Richard wrote:
The default user name and password is a huge issue in some cases. For
example, hackers can get into the server, grab the configuration, program
their own phone and make free calls. Another example, if you have multiple
domains, then you want different username/passwd for each domain.
Paulo Adriano wrote:
Hi,
I m looking for some real help in getting Zaptel to run on a SUSE 9.1
linux 2.6.4-52-smp gcc 3.3.3.
Asterisk is running but now it s time to add digium fxs boards.
That's what I've got running on my boxes. What specifically are you
looking for? Required packages
On Wed, 2004-10-27 at 15:27, Jim Van Meggelen wrote:
> [EMAIL PROTECTED] wrote:
> > See http://www.websitemanagers.com.au/asterisk/
> > I have run/maintained www.deadcat.net for a long time (6 or
> > more years I
> > think) which is basically the same sort of thing for Big
> > Brother (cross platfo
Hi,
Il mer, 2004-10-27 alle 05:35, Anton Tinchev ha scritto:
> Will be there new card?
> I'm asking it, 'couse i'm going to buy 3-4 cards?
> Or i should wait for the new one?
the card is here already. our latest shipment
of E1 single span cards was of te110p
Is a new card, that does E1 or T1 (lik
[EMAIL PROTECTED] wrote:
> On Wed, 2004-10-27 at 13:37, Jim Van Meggelen wrote:
>> People will want to pay for your expertise because you wrote (or at
>> least contributed to) the base platform, or language, or
>> what-have-you. The more one contributes, the more their credibility
>> is established
Correction, with the real 1.0.2 (not the .9 or whatever got accidentally
released)
It works fine.
Check it out, look it up on the wiki.
Donny
-Original Message-
From: Donny Kavanagh
Sent: October 27, 2004 1:02 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE:
On Wed, 2004-10-27 at 06:50, Andrew Kohlsmith wrote:
> On October 26, 2004 04:15 pm, Richard Lyman wrote:
> > the issue i think is being discussed is when all participants can
> > talk. if you were simply retransmitting client side muted then
> > you could fit alot of listeners amoung one or more
On Wed, 2004-10-27 at 13:37, Jim Van Meggelen wrote:
> People will want to pay for your expertise because you wrote (or at
> least contributed to) the base platform, or language, or what-have-you.
> The more one contributes, the more their credibility is established --
> their services gain value.
App_conference worked well for me, but after upgading to 1.0.2 this
evening, it would no longer compile. I will take a closer look at it
soon. But it was much better then app_meetme the most important thing
being AGC. (automatic gain control)
-Original Message-
From: Kenneth Shaw [mailt
That's exactly what I need...going there now!!!
-Mark Halverson
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Tuesday, October 26, 2004 9:56 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Asterisk Intro for newbies
Hi all.
Just for more info. I had to put 6 filters on a line for a customer with a
X100P.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Stewart Nelson
Sent: Tuesday, October 26, 2004 7:49 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] X100P noise on ADS
Hi all...
Just got Asterisk to work well with a SIP phone (Budgetone 101) and
TDM400P Digium card. I'm really happy with the whole setup. Anyway, I
put together an article describing what worked for me. This is very
beginner stuff, so any newbies out there, please take a look I hope it
he
Put the same lines as mentioned below in your sip.conf file. You'll need
to add the lines for every phone you want to be in the pickup group.
You can then make different pickup groups for different teams etc.
This assumes that your phones are SIP based.
Now, what I want to have a line ring on al
Jim Van Meggelen wrote:
--> HUGE SNIP <--
We all remember Commodore, right?
--> HUGE SNIP <--
Sorry just a quick point...
There is still quite a large C64 community and someone has managed to
make a new one. This uses FPGAs and allows the main chip to be set up
from compact flash.
