Title: Message
Hi
1)Could anyone
please tell me how to simulate a commercial level PBX (handling about 300
simultaneous calls) using Asterix.
2)Assume thatI
have installed Asterix on a machine , now what hardware doI use for
supplying as input a large number of simultaneous calls?
From: Andrew Kohlsmith [EMAIL PROTECTED]
Either way a decision needs to be made. There's no magic fairy gonna come
down and wiggle her pretty lil' ass over the walls and you magically have
dual Cat5e to every desk and some great POE-injecting switches upstairs.
:-)
Those fairy's do actually
On Wed, 18 Aug 2004, Darryl Ross wrote:
Oh, so I how does Asterisk knows when to start dialing out the
numbers, if there are no rules?
Have a look at http://www.voip-info.org/wiki-Asterisk+Extension+Matching
It doesn't actually tell the whole picture. There are two ways to handle
On Tue, 17 Aug 2004, Francis Augusto Medeiros wrote:
Actually, we also have non-fixed phone numbers in Germany. I think this is
not weird, I think this is very good. And again, Asterisk supports this.
Oh, so I how does Asterisk knows when to start dialing out the
numbers, if there are no
On Wed, 18 Aug 2004, Damjan wrote:
Sory if this is offtopic or rude, but I guess some people here might
have the experience...
I'm looking for a product to connect my 30-some DECT users to the voip
(SIP) network.
Some kind of DECT base station that connects to ethernet and supports
the
On Tue, 17 Aug 2004, Steven Critchfield wrote:
Asterisk + E100P + a E1 channel bank.
Which services DECT phones exactly how?
Steve
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To
[On a general note, the original post should not have been sent to
asterisk-dev. That list is for development of the Asterisk code itself,
not solutions based on it. That topic is for asterisk-users.]
On Wed, 18 Aug 2004, Mayank Mishra wrote:
1)Could anyone please tell me how to simulate a
Hello
This Question maybe ask many times.
I'm From the Netherlands and looking for an sip/aix pstn provider in the
US.I asked wipphone, vonage but they don't work for europe.
I've visit many not finished website's about sip pstn
Can somebody recomed me any good sip/aix provider in the us.
Hi,
I'm running the latest CVS HEAD version of asterisk, and I'm experiencing
hangups during voice conversation. This happens quite regularely and
often.
The problem is in dsp.c, line 1235, where it says
accum /= len;
But `len', at this point, is 0, resulting in a SIGFPE. The routine
Hey All,
Thought I'd take a bash at trying to get Festival to work here on my lab
system with the aim of using it to create our IVR menu prompts. I've
spent most of the afternoon searching through the Wiki, the Festival
website and Google and I've got a couple of questions.
First one is that
Hi,
We have a case where we need of 16 x FXS, 12 x FXO and 1 x E1. To
do this using Digium products I need of 8 PCI slots. This is not
possible to be done in one computer and that's why I try to start
using TDMoE. Unfortunately all my tries are without success. The
network is crashed
Darryl Ross wrote:
Hey All,
Thought I'd take a bash at trying to get Festival to work here on my
lab system with the aim of using it to create our IVR menu prompts.
I've spent most of the afternoon searching through the Wiki, the
Festival website and Google and I've got a couple of questions.
The other way is to hand off the call to the pstn as soon as you know
that is where it is headed and just pass the digits to the pstn after
that. This is called overlap dialing and is how most users are used
to the pstn working. This works for at least the zap channels.
It works also for
I suggest you go the channel bank route.
On Wed, 18 Aug 2004 10:16:01 +0200
Miroslav Nachev [EMAIL PROTECTED] wrote:
Hi,
We have a case where we need of 16 x FXS, 12 x FXO and
1 x E1. To
do this using Digium products I need of 8 PCI slots. This
is not
possible to be done in one
Sebastian Sporleder wrote:
Darryl Ross wrote:
Assuming that the debian packages are not compatible, which version of
Festival do I need? The Wiki page mentioned above says to grab the
tarball of 1.4.3, which is no longer available from the website. Only
1.95 is available. Will that work? Does
By call retrieval, I mean this: when the phone rings on an extension
(incoming call), but I'm far from it, then, dialing a certain prefix
would make me pick up that call from the extension that's nearby me.
