Re: [asterisk-users] Asterisk stopping for no reason

2010-04-29 Thread Motiejus Jakštys
Hi, please always add asterisk version to your query. I managed to run internet radio (that streams MP3) within asterisk. Minor change is nescesarry to make it work with random MP3s. My Dialplan: exten => _X.,n,Answer() exten => _X.,n,MP3Player(http://stream.m-1.fm/m1/mp3) $ cat /usr/bin/mpg123

[asterisk-users] Caller ID on Asterisk and Astribank

2010-04-29 Thread frangky robert
Hi all... I have a problem with caller id on my asterisk server. here is my configuration : centos-5, asterisk 1.6.2.1, dahdi-linux-complete-2.2.1, libpri-1.4.10.2 ibm X-3200 series, xorcom astribank (16fxo, 8fxs), 16 line telco (hunting) everything fine until I try to feed my app with caller id

[asterisk-users] Friday 12 Noon EDT: Media5fone Mobile SIP Client Symbian S60 & iPhone

2010-04-29 Thread Randy R
Hi, If I was going to post this as an iPhone-only SIP client, I'd expect loud booing and hissing, but Media5 mobile SIP client is available for the Symbian S60 platform, too, or will be shortly. Interested? To join us and hear about Media5 form Pascal Dore, see http://vuc.me Speaking of mobile,

[asterisk-users] Continuing after a TIMEOUT(absolute)

2010-04-29 Thread Brendan Sterne
Greetings, I'm trying to continue to do some processing after a TIMEOUT (absolute). In my dialplan below, when a call comes in to [default], I call macro-phonenum and pass it a timeout of 20 seconds. macro- phonenum sets TIMEOUT(absolute), then loops saying the phone number that was called

Re: [asterisk-users] Duplicated DTMF with bridged IAX channels maybe?

2010-04-29 Thread James Lamanna
On Wed, Apr 28, 2010 at 6:57 PM, James Lamanna wrote: > Hi, > I have a duplicated DTMF issue with, it appears, bridged IAX channels. > I have the following setup: > >   PRI                  IAX > <>* PSTN <--->* Dialplan > > I've configured a number on the dialplan server to make and o

Re: [asterisk-users] Follow-me to my answering machine :-(

2010-04-29 Thread Bryan Jacobs
On Thu, 29 Apr 2010 18:03:26 -0400 "Barry L. Kline" wrote: > Bryan Jacobs wrote: > > > I can't "just call the car" - the car is my cell phone DID with a > > bluetooth kit. > > I did this same thing you're attempting. I have a desk set at home, a > Polycom in my office and my cell phone all bei

Re: [asterisk-users] Issue with (pattern) matching extension

2010-04-29 Thread Andrew Latham
Are you guys talking about the Asterisk Cookbook Because that could be released in the next 20 years at this point... ~ Andrew "lathama" Latham lath...@gmail.com * Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software * Learn more about Linux http://en.wikipedia.org/wiki/Li

Re: [asterisk-users] Issue with (pattern) matching extension

2010-04-29 Thread Philip A. Prindeville
Certainly getting people to email in updated examples would speed the book along... On 04/29/2010 04:06 PM, Danny Nicholas wrote: > Not really complaining, but AKAIK, this document is current as of about > 1.4.10? > > -Original Message- > From: asterisk-users-boun...@lists.digium.com > [m

Re: [asterisk-users] Issue with (pattern) matching extension

2010-04-29 Thread Steve Howes
On 29 Apr 2010, at 22:56, Leif Madsen wrote: > Danny Nicholas wrote: >> Good snippet, Leif. It's easier to read 100 threads on this forum than the >> 100 pages of the infamous "Asterisk Book" PDF. > Infamous? Ouch :) He's insulting our holy book! Stone him! ;) S --

Re: [asterisk-users] Issue with (pattern) matching extension

2010-04-29 Thread Danny Nicholas
Not really complaining, but AKAIK, this document is current as of about 1.4.10? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Philip A. Prindeville Sent: Thursday, April 29, 2010 4:59 PM To: asterisk-users@lis

