Re: [asterisk-users] Re: Audiocodes GPL

2007-01-19 Thread Carla Schroder

  On 12/4/06, Yaniv Nizan [EMAIL PROTECTED] wrote:

 
  2) What does one go about doing to correct GPL violations? Perhaps
  someone has a generic legal letter that can be used in these
  situations?

This should help answer that question:

http://www.fsf.org/licensing/licenses/gpl-faq.html#ReportingViolation

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Re: [asterisk-users] MusicOnHold Files

2007-01-04 Thread Carla Schroder
On Thursday 04 January 2007 18:01, Forrest Beck wrote:
 I was just wondering what you all are doing for music on hold files
 for best quality.  I am not much of an expert on sound rates, bits,
 stereo, mono, tracks, and all that jazz.  Currently I am taking music
 from a CD (our campus jazz band has recorded a CD), converting to WAV,
 using Audacity to convert the stereo tracks into mono, drop the gain
 to -15db, then I use sox to convert to GSM and 8 bit (by typing #
 /usr/bin/sox file1.wav -r 8000 -c1 file2.gsm resample -ql )

 The audio while on hold is OK.  I wonder if there is a way to get
 better audio.  I noticed that asteriskguru.com has a audio conversion
 on their website.

 
Since telephones are so lo-fi, is there any reason to invest a lot of energy 
in hi-fi hold music? .wav is uncompressed and lossless, so it should sound 
better than .gsm which is a lossy format. But I doubt anyone can really tell 
the difference on a telephone. anyway, how long are you leaving your callers 
on hold? :)
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Re: [asterisk-users] Asterisk and MiniITX setups

2006-12-29 Thread Carla Schroder
On Friday 29 December 2006 14:46, Mark Greene wrote:
 How well do you think asterisk could run on a miniITX board like the ones
 linked below with the call volume of say a small doctors office or
 something?

 http://www.mini-box.com/s.nl/sc.8/category.15/.f

 - Mark

Have you looked at AstLinux? It's perfect for small form-factor boxes. The 
hardware should do what you want, provided you're not doing a lot of 
transcoding. 
http://www.astlinux.org/

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[asterisk-users] Answering Machine Detect (AMD) time values

2006-12-22 Thread Carla Schroder
Does anyone know what the time values in amd.conf are? Are they seconds, 
fractions of seconds, heartbeats, what?

 ;'initialSilence' is the maximum silence duration before the greeting
initial_silence = 25; Maximum silence duration before the greeting.

It doesn't say in amd.conf or at 
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+AMD

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Re: [asterisk-users] Agentcallbacklogin deprecation

2006-12-20 Thread Carla Schroder
Sooo what replaces it? Convoluted dialplans? AEL stuff? Agentcallbacklogin was 
simple and easy. Hardly anything in Asterisk is simple or easy. Maybe that's 
why it was removed. :)

On Wednesday 20 December 2006 2:21 pm, Lenz wrote:
 I would not say that * has been taken out of the realm of reasonable CC
 solutions, luckily there are still a lot of ways to configure * to meet
 the most diverse needs, but it's surely a fact that a very convenient and
 used approach being deprecated will be an annoyance for CC designers and
 will confuse the novice users. I am sure the developers who took this
 decision had the best intentions and the soundest reasons, but the result
 is IMHO a bit of unnecessary complexity.
 I happen to look everyday at a lot of queue_log files from a lot of
 different CCs, and I'd say that 90% of non-TrixBox users use
 Agentcallbacklogin as of today.
 l.

 On Wed, 20 Dec 2006 22:54:16 +0100, Jordan Novak

 [EMAIL PROTECTED] wrote:
   I agree with these fella's, this is a piss poor way of fixing it. I
  only know of one call center that used static agents, mostly because
  they were sold a peice of crap and they had no idea how to use it the
  other way. I think you will find the majority of call centers are
  callback centers. This decision has taken Asterisk out of the realm of
  providing reasonable call center solutions. VIVA ZAPATA!!!  Let's start
  a revolution!