That should work via dialplan. In bri-stuff-0.1.0-RC4 is app_pickup, a
channel
I don't have a clue about MEGACO, but isn't this just another name for
MGCP ? If yes, then try chan_mgcp
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
Hi,
C18 I suggest you go the channel bank route.
Can you be more detailed? Any URL? What is this and how to do it?
On Wed, 18 Aug 2004 10:16:01 +0200
Miroslav Nachev [EMAIL PROTECTED] wrote:
Hi,
We have a case where we need of 16 x FXS, 12 x FXO and
1 x E1. To
do this using
Miroslav Nachev wrote:
Hi,
C18 I suggest you go the channel bank route.
Can you be more detailed? Any URL? What is this and how to do it?
You can start by looking at the WiKi pages:
http://www.voip-info.org/wiki-Asterisk+Hardware (under the Channel Bank
section)
Hi again,
after two hours of testing, the following patch seems to fix my problem,
it may not be the best way to do this, however.
It seems to be an i4l issue, so this problem should better be fixed in
chan_modem_i4l. Btw, I use a vanilla fedora core 2.0 kernel
(2.6.7-1.494.2.2). with one HFC
HelloI am using the asterisk of the new
version. Please give me some help toconfigure the IAXsoftphone to
IAXsoftphone with asterisk.Thanks in
advanceddipak
Sjaak Nabuurs [EMAIL PROTECTED] wrote:
This Question maybe ask many times.
I'm From the Netherlands and looking for an sip/aix pstn provider in the
US.I asked wipphone, vonage but they don't work for europe.
I've visit many not finished website's about sip pstn
Can somebody recomed me any
Hi,How to accepts the call and plays a voice message on the line without
billing the caller ? This may
be necessary for IVR applications that want to explain features of the
service offered .
Regards
_
MSN 8 with e-mail virus
Dipak wrote:
I am using the asterisk of the new version. Please give me some help to
configure the IAXsoftphone to IAXsoftphone with asterisk.
Can you at least give more details? What is not working? What is
working? What are you trying to do? Any samples of your extensions.conf
and iax.conf
Good day all
I want to know how to configure asterisk so that for instance if you
press *5 it will pickup any ringing(unanswered) calls.
My problem is this,at lunch time a bunch of people go out for lunch and
when a call comes in it just ring and go threw the whole step.
I want someone,whoever is
Hi all.
Can you reccomend a SIP / IAX provider in the Netherlands ?
I need a few Numbers, and of course cheap rates :)
/Regards Mike
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To
OK, but there are just FXS ports. What about FXO ports?
Miroslav Nachev wrote:
Hi,
C18 I suggest you go the channel bank route.
Can you be more detailed? Any URL? What is this and how to do it?
You can start by looking at the WiKi pages:
Miroslav Nachev wrote:
OK, but there are just FXS ports. What about FXO ports?
You CAN actually get channel banks in various different configurations -
FXO-only, FXS-only or any combination of both. Look at the WiKi for the
Adtran channel banks and follow the link from there to their page -
[EMAIL PROTECTED] wrote:
Can you reccomend a SIP / IAX provider in the Netherlands ?
I need a few Numbers, and of course cheap rates :)
We van provide SIP termination, send an email to
[EMAIL PROTECTED] about your needs.
--
Andreas SikkemaRits tele.com
Scheepmakersstraat 11
Jeff Borders [EMAIL PROTECTED] wrote:
I'm very interested in the Digium/Asterisk combination but need some
clarification. I would like to setup a SOHO for business and home use.
Scenario One:
I have one analog line, 4 analog telephones.
Do I need a TDM400P + 4 FXS modules (Green) + X100P?
Hi, i have a problem when starting asterisk. in fact, once audio module
is loaded, it produces very much noise on my speakers. i tried both alsa
and oss driver with the same results. does anyone have a solution? i
have a Mandrake 10 and a Fedora core 2.
thanks very much,
Bob
Hi there,
Does anyone on the list know what sort of power the Adtran Total Access
850 channel bank consumes? I'm trying to put together 10 of them and
need to know what sort of UPS should be hooked up to them. Client is
asking for 10 hours backup time... or should I just go with the 850's
I am trying it this evening. It is sitting next to my desk, but in
white.