Re: [asterisk-users] Follow-me to my answering machine :-(

2010-04-29 Thread Barry L. Kline
Bryan Jacobs wrote: > I can't "just call the car" - the car is my cell phone DID with a > bluetooth kit. I did this same thing you're attempting. I have a desk set at home, a Polycom in my office and my cell phone all being called at the same time. I called Verizon and had them disable voice ma

Re: [asterisk-users] Issue with (pattern) matching extension

2010-04-29 Thread Philip A. Prindeville
On 04/29/2010 03:56 PM, Leif Madsen wrote: > Danny Nicholas wrote: > >> Good snippet, Leif. It's easier to read 100 threads on this forum than the >> 100 pages of the infamous "Asterisk Book" PDF. >> > Infamous? Ouch :) > > Leif. > Danny: Well, there is an effort to improve the docume

Re: [asterisk-users] Issue with (pattern) matching extension

2010-04-29 Thread Leif Madsen
Danny Nicholas wrote: > Good snippet, Leif. It's easier to read 100 threads on this forum than the > 100 pages of the infamous "Asterisk Book" PDF. Infamous? Ouch :) Leif. -- _ -- Bandwidth and Colocation Provided by http://ww

[asterisk-users] Asterisk stopping for no reason

2010-04-29 Thread Alexandre Vézina
Hi, Few days ago, my asterisk began to stop unexpectedly What I did: - Added a mp3 to the musiconhold directory - Adjusted the permissions (chown asterisk:asterisk + chmod 755) - Reconfigured the musiconhold.conf to the deprecated format (found the example on the internet) [classe

[asterisk-users] Calls Dropping

2010-04-29 Thread Dan Journo
Hi, I'm having a major problem with random calls dropping. After spending weeks trying to figure it out, i've finally spotted the issue but don't know how to resolve it. I run a sip server that's hosted in a data centre. It has a public IP address with no nat involved. My provider also has a p

Re: [asterisk-users] ATA shootout: PAP2T versus Grandstream Handytone 286

2010-04-29 Thread Geoff Lane
On Thursday, April 29, 2010, David Backeberg wrote: > What do people think about both products? > Bonus points for if people have bulk deployed these, either with TFTP > and configs pushed from a server, or some other good idea. I can't claim the bonus points. However, I did have a couple of Gra

Re: [asterisk-users] ATA shootout: PAP2T versus Grandstream Handytone 286

2010-04-29 Thread Jian Gao
Dan Journo wrote: > On Thu, 29 Apr 2010, David Backeberg wrote: > > >> I'm considering a situation where I buy about twenty ATA devices. >> >> I've played with the Linksys / Cisco PAP2T, and got that working fine >> with some inbound and outbound faxing. The web GUI was okay. I'm >> seeing pri

Re: [asterisk-users] ATA shootout: PAP2T versus Grandstream Handytone 286

2010-04-29 Thread Dean Hoover
On 4/29/2010 3:10 PM, Jeff LaCoursiere wrote: > > On Thu, 29 Apr 2010, David Backeberg wrote: > >> I'm considering a situation where I buy about twenty ATA devices. >> >> I've played with the Linksys / Cisco PAP2T, and got that working fine >> with some inbound and outbound faxing. The web GUI wa

Re: [asterisk-users] ATA shootout: PAP2T versus Grandstream Handytone 286

2010-04-29 Thread Dan Journo
On Thu, 29 Apr 2010, David Backeberg wrote: > I'm considering a situation where I buy about twenty ATA devices. > > I've played with the Linksys / Cisco PAP2T, and got that working fine > with some inbound and outbound faxing. The web GUI was okay. I'm > seeing prices around $45 to $50 for this th

Re: [asterisk-users] ATA shootout: PAP2T versus Grandstream Handytone 286

2010-04-29 Thread Jeff LaCoursiere
On Thu, 29 Apr 2010, David Backeberg wrote: > I'm considering a situation where I buy about twenty ATA devices. > > I've played with the Linksys / Cisco PAP2T, and got that working fine > with some inbound and outbound faxing. The web GUI was okay. I'm > seeing prices around $45 to $50 for this t