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Re: [asterisk-users] Agentcallbacklogin deprecation

2006-12-20 Thread Carla Schroder
Ick. Don't WANT to do all that. no. Anyway, according to this it should 
still work in 1.4:
http://www.mail-archive.com/asterisk-users%40lists.digium.com/msg166873.html
Yes, AgentCallbackLogin is deprecated, but it will not be removed until after 
1.4. 

On Wednesday 20 December 2006 3:25 pm, Lenz wrote:
 You can use Local channels, plus some custom queue_log logging, unless you
 want fancy agent/extension association; in that case you'll have to have
 your fun with the dialplan :)
 l.



 On Wed, 20 Dec 2006 23:51:31 +0100, Carla Schroder [EMAIL PROTECTED]

 wrote:
  Sooo what replaces it? Convoluted dialplans? AEL stuff?
  Agentcallbacklogin was
  simple and easy. Hardly anything in Asterisk is simple or easy. Maybe
  that's
  why it was removed. :)
 
  On Wednesday 20 December 2006 2:21 pm, Lenz wrote:
  I would not say that * has been taken out of the realm of reasonable CC
  solutions, luckily there are still a lot of ways to configure * to meet
  the most diverse needs, but it's surely a fact that a very convenient
  and
  used approach being deprecated will be an annoyance for CC designers and
  will confuse the novice users. I am sure the developers who took this
  decision had the best intentions and the soundest reasons, but the
  result
  is IMHO a bit of unnecessary complexity.
  I happen to look everyday at a lot of queue_log files from a lot of
  different CCs, and I'd say that 90% of non-TrixBox users use
  Agentcallbacklogin as of today.
  l.


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Re: [asterisk-users] 2 devices using same sip account

2006-12-19 Thread Carla Schroder
Your phones only register once, when they first start up. Seems to me that 
having multiple phones on the same account is asking for trouble- why not set 
up multiple accounts in the usual way, and create a ring group for all the 
phones you want to use? Like this example that rings two phones at the same 
time:

exten = 100,1,Dial(SIP/101SIP/102,30,t)
exten = 100,2,VoiceMail([EMAIL PROTECTED])

There are all kinds of fancy variations on this theme, but the idea is the 
same: one user with many phones, one extension, one voicemail box.

On Tuesday 19 December 2006 8:18 am, rilawich ango wrote:
 It seems that Greg is truth for the case.  Asterisk doesn't care how
 many devices register to the same account as it is a feature of sip
 protocol (please let me know if there is a method to restrict it).

 In my case, I use a soft phone an hard phone using the same sip
 account information to register to the same asterisk.  Soft phone
 register first and then hard phone register later.  I dial the number
 and hard phone ring.  Then I disconnect hard phone and expect soft
 phone will be ring after a couple of time.  However, soft phone didn't
 ring as the call is failed.  I issue database showkey
 SIP/Registry/sip account in CLI.  It displays the information which
 belongs to hard phone.  That's mean asterisk will keep the information
 of hard phone even it is disconnected with ignoring the soft phone
 registration.  Does asterisk can be set to refresh its registry in a
 couple of time to remove the old registry record?

 On 12/19/06, Johansson Olle E [EMAIL PROTECTED] wrote:
  19 dec 2006 kl. 11.58 skrev Gregory Duchatelet:
   Hi,
  
   It seems that they both can make calls, but only one can receive
   call: the
   last registered...
  
   Greg
  
   Hi all,
 What will happen if 2 devices using the same set of sip account to
   connect to the same asterisk?  Do they both can make call?  Can they
   receive call as normal?
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  In Asterisk, you should only have one phone per account. We do not
  support
  multiple devices per account. The PBX core needs to know how many
  devices
  that we are calling each time we access it.
 
