[Asterisk-Users] gumstix

2004-12-22 Thread Tim Sailer
It would be a real kick to get one of these to run Asterisk. :)

http://www.gumstix.com/

Tim

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[Asterisk-Users] Centrex

2004-11-01 Thread Tim Sailer
OK folks,
  I'm trying to help get another remote Asterisk box up and running.
The system currently has a single FXO card, but it doesn't seem to 
be working, my guess is because the inbound line is CENTREX. Knowing
nothing about Centrex, can someone tell me if I'm right, and need an
FXS card?

Thanks,
Tim

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Re: [Asterisk-Users] Centrex

2004-11-01 Thread Tim Sailer
On Mon, Nov 01, 2004 at 07:13:07PM -0200, Marcelo Pacheco wrote:
 If you plug a plain analog phone to the line, then you need an FXO card in 
 it's place.

Yup. It works with a plain ol' phone, Fax, and modem, but you need to
dial 9 to get out. Just a generic install of Asterisk and having the
FXO card configured, asterisk never sees the line ring.

 Centrex usually means that you have hired a virtual PBX system. That in itself 
 doesn't mean FXS/FXO/T1/E1, a Centrex could be shipped to you in many 
 diferent ways.

sigh. I hate phone companies.
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[Asterisk-Users] tdm400p lockups

2004-08-06 Thread Tim Sailer
I have a system that has a TDM400 with 3 FXO modules. Recently, the 
system started not seeing the ports. No answer, no lines available
for oubound dial, and only a reboot of the machine will bring it back.
Any ideas to try?

Thanks,
Tim

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Re: [Asterisk-Users] tdm400p lockups

2004-08-06 Thread Tim Sailer
On Fri, Aug 06, 2004 at 11:47:31AM -0400, Joshua M. Thompson wrote:
 On Fri, 2004-08-06 at 11:19 -0400, Tim Sailer wrote:
  I have a system that has a TDM400 with 3 FXO modules. Recently, the 
  system started not seeing the ports. No answer, no lines available
  for oubound dial, and only a reboot of the machine will bring it back.
  Any ideas to try?
 
 You too? Mine (two FXO/two FXS) started doing that, and I had to keep
 unloading and reloading the driver to bring it back. Then about a week
 ago it just stopped seeing modules period, no matter what I do. I'm
 about to test it on another machine to make sure it's really dead and
 then RMA it.

Ew.

 Maybe a bad batch of cards? When did you buy yours? I got mine mid-June,
 right after FXO modules became (briefly) available again.

We bought this one when the fxo modules first became available.

Tim

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Re: [Asterisk-Users] iax codec problem

2004-06-06 Thread Tim Sailer
On Tue, Jun 01, 2004 at 07:49:29PM -0500, Yelson Vivas wrote:
 Hi everybody
 
 i have a problem trying to connect an incomming phone call from pstn to my
 (soft phone) iaxcomm, the  phone rings but when i try to answer the call,
 asterisk sends a message like this.
 
 Jun 1 19:33:17 NOTICE[5013528]: channel.c:1223 ast_read: Dropping
 incompatible voice frame on IAX2[192.168.222.99:4569]/16 of format GSM since
 our native format has changed to ALAW

I have the same problem. IAXCOMM works fine with * 0.7.2, but not 0.9.
However, you can make calls fine, just not pick up inbound calls.

Tim

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Re: [Asterisk-Users] UIP 200

2004-05-21 Thread Tim Sailer
Yes, I'm very familiar with Brian's review. Review != Tips or Tricks.

To quote from it: Once I gained access to the appropriate admin guide, I whipped up a 
few
of the configuration files on my TFTP server

How did he gain access? What were the config files? Basic examples? You see,
he plainly said that there were issues. Not what they were and how he
solved them, just that they were there..

Tim

On Thu, May 20, 2004 at 06:02:40PM -1000, James H. Thompson wrote:
 http://www.voip-info.org/tiki-index.php?page=Uniden
 
 Jim
 
 James H. Thompson
 [EMAIL PROTECTED]
 
 - Original Message - 
 From: Tim Sailer [EMAIL PROTECTED]
 To: Asterisk Users [EMAIL PROTECTED]
 Sent: Thursday, May 20, 2004 5:21 PM
 Subject: [Asterisk-Users] UIP 200
 
 
  I have a UIP200 on the way for eval. Does anyone have tips or tricks
  to get it working right away with * ? I hate having to go through the
  pain someone else braver than I went through already. :)
  
  Tim
  
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[Asterisk-Users] UIP 200

2004-05-20 Thread Tim Sailer
I have a UIP200 on the way for eval. Does anyone have tips or tricks
to get it working right away with * ? I hate having to go through the
pain someone else braver than I went through already. :)

Tim

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Re: [Asterisk-Users] Virbiage FT201 IAX Hard Phone

2004-05-10 Thread Tim Sailer
On Mon, May 10, 2004 at 10:53:14AM -0700, Brian D'Arcy wrote:
 Does anyone have any recent news on the Virbiage FT201 IAX Hardphone?
 I'd *really really* like to deploy these phones instead of SIP
 hardphones, and I can't help but wonder if I'm going to shoot myself in
 the foot (or another sensitive area) by deploying a ton of SIP phones
 just to find the IAX Hardphones were released a week later...

The response from them less than 10 days ago was 'about 2 months'. Which
is what they have been saying for the last *3* months. I'm giving up
waiting. They seem to be vaporware. :( Does anyone have a supplier for
the Uniden UIP200 phones?

Tim

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Re: [Asterisk-Users] Newbie x100p install question

2004-05-07 Thread Tim Sailer
On Fri, May 07, 2004 at 10:45:13AM -0400, Rick Beasley wrote:
 PROBLEM: My x100p only does bad things. When I plug the line from the 
 wall into the card, other phones in my house go nuts, no dialtone, crazy 
 clicking, random tones, etc. Following the steps below, the line is 
 screwed up any time I test after step 2. Just plugging the line from the 
 wall into the x100p takes my other phones off line. External calls to my 
 line hear busy. Obviously, can't do the first testing with asterisk+x100p.
 
 1. Installed asterisk
 2. Installed card in slot (no shared IRQs)
 3. Installed (modprobed) drivers
 4. Modified config files to just basics for my environment
 5. Ran ztcfg
 6. Started asterisk
 
 No errors, asterisk starts fine, modprobe -q shows drivers. Next I'm 
 going to experiment with context since I've seen several 
 recommendations including incoming and bell.

This sounds like a hardware error, not software. Change the line cord
from the FXO port to the wall. If that doesn't make a difference, change
jacks. Try that jack with another phone.

Tim

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[Asterisk-Users] Syntax

2004-05-04 Thread Tim Sailer
I've been wondering what the difference is in the syntax of things,
like Dial.