Re: [asterisk-users] Issue with (pattern) matching extension

2010-04-29 Thread Philip Prindeville
On 4/29/10 1:55 PM, Tilghman Lesher wrote: > On Thursday 29 April 2010 13:11:17 Philip A. Prindeville wrote: > >> Doesn't quite make it 'deterministic' if you have to test it to see what >> it's going to do. >> > The code is deterministic. The human who wrote the example is not. Are > y

Re: [asterisk-users] Issue with (pattern) matching extension

2010-04-29 Thread Danny Nicholas
Worse things have been proposed for humans; many readers would like to see this done to posters such as I. :) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman Lesher Sent: Thursday, April 29, 2010 2:55 P

Re: [asterisk-users] Issue with (pattern) matching extension

2010-04-29 Thread Tilghman Lesher
On Thursday 29 April 2010 13:11:17 Philip A. Prindeville wrote: > Doesn't quite make it 'deterministic' if you have to test it to see what > it's going to do. The code is deterministic. The human who wrote the example is not. Are you proposing a genetic modification to make humans deterministic?

Re: [asterisk-users] Code in extensions.conf to leave a voice mail in another PBX ?!

2010-04-29 Thread khalid touati
Hi Guys, Danny: as i said from pbx1 (1.4) to pbx2 (1.2) it's working fine. Peder: i just didn't want to put a lot of lines, (by the way it's dialing talking fine), but here you are: [macro-stdexten] exten => s,n,Dial(SIP/${ARG1}&IAX2/${ar...@${arg1},20,tTrWw);Ring phone for 20 seconds exten =

[asterisk-users] ATA shootout: PAP2T versus Grandstream Handytone 286

2010-04-29 Thread David Backeberg
I'm considering a situation where I buy about twenty ATA devices. I've played with the Linksys / Cisco PAP2T, and got that working fine with some inbound and outbound faxing. The web GUI was okay. I'm seeing prices around $45 to $50 for this thing. It comes with two FXS ports, but I only need one

Re: [asterisk-users] Asterisk Query

2010-04-29 Thread Juan David Diaz
2010/4/29 garge rama > > > Hi, > > > > I am new to asterisk and trying to make calls with TDM400P asterisk digium > card. > > > > I am using asterisk-1.6.2.4, dahdi-linux-complete-2.3.0+2.3.0 and > libpri-1.4.10.2 packages which are downloaded from asterisk website ( > www.asterisk.org) > > and a

Re: [asterisk-users] Code in extensions.conf to leave a voice mail in another PBX ?!

2010-04-29 Thread Peder
In PBX1, where are you actually dialing the phone? The first line of the macro just says "goto dialstatus" with no Dial statement. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of khalid touati Sent: Thursday, April 29, 2010 2:03 PM

Re: [asterisk-users] Code in extensions.conf to leave a voice mail inanother PBX ?!

2010-04-29 Thread Danny Nicholas
If you dial 8029 from PBX1, does VM work? In my experience, cross-version IAX is tricky. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of khalid touati Sent: Thursday, April 29, 2010 2:03 PM To: Asterisk Users Mailing List -

[asterisk-users] Code in extensions.conf to leave a voice mail in another PBX ?!

2010-04-29 Thread khalid touati
Hi Guys, i spent some time to figure this out (since i love how dialplan is written) but i decided to ask for your help guys. i have two asterisk servers one is 1.2 the other is 1.4, from 1.4 (pbx1) to 1.2 (pbx2) i can leave a voice mail without any pb, but from pbx2 to pbx1 it just hang up. in p

Re: [asterisk-users] Asterisk 1.4.30 is slow sending STDIN to AGI script

2010-04-29 Thread Steve Edwards
Un-top-posting... >>> On Wed, 28 Apr 2010, Ryan Bullock wrote: >>> Looking at the Asterisk::AGI docs, maybe try calling ReadParse() early in the script to read in anything from stdin? (From the docs) # pull AGI variables into %input %input = $AGI->ReadParse(); > >> O

Re: [asterisk-users] Issue with (pattern) matching extension

2010-04-29 Thread Philip A. Prindeville
On 04/29/2010 12:09 PM, Tilghman Lesher wrote: > On Thursday 29 April 2010 11:46:39 Philip Prindeville wrote: > >> On 4/29/10 4:22 AM, Jim Dickenson wrote: >> >>> I banged my head with a like problem a few days ago. >>> >>> > exten => _fn-.,1,NoOp(ISN: ${DIALSTATUS}) >