  /O

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Re: [asterisk-users] 2 devices using same sip account

2006-12-19 Thread Carla Schroder
Hmm, I don't know what happens when one of the lines is busy and none of the 
lines get answered. It's easy enough to test. If it doesn't go to voicemail, 
then perhaps this is what you want:

http://www.voip-info.org/wiki/view/Asterisk+tips+findme

On Tuesday 19 December 2006 9:58 am, René Enskat wrote:
 how isit possible to get the VM there when one line is busy?

 regards rene

 On Tue, 19 Dec 2006 09:48:01 -0800

   Carla Schroder [EMAIL PROTECTED] wrote:
  Your phones only register once, when they first start up. Seems to me
  that having multiple phones on the same account is asking for trouble-
  why not set
  up multiple accounts in the usual way, and create a ring group for all
  the phones you want to use? Like this example that rings two phones at
  the same time:
 
  exten = 100,1,Dial(SIP/101SIP/102,30,t)
  exten = 100,2,VoiceMail([EMAIL PROTECTED])
 
  There are all kinds of fancy variations on this theme, but the idea is
  the same: one user with many phones, one extension, one voicemail box.

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Re: [asterisk-users] Voicemail delivery

2006-12-18 Thread Carla Schroder
On Monday 18 December 2006 9:24 am, Ejay Hire wrote:
 Hi.

 How do I cause voicemails that land in one mailbox to be delivered to
 another?

 I.e. I have a incoming call extension that rings all the phones.  If it
 times out, the caller drops into the general mailbox.  I would like
 messages dropped in the general mailbox to fall into another users
 mailbox.

Maybe this is what you want:

[ringallphones]
exten = 100,1,Dial(SIP/200SIP/201SIP/202,30,t)
exten = 100,2,VoiceMail([EMAIL PROTECTED])

This specifies which voicemail box to use if no one answers.

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Re: [asterisk-users] Linux distro + Asterisk or Trixbox?

2006-12-17 Thread Carla Schroder
On Sunday 17 December 2006 10:47 pm, Andrew Joakimsen wrote:
 I've used Asterisk on a bunch of RH 7.3 machines which were then replaced
 by RHEL 4. It is very stable, my biggest compliant is that RHEL(or CentOS,
 which is a direct rip-off) uses outdated packages (Linux 2.4.x, Apache 1,
 Mysql 4, php 4, etc) and Linux 2.4.x requires certain USB hardware to use
 zaptel timing without a hardware card, so we have a bunch of these dual
 xeon machines with the wrong USB hardware and can only run MeetMe on the
 one with the t1 cards.

CentOS 4 was released May 2005 with a 2.6 kernel, Apache 2, and all other 
similarly current packages. The current kernel is 2.6.9-something. 

CentOS is a legal re-distribution of RHEL 4 rebuilt from source RPMs. Just 
like Pie Box, White Box, Tao, Lineox, and all the other Red Hat clones.

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Re: [asterisk-users] Linux distro + Asterisk or Trixbox?

2006-12-16 Thread Carla Schroder
On Saturday 16 December 2006 5:14 am, Phil Finkler wrote:
 Hey all,



 I've been doing a lot of playing, and a lot of reading, and it seems
 people are split as to whereas if they're running their favorite Linux
 distro and asterisk or Trixbox.  I'm getting closer to really looking at
 a production environment and I'm just looking for any opinions.  I'm
 really enjoying learning linux and asterisk, so initial ease of use
 isn't really a huge benefit to me.  In the end stability and
 upgradeability will be my main concerns.



Trixbox is HUGE. If you need all the bells and whistles- a MySQL backend, the 
CentOS operating system, AMP, SugarCRM, Festival, monitoring consoles, 
everything pointy-clicky, and on and on and on, then Trixbox is for you. It 
has some disadvantages. There is not a clear correlation between the 
graphical admin tools and the underlying text configuration files, so 
debugging problems is harder, and you have to know two ways of doing things. 
The documentation sucks rocks- there isn't any to speak of. When 
[EMAIL PROTECTED] changed the name to Trixbox, they moved to a new web site and 
didn't bring any of the help docs or forums with them. There is a book you 
can buy, 'Trixbox Made Easy'.

I think it's better to learn plain-vanilla Asterisk first. Then if you move on 
to some other implementation you'll be better prepared to understand what 
it's doing.