Some examples show things like:
exten = 500,1,Dial,SIP/${EXTEN}|10

but other examples show:
exten = 500,1,Dial(SIP/${EXTEN}|10)
or
exten = 500,1,Dial(SIP/${EXTEN},10)

Which one is correct? Or most correct? Which one is preferred, and why?
I'm sure I'm not the only one with this question... :)

Tim

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[Asterisk-Users] Linux IAX client

2004-05-04 Thread Tim Sailer
Folks,
  It seems like the * v 0.9 and iaxcomm won't speak to each other. Is there
another IAX2 client that is usable under Linux (Debian preferred)?

Thanks,
Tim

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Re: [Asterisk-Users] 0.7.2 debs

2004-05-04 Thread Tim Sailer
On Tue, May 04, 2004 at 03:14:43PM -0400, Jeremy McNamara wrote:
 Tim Sailer wrote:
 
 Does anyone still have the 0.7.2 debs hanging around? I need to revert
 a recent upgrade. We're having too may flaky problems (like softphones
 being able to dial out fine, but GrandStreams failing to dial every
 other time), and iaxcomm not working with gsm.
 
  
 
 
 Why not diagnose the problem and then assist in solving it, if there 
 really is a problem?

Well, there really *is* a problem. I'll try to debug it, but NOT on the 
client's production system! I need to get them working first.

Tim

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[Asterisk-Users] quad fxo

2004-05-03 Thread Tim Sailer
Folks,
  I'm trying to install one of the new quad fxo cards remotely. I know
the existing machine was too old to have a PCI 2.2 bus, so I had my
helper at the other end try a few boxes that were sitting on a shelf
with the new card and a Knoppix cd. He found one that reported the
card as the Tiger Jet. Good. Now, we moved the HD from the existing
machine, loaded with Debian, and the card is just seen as the generic
communications device, like the bus is wrong. Any pointers on this?
The machine is ~500 miles away at the moment, and off the network,
so most of this is done by phone. :(

Thanks,
Tim

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Re: [Asterisk-Users] quad fxo

2004-05-03 Thread Tim Sailer
On Mon, May 03, 2004 at 07:39:06PM +0200, Michael Sandee wrote:
 The bus isn't wrong... debian is wrong. Like everything in debian... it 
 ships with an old pci.ids
 (No flames intended... but still :P )
 
 replace yours with one from:
 http://pciids.sourceforge.net/
 
 And It *should* report it better... (Didn't verify)
 Not that any of this matters... Just load the driver and get on with it.

Driver didn't load either. So I'm supposing that it's the bus. Hmm. MAybe that
was on another machine we tried. I'll give it a whirl.

Thanks,
Tim

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Re: [Asterisk-Users] quad fxo

2004-05-03 Thread Tim Sailer
On Mon, May 03, 2004 at 03:13:56PM -0400, Tim Sailer wrote:
 On Mon, May 03, 2004 at 07:39:06PM +0200, Michael Sandee wrote:
  The bus isn't wrong... debian is wrong. Like everything in debian... it 
  ships with an old pci.ids
  (No flames intended... but still :P )
  
  replace yours with one from:
  http://pciids.sourceforge.net/
  
  And It *should* report it better... (Didn't verify)
  Not that any of this matters... Just load the driver and get on with it.
 
 Driver didn't load either. So I'm supposing that it's the bus. Hmm. MAybe that
 was on another machine we tried. I'll give it a whirl.

Getting the latest CVS zapatel seems to have done the trick. Now to
test this live. :(

Tim

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Re: [Asterisk-Users] smallest phone

2004-04-23 Thread Tim Sailer
On Fri, Apr 23, 2004 at 08:37:42AM -0500, Eric Wieling wrote:
 On Fri, 2004-04-23 at 00:39, James H. Thompson wrote:
  A standard butt set (e.g. http://www.sandman.com/pdf/page81.pdf) combined with a 
  Grandstream (very
  small) or Sipura ATA would make a pretty small combination and be useful for 
  analog PSTN POTS line
  testing too.
 
 We use a buttset and the beta release of the IAXy  (see the mailing list
 archives if you don't know what an IAXy is)

I know what the IAXy is, but I've love to know where  to get one!

Tim

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[Asterisk-Users] smallest phone

2004-04-22 Thread Tim Sailer
Folks,
  I'm looking for a SIP or IAX phone for field techs to take with them
when out on service calls. The regular desktop phones are just way too
big. Is there anything like the size of a full-sized cell phone? Or 
smaller, not I doubt that...

Tim

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Re: [Asterisk-Users] Immixtel VOIP Adapters

2004-03-25 Thread Tim Sailer
On Wed, Mar 24, 2004 at 06:25:01PM -0600, Michael Graves wrote:
 My wife has a business as well as I. I had envisioned the three line
 comming in to * in such a manner that her line would be handled in a
 different manner than my two.

Sure. Just change the context for that Zap channel, and have that
context set up in extensions.conf.

Tim

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Re: [Asterisk-Users] Immixtel VOIP Adapters

2004-03-25 Thread Tim Sailer
On Thu, Mar 25, 2004 at 01:58:54PM -0600, Michael Graves wrote:
 On Thu, 25 Mar 2004 13:51:15 -0500, Tim Sailer wrote:
 
 On Wed, Mar 24, 2004 at 06:25:01PM -0600, Michael Graves wrote:
  My wife has a business as well as I. I had envisioned the three line
  comming in to * in such a manner that her line would be handled in a
  different manner than my two.
 
 Sure. Just change the context for that Zap channel, and have that
 context set up in extensions.conf.
 
 Tim
 
 Tim,
 
 I understand that as it relates to my present installation with 3 X101p
 cards in my server. I was wondering if the 4 port FXO adapters from
 Welltech/Clipcomm/Mediatrix/add-your-favorite-here would work the same
 way. They're not Zap channels...I think? Do they present themselves as
 4 separate ports to *?

What does ztcfg -v show you? How many channels? You should see info
in 'dmesg' also.

Tim

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Re: [Asterisk-Users] Problems with asterisk and gnophone on Gentoo box

2004-03-19 Thread Tim Sailer
On Fri, Mar 19, 2004 at 12:53:04PM -0500, Kevin wrote:
 gnophone still seems iffy.  Sometimes I start it and get:
 bash-2.05b# gnophone
 Loaded and activated '/usr/lib/gnophone/modules/audio-oss.so'
 Loaded and activated '/usr/lib/gnophone/modules/audio-phone.so'
 iax.c line 654 in iax_init: Started on port 5036
 Listening on port 5036
 Initialized phone core
 No audio devices found

I'm master of the obvious, so forgive me if this sounds like a simple
or stupid question, but you're not trying to run asterisk and gnophone
on the same system, with only one sound device, are you?

Tim

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[Asterisk-Users] Monastery Devel snapshot

2004-03-18 Thread Tim Sailer
I just pushed out a snapshot of the -devel version of monastery.

ftp://ftp.buoy.com/pub/asterisk/monastery-devel.tgz

Tim

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Re: [Asterisk-Users] Crisco Softphone

2004-03-17 Thread Tim Sailer
On Tue, Mar 16, 2004 at 05:20:28PM -0800, Andrew Gillham wrote:
 Derek Bruce wrote:
 
 depends on which Cisco softphone you are refering to... they have a few
 different versions... including an NBX version which will not work with
 Asterisk...
 
