility from isdn q931?
Hi group,
how can I decode Facility info from "debug isdn q931" received form qsig
trunk? The message is like this:
*Mar 4 21:11:42.492: ISDN Se0/0:15 Q931: RX <- ALERTING pd = 8 callref =
0x8004
Facility i =
0x91AA068001008201008B0100A11402010C06042B0C09018009564552534F20313032
Shift to Codeset 5
Codeset 5 IE 0x32 i = 0x81
I would like to know does it carry dial reason (it should be Fwd
(NoReply))
and original number? Any tools or documentation available on this subject?
Thanks,
Ivan
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------------------------------
Message: 6
Date: Tue, 10 May 2005 15:27:01 +0200
From: "Erik Erasmus" <ErikE at kss.co.za>
Subject: [cisco-voip] web attendant
To: <cisco-voip at puck.nether.net>
Message-ID:
<3E110434E2419D47883D119DD226ACA170D6D3 at kss-jhb-exch-01.kss.co.za>
Content-Type: text/plain; charset="us-ascii"
Question regarding Web attendant. Is webattendant supported in call manager
version 4 xxx. I searched Cisco's web site and found only references to
documents explaining web attendant on call manager 3.0,
3.1 and even 2.4.
I also looked at our own call manager 4 admin interface and could not find
the area to configure web attendant.
Erik Erasmus
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Message: 7
Date: Tue, 10 May 2005 09:56:01 -0400
From: "Wes Sisk" <wsisk at cisco.com>
Subject: RE: [cisco-voip] web attendant
To: "Erik Erasmus" <ErikE at kss.co.za>, <cisco-voip at puck.nether.net>
Message-ID: <ELEMLGDGDBLINAJEIOCLOEDFPIAA.wsisk at cisco.com>
Content-Type: text/plain; charset="us-ascii"
webattendant is replaced by attendant console in all CM 4.x.
/Wes
-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net]On Behalf Of Erik Erasmus
Sent: Tuesday, May 10, 2005 9:27 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] web attendant
Question regarding Web attendant. Is webattendant supported in call manager
version 4 xxx. I searched Cisco's web site and found only references to
documents explaining web attendant on call manager 3.0, 3.1 and even 2.4.
I also looked at our own call manager 4 admin interface and could not find
the area to configure web attendant.
Erik Erasmus
------------------------------------------------------------------------
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Message: 2
Date: Tue, 10 May 2005 16:57:11 +0200
From: Maciej Bylica <voip at bylica.it.pl>
Subject: Re: [cisco-voip] AS5350 SER and CLIP
To: Gerd Feiner <g.feiner at cablesurf.de>
Cc: "'cisco-voip at puck.nether.net' <cisco-voip at puck.nether.net>"
<cisco-voip at puck.nether.net>
Message-ID: <4280CBC7.6060105 at bylica.it.pl>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Gerd Feiner wrote:
> Hi,
>
> I can't seem to get CLIP working with my SIP phones? The TDM-switch
> (Softswitch) sees the CLI of the calling (SIP-)party, but it get's not
> displayed on the callees PSTN-side. What's causing this and how to
> avoid it? We want CLIP to be enabled per default.
>
> Brgds,
> Gerd
>
Hi
I had the same problem on Cisco AS5300 with IOS ver 12.2(15) See here for
details http://www.erlang.com/forum/voip/thread.htx?thread=2915#10699.
I was struggling with this problem a few days and finally changed the IOS.
I hope this can help, but still i am not sure if this is a kind of bug on
ver 12.2(15).
Cheers,
Maciej.
------------------------------
Message: 3
Date: Tue, 10 May 2005 11:02:41 -0400
From: Candace Holman <candace_holman at harvard.edu>
Subject: Re: [cisco-voip] CME, VPN, PIX IOS
To: "Jim McBurnett" <jim at tgasolutions.com>,
<cisco-voip at puck.nether.net>
Message-ID: <6.1.0.6.2.20050510110032.01efac50 at camail.harvard.edu>
Content-Type: text/plain; charset="us-ascii"; format=flowed
It might not hurt to check the PIX configuration. I believe there are a few
voip fixups (skinny, sip, etc) that you can apply to make sure that audio
flows in both directions dynamically. Also check to make sure tftp, the
other udp firewall nasty with a fixup, is working properly.
Candace
At 08:57 PM 5/9/2005, Jim McBurnett wrote:
>Hello!
>I have been testing a little something and I was wondering if anyone
>else has seen this..
>
>CME ver 3.2
>PIX 6.3.3(132) on a 501
>
>IP phone
> |
>PIX 501
> |
>Internet
> |
>2811 with CME/VPN
> |
>POTS lines
>
>I have one way audio on the IP phone at the PIX location.
>It does not matter how I try to call- To POTS or to other IP phones at
>the 2811 site, the other end can hear me, but not the other may around.
>
>I have good ping times (sub 50ms)
>Packet size has been adjusted up and down from 1400 to 1492
>
>The VPN works fine for other DATA.. The Circuit was monitored on both
>ends, neither was even approaching 30% capacity
>
>I have check the VPN access-list and it all seems fine Traffic is
>flowing for all the correct IP addresses.