You might give AstLinux a try. It's a complete Linux distribution + Asterisk 
1.2.-something, but tiny, about 40 megabytes. No wasted bits. It has a 
nicely-organized Web GUI for those who like such. You can switch between the 
Web interface and editing the config files directly without getting in 
trouble. It runs on single-board computers and ordinary old PCs. It's my 
current fave, though I'm also running Asterisk 1.4/CentOS on a test box.

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Re: [asterisk-users] New installation CentOS 4 x86 or X86_64

2006-12-11 Thread Carla Schroder
On Sunday 10 December 2006 11:18 pm, Remco Barendse wrote:
 Hi list!

 I have to do a new bare metal installation of a box running Asterisk with
 bristuff or vzaphfc.

 The box will be used as a really lightly loaded file server and pbx.

 Any advise on which architecture I should use? The cpu is a 64 bit capable
 AMD (the box is running x86_64 now) but is still suffering from echo on
 the BRI lines.

 Should I go with the normal x86 or the 64 bit x86_64 arch.?


x86-32 isn't as fun as x86_64, but it's fewer hassles. With 64-bit systems 
you'll run into the odd app or driver that hasn't been ported to 64-bit 
architectures yet. You can run 32-bit code on 64-bit systems in chroots, 
which I think is a horrid pain, but some folks don't mind. :)

The big advantage of a 64-bit system is being able to handle huge amounts of 
memory (over 4 gigabytes) and gigantic files (up to 4 exabytes, wheee!), 
which doesn't really apply to an Asterisk server.
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Re: [asterisk-users] CLI History

2006-12-11 Thread Carla Schroder
On Monday 11 December 2006 9:31 am, Douglas Garstang wrote:
 What's wrong with the Asterisk CLI history? When I exit the CLI, and
 re-enter, the last command in the history always defaults to 'stop now'.
 This is very bad, and it's caused accidental shutdowns more than once.

 Connected to Asterisk 1.2.9.1 currently running on hera (pid = 17399)
 Verbosity is at least 3
 hera*CLI A
 No such command 'A' (type 'help' for help)
 hera*CLI B
 No such command 'B' (type 'help' for help)
 hera*CLI C
 No such command 'C' (type 'help' for help)
 hera*CLI D
 No such command 'D' (type 'help' for help)
 hera*CLI E
 No such command 'E' (type 'help' for help)
 hera*CLI
 [10:[EMAIL PROTECTED](pbx3):~]# asterisk -trv
 Asterisk 1.2.9.1, Copyright (C) 1999 - 2006 Digium, Inc. and others.
 Created by Mark Spencer [EMAIL PROTECTED]
 Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for
 details. This is free software, with components licensed under the GNU
 General Public License version 2 and other licenses; you are welcome to
 redistribute it under certain conditions. Type 'show license' for details.
 =
 Connected to Asterisk 1.2.9.1 currently running on hera (pid = 17399)
 Verbosity is at least 3
 hera*CLI stop now -- I pressed the UP arrow upon re-entering the console!


 I'm a bit confused by your example. What are A,B,C, etc? To exit the Asterisk 
console, I type 'exit'. Asterisk continues to run, as it should. To re-enter 
the console I use asterisk -rvvv.

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good Linux references (was: Re: [asterisk-users] Re: Switching from FreeBSD to Linux - which distro?)

2006-12-09 Thread Carla Schroder
On Friday 08 December 2006 14:03, John Novack wrote:
 David Thomas wrote:
 
  Note: a default install of CentOS installs a bunch of unnecessary
  services that you will want to turn off using chkconfig service_name
  off.
 
  David

 It MIGHT be useful for SOMEONE to specify what those unnecessary
 services are

 John Novack

That depends entirely on what services you need to have running. No one will 
hold being inexperienced against you- but unwilling to learn is something 
else. The world is crammed to the gills with excellent Linux system and 
network administration references. A good resource for anyone new to CentOS 
is the Red Hat manuals:
http://www.redhat.com/docs/

And of course as all good computer geeks know, go to the distribution's home 
page, which is http://www.centos.org/. Wikis, forums, mailing lists are all 
here.

Running a complex server like Asterisk has several interdependent parts: 
Asterisk itself, provisioning phones, operating system administration, and 
network administration. You already know about this here excellent list, and 
hopefully are aware of http://www.voip-info.org/wiki/, the home of all things 
Asterisk.