 
 - Original Message -
 From: Tim Sailer [EMAIL PROTECTED]
 To: Asterisk Users [EMAIL PROTECTED]
 Sent: Tuesday, March 16, 2004 2:57 PM
 Subject: [Asterisk-Users] Crisco Softphone
 
 
  
 
 I was given an eval of the Cisco softphone to try out. Has anyone
 gotten this to work with * yet?
 
 Tim
 

 
 From the subject I would have guessed some sort of vegetable shortening 
 based phone,
 not an NBX one.

Callmanager based one V1.3(3)

Tim

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[Asterisk-Users] NuFone?

2004-03-17 Thread Tim Sailer
Does anyone have any other information on nufone? nufone.net has got
to be the most singularly uninformative and annoying web page I've seen
for a business in quite a while...

Things like prices would be nice, also areas of coverage, etc.

Tim

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Re: [Asterisk-Users] NuFone?

2004-03-17 Thread Tim Sailer
On Wed, Mar 17, 2004 at 09:10:42AM -0600, Matthew Marlowe wrote:
 Try searching the forums or emailing sales, or calling sales.

Nah. I'll go elsewhere. Me having to search hard for information like
that tells me they're not looking for new business.

Thanks,
Tim

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[Asterisk-Users] local VoIP in Florida

2004-03-17 Thread Tim Sailer
Is there *any* providers that have local numbers in the Tampa/St. Pete
areas? St. Petersburg is 727 areacode. We're trying to test the market
for some services in other areas, and want local numbers...

Tim

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Re: [Asterisk-Users] local VoIP in Florida

2004-03-17 Thread Tim Sailer
On Wed, Mar 17, 2004 at 08:54:57PM -0600, Matthew Marlowe wrote:
 727 or 772? There is 772 in FL available. 

727. That's St. Pete/Clearwater. What area is 772?

Tim

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[Asterisk-Users] Crisco Softphone

2004-03-16 Thread Tim Sailer
I was given an eval of the Cisco softphone to try out. Has anyone
gotten this to work with * yet?

Tim

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[Asterisk-Users] MySQL VM config

2004-03-11 Thread Tim Sailer
In Monastery, I'm using the show voicemail users command to get a list
of defined users, and how many VM messages they have. It seems that this
doesn't work when MySQL is used for the VM config. I can get the mbox
info out of the correct table, but where can I find the number of unread
messages?

Tim

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[Asterisk-Users] monastery devel page

2004-03-08 Thread Tim Sailer
I have had quite a number of people asking if I would incorporate certain
features into monastery. I plan to do that, as time permits, *after* I
get done with the enhancements I want to do. If you would like to watch
what the latest looks like, you can point your browser to
http://pbx.unslept.com/newstatus.php . This may be broken at times, so
please *don't* send me email tellimg me so, I know already. I did it. :)

The latest change is obvious at the top of the page. User turns green
when their client registers, extension goes red when they are on the
phone. Voicemail count is next to that.

Tim

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Re: [Asterisk-Users] monastery devel page

2004-03-08 Thread Tim Sailer
On Mon, Mar 08, 2004 at 10:35:54AM -0600, Matthew Marlowe wrote:
 When is this going to be available? 

Sometime this week. I want to make a few other changes before I release
this version. I need to make it look better. My UI skills suck royally,
so I want to make it look a bit better, which will make it easier to
watch.

Tim

PS: I'll make a monastery-dev.tgz snapshot available, and leave the stable
one. I'm amazed at the interest in this little project.

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Re: [Asterisk-Users] message lights and stutter tones

2004-03-08 Thread Tim Sailer
On Mon, Mar 08, 2004 at 08:14:00PM -, Simon Chappell wrote:
 Hi al
 
 I have 3 GS 101's plugged into asterisk.
 They work great and teh quality of sound I can not fault. Most people I am
 speaking to now ask if I have a new phone because the quality is so much
 better.
 My latest quandry is to do with the message button and stuttertones. I
 dont get either.. If i have a message waiting the only way we know is by
 email..
 
 I have added mailbox=mailboxNumber for each extension but still no avail.
 
 Is there any magic tweak I am missing ?

Yup. [EMAIL PROTECTED]

Tim

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Re: [Asterisk-Users] message lights and stutter tones

2004-03-08 Thread Tim Sailer
On Mon, Mar 08, 2004 at 03:12:52PM -0600, [EMAIL PROTECTED] wrote:
 Simon,
 Do the GS phones support stutter tone as-well-as 
 the message light?

I'm not Simon, but yes, they do. At least my -100 does. The display
backlight flashes, and you get the stutter dialtone.

Tim

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Re: [Asterisk-Users] Monastery - question

2004-03-08 Thread Tim Sailer
On Mon, Mar 08, 2004 at 10:06:22PM -0300, Bartosz Jozwiak wrote:
 Hello,
 
 For I would like to say that this is great tool!!!

Thanks!

 I have one question. Why my busy indication
 is turned on even when person hangup a call ?

IAX or SIP?

 I have noticed that this happens only to phones
 extension where a lot of incomming call trafic is
 generated. Is it something to do with call waiting?

Nope. Since this is starting to clog the * list, I'll set up a
monastery list on buoy.com to handle the support, etc. Contact me
off-list to work through this.

Tim

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[Asterisk-Users] monastery mailinglist

2004-03-08 Thread Tim Sailer
http://www.buoy.com/mailman/listinfo/monastery is where you can sign up
and join the chatter about Monastery. Support and announcements will be 
on this list

Tim

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[Asterisk-Users] monastery

2004-03-06 Thread Tim Sailer
MONitor ASTERisk easilY 

This is this perl and PHP thing I was putting together over the last
day or so.

You can see it working on http://pbx.unslept.com/status.php , and you
can download it from ftp://buoy.com/pub/asterisk/monastery.tgz

Tim

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Re: [Asterisk-Users] monastery

2004-03-06 Thread Tim Sailer
On Sat, Mar 06, 2004 at 09:26:28PM -, Scott Stingel wrote:
 cool!  I like the light bulbs.   (and the name)

They were something I had in an old image collection. I'll
hvae to find something a bit more 'modern'. :)
 nice job 

Thanks. My wife, a very accomplished graphics designer, just laughs
at my UI attempts. Hey, they work, which is all I care about.

Tim

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Re: [Asterisk-Users] monastery

2004-03-06 Thread Tim Sailer
On Sat, Mar 06, 2004 at 04:38:18PM -0500, [EMAIL PROTECTED] wrote:
 What about Zap Channels?

That's next on the list. I have to figure out what to do with them.

Tim

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Re: [Asterisk-Users] Simple * status

2004-03-05 Thread Tim Sailer
On Fri, Mar 05, 2004 at 08:27:59PM +0100, [EMAIL PROTECTED] wrote:
 Tim,
 It looks interesting.. Are you willing to release the  source code?