Re: [asterisk-users] Issue with (pattern) matching extension

2010-04-29 Thread Tilghman Lesher
On Thursday 29 April 2010 11:46:39 Philip Prindeville wrote: > On 4/29/10 4:22 AM, Jim Dickenson wrote: > > I banged my head with a like problem a few days ago. > > > >>> exten => _fn-.,1,NoOp(ISN: ${DIALSTATUS}) > > > > n does not mean the letter n in a pattern it has a special meaning! > > That'

Re: [asterisk-users] Asterisk 1.4.30 is slow sending STDIN to AGI script

2010-04-29 Thread Danny Nicholas
Speaking from a "Perl'er" perspective, there's no good reason that Asterisk::AGI shouldn't do the ReadParse automatically except that it requires the module author to do something that the user should be doing as a "best practice" and could lead to unexpected errors in a reuse environment. IMO the

Re: [asterisk-users] Asterisk 1.4.30 is slow sending STDIN to AGI script

2010-04-29 Thread Steve Edwards
>> On Wed, 28 Apr 2010, Ryan Bullock wrote: >> >>> Looking at the Asterisk::AGI docs, maybe try calling ReadParse() early >>> in the script to read in anything from stdin? >>> >>> (From the docs) >>> # pull AGI variables into %input >>> %input = $AGI->ReadParse(); > On Wed, Apr 28, 2010 at 09:34:

Re: [asterisk-users] Asterisk 1.4.30 is slow sending STDIN to AGI script

2010-04-29 Thread Tzafrir Cohen
On Wed, Apr 28, 2010 at 09:34:18AM -0700, Steve Edwards wrote: > On Wed, 28 Apr 2010, Ryan Bullock wrote: > > > Looking at the Asterisk::AGI docs, maybe try calling ReadParse() early > > in the script to read in anything from stdin? > > > > (From the docs) > > # pull AGI variables into %input > >

Re: [asterisk-users] Odd Issue With Polycom Phones]

2010-04-29 Thread Gord Urquhart
The phone is only making one call, notice the call-id did not change. The second INVITE is sent in responce to a 401 Authentication Required. The 401 will contain the necessary authentication information for the phone to use to build the Authorization header that it inserts in the second invite. TH

Re: [asterisk-users] Issue with (pattern) matching extension

2010-04-29 Thread Danny Nicholas
It's a pattern matching thing; the asterisk module "knows" how to process stdexten, but thinks that "n" or "N" is a digit substitution. When "n" or "N" is "escaped" ([n] or [N]) the program knows to treat it as a literal and not a pattern match. -Original Message- From: asterisk-users-bo

Re: [asterisk-users] Issue with (pattern) matching extension

2010-04-29 Thread Philip Prindeville
On 4/29/10 4:22 AM, Jim Dickenson wrote: > I banged my head with a like problem a few days ago. > > >>> exten => _fn-.,1,NoOp(ISN: ${DIALSTATUS}) >>> > n does not mean the letter n in a pattern it has a special meaning! > That's capital N, isn't it? Also, the prefix "_stdexten-."

Re: [asterisk-users] Strange Invite issue

2010-04-29 Thread Gareth Blades
Can you post a sip debug Tarek Sawah wrote: > Greetings List. > I'm facing a strange issue with one of my providers.. after sending an INVITE > request my server places the call on hold.. until the call is answered.. > this is happening only with this provide although i have 3 other providers i

[asterisk-users] Strange Invite issue

2010-04-29 Thread Tarek Sawah
Greetings List. I'm facing a strange issue with one of my providers.. after sending an INVITE request my server places the call on hold.. until the call is answered.. this is happening only with this provide although i have 3 other providers i route calls through.. can anyone explain what is g

[asterisk-users] Dropping incompatible voice frame

2010-04-29 Thread Vieri
Hi, What does this message imply? [Apr 29 14:46:30] NOTICE[32175] channel.c: Dropping incompatible voice frame on IAX2/trunk1-9085 of format alaw since our native format has changed to 0x4 (ulaw) If voice frames have been dropped then I suppose that the call quality may be affected? Vieri