The project
What is the point of a password anyways? Just curious
Thanks,
Steve Totaro
[EMAIL PROTECTED]
www.totarotechnologies.com
- Original Message -
From: "Darren Wiebe" <[EMAIL PROTECTED]>
To: "Nahuel Alejandro Ramos" <[EMAIL PROTECTED]>; "Asterisk Users Mailing
List - Non-Commercial Discussi
People will want to pay for your expertise because you wrote (or at
least contributed to) the base platform, or language, or what-have-you.
The more one contributes, the more their credibility is established --
their services gain value. This holds true not only for individuals, but
for companies a
In our setup we tried to use a 4 Port T1 card in a Compaq
proliant. We had no issues with processor load or dropped
calls but major echo problems. I figured that since I was
using a PRI it was digital and should have no echo
issuesI was wrong. I had just as many problems with my
pri's as I had
On Tue, 26 Oct 2004 16:44:01 -0600, Michael Loftis <[EMAIL PROTECTED]> wrote:
> Anyone have a way to get/know if these phone support anything other than
> the default Chirp 1 ringer for these phones (like the 7940/7960 where you
> can load a fairly arbitrary number of ringers...)
>
> TIA
>
Unfor
I have setup a gentoo box with asterisk and I second these suggestions.
The latest portage tree has the latest release of *. However if you
plan on keeping up to date with CVS head, I suggest you for-go using
the portgage install, and use the source instead.
Ed Brady
Brian West wrote:
On Wednesday 27 October 2004 11:00, Matt Riddell wrote:
> Ronald Wiplinger wrote:
> > I am not sure how it is called, but I would like to pick up the ringing
> > phone of my co-worker on an Asterisk PBX system.
> >
> > Can I do that, and if how?
>
> Sure you can. In your /etc/asterisk/zapata.conf
Mark Halverson wrote:
> Thank you soo muchI can get it running now!!!
:-)
> So now comes the fun part...trying to understand the iax, sip and
> extensions .conf files.
The wiki (www.voip-info.org) is really good for this, they have examples
for pretty much every combination.
> Basically,
Hey Stewart,
Thanks for the info..
Funny, thing, since 1.0.2 came out today, I went ahead and upgraded to
that, now everything works. Must have been some bug with the CVS
version I was on..
If you're having broadvoice problems, upgrade.
Stewart Nelson <[EMAIL PROTECTED]> [2004-10-27 02:05:09
Anton Tinchev wrote:
Will be there new card?
I'm asking it, 'couse i'm going to buy 3-4 cards?
Or i should wait for the new one?
Please don't crosspost to both lists as most of the people on
asterisk-dev are on asterisk-users. I have already replied to your
other post.
BTW: for others the answe
Ronald Wiplinger wrote:
I am not sure how it is called, but I would like to pick up the ringing phone
of my co-worker on an Asterisk PBX system.
Can I do that, and if how?
Sure you can. In your /etc/asterisk/zapata.conf file you can add:
callgroup=1
pickupgroup=1
before your channel lines.
Then
Currently, astcc does not have builtin support to handle passwords
also. If somebody would like to code in the option would be cool. I've
though of doing it but it is pretty much on the bottom of my todo list
as I'm currently rather far behind.
Darren
Nahuel Alejandro Ramos wrote:
Thanks Stev
Jay Milk wrote:
Thanks for the help-offer, but I think the disadvantages of being a
linux-idiot and attempting a very ambitious install, such as Gentoo,
outweigh the few advantages which Gentoo may or may not award me. Since
this is a home-office PBX, I can afford a slightly less than optimal
kern
I am not sure how it is called, but I would like to pick up the ringing phone
of my co-worker on an Asterisk PBX system.
Can I do that, and if how?
bye
Ronald
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Will be there new card?
I'm asking it, 'couse i'm going to buy 3-4 cards?
Or i should wait for the new one?
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Harry McGregor wrote:
I would love to see a good cheap phone with 802.3af, but they have yet
to come out. We have been looking at the Zip 4x4, as it's about the
best of the lot. We tried to look at the UIP200, but it was not even
readily available.