Sure. let me clean it up a bit... OK, a LOT... and finish the comments,
and I'll have a download link for it sometime this weekend. I'll keep
the downloadable stuff up-to-date with the running version.

Tim

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[Asterisk-Users] Meetme

2004-03-04 Thread Tim Sailer
OK, maybe I need more coffee. Or less. Either way, I'm stumped.

I have a Meetme conference room configured. Meetme(|M|) to enable
the MOH. When you are the first one to go into the conf, you get the
announcement that you are the only one, and then a *male* voice gives
a little talk about 'Why are we putting you on hold?'. Where does that
come from and how do I get rid of it? It's not any of the sound files 
that I can see/hear, and running 'asterisk -vvvr' doesn't show any
file being played...

Tim

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Re: [Asterisk-Users] Re: Meetme

2004-03-04 Thread Tim Sailer
On Thu, Mar 04, 2004 at 02:53:20PM -, Tony Mountifield wrote:
 In article [EMAIL PROTECTED],
 Tim Sailer [EMAIL PROTECTED] wrote:
  OK, maybe I need more coffee. Or less. Either way, I'm stumped.
  
  I have a Meetme conference room configured. Meetme(|M|) to enable
  the MOH. When you are the first one to go into the conf, you get the
  announcement that you are the only one, and then a *male* voice gives
  a little talk about 'Why are we putting you on hold?'. Where does that
  come from and how do I get rid of it? It's not any of the sound files 
  that I can see/hear, and running 'asterisk -vvvr' doesn't show any
  file being played...
 
 Have a look in /var/lib/asterisk/mohmp3 - there is a sample file in
 there with what you describe. Delete or move that file, and put your
 own MP3s in that directory.

OK, that's strange. I took out the sample file a long time ago! I guess
an upgrade put it back... sheesh. Gotta be careful about that. That's 
strange. I must have hit the timing *perfectly*... the 3 times I tried this,
that mp3 played as soon as I entered the room, and it's supposed to be
a random play... I think it's time I played lotto this week! :)

Tim

PS: Thanks. That was driving me nuts!

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[Asterisk-Users] 3com NBX phones

2004-03-04 Thread Tim Sailer
Does anyone know if the 3Com NBX 2102 series phones with with * ? There
are a crapload (a very precise measurement) on eBay, but I can't figure
out what protocols they talk.

Tim

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Re: [Asterisk-Users] 3com NBX phones

2004-03-04 Thread Tim Sailer
Just found on nwfusion.com:

3Com plans to announce the 3102 Business Phone, a SIP-based handset that 
works with the vendor's VCX IP PBX, technology borrowed from 3Com's 
now-defunct carrier softswitch business.

The phone is also compatible with 3Com's small- and midsize-site NBX IP PBX. 
The phones support a G.723 wideband audio codec, which 3Com says provides 
clearer voice than previous 3Com IP phones. The phone is expected to be 
available March 19 for $310. 




Maybe they'll have the SIP NBX images out. It seems like you can buy the phones
for $35-60

Tim

On Thu, Mar 04, 2004 at 01:28:52PM -0700, Derek Bruce wrote:
 The 3com NBX phones use a protocol called H3, developed by graduate students
 at MIT. The protocol has no documentation at all (except by a very P.O.ed
 guy in Australia. 3com can by the protocol when they aquired NBX corp in
 1999. (NBX corp was the company formed by MIT to market the system).
 
 So
 Short answer... NO, the NBX phones can not be used by Asterisk or any other
 system...
 Long answer... There is supposed to be a beta SIP firmware for the NBX
 phones, derived from the 3com SIP phones (which were apparently very similar
 hardware). 3com disavowes any knowledge of this firmware, and their SIP
 phone have been discontinued. There are people on this list that have used
 said firmware, but they have not indicated a lot of success. 3com has
 recently (weell, over a year ago.. but that's recent in terms of 3com
 development) announced a system called the VBX (which is supposed to support
 SIP) and SIP support for the NBX, which means that there may be a sip image
 available eventually for the NBX phones. If you are looking at buying the
 phones to use with Asterisk, you would be better off buying another vendors
 product. If you already have an NBX... the landscape is going to be changing
 soon (we hope).
 
 BTW, the NBX was supposed to be, 4 years ago, what Asterisk is today... Open
 Source, expandable and extensible. We did our research and asked our
 questions and purchased the system. spent $14000 Canadian. It still sells
 for about the same price. We even joined their Solutions Provider Program...
 what we got was a system that will do what we want... but will cost us
 another $1 CDN.
 
 So... cheers to Mark for starting, and continuing, a system that is all, and
 much more, that the NBX wanted to, be but couldn't...
 
 My appologies to all who feel this is off topic...
 
 Derek Bruce
 Calgary Telecom
 
 - Original Message -
 From: Tim Sailer [EMAIL PROTECTED]
 To: Asterisk Users [EMAIL PROTECTED]
 Sent: Thursday, March 04, 2004 12:38 PM
 Subject: [Asterisk-Users] 3com NBX phones
 
 
  Does anyone know if the 3Com NBX 2102 series phones with with * ? There
  are a crapload (a very precise measurement) on eBay, but I can't figure
  out what protocols they talk.
 
  Tim
 
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Re: [Asterisk-Users] VTGO-PG and IPP200

2004-03-03 Thread Tim Sailer
Email back from ipblue:

Thank you for your inquiry. At the present time we work under SCCP or
H.323 protocols. We are developing a SIP phone which will be available
sometime in Q2.

Regards,

Andrew Schecter
Vice President of Sales
IP blue Software Solutions
15 East 26th Street
New York, NY  10010
212.485.1225 (tel)
212.485.1380 (fax)
[EMAIL PROTECTED]

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[Asterisk-Users] VTGO-PG and IPP200

2004-03-02 Thread Tim Sailer
First, thank you to whomever it was that pointed me towards the 
IPP200. It works great, both under Windows with X-lite and Linux
with iaxcomm. It took me about 3 minutes on each to get it
configured and working.

Second, has anyone tried the VTGO-PC (any version) from ipblue.com? I'd
dearly love a softphone with MWI...

Tim

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Re: [Asterisk-Users] VTGO-PG and IPP200

2004-03-02 Thread Tim Sailer
On Tue, Mar 02, 2004 at 12:11:22PM -0600, Steven Sokol wrote:
 IAX Phone has MWI and, with a small change in your Asterisk, can give you a
 count of messages waiting.  Plus it has hook-switch integration with the
 IPP200.

I'd like to try it, but I'm unable to get it to even start on my Lose2k
system...

Tim

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Re: [Asterisk-Users] VTGO-PG and IPP200

2004-03-02 Thread Tim Sailer
On Tue, Mar 02, 2004 at 10:40:24AM -0800, Ed Rubright wrote:
 The VTGO-PC appears to be Cisco Call Manager only or did you find somewhere
 on there site that they have a SIP version?