>
>Thanks,
>Jim
>
>
>
>_______________________________________________
>cisco-voip mailing list
>cisco-voip at puck.nether.net
>https://puck.nether.net/mailman/listinfo/cisco-voip
------------------------------
Message: 4
Date: Tue, 10 May 2005 08:08:49 -0700
From: "Voll, Scott" <Scott.Voll at wesd.org>
Subject: RE: [cisco-voip] CME, VPN, PIX IOS
To: "Jim McBurnett" <jim at tgasolutions.com>,
<cisco-voip at puck.nether.net>
Message-ID: <D713462ED535184D830F6F6301753E7E01511DE4 at VISHNU.wesd.org>
Content-Type: text/plain; charset="us-ascii"
Jim--
This sounds like either an ACL problem, routing problem, or a Fixup problem.
Check all your fixup protocols.
Then check all your routing tables (make sure all subnets that need to go
across the VPN are there).
And make sure that the VPN ACL includes everything. Sounds like the setup
is going to the CME but when the RTP stream starts you are not getting the
phone to phone or phone to VGW RTP back (check IP addresses for being
blocked).
Hope it helps
Scott
-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jim McBurnett
Sent: Monday, May 09, 2005 5:57 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] CME, VPN, PIX IOS
Hello!
I have been testing a little something and I was wondering if anyone else
has seen this..
CME ver 3.2
PIX 6.3.3(132) on a 501
IP phone
|
PIX 501
|
Internet
|
2811 with CME/VPN
|
POTS lines
I have one way audio on the IP phone at the PIX location.
It does not matter how I try to call- To POTS or to other IP phones at the
2811 site, the other end can hear me, but not the other may around.
I have good ping times (sub 50ms)
Packet size has been adjusted up and down from 1400 to 1492
The VPN works fine for other DATA.. The Circuit was monitored on both ends,
neither was even approaching 30% capacity
I have check the VPN access-list and it all seems fine Traffic is flowing
for all the correct IP addresses.
Thanks,
Jim
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------------------------------
Message: 5
Date: Tue, 10 May 2005 18:20:31 +0300
From: Yura Monchak <yurun at portaone.com>
Subject: Re: [cisco-voip] AS5350 SER and CLIP
To: cisco-voip at puck.nether.net
Message-ID: <4280D13F.3010806 at portaone.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Maciej Bylica wrote:
>Gerd Feiner wrote:
>
>
>>Hi,
>>
>>I can't seem to get CLIP working with my SIP phones? The TDM-switch
>>(Softswitch) sees the CLI of the calling (SIP-)party, but it get's not
>>displayed on the callees PSTN-side. What's causing this and how to
>>avoid it? We want CLIP to be enabled per default.
>>
>>Brgds,
>>Gerd
>>
>>
>>
>Hi
>
>I had the same problem on Cisco AS5300 with IOS ver 12.2(15) See here
>for details
>http://www.erlang.com/forum/voip/thread.htx?thread=2915#10699.
>
>I was struggling with this problem a few days and finally changed the IOS.
>
>I hope this can help, but still i am not sure if this is a kind of bug
>on ver 12.2(15).
>
>Cheers,
>Maciej.
>
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>cisco-voip mailing list
>cisco-voip at puck.nether.net
>https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
I can suggest you trying translation rules. It is possible that your ISDN
termination partner does not support CLI in the format you are trying to
send. National usually works. Here is an example:
translation-rule 100
rule 0 123 456 any national
...
apply the rule to incoming voip CLIs:
voip-incoming translation-rule calling 100
Good luck.
--
Yura Monchak
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End of cisco-voip Digest, Vol 27, Issue 31
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------------------------------
Message: 2
Date: Tue, 10 May 2005 11:56:41 -0400
From: "Wes Sisk" <wsisk at cisco.com>
Subject: RE: [cisco-voip] CME, VPN, PIX IOS
To: "Voll, Scott" <Scott.Voll at wesd.org>, "Jim McBurnett"
<jim at tgasolutions.com>, <cisco-voip at puck.nether.net>
Message-ID: <ELEMLGDGDBLINAJEIOCLCEDHPIAA.wsisk at cisco.com>
Content-Type: text/plain; charset="us-ascii"
if you are only using the router and pix for VPN, then do not use the fixup.
Fixup is primarily needed when you need to dynamically open/close tcp/udp
sessions or when you are NAT'ing the connection.
/Wes
-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net]On Behalf Of Voll, Scott
Sent: Tuesday, May 10, 2005 11:09 AM
To: Jim McBurnett; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] CME, VPN, PIX IOS
Jim--
This sounds like either an ACL problem, routing problem, or a Fixup problem.
Check all your fixup protocols.
Then check all your routing tables (make sure all subnets that need to go
across the VPN are there).
And make sure that the VPN ACL includes everything. Sounds like the setup
is going to the CME but when the RTP stream starts you are not getting the
phone to phone or phone to VGW RTP back (check IP addresses for being
blocked).
Hope it helps
Scott
-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jim McBurnett
Sent: Monday, May 09, 2005 5:57 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] CME, VPN, PIX IOS
Hello!
I have been testing a little something and I was wondering if anyone else
has seen this..
CME ver 3.2
PIX 6.3.3(132) on a 501
IP phone
|
PIX 501
|
Internet
|
2811 with CME/VPN
|
POTS lines
I have one way audio on the IP phone at the PIX location.
It does not matter how I try to call- To POTS or to other IP phones at the
2811 site, the other end can hear me, but not the other may around.
I have good ping times (sub 50ms)
Packet size has been adjusted up and down from 1400 to 1492
The VPN works fine for other DATA.. The Circuit was monitored on both ends,
neither was even approaching 30% capacity
I have check the VPN access-list and it all seems fine Traffic is flowing
for all the correct IP addresses.
Thanks,
Jim
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End of cisco-voip Digest, Vol 27, Issue 32
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