The Polycom SoundPoint IP300 is around $130 and
>From the sample queues.conf:
member => Agent/@1 ; Any agent in group 1
member => Agent/:1,1 ; Any agent in group 1, wait for first
; available, but consider with penalty
Can anyone elaborate on "wait for first available" and a scenario in which
that might be used? I'm u
If the call goes through, is voice quality good and free of pops and clicks?
(esp for the receiving party)
I had a problem with a TDM card and moving it to a different slot cleared
the clicks and pops and distortions and was having the same problem dialing
out. I had figured out that both were r
Thank you soo muchI can get it running now!!!
So now comes the fun part...trying to understand the iax, sip and extensions
.conf files.
Basically, here is my setup:
- One SIP account with broadvoice and want to limit that to only ONE
outgoing call at a time
- One Freeworld DialUp Accoun
I am experiencing a rather strange problem.
exten => s,1,Answer
exten => s,2,Background(welcome)
... other auto attendant stuff ...
will result in the calling party not getting any audio (neither
ringing feedback nor IVR audio) if callling in from the PSTN on an
X100P.
However ...
exten => s,1,
You don't need to spend anything approaching $40,000 to use a Cisco router
for four PRIs.
In your evaluation, consider also that the router is more reliable, has a
worldwide service organization behind it, and let's a single PBX handle many
more calls when it's relieved of transcoding (put voicem
Hello again,
I sent out an email last night that indicated that Asterisk 1.0.2 was
available for download. It turns out that the script that Mark and I used
to make the release checked out the code from the v1-0_stable branch
instead of v1_0. The v1-0_stable branch was from a while before 1.0 wa
Hi,
I'm a Asterisk newb, but have you adjusted your txgain? If you adjust
this too far down, it can cause dropped numbers like that. Or if you
have not adjusted it, perhaps you need to increase the txgain to make
sure the telco is "hearing" your dialing.
Jeremy
On Tue, 26 Oct 2004 21:20:07 -0
Steve Underwood wrote:
Kevin Walsh wrote:
Brian McSpadden [EMAIL PROTECTED] wrote:
I would buy a $10 license of Digium's g.729 and do some testing with
that. From what people have told me, that "open source" g.729
implementation causes crashes, performs more poorly, and just isn't a
good idea. T
The default user name and password is a huge issue in some cases. For
example, hackers can get into the server, grab the configuration, program
their own phone and make free calls. Another example, if you have multiple
domains, then you want different username/passwd for each domain.
> -Origin
I'm looking to build an Asterisk system to place in front of my call
center switch. My plan is to eat up 4 PRIs in one office, send to my
other office and convert back to PRI for my legacy switch. Voice
quality is critical, would I be better off going with a Cisco AS5350
using hardware DSPs
When I dial out, from both my analog and my SIP phones, it fails roughly
80% of the time with various telco messages (eg., "The number you have
dialed..."). The messages take a good 30 seconds before I hear them,
which makes me think the telco isn't seeing enough digits. 20% of the
time, my calls
The hard disk on an old system with an ISA card just died and I
reloaded a more modern OS (Fedora Core 1) and asterisk, but I wonder
what I need to do to get ISA support working.
# /etc/init.d/zaptel start
Loading zaptel framework: [ OK ]
Loading zaptel hardware
Hi,
Is it possible to perform Call Deflection, if that's the correct
term, to forward incoming phone calls on an E100P to another remote
phone number? I've found support for the CAPI cards, but nothing
that seems to refer to the Zaptel side of things?
Unless I'm barking up the wrong tree...
I looking for some compatibility information about E1 channel banks working with *.
Some conf files will be great too.
Thanks
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I have tried another microfilter, the long cable and the cascaded
microfilter and all made no difference at all..
I dont think it is the microfilter or the internal house cabling.. Also
the fact that a standard analog phone doesn't do it also points to the
X100P..