I'm bugging them about the SIP, but so far, no answer. They claim it's
a software close of the Cisco hardphone, so...

 If it is Call Manager only...how far along is Asterisk support for Call
 Manager?  I looked on the Wiki pages and saw mentioned the 2 implementations
 of the skinny protocol, but nothing about its current feature state.

I'm not sure what state it's in. Or how to configure it for that matter.
I may grab the eval copy and try to crash * with it... :)

Tim

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Re: [Asterisk-Users] Tiny install with Solid State Storage

2004-03-01 Thread Tim Sailer
On Mon, Mar 01, 2004 at 10:01:48AM -0600, Matt wrote:
 Hello All,
 I was wondering if anyone is successfully running asterisk on a system
 with solid state storage, such as a compact flash card? I'm looking for some
 pointers on doing this.

Look for pebble linux, based on Debian, which has asterisk packaged for it
already. You can find it somewhere on www.nycwireless.org. This is what
they use for the Linux based APs scattered across the city.

Tim

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[Asterisk-Users] USB Phones

2004-02-27 Thread Tim Sailer
I have some mobile users that would prefer to have a 'real phone' instead
of a computer headset. I've been looking around at the USB phone setups,
which is (it seems) simply a softphone with a USB handset. The only
ones I've found seen to be locked to a particular service provider.
Has anyone used these, and are there any ones that can work as a
general softphone, like X-Lite?

Tim

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Re: [Asterisk-Users] USB Phones - Proper URL

2004-02-27 Thread Tim Sailer
On Fri, Feb 27, 2004 at 10:09:49AM -0600, Steven Sokol wrote:
 Again, I hosed up some thing.  Wrong URL.  Here's the proper URL for
 Eutectics:
 
 http://www.eutecticsinc.com/usbPhones/usbPhones.html

I figured it out. :)

 Perhaps I should try sleeping.  They say it's good for you...

Really? I wouldn't know... (my home domain is unslept.com) :)

Tim

PS: You just got the driver only option?

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Re: [Asterisk-Users] USB Phones - Proper URL

2004-02-27 Thread Tim Sailer
On Fri, Feb 27, 2004 at 10:48:00AM -0600, Steven Sokol wrote:
  PS: You just got the driver only option?
 
 [Steven Sokol] 
 Yep.  I did order the API as well.  They make you sign an NDA (pretty basic
 one).  The API covers the hook-switch integration and the keypad integration
 for their IPP5xx series phones.
 
 What client are you going to use it with?

Something SIP. Most likely start with X-Lite, and see how that goes over
with the folks in the field.

Tim

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[Asterisk-Users] gnophone

2004-02-27 Thread Tim Sailer
Does anyone actually have a 2.4 version of gnophone that will compile?
All the copies on the ftp site have a corrupt file, as does CVS...

Tim

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Re: [Asterisk-Users] voicemail not working with mysql!!!!

2004-02-25 Thread Tim Sailer
On Wed, Feb 25, 2004 at 01:46:39PM +, Fran Boon wrote:
 one more thing which one is newer versionand has mysql support
 voicemail or voicemail2
 
 'voicemail' is deprecated.
 When people talk about voicemail these days, they mean 'voicemail2'

Really? In the docs somewhere, it shows voicemail2 as deprecated.

Tim

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Re: [Asterisk-Users] voicemail not working with mysql!!!!

2004-02-25 Thread Tim Sailer
On Wed, Feb 25, 2004 at 03:22:30PM +, Fran Boon wrote:
 When people talk about voicemail these days, they mean 'voicemail2'
 Really? In the docs somewhere, it shows voicemail2 as deprecated.
 
 ok, 'voicemail2' has been renamed as 'voicemail'

Ah. Non-obvious. Now I'll have to go back and play with it to see
if I can get it to work.

Tim

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Re: [Asterisk-Users] Simulating the lighted line in use type of phone

2004-02-25 Thread Tim Sailer
On Wed, Feb 25, 2004 at 05:46:55PM -0500, Andrew Thompson wrote:
  I looked at it, and it won't work for me the way you have written it
  without me hacking at it. You insist on having MySQL run in network
  mode. I really don't want another port/service running on a box that
  is internet-facing... call me silly. :) For things like this, I'll
  only have the service running locally.
  
  Tim
 
 Just because a port is active doesn't mean that the world has to be able
 to see it.

True, but my * server is on the public Internet, not behind a firewall.

 I've not used this script(astguiclient), is it intended to be ran from
 the client, or from the server?

Both, it seems.

Tim

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Re: [Asterisk-Users] Simulating the lighted line in use type of phone

2004-02-25 Thread Tim Sailer
On Tue, Feb 24, 2004 at 10:37:21PM -0500, Tim Sailer wrote:
 On Tue, Feb 24, 2004 at 04:33:41PM -0500, mattf wrote:
  Take a look at my GUI app:
  
  http://sourceforge.net/projects/astguiclient/
  
  It'll run on Linux and Windows, it's written in perl and it'll list every
  channel(Zap/SIP/Local) that is active on your system updated every second.
  
  You can also do a lot of other things with it too.
  
  We've been using it here at my company for months on over 60 computers and
  it scales rather well.
 
 I looked at it, and it won't work for me the way you have written it
 without me hacking at it. You insist on having MySQL run in network
 mode. I really don't want another port/service running on a box that
 is internet-facing... call me silly. :) For things like this, I'll only
 have the service running locally.

OK, following up on my own e-mail, I have the gui working with the
unix socket by doing the following:

in libs/AST_VICI_conf.pl:

$DB_server = '';# MySQL server IP
#added unix socket to support local use - tps 02252004
$DB_unixsocket = '/var/run/mysqld/mysqld.sock'; # path to local socket

in AST_WINphoneAPP_0.9.pl and AST_update.pl, changed every instance of:

my $dbh = Net::MySQL-new(hostname = $DB_server, database = $DB_database, user 
= $DB_user, password = $DB_pass) 
or  die Couldn't connect to database: \n;

to 

my $dbh = Net::MySQL-new(unixsocket = $DB_unixsocket, hostname = $DB_server, 
database = $DB_database, user = $DB_user, password = $DB_pass) 
or  die Couldn't connect to database: \n;


Now it starts up and runs, and the Zap channels are seen, but not the SIP
ones yet...

Tim

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Re: [Asterisk-Users] Simulating the lighted line in use type of phone

2004-02-24 Thread Tim Sailer
On Tue, Feb 24, 2004 at 03:49:36PM -0500, Chris Clifton wrote:
 That's fine for outbound lines, but what if I want to call the guy in the
 next office ? I have to call him and get redirected to his busy vm just to
 know that he's on the phone.
 
 This is a huge issue with the recepetionist with the 'master console'. How
 does he/she know whether a user is busy or not ?
 
 20 year old phone systems offer this capability.
 
 I see this as a serious shortfall of asterisk / currently available ip
 phones right now.