Okay, I tried this phone.
From what I know so far, this is just another phone based on the PA168
chip from Centrality Comm, so it has it's pro's and con's. For example,
the ATCOM AT-323 is very similar.
Now I know
Anyone knows where can I find spandsp? Official site seems permanently
down...
TIA,
Simone.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
A photo of the phone is also at
Ahh, fast finger.
A photo of the phone is also at
http://www.yntx.com/en/
All in all, the phone is much nicer as the AT323. It's still annoying that
the Ethernet-Traffic LED is at the front, so you have some constant
blinking.
The 8 function buttons at the
Hi,
Since DID's are a topic of conversation at the moment... (and I'm in
the market..)
I'm looking for a DID termination in the UK (London), USA (Los Angeles)
and China (Beijing).
Does anyone presently use companies providing these have suggestions
on who to use? We need a quality service,
-- {Rene wrote:}
As an example: I set the gateway's telephone number both in
extensions.conf, oh323.conf and phones.conf but I don't want to change
the same data in three separate files whenever I want to set a different
telephone number for the gateway.
- {Darryl Ross wrote:}
I have no idea
Title: Message
My Requirements
are
1) I need to
simulate a commercial level PBX ( Handling about 250 simultaneous calls )
using Asterisk.
Is this possible?
If yes then what is the hardware I need to procure for such an
Installation.
Moreover would Asterisk be able to handle such a
On Tue, 17 Aug 2004 01:16:21 +0200, Patrick [EMAIL PROTECTED] wrote:
On Mon, 2004-08-16 at 22:13, Markus Engelbrecht wrote:
Hello,
so I decided to update to the latest CVS version of asterisk and of
chan_capi.
You are compiling the wrong version of chan_capi to get chan_capi to
work
On Wed, 18 Aug 2004 12:29:24 +0200, Kramer, R.D.J. [EMAIL PROTECTED] wrote:
Thanks for your suggestion, I like that idea. It works for
extensions.conf, where under [globals] I
#include vars.conf
and in vars.conf I set
MY_E164=311
and in the remainder of the extensions.conf file I can
On Wednesday 18 August 2004 07:14, Eran Gal wrote:
Does anyone know which channel banks work well with asterisk.
I've used the Carrier Access Access Bank I and the Carrier Access Adit600. I
*far* prefer the Adit600, even though it has an oddball form factor. (It's
about 2U tall but only
On Wednesday 18 August 2004 04:54, Altus Snyman wrote:
I want to know how to configure asterisk so that for instance if you
press *5 it will pickup any ringing(unanswered) calls.
Yup this can be done -- just make sure your extension is in a pickupgroup that
matches the incoming line's
On Wednesday 18 August 2004 04:45, steven louse wrote:
Hi,How to accepts the call and plays a voice message on the line without
billing the caller ? This may
be necessary for IVR applications that want to explain features of the
service offered .
With PRI you can do this, and that is more or
Andrew Yager wrote:
Hi,
Since DID's are a topic of conversation at the moment... (and I'm in the
market..)
Please take this kind of business questions to the asterisk-biz list.
/O
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
and in a vpb card?
Thanks
Altus
On Wed, 2004-08-18 at 13:50, Andrew Kohlsmith wrote:
On Wednesday 18 August 2004 04:54, Altus Snyman wrote:
I want to know how to configure asterisk so that for instance if you
press *5 it will pickup any ringing(unanswered) calls.
Yup this can be done --
Greg,
Yes, it helps quite a bit. It shows me where Comedian Mail spawns the
external app.
Do you have a copy of your SIP MWI script? I may be able to use it as a
starting point.
FWIW, I've been using my extensions.conf to set/unset MWI on phones
attached to Cisco Call Manager - it's a bit
Hi,
We try to start TDMoE but the result is that the Asterisk and the
Network are crashed.
Are there some successful stories with TDMoE? Any help will be very
useful.
Best Regards,
Miroslav Nachev
___
Asterisk-Users mailing list
On Wednesday 18 August 2004 07:51, Altus Snyman wrote:
and in a vpb card?