Re: [asterisk-users] Dropping incompatible voice frame

2010-04-29 Thread Danny Nicholas
Possibly or possibly not. Most (IMO) calls are placed initially with the choice 2-3 or more codecs. Normally one codec is negotiated and life goes on, but IAX is a little different from a SIP/DAHDI call. The most certain remedy I can think of for this it to just "unallow" the alaw codec on IAX ca

Re: [asterisk-users] Issue with (pattern) matching extension

2010-04-29 Thread Danny Nicholas
Good snippet, Leif. It's easier to read 100 threads on this forum than the 100 pages of the infamous "Asterisk Book" PDF. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leif Madsen Sent: Thursday, April 29, 20

Re: [asterisk-users] Issue with (pattern) matching extension

2010-04-29 Thread Leif Madsen
Jim Dickenson wrote: > I banged my head with a like problem a few days ago. > >>> exten => _fn-.,1,NoOp(ISN: ${DIALSTATUS}) > > n does not mean the letter n in a pattern it has a special meaning! Right! Be very careful about what you're matching! When it comes to matching things like 'N', 'X',

Re: [asterisk-users] incoming call should ring on several dahdi channels

2010-04-29 Thread Gareth Blades
typo ... OFFICE1=DAHDI/13 OFFICE2=DAHDI/14 exten => 12345678,1,Dial(${OFFICE1}&${OFFICE2},,rtT) Gareth Blades wrote: > Try this. > > OFFICE1=DAHDI/13 > OFFICE2=DAHDI/14 > exten => 12345678,1,Dial(${OFFICE1}&{OFFICE2},,rtT) > > > Peter Gelencser wrote: >> Hi, >> >> >> I need a feature from aste

Re: [asterisk-users] incoming call should ring on several dahdi channels

2010-04-29 Thread Gareth Blades
Try this. OFFICE1=DAHDI/13 OFFICE2=DAHDI/14 exten => 12345678,1,Dial(${OFFICE1}&{OFFICE2},,rtT) Peter Gelencser wrote: > Hi, > > > I need a feature from asterisk with dahdi channels, if there is an > incoming call, it should ring on several dahdi channels. > > My channels look like: > > OFF

Re: [asterisk-users] B400P and A1200P changes card order

2010-04-29 Thread Peter Gelencser
2010.04.20. 16:50 keltezéssel, Shaun Ruffell írta: > On 04/19/2010 03:48 AM, Peter Gelencser wrote: >> I've run into a veird problem. I'm using a B400P BRI and an A1200P card >> with dahdi (2.2.1) driver. The dahdi_scan shows the each moduls and >> spans, everything seems fine. With dahdi_genconf

[asterisk-users] incoming call should ring on several dahdi channels

2010-04-29 Thread Peter Gelencser
Hi, I need a feature from asterisk with dahdi channels, if there is an incoming call, it should ring on several dahdi channels. My channels look like: OFFICE1=DAHDI/13,,rtT OFFICE2=DAHDI/14,,rtT If I add this line: exten => 12345678,1,Dial(${OFFICE1}&{OFFICE2}) only OFFICE1 rings. If I cha

Re: [asterisk-users] No change in payload. (SDP)

2010-04-29 Thread Kevin P. Fleming
Aditya Kumar wrote: > re-posting the question. > --- > use case: > when some one in my pbx calls 100.200, I have translations well defined, > Media also (media via asterisk) --Works. > when some one calls bob, or for any names I am adding Domain and call is > been sent to the other party

[asterisk-users] Polycom 330 not connecting

2010-04-29 Thread Tony LaMear
I was just wondering if anyone is having the same problem will Polycom 330 ip phone. The phone looses the network and when you reboot the phone it can no longer find the DHCP server. I put an address in manually, but the phone is still not able to connect to the network. I replaced the phone wit

Re: [asterisk-users] Starting call recording using a dynamic feature to call a macro

2010-04-29 Thread Gareth Blades
Ignore me I figured it out. The dangers of copy and paste. After looking through the code line by line I noticed the 'b' parameter to monitor(). Fine to use before the dial command but shouldnt be used when a call is in progress. Gareth Blades wrote: > I have got call recording working on our 1