I'm scratching out a way of doing this using the manager port. There is
a TON of info that is passed to the manager port once you are authenticated.
I'd love the interface to be Java (portablilty), but it could be web based
with a short refresh (say 5 seconds) from a database.

Tim

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Re: [Asterisk-Users] Can't compile Zaptel drivers

2004-02-24 Thread Tim Sailer
I believe this is a PPP issue. Either make sure you have
ppp support in your kernel (even as a module) or look in the
Makefile for 'PPP'. It should be obvious when you see it.

Tim

On Tue, Feb 24, 2004 at 05:47:24PM -0500, Mark Phillips wrote:
 Hi folks,
 
 I'm trying to compile the zaptel driver and am getting nowhere.
 
 I've got 2.4.22-10mdk sources loaded.
 
 I've edited the Makefile to reflect my multiprocessor kernel (although I'm
 only running 1 P4 - Mandrake 9.2 installed it this way) and I've run a
 make menuconfig in the linux source area and loaded the multi processor
 config and done a make dep.
 
 When I run the zaptel Makefile I get the following unresolved symbols;
 
 /sbin/depmod -a
 depmod: *** Unresolved symbols in
 /lib/modules/2.4.22-10mdk-p3-smp-64GB/misc/tor2.o
 depmod: *** Unresolved symbols in
 /lib/modules/2.4.22-10mdk-p3-smp-64GB/misc/torisa.o
 depmod: *** Unresolved symbols in
 /lib/modules/2.4.22-10mdk-p3-smp-64GB/misc/wcfxo.o
 depmod: *** Unresolved symbols in
 /lib/modules/2.4.22-10mdk-p3-smp-64GB/misc/wcfxs.o
 depmod: *** Unresolved symbols in
 /lib/modules/2.4.22-10mdk-p3-smp-64GB/misc/wct1xxp.o
 depmod: *** Unresolved symbols in
 /lib/modules/2.4.22-10mdk-p3-smp-64GB/misc/wct4xxp.o
 depmod: *** Unresolved symbols in
 /lib/modules/2.4.22-10mdk-p3-smp-64GB/misc/wcusb.o
 depmod: *** Unresolved symbols in
 /lib/modules/2.4.22-10mdk-p3-smp-64GB/misc/zaptel.o
 depmod: *** Unresolved symbols in
 /lib/modules/2.4.22-10mdk-p3-smp-64GB/misc/ztd-eth.o
 depmod: *** Unresolved symbols in
 /lib/modules/2.4.22-10mdk-p3-smp-64GB/misc/ztdynamic.o
 
 
 Any ideas anyone?
 
 
 
 -- 
 Mark Phillips, G7LTT/KC2ENI
 Randolph, NJ
 http://www.g7ltt.com/
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Re: [Asterisk-Users] Simulating the lighted line in use type of phone

2004-02-24 Thread Tim Sailer
On Tue, Feb 24, 2004 at 04:33:41PM -0500, mattf wrote:
 Take a look at my GUI app:
 
 http://sourceforge.net/projects/astguiclient/
 
 It'll run on Linux and Windows, it's written in perl and it'll list every
 channel(Zap/SIP/Local) that is active on your system updated every second.
 
 You can also do a lot of other things with it too.
 
 We've been using it here at my company for months on over 60 computers and
 it scales rather well.

I looked at it, and it won't work for me the way you have written it
without me hacking at it. You insist on having MySQL run in network
mode. I really don't want another port/service running on a box that
is internet-facing... call me silly. :) For things like this, I'll only
have the service running locally.

Tim

-- 

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 Network and Systems Operations PO Box 726  
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Re: [Asterisk-Users] Re: vm to email ?

2004-02-20 Thread Tim Sailer
On Thu, Feb 19, 2004 at 01:11:31AM -0500, Netlabz - Chris Clifton wrote:
 duuh .. figured it out ... didn't have sendmail running.
 
 Too much * hacking at the end of a long day.

Sendmail should not have to be running to be able to send outbound mail.
It only needs to be running to receive connections. Take a look at
'sendmail -q'. Having sendmail-hack-dujour running opens you machine
to cmopromise, if you are running on the public network.

Tim

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[Asterisk-Users] POTS Distinctive Ring

2004-02-20 Thread Tim Sailer
Is there any way for * to tell the difference in inbound ring patterns?
I'm finding docs on how to make it ring Zap channels differently, but
I have a situation where one inbound line has distinctive ring for a
another purpose, and I want * to do something different what that
ring pattern is seen...

Tim

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Re: [Asterisk-Users] Digium connectivity issue?

2004-02-15 Thread Tim Sailer
On Sun, Feb 15, 2004 at 12:18:03PM -0500, John Fraizer wrote:

 There are several entities out there who will do secondary DNS for free. 

I'll do secondary DNS if they want.

Tim

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Re: [Asterisk-Users] Digium connectivity issue?

2004-02-14 Thread Tim Sailer
On Fri, Feb 13, 2004 at 07:54:50AM -0600, Rich Adamson wrote:
 
 Are others seeing hugh delays and/or lack of connectivity to Digium?

Yeah. Traceroutes stop at their provider. They are either having
connectivity issues (if they have dsl/T1/whatever into their facility)
or server problems if colo-ed. Digium and * websites are down at the
same time...

Tim

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Re: [Asterisk-Users] HELP!!!! Having problems Starting Asterisk

2004-02-14 Thread Tim Sailer
On Fri, Feb 13, 2004 at 10:37:01PM +, Robert Boardman wrote:
 I have been trying to start asterisk all night after a reboot
 
 I keep getting this error scrolling up the screen
 
 ouch: error while writing audio data broken pipe
 
 when I go to another console there are 4 instances of mpg123  running 
 and  when I do TOP they are taking 100% CPU between them
 
 I have re installed mgp123 but it still doesn't help
 
 any Ideas?

Try shutting down all * processes (including mpg123). Now, see if your
audio works normally. If not, rmmod the zaptel/fx? modules, and see if that
works. If not, you should start by getting your audio on the consloe to
work normally first, then, check with the zap/etc modules loaded, then
try * . One step at a time.

Tim

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[Asterisk-Users] GS BT-100 echo

2004-02-13 Thread Tim Sailer
I picked up a GS 100 phone based on the overall good response I've heard
of these phones. One thing I'm fighting with, which I can't find any
info on, is a *real* bad local echo on the GS. The remote end doesn't
hear it, and all the docs I see about echocancel deal with hardwired 
phones/ports (fxs/fso).

Phone software is:
Software Version: Program--1.0.4.45Bootloader--1.0.0.13HTML--1.0.0.20

if that matters.

sip.conf for the phone is:

[gs1]
type=friend
username=gs1
secret=
host=dynamic
canreinvite=no
nat=yes
qualify=1000
disallow=all
allow=alaw
allow=ulaw

Tim

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[Asterisk-Users] Database items

2004-02-12 Thread Tim Sailer
I have the MySQL CDR working, and, being a long-time DBMS programmer,
I'm looking to put other things into the database. I see that Voicemail
*used* to be supported in MySQL, but that seems deprecated. Is there
anything else that has DB support? I'd sure like to be able to do
more from scripts, web, etc.