I can't move the X100P to ano
This is sorta OT, but I'd like to thank you for the last line in your
comments on Gentoo. For some users (particularly beginners) the setup
time and effort of Gentoo is not worth the return. For others it is,
me being one. I love portage with a passion.After you do one or
two Gentoo installs
Steve Underwood [EMAIL PROTECTED] wrote:
> Kevin Walsh wrote:
> > Brian McSpadden [EMAIL PROTECTED] wrote:
> > > I would buy a $10 license of Digium's g.729 and do some testing with
> > > that. From what people have told me, that "open source" g.729
> > > implementation causes crashes, performs mor
Use something like ProFTPD or something that is "supported" under their
manual (These are better FTP daemons anyway).
The default username/pass is PlcmSpIp btw.
-Tim
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bob Knight
Sent: Tuesday, October 26, 2
Kristian Kielhofner wrote:
Richard wrote:
Hi Kristian,
I'd like to use ftp because of several advantages it has. For example,
ability to change the time stamp and reload the phone. But the default
password is a big issue. I'd like to change it but don't want to go to
each
phone and reset it. Any w
Mark Halverson wrote:
Thank you Brianonly one problem...how?
Do I use the RPM command or what...and from what directory?
Hint, newbie here ;-)
1) Select a mirror to download the RPM file from:
http://rpmseek.com/rpm-dl/zlib-devel-1.1.4-8.i386.html?hl=com&cs=zlib-devel:PN:0:0:0:0:558969
2) Once
Jeff,
I did a cut-n-paste of your configuration straight into my sip.conf,
updated the username and password. Still getting the same result as
before, audio in only one direction. Can can call between my local
SIP extensions fine, so I know my sipura box is working and configured
correctly.
I'
Mr. James W. Laferriere wrote:
Hello Brian , Why wouldn't 'make clean' do just that ?
Tia , JimL
Make clean cleans them from the buildroot. Brian is talking about
/usr/lib/asterisk/modules - those .so's need to go when going from CVS
-> 1.0.x
rm /usr/lib/asterisk/modules/*.so
--
Richard wrote:
Hi Kristian,
I'd like to use ftp because of several advantages it has. For example,
ability to change the time stamp and reload the phone. But the default
password is a big issue. I'd like to change it but don't want to go to each
phone and reset it. Any way to change it?
Thanks,
I
Thank you Brianonly one problem...how?
Do I use the RPM command or what...and from what directory?
Hint, newbie here ;-)
-Mark Halverson
1-800-698-5856
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian West
Sent: Tuesday, October 26, 2004 4:4
Try OpenNA
www.openna.com
Very secure (and fast) out of the box. Downright paranoid sometimes.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian West
Sent: Tuesday, October 26, 2004 5:12 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussio
Kevin Walsh wrote:
Brian McSpadden [EMAIL PROTECTED] wrote:
I would buy a $10 license of Digium's g.729 and do some testing with
that. From what people have told me, that "open source" g.729
implementation causes crashes, performs more poorly, and just isn't a
good idea. There are people that ha
Install zlib-devel.
bkw
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Mark Halverson
> Sent: Tuesday, October 26, 2004 6:43 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Should I be worried? Newbie Warning
>
> Double Warn
THIS IS WRONG:
exten => s,1,SetMusicOnHold,default
exten => s,2,responsetimeout,20
exten => s/_20,3,goto(9000,1)
exten => s/_973111,4,goto(9000,1)
exten => s,5,Dial(Zap/1&Zap/2&Zap/3&SIP/2002&SIP/2102,17,tr)
THIS IS CORRECT (example):
exten => s/_40,1,Goto(9000,1)
exten => s/_9
Double Warning - LINUX & ASTERISK newbie
I downloaded Red Hat 9.0 from www.linuxiso.org today and installed it
according to the pdf found at:
http://members.lycos.co.uk/wipe_out/asterisk/asterisk_rh9_install-v1.3.pdf
After doing the cvs checkout thing, I followed the instructions and changed
Think of the ex g/f logic as a 'optional' match. If you don't match at
three you'll never see 4, so you'll want to have multiple 3 entries, and
move your 4 entry to 3, and your 5 up to 4, then on 4 you'll need some sort
of 'catch' like an _NXXNXX entry.