Re: [asterisk-users] Issue with (pattern) matching extension

2010-04-29 Thread Jim Dickenson
I banged my head with a like problem a few days ago. >> exten => _fn-.,1,NoOp(ISN: ${DIALSTATUS}) n does not mean the letter n in a pattern it has a special meaning! -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Apr 29, 2010, at 1:33 AM, Ishfaq Malik wrote: > Phili

[asterisk-users] Starting call recording using a dynamic feature to call a macro

2010-04-29 Thread Gareth Blades
I have got call recording working on our 1.4.30 asterisk box together with a recording pause ability and being able to play different audio to each party at the start and end of the pause. This all works perfectly but one wish is to have the audio files have a beep or something in them so when

[asterisk-users] mysql realtime schema

2010-04-29 Thread Vasiliy G Tolstov
Hello. Where i can find complete realtime mysql schema for asterisk 1.6? Google get results to some tables. I want to do all iaxusers iaxpeers sipusers sippeers sipregs voicemail extensions meetme queues queue_members musiconhold queue_log in separate mysql tables. -- Vasiliy G Tolstov Selfip

Re: [asterisk-users] AGI <==> DeadAGI

2010-04-29 Thread Ishfaq Malik
Redouane Zerargui wrote: > Hello, i have this problem : > i phone person B . > _/*if i hang up*/_, i have this "h" extension : exten => h,1,AGI(ende.agi) > _/*if the person B hangs up*/_ , i have this "h" extension : exten => > h,1,DeadAGI(ende.agi) > > The problem is, i do not know where hangs

Re: [asterisk-users] Detect if a Number is up or not

2010-04-29 Thread ABBAS SHAKEEL
Thanks Loan Indreias ... Nice Idea Thanks Danny Nicholas. Cheers On Tue, Apr 27, 2010 at 7:17 PM, Danny Nicholas wrote: > This is probably a good idea, BUT it is likely that the dialed phone will > never ring (Perhaps that is the desired effect); In my experience it takes > Zap/DAHDI about 2-

[asterisk-users] AGI <==> DeadAGI

2010-04-29 Thread Redouane Zerargui
Hello, i have this problem : i phone person B . *if i hang up,* i have this "h" extension : exten => h,1,AGI(ende.agi) *if the person B hangs up* , i have this "h" extension : exten => h,1, DeadAGI(ende.agi) The problem is, i do not know where hangs up the first . How kann i combine AGI and DeadA

[asterisk-users] AGI <==> DeadAGI

2010-04-29 Thread Redouane Zerargui
Hello, i have this problem : i phone person B . *if i hang up*, i have this "h" extension : exten => h,1,AGI(ende.agi) *if the person B hangs up* , i have this "h" extension : exten => h,1, DeadAGI(ende.agi) The problem is, i do not know where hangs up the first . How kann i combine AGI and DeadA

[asterisk-users] Asterisk Query

2010-04-29 Thread garge rama
Hi, I am new to asterisk and trying to make calls with TDM400P asterisk digium card. I am using asterisk-1.6.2.4, dahdi-linux-complete-2.3.0+2.3.0 and libpri-1.4.10.2 packages which are downloaded from asterisk website ( www.asterisk.org) and able to compile successfully. TDM400P Digium card

Re: [asterisk-users] Issue with (pattern) matching extension

2010-04-29 Thread Ishfaq Malik
Philip A. Prindeville wrote: > Here's a segment of my dialplan, I'm working on the freenum/ISN > functionality: > > > same => n,Set(isnresult=${ENUMLOOKUP(${EXTEN},sip,,1,freenum.org)}) > same => n,GotoIf($["${isnresult}" != ""]?:fn-CONGESTION,1) > ; set up our outgoing call state > same => n,Set(S

[asterisk-users] SpiderMux?

2010-04-29 Thread Tim Nelson
Greetings all- I've stumbled upon a TDMoE gateway for FXO/FXS called the SpiderMux. It looks rather interesting. Has anyone used one? Where did you purchase it? Pricing? Operational issues? http://spidermux.com/ Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 --