Tim

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[Asterisk-Users] Log entry

2004-02-10 Thread Tim Sailer
Feb 10 09:57:37 WARNING[98311]: db.c:46 dbinit: Unable to open Asterisk database

I'm seeing this in my logs, 0.7.2 (debian package). I looked through the
source, and thought it was looking for a RDBMS (mysql/postresql), so
I set both of them up (one at a time), and I'm still getting it.
What is this talking about?

Tim

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[Asterisk-Users] Log entry - solved

2004-02-10 Thread Tim Sailer
Sigh. As soon as I send e-mail, I find the answer. /var/lib/asterisk/astdb
is the file. Was owned by root, not asterisk (at least on Debian).
Time to file a package bug report for Debian.

Tim

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Re: [Asterisk-Users] Wait command in auto attendant causes sched.c error

2004-02-10 Thread Tim Sailer
On Tue, Feb 10, 2004 at 03:04:41PM -0600, Tilghman Lesher wrote:
 On Tuesday 10 February 2004 14:36, Christopher J. Wolff wrote:
  Feb 10 13:19:36 NOTICE[17743913]: Request to schedule in the
  past?!?!
 
 Your machine is heavily loaded.  A thread which was interrupted to
 schedule another thread or process was not able to complete its task
 in time.  Your solution is either to reduce the load on that machine
 or get yourself a faster machine.  Low memory is another possible
 culprit (i.e. your processes are swapping, which slows down context
 switches).

I get this on my brand new machine, 2.4 GHz, 512M RAM, .01 load

Tim

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Re: [Asterisk-Users] Calling SIP

2004-02-09 Thread Tim Sailer
On Mon, Feb 09, 2004 at 01:37:55PM -0600, Eric Wieling wrote:
 That's just the way Asterisk's dial command works.

Hmm. I see. If it can't create the channel for either reason
(busy or not registered), it's handled the same. I think I'll
kludge up a perl script to watch the SIP channels register and
unregister, and update a database table, which will be displayed
on a web page to show who is actually active.

Tim

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Re: [Asterisk-Users] modprobe wcfxs

2004-02-07 Thread Tim Sailer
Replying to my own e-mail, how tacky.

Anyway, just for the record, I went out to the local computer shop
looking for a new power supply, and ended up walking out $255
poorer, but with a new motherboard, 2.4Ghz CPU and 256M DDR memory,
*and* a new case and power supply.

2 hours later, Debian was isntalled on an old disk, the kernel recompiled,
zaptel recompiled, modprobes worked, and all is right with the world.

So, solution *was* the motherboard, as the 2.2 PCI wasn't really
supported until late 2000, it seems.

Thanks for all the help,
Tim

On Fri, Feb 06, 2004 at 06:52:54PM -0500, Tim Sailer wrote:
 On Fri, Feb 06, 2004 at 05:36:47PM -0600, Tilghman Lesher wrote:
  On Friday 06 February 2004 16:26, Tim Sailer wrote:
   OK,  folks... I'm having the same problem as a few people. device
   not found when I do the modprobe wcfxs. I looked in the archives,
   and I see 4 or 5 people have had the same problem. I even foudn the
   reply to a post like mine that said look in the archives, others
   have had the same problem. Very true, but I can't find the answer.
   If someone can simply point me to the archive with the solution, I
   can go from there. :)
  
  A common problem is forgetting to connect the 4-pin molex on the side
 
 Nope, plugged in. I even tested it to make sure the voltages were
 right.
 
  of the card.  If you're still having problems, you could try a greater
  wattage power supply or a different motherboard.
 
 Hrm. A different mother board is out of the question right now. The
 power supply, maybe. ATX...
 
  It's strange, but while the TDM400P is up to the PCI spec, some
  motherboards are deficient.  It is only when inserting a card which
  stresses the PCI spec to the max that you may wind up discovering
  this.
 
 This is an older motherboard, but it never had any problems driving
 things like video capture which tends to stress the bus.
 
  Also, if your TDM400P does not have a molex connector, you can get
  a free upgrade from the company that sold you the TDM400P.
 
 Brandy new from Digium. It had better be right. :)
 
 Tim
 
 -- 
 
  Tim Sailer Coastal Internet, Inc.  
  Network and Systems Operations PO Box 726  
  http://www.buoy.comMoriches, NY 11955  
  [EMAIL PROTECTED]   (631) 399-2910  (888) 924-3728  
 
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[Asterisk-Users] ringing

2004-02-07 Thread Tim Sailer
When an extention (either ZAP or SIP) is dialed, the calling person
hears just silence. Is it possible to get Ringing to work with this?
It seems to cause people to hang up when there is silence.

Tim

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 Network and Systems Operations PO Box 726  
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Re: [Asterisk-Users] ringing

2004-02-07 Thread Tim Sailer
On Sun, Feb 08, 2004 at 12:04:09AM -0500, Lance Arbuckle wrote:
 
 
 Tim Sailer wrote:
  
  When an extention (either ZAP or SIP) is dialed, the calling person
  hears just silence. Is it possible to get Ringing to work with this?
  It seems to cause people to hang up when there is silence.
  
  Tim
 
 
 
 http://www.voip-info.org/wiki-Asterisk+cmd+Dial

Jeez. I'm tired. Thank you. I read that 3 times, and, only on the
fourth time did I pick up on the 'r' option. What would be nice would
be to have a different ring (like our big seimens), regular ring for
an outside line, short-short for an inside line...

Tim

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 Network and Systems Operations PO Box 726  
 http://www.buoy.comMoriches, NY 11955  
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[Asterisk-Users] modprobe wcfxs

2004-02-06 Thread Tim Sailer
OK,  folks... I'm having the same problem as a few people. device not
found when I do the modprobe wcfxs. I looked in the archives, and I see
4 or 5 people have had the same problem. I even foudn the reply to a post
like mine that said look in the archives, others have had the same problem.
Very true, but I can't find the answer. If someone can simply point me to  
the archive with the solution, I can go from there. :)

Thanks,
Tim

-- 

 Tim Sailer Coastal Internet, Inc.  
 Network and Systems Operations PO Box 726  
 http://www.buoy.comMoriches, NY 11955  
 [EMAIL PROTECTED]   (631) 399-2910  (888) 924-3728  

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Re: [Asterisk-Users] modprobe wcfxs

2004-02-06 Thread Tim Sailer
On Fri, Feb 06, 2004 at 05:36:47PM -0600, Tilghman Lesher wrote:
 On Friday 06 February 2004 16:26, Tim Sailer wrote:
  OK,  folks... I'm having the same problem as a few people. device
  not found when I do the modprobe wcfxs. I looked in the archives,
  and I see 4 or 5 people have had the same problem. I even foudn the
  reply to a post like mine that said look in the archives, others
  have had the same problem. Very true, but I can't find the answer.
  If someone can simply point me to the archive with the solution, I
  can go from there. :)
 
 A common problem is forgetting to connect the 4-pin molex on the side

Nope, plugged in. I even tested it to make sure the voltages were
right.

 of the card.  If you're still having problems, you could try a greater
 wattage power supply or a different motherboard.