--On Tuesday, October 26, 2004 13:55
I'm currently running an office (about 25 phones) of Cisco 7960G's
running SIP back to my asterisk box running 1.0.1. The asterisk box is
attached to the Telco via a PRI (via a T100P).
I'm getting complaints that the phone calls are cutting out on people
(both parties) at random intervals, for
> Who is going to accept the SIP calls, Asterisk ou SER?
Either one of them can accept/register SIP agents, although what I
have been reading SER is more scalable for large implementations.
> Who is going to redirect calls to PSTN? Asterisk or SER?
Only Asterisk can interface with PSTN using spec
Hi Kristian,
I'd like to use ftp because of several advantages it has. For example,
ability to change the time stamp and reload the phone. But the default
password is a big issue. I'd like to change it but don't want to go to each
phone and reset it. Any way to change it?
Thanks,
> -Origina
[EMAIL PROTECTED] wrote:
>>> What is with that thought?
>>
>> Technically you could probably make something like that work, but I
>> would think you'd scare away a lot of customers with such a scheme.
>> If they want to get locked into a
>> "you-don't-really-own-your-software-we-do" kind of licens
On Tue, 2004-10-26 at 16:42 -0400, Chris Stratton wrote:
> I have been working with FastAGI. I really like it and the idea. But I
> was curious why the decision was made to not follow a standard http
> protocol approach? If it had been http, I could then use all the
> conventional tools that alread
--On Tuesday, October 26, 2004 07:22 -0400 Steve Totaro
<[EMAIL PROTECTED]> wrote:
I am sure the problem could be fixed within minutes if you post the
console output.
I got pushed in the right direction, two part problem * accepting a SIP
INVITE transfer that it wasn't going to accept (thus the
Hello,
Wouldn't it be
good to have sipfriends peers included in "sip show peers" and "sip show users"?
And iaxfriends also.
Tomica
Crnek
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To
Matthew Boehm wrote:
Tried. No dice. What next?
You could try typing asterisk -vv
(i.e. no r)
It should say "Asterisk already running on...". Does it?
--
Cheers,
Matt Riddell
___
http://www.sineapps.com/news.php (Daily Asterisk News - html)
http:/
Hi,
I'm just trying to register our codecs but it keeps giving the following error.
[EMAIL PROTECTED]/usr/home#chmod 777 register
[EMAIL PROTECTED]/usr/home#./register
ELF binary type "0" not known.
./register: Exec format error. Binary file not executable.
Any ideas?
Cheers,
Sahil
_
Tried. No dice. What next?
Matthew
- Original Message -
From: "Nicolás Gudiño" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<[EMAIL PROTECTED]>
Sent: Tuesday, October 26, 2004 4:30 PM
Subject: Re: [Asterisk-Users] Can't connect even though its running.
Chris Bshaw wrote:
Hi
I have a Cisco 8 port E1 gateway module (x6608) in a Cisco 6509
Chassis. I would like if possible to configure Asterisk to use this
using MGCP
IIRC, Cisco E1 PRI MGCP gateways require a separate network socket to
relay the Q.931 signals from the gw to the call agen
On Mon, 2004-10-25 at 16:15, Reid A. Forrest wrote:
> I may be wrong, but from what I've seen so far, an FXS port will run you
> about $100/port anyway, plus the cost of the analog device. At this price, I
> can't see any reason not to dump the analog and go with a cheap VOIP device.
> Even the lo
Hello Brian , Why wouldn't 'make clean' do just that ?
Tia , JimL
On Tue, 26 Oct 2004 [EMAIL PROTECTED] wrote:
I did the trick, Wonderfull!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian West
Sent: dinsdag 26 oktober 2004 23
--On Monday, October 25, 2004 20:58 -0400 Robert Jackson
<[EMAIL PROTECTED]> wrote:
We had a similar problem using AgentCallBackLogin. We were
specifying a context to AgentCallbackLogin and that context
only had extensions defined for the agents' extensions.