These are my fave resources for Linux system and network administration:

TCP/IP Network Administration, Third Edition
http://www.oreilly.com/catalog/tcp3/
TCP/IP is fundamental to VoIP and computer networking. If you don't understand 
TCP/IP everything else will remain mysterious

bash Quick Reference
http://www.oreilly.com/catalog/bashqr/
get up to speed quickly on the primary Linux command shell

Open Sources: Voices from the Open Source Revolution
http://www.oreilly.com/catalog/opensources/book/toc.html
Free to read online. My favorite inspirational book with essays by the movers 
and shakers of the free software movement

And of course, may I modestly point to my own book (see sig.) Though it's 
getting a bit old and needing an update.

Carla
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Re: [asterisk-users] Switching from FreeBSD to Linux - which distro?

2006-12-07 Thread Carla Schroder
On Wednesday 06 December 2006 20:12, Lacy Moore - Aspendora wrote:
 On 12/6/06, John Novack [EMAIL PROTECTED] wrote:
  Go get the ISO's, and remember to INSTALL EVERYTHING, then you won't run
  into some gotcha down the road where there is some missing file that
  needs to be put who knows where.

 Wow!  Are you sure about that?

Doesn't seem like an issue to me. yum install foo is easy, and I've always 
preferred servers that are as lean as possible, rather than all porky with 
unnecessary packages and services.

Someone else mentioned AstLinux, and it is very nice. About 40 megabytes. No 
lard at all.

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Re: [asterisk-users] Switching from FreeBSD to Linux - which distro?

2006-12-07 Thread Carla Schroder
On Thursday 07 December 2006 17:42, John Novack wrote:
 Carla Schroder wrote:
  On Wednesday 06 December 2006 20:12, Lacy Moore - Aspendora wrote:
  On 12/6/06, John Novack [EMAIL PROTECTED] wrote:
  Go get the ISO's, and remember to INSTALL EVERYTHING, then you won't
  run into some gotcha down the road where there is some missing file
  that needs to be put who knows where.
 
  Wow!  Are you sure about that?
 
  Doesn't seem like an issue to me. yum install foo is easy, and I've
  always preferred servers that are as lean as possible, rather than all
  porky with unnecessary packages and services.
 
  Someone else mentioned AstLinux, and it is very nice. About 40 megabytes.
  No lard at all.

 That may be true for you and those that know Linux and how to respond to
 a missing file because it wasn't initially installed.
 For those who don't practice Linux as a religion but simply want to use
 a telephony application, it works to install everything, and move on to
 learning Asterisk and all IT'S warts and gotchas,


You're saying it's a religion to understand how to administer the operating 
system on a server? What a novel concept. 

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Re: [asterisk-users] Switching from FreeBSD to Linux - which distro?

2006-12-05 Thread Carla Schroder
On Tuesday 05 December 2006 15:36, Phil Finkler wrote:
 Does there seem to be a popular Linux distro folks use specifically for
 Asterisk?  I'd like to move off of FreeBSD but I'm not too familiar with
 Linux distros.  In particular, I'm looking for a free, stable, well
 supported distro that has a friendly community.  Any advice appreciated.
 Sorry for asking a question that I'm sure has been asked thousands of
 times.



Debian is my fave, but for Asterisk I use CentOS. It's a free-of-cost clone of 
Red Hat Enterprise Linux, so it's very stable and reliable, and Asterisk runs 
great on it. Debian is good too. They have Asterisk packages, but they're 
generally a little bit old. Source installations work fine. Both have large, 
active developer and user communities.

Avoid Fedora- it's too much of a moving target, and Asterisk installations are 
always a nightmare. Ubuntu is the darling of the Linux world, but I've had 
problems with Asterisk on it too.

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Re: [asterisk-users] names of SIP aware firewalls

2006-11-05 Thread Carla Schroder
On Sunday 05 November 2006 13:54, Erick Perez wrote:
 Besides ranch networks and borderware, what other SIP aware firewalls
 for the SOHO/medium market exists?