Hrm. A different mother board is out of the question right now. The
power supply, maybe. ATX...

 It's strange, but while the TDM400P is up to the PCI spec, some
 motherboards are deficient.  It is only when inserting a card which
 stresses the PCI spec to the max that you may wind up discovering
 this.

This is an older motherboard, but it never had any problems driving
things like video capture which tends to stress the bus.

 Also, if your TDM400P does not have a molex connector, you can get
 a free upgrade from the company that sold you the TDM400P.

Brandy new from Digium. It had better be right. :)

Tim

-- 

 Tim Sailer Coastal Internet, Inc.  
 Network and Systems Operations PO Box 726  
 http://www.buoy.comMoriches, NY 11955  
 [EMAIL PROTECTED]   (631) 399-2910  (888) 924-3728  

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Re: [Asterisk-Users] modprobe wcfxs

2004-02-06 Thread Tim Sailer
On Fri, Feb 06, 2004 at 07:58:09PM -0500, Sean Cheesman wrote:
 Now we're getting somewhere!  The TDM400P is a PCI 2.2 card.  So
 depending on what you mean by an older motherboard, that might be your
 problem.  

It's a dual Celery (A-Bit, I think) board from around 1999-2000.

Tim

-- 

 Tim Sailer Coastal Internet, Inc.  
 Network and Systems Operations PO Box 726  
 http://www.buoy.comMoriches, NY 11955  
 [EMAIL PROTECTED]   (631) 399-2910  (888) 924-3728  

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[Asterisk-Users] zaptel on Debian

2004-02-05 Thread Tim Sailer
Does anyone have the zaptel modules built for Debian 2.4.24 kernel?

When I try to compile I get :

./gendigits
gcc -I/include -O2 -DMODULE -D__KERNEL__ -DEXPORT_SYMTAB -Wall -I. -Wstrict-prototypes 
-fo
mit-frame-pointer   -DSTANDALONE_ZAPATA -c zaptel.c
In file included from /usr/include/asm/smp.h:18,
 from /usr/include/linux/smp.h:17,
 from /usr/include/linux/sched.h:23,
 from /usr/include/linux/module.h:10,
 from zaptel.c:44:
/usr/include/asm/mpspec.h:6:25: mach_mpspec.h: No such file or directory
In file included from /usr/include/asm/smp.h:18,
 from /usr/include/linux/smp.h:17,
 from /usr/include/linux/sched.h:23,
 from /usr/include/linux/module.h:10,
 from zaptel.c:44:
/usr/include/asm/mpspec.h:8: error: `MAX_MP_BUSSES' undeclared here (not in a function)
/usr/include/asm/mpspec.h:9: error: `MAX_MP_BUSSES' undeclared here (not in a function)
/usr/include/asm/mpspec.h:10: error: `MAX_MP_BUSSES' undeclared here (not in a 
function)
/usr/include/asm/mpspec.h:12: error: `MAX_MP_BUSSES' undeclared here (not in a 
function)
/usr/include/asm/mpspec.h:19: error: `MAX_APICS' undeclared here (not in a function)
/usr/include/asm/mpspec.h:20: error: `MAX_MP_BUSSES' undeclared here (not in a 
function)
/usr/include/asm/mpspec.h:20: error: conflicting types for `mp_bus_id_to_type'
/usr/include/asm/mpspec.h:8: error: previous declaration of `mp_bus_id_to_type'
/usr/include/asm/mpspec.h:22: error: `MAX_IRQ_SOURCES' undeclared here (not in a 
function)
/usr/include/asm/mpspec.h:24: error: `MAX_MP_BUSSES' undeclared here (not in a 
function)
/usr/include/asm/mpspec.h:24: error: conflicting types for `mp_bus_id_to_pci_bus'
/usr/include/asm/mpspec.h:12: error: previous declaration of `mp_bus_id_to_pci_bus'
/usr/include/asm/mpspec.h:54: error: `MAX_APICS' undeclared here (not in a function)
In file included from /usr/include/asm/smp.h:20,
 from /usr/include/linux/smp.h:17,
 from /usr/include/linux/sched.h:23,
 from /usr/include/linux/module.h:10,
 from zaptel.c:44:
/usr/include/asm/io_apic.h:120: error: `MAX_IRQ_SOURCES' undeclared here (not in a 
functio
n)
/usr/include/asm/io_apic.h:120: error: conflicting types for `mp_irqs'
/usr/include/asm/mpspec.h:22: error: previous declaration of `mp_irqs'
make[2]: *** [zaptel.o] Error 1
make[2]: Leaving directory `/usr/src/modules/zaptel'
make[1]: *** [binary-modules] Error 2
make[1]: Leaving directory `/usr/src/modules/zaptel'
make: *** [kdist_image] Error 2


I'm too tired to fight with this. If someone has the modules for this kernel,
can you tar them up and send them to me?

Tim

-- 

 Tim Sailer Coastal Internet, Inc.  
 Network and Systems Operations PO Box 726  
 http://www.buoy.comMoriches, NY 11955  
 [EMAIL PROTECTED]   (631) 399-2910  (888) 924-3728  

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Re: [Asterisk-Users] zaptel on Debian

2004-02-05 Thread Tim Sailer
On Thu, Feb 05, 2004 at 10:52:00PM -0500, Sean Cheesman wrote:
 do you have the kernel source installed?

Yup, in usr/src/linux .

Tim

-- 

 Tim Sailer Coastal Internet, Inc.  
 Network and Systems Operations PO Box 726  
 http://www.buoy.comMoriches, NY 11955  
 [EMAIL PROTECTED]   (631) 399-2910  (888) 924-3728  

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[Asterisk-Users] voip phones

2004-02-03 Thread Tim Sailer
What is the best inexpensive voip phone out there? I want to try
a few with *, but don't want to go broke while I'm just playing
around...

Tim

-- 

 Tim Sailer Coastal Internet, Inc.  
 Network and Systems Operations PO Box 726  
 http://www.buoy.comMoriches, NY 11955  
 [EMAIL PROTECTED]   (631) 399-2910  (888) 924-3728  

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[Asterisk-Users] diax softphone

2004-02-03 Thread Tim Sailer
I have my asterisk box on the public network. I have a winders box on the
public network, running diax. I have a winders box, same setup, behind
my Linux iptables firewall, on a private network. Both boxes cann register
iax2 to asterisk, and dial, but as soon as asterisk tries to do the native
bridge, the call fails. I've poked at this way too much today... what ports
do I need to open for this to work?

Tim

-- 

 Tim Sailer Coastal Internet, Inc.  
 Network and Systems Operations PO Box 726  
 http://www.buoy.comMoriches, NY 11955  
 [EMAIL PROTECTED]   (631) 399-2910  (888) 924-3728  

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