Since * uses the current context for
Rob Fugina wrote:
Is it just me, or is the Dial timeout off by a factor of two?
This is not something that has appeared recently, but I finally
decided to do some timings and see if it was just my perception or
something...
With Dial(Blah/999,20), the phone will ring for 40 seconds. With a
timeout
Anyone have a way to get/know if these phone support anything other than
the default Chirp 1 ringer for these phones (like the 7940/7960 where you
can load a fairly arbitrary number of ringers...)
TIA
--
GPG/PGP --> 0xE736BD7E 5144 6A2D 977A 6651 DFBE 1462 E351 88B9 E736 BD7E
__
> Given that this is the case, I thought that someone else must have
> released an "open source G.729 codec".
You can't the patent issues will keep this from taking place.
bkw
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Brian McSpadden [EMAIL PROTECTED] wrote:
> On Tue, 26 Oct 2004 17:50:55 -0400, Kanuri, Seshu (Company IT)<[EMAIL PROTECTED]>
> wrote:
> He's just playing like he doesn't know what we're talking about. He
> knows good and well that there's no free open g.729 implementation. It
> is open source only
I have delete=yes and attach=yes. But my messages are not getting
deleted after they're sent. I'm running asterisk as root so it can't be
a permission issue. Any ideas?
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I have a system with VIA chipsets with one T400P (3FXS,1FXO) and 2 E100P (for
testing with a cross over cable between them), and the only problem I have is
when I do something like:
exten => 5700,1,Dial(Zap/g3/5701)
exten => 5701,1,Dial(Zap/g3/5702)
exten => 5702,1,Dial(Zap/g3/5703)
exten => 570
Em Dom 24 Out 2004 04:04, Benjamin on Asterisk Mailing Lists escreveu:
> > This is probably a good time to ask if there is any
> > planned support for a g729 binary for YDL and
> > G3/G4, etc. I would love to start playing with
> > apple hardware, YDL, and asterisk.
> > But I need that binary!
> I
On Tue, 26 Oct 2004 17:50:55 -0400, Kanuri, Seshu (Company IT)
<[EMAIL PROTECTED]> wrote:
> Kevin,
>
> Are you looking for references for an actual "implementation" by someone
> on Asterisk?
>
> I think those who replied to your post do not understand the meaning of
> "Where can I find the open s
Thomas Hupfeldt wrote:
- Original Message -
From: "Kristian Kielhofner"
10.x uses kernel 2.6. You should be able to use ztdummy much easier
than zaprtc. use make linux26 (I think) with it. I still am not sure
how you toasted your last install, but that MAY help... Actually I
don't remem
Is it just me, or is the Dial timeout off by a factor of two?
This is not something that has appeared recently, but I finally
decided to do some timings and see if it was just my perception or
something...
With Dial(Blah/999,20), the phone will ring for 40 seconds. With a
timeout value of 10, it
Kevin,
Are you looking for references for an actual "implementation" by someone
on Asterisk?
I think those who replied to your post do not understand the meaning of
"Where can I find the open source G.729 implementation?".
Call me stupid, but I too don't understand the difference. As far as I
re
I did the trick, Wonderfull!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian West
Sent: dinsdag 26 oktober 2004 23:21
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Asterisk 1.0.2
rm all the .so's and t
was my fault!!
I have repaired my error. The problem was on my sip.conf i would not
add reinvite=yes and canreinvite=yes.
Nahuel Ramos.