Netfilter's SIP connection-tracking module is ready for prime time, and will 
be included in 2.6.18 Linux kernels. Early birds can patch older kernels and 
not wait. http://www.enterprisenetworkingplanet.com/netos/article.php/3638441

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Re: [asterisk-users] Polycom SP4000 ftp problem

2006-10-23 Thread Carla Schroder
On Monday 23 October 2006 15:51, Edwin Lam wrote:
 i recently bought an SP4000 conference phone but having problem
 provisioning it using ftp, every time it just hangs at
 Updating initial configuration... screen. when i switch it
 to tftp, it'll work fine. i though it was bootrom/firmware issue
 so i upgrade it to bootrom 3.2.2/sip 2.0.1 but it makes no
 difference. any thoughts?
 
 p.s. i'm using debian sarge proftpd 1.2.10 and the setting works
 fine w/ SP501 with bootrom 3.1.2/sip 1.6.3
 

Sooo...stick with tftp? :) Seriously, that's what it's for. tftp isn't really 
an FTP server; it uses a different protocol, and uses only a single port (UDP 
69). You can't use real FTP servers for this.

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[asterisk-users] make menuselect question- Module Embedding

2006-10-23 Thread Carla Schroder
What does option '11. Module Embedding' do in Asterisk 1.4? The default is 
none of them are selected:

 [ ] 1.  apps
 [ ] 2.  cdr
 [ ] 3.  channels
 [ ] 4.  codecs
 [ ] 5.  formats
 [ ] 6.  funcs
 [ ] 7.  pbx
 [ ] 8.  res


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Re: [asterisk-users] Polycom SP4000 ftp problem

2006-10-23 Thread Carla Schroder
On Monday 23 October 2006 17:38, Edwin Lam wrote:
 Re: [asterisk-users] Polycom SP4000 ftp problem
  From: Edwin Lam [EMAIL PROTECTED]
  To: asterisk-users@lists.digium.com
  
 Carla Schroder wrote:
 
  Sooo...stick with tftp? :) Seriously, that's what it's for. tftp isn't 
really 
  an FTP server; it uses a different protocol, and uses only a single port 
(UDP 
  69). You can't use real FTP servers for this.
 
 sure if there's a tftp server that can provide the security and flexibility
 of ftp server.

The difference between tftp and 'real' FTP servers is it does not ask for a 
login- that's why it's used for diskless clients and PXE net installs. ProFTP 
(and all other FTP servers) require a login authorization. This is usually 
invisible to the end-user on public FTP servers, but it's still there. So I'd 
look for how the phone authorizes itself to the ftp server.

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Re: [asterisk-users] identifying Eicon Diva Server V-4BRI-8M vs 4BRI-8M

2006-10-18 Thread Carla Schroder
On Wednesday 18 October 2006 04:52, Klaus Darilion wrote:
 
 Hi (Armin)!
 
 Does someone knows how to identify the type of the card? The delivery 
 note says it is a V-4BRI-8M, whereas lspci reports a 4BRI-8M.
 
 What is it really? Are there any Eicon tools to identify the card type?
 
 thanks
 klaus
 
 :0a:03.0 Network controller: Eicon Networks Corporation Diva Server 
 4BRI-8M Rev 2 (rev 01)
          Subsystem: Eico

Have you tried running the update-pciids command to make sure your pci-id 
database is current? It's part of the pciutils package.

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Re: [asterisk-users] 1.4 downgrade

2006-10-18 Thread Carla Schroder
On Wednesday 18 October 2006 14:31, Jason Walker wrote:
 I am having a bunch of issues with 1.4 and want to go back to 1.2 any 
 ideas on the best way I saw someone say apt-get remove will this work 
 for asterisk or do I need to do it for each libpri, addons, zaptel and 
 asterisk? 