On Tue, 26 Oct 2004 18:14:57 -0300, Nahuel Alejandro Ramos
<[EMAIL PROTECTED]> wrote:
> Hi everyone,
> I am using astcc for my prepaid calling card and th
- Original Message -
From: "Kristian Kielhofner"
>
> 10.x uses kernel 2.6. You should be able to use ztdummy much easier
> than zaprtc. use make linux26 (I think) with it. I still am not sure
> how you toasted your last install, but that MAY help... Actually I
> don't remember you ever
> > (Mandrake 10 does a good job of locking down a box. Minus whatever You
> > install
> > and run that might be a liability.)
>
> RIIIGHT a box is only as secure as the admin of the box makes it... not how
> secure the distro is. I don't know one linux distro that does it right.
Part of the mis
Hi,
I wish to determine whether a call is original call or it was transferred by someone.
I need to do it within AGI script.
Can I consider the following statements true:
1. if agi_extension != agi_dnid then the call is a transferred call
2. the call was transferred by extension agi_extension
My
Hi,
I´m looking for some real help in getting Zaptel to run on a SUSE
9.1 linux 2.6.4-52-smp gcc 3.3.3.
Asterisk is running but now it´s time to add digium fxs boards.
Thanks in advance,
Paulo Adriano
Francisco Paulo AdrianoWaveLIS LDAMobile +351 91 870 87
98Office + 351 21 989
Title: Message
Try
resetting user settings from the phone menu.
Peter
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
RichardSent: Wednesday, 27 October 2004 7:24 AMTo:
[EMAIL PROTECTED]Subject: [Asterisk-Users] polycom IP
500/600
Richard wrote:
Hi,
A few questions about polycom IP 500/600.
. how do I reset everything to factory default? The combination of
4,6,8,* seems only reset the network setup. Other settings, e.g. time
server, SIP configure are still there.
. is a way to push ftp username/password via dhcp? I’d like
What is the best guide for the following situation?
When a client dials a long distance # ( ie 5554441212) I
want Asterisk to use my channel 4 on my TDM400 to dial. But , I want it to dial
our local long distance exchange first.
I have it dialing fine , I just need to know how to ha
Thomas Hupfeldt wrote:
The reason to why i sad that i need to install it from scratch, was that i
didnt knew to the rescue function on the cd's.
Do any of you know, wheter it would help to installe the newest mandrake
10.1 instead of the v. 9.2 ?
And as I said before, im a really newbe to linux, an
Hi,
On Tue, 26 Oct 2004 14:01:14 -0500, Matthew Boehm <[EMAIL PROTECTED]> wrote:
> -r and -R are the same thing except -R tries to reconnect if dropped.
>
> regarless, -r didn't work either.
su root and try again...
--
Nicolás Gudiño
Buenos Aires - Argentina
___
Josh Krueger [EMAIL PROTECTED] lazily top-posted:
> Unlike some here, I wont say "Wiki" or "Google".
>
> The wiki page that references it is:
> http://www.voip-info.org/tiki-index.php?page=Asterisk%20G.729%20Licensing.
> and the actual site is: http://www.readytechnology.co.uk/open/g729/
>
I'm aw
Title: polycom IP 500/600
Hi,
A few questions about polycom IP 500/600.
. how do I reset everything to factory default? The combination of 4,6,8,* seems only reset the network setup. Other settings, e.g. time server, SIP configure are still there.
. is a way to push ftp username/password v
rm all the .so's and try again.
bkw
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
> Sent: Tuesday, October 26, 2004 4:15 PM
> To: [EMAIL PROTECTED]
> Subject: RE: [Asterisk-Users] Asterisk 1.0.2
>
> Hello
>
> I
- Original Message -
From: "Kristian Kielhofner"
> Dave Cotton wrote:
>
> > If I read it right that's what he did, the Mandrake CDs have a rescue
> > kernel which could well be totally different once running it let's you
> > straighten out problems.
I used the cd to rescue my mandrake. :
Hi everyone,
I am using astcc for my prepaid calling card and the billing goes
perfect, but when I make a call I do not hear any voice. The strange
thing is that when I press a number on my Xten, on my ATA I hear a
simple "beep" for all the numbers. I have tryed to change the codecs
but I still h
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