That works only if you are running Debian or some flavor thereof like Ubuntu, 
and installed it with apt-get. Otherwise it's just source installs. 1.4 has a 
make-uninstall target:

[EMAIL PROTECTED] asterisk-1.4.0-beta2]# make uninststall
...
[stuff and more stuff]
...
 +- Asterisk Uninstall Complete -+
 + Asterisk binaries, sounds, man pages, +
 + headers, modules, and firmware builds,+
 + have all been uninstalled.+
 +   +
 + To remove ALL traces of Asterisk, +
 + including configuration, spool+
 + directories, and logs, run the following  +
 + command:  +
 +   +
 +make uninstall-all +
 +---+

I haven't looked in the other source directories, but it's easy enough to try 
out. You can always read the Makefiles to see what they can do.

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[asterisk-users] 1.4 gsm files changed??

2006-10-17 Thread Carla Schroder
I like to use zgsmplay to preview Asterisk's gsm files. Works like a charm in 
1.2. But something changed in 1.4- they play at the wrong speed, very slowly. 
They sound fine in the normal course of using the Asterisk server.

The file sizes seem wacko. The first example is from 1.2, the second from 1.4:

$ ls -l im-sorry.gsm
-rw-r--r--  1 1000 1000 1419 Dec  5  2005 im-sorry.gsm


$ ls -l im-sorry.gsm
-rw-r--r--  1 1000 1000 9240 Sep 20 09:36 im-sorry.gsm

All of them are several times larger in 1.4- is it supposed to be like this? 
Should I file a bug report? 

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Re: [asterisk-users] 1.4 gsm files changed??

2006-10-17 Thread Carla Schroder
On Tuesday 17 October 2006 18:58, Carla Schroder wrote:
 I like to use zgsmplay to preview Asterisk's gsm files. Works like a charm 
in 
 1.2. But something changed in 1.4- they play at the wrong speed, very 
slowly. 
 They sound fine in the normal course of using the Asterisk server.
 
 The file sizes seem wacko. The first example is from 1.2, the second from 
1.4:
 
 $ ls -l im-sorry.gsm
 -rw-r--r--  1 1000 1000 1419 Dec  5  2005 im-sorry.gsm
 
 
 $ ls -l im-sorry.gsm
 -rw-r--r--  1 1000 1000 9240 Sep 20 09:36 im-sorry.gsm
 
 All of them are several times larger in 1.4- is it supposed to be like this? 
 Should I file a bug report? 
 

I've narrowed it down to asterisk-extra-sounds-en-gsm-1.4.1.tar.gz. They're 
all way too big and play too slowly. I suppose a bit of resampling will take 
care of that until they're fixed in some future release.

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[asterisk-users] 1.4 beta voicemail warning

2006-10-16 Thread Carla Schroder
hey all,

I'm getting this warning on the console when I leave a voicemail on my test 
server:

[Oct 16 20:56:36] WARNING[3853]: app_voicemail.c:6552 vm_exec: Prefixing the 
mailbox with an option is deprecated ('[EMAIL PROTECTED]').
[Oct 16 20:56:36] WARNING[3853]: app_voicemail.c:6553 vm_exec: Please move all 
leading options to the second argument.

This is what voicemail.conf looks like:

[local-vm-users]
250 = 1234,User1 One
251 = 3456,User2 Two
252 = 4567,User3 Three

This is what extensions.conf says:

[local-users]

exten = 250,1,Dial(SIP/User1,10,rt)
exten = 250,2,VoiceMail([EMAIL PROTECTED])

It works fine despite the warnings. I've been trying to change the configs to 
make the warnings go away, but nothing I've tried works. Anyone know what to 
do?

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[asterisk-users] macros reference?

2006-10-16 Thread Carla Schroder
One more question- is there a reference somewheres for all the built-in 
Asterisk macros?

thanks!

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Re: [asterisk-users] macros reference?

2006-10-16 Thread Carla Schroder
On Monday 16 October 2006 21:20, Kristian Kielhofner wrote:
 Carla Schroder wrote:
  One more question- is there a reference somewheres for all the built-in 
  Asterisk macros?
  
  thanks!
  
 
 Carla,
 
   There aren't any built in Asterisk macros.  Are you asking about 
 functions?
 
 http://www.voip-info.org/wiki/index.php?page=Asterisk+functions
 

Yes, thank you!

Carla

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