Pardon my ignorance, but what's a vpb card?
-A.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or
sorry,using the vpb.conf so card like voicetronix openline 4 card.
Sorry my bad
On Wed, 2004-08-18 at 14:32, Andrew Kohlsmith wrote:
On Wednesday 18 August 2004 07:51, Altus Snyman wrote:
and in a vpb card?
Pardon my ignorance, but what's a vpb card?
-A.
* Please DO NOT post the same message to two lists. We have divided the lists to be
able to stay focused and lesser the burden. You are not raising the chances of getting
a reply, you are instead annoying a lot of people. Most of the people on the -dev list
are reading all other lists.
* Please
Hi,
I want to setup an opensource system for voip and traditional pstn calls;
I have an adm with 15 outgoing lines that is connected to al alcatel box, I
want to try asterisk and connect a linux box with asterisk to the adm instead
of alcatel box;
I have read some documentation and seems that
The problem appears to be that a 7960/7940 running P003AM30, the load
shipped from the factory, cannot load a new load file that is more than
393216 bytes in size.
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_field_notice09
186a008009451b.shtml
My 7960 is hosed. FedEx just
Well,
I've temporary connected asterisk to a Cisco E1 (which is configured to use
QSIG). Cisco debugs show:
Aug 18 13:15:05.973: %CONTROLLER-5-UPDOWN: Controller E1 1/1, changed state
to up
Aug 18 13:15:05.989: ISDN Se1/1:15: TX - SABMEp c/r=1 sapi=0 tei=0
Aug 18 13:15:06.330: ISDN Se1/1:15: RX
Hi!
Hi,How to accepts the call and plays a voice message on the line without
billing the caller ? This may
be necessary for IVR applications that want to explain features of the
service offered .
Usually this is done on the network side of the setup. You should ask
your telco for this.
Chris
On Wednesday 18 August 2004 08:34, Altus Snyman wrote:
sorry,using the vpb.conf so card like voicetronix openline 4 card.
Sorry my bad
That I'm not sure -- I have never used a Voicetronix card. The
callgroup/pickupgroup stuff is in zapata.conf though, but I believe similar
mechanisms exist
Hi everybody,
can anyone show me a exemple config for call-back?
I need something like:
1) I call asterisk server from my cellular
2) asterisk hang up my call (on d-channel)
3) asterisk recall to my cellular and give me a PSTN tone, so
I can to pick up a call and to dial new phone number
Occasionally my /etc/hosts file gets corrupted. The IP address and the
host name switch positions with the host name to the left.
What this happens, my 7940 phones won't register. Fixing /etc/hosts
allows the phones to register.
Do any of you Linux gurus know who is corrupting the hosts
Andrew Kohlsmith wrote:
I use it here all the time and it works very well. I have my home # ring a
dummy line at work (it doesn't ring anywhere, just gets the call in to the
office asterisk server) and then when my IM tells me I have an incoming call
I can *8 it and receive calls to my number
Am Mi, 2004-08-18 um 15.36 schrieb Andrew Kohlsmith:
On Wednesday 18 August 2004 08:34, Altus Snyman wrote:
sorry,using the vpb.conf so card like voicetronix openline 4 card.
Sorry my bad
That I'm not sure -- I have never used a Voicetronix card. The
callgroup/pickupgroup stuff is in
Yes, I know this is lame, but my location limits me to using a dial-up ISP. I am
running asterisk with a T100P and a TDM400 card. I currently have dial-on-demand
setup on the same box, using diald and an external modem. To prevent DOD from trying
to dial out during an external call, I have
Hello All,
Currently my setup uses Xlite and Asterisk and i found that all the RTP
voice packets are transfered via the asterisk server from one xlite to
another. Is there any possibility that we can make all the RTP Packets to be
transfered directly between the two clients once the connection is
On Wednesday 18 August 2004 09:47, Jeff Roberts wrote:
Hey Andrew, what kind of extension logic,etc to you use to get the im to
tell you the call is there?
http://www.mixdown.ca/~andrew/astbot/
I've found that, depending on the version of Asterisk, you have to escape
${CALLERID} differently.
Hello!
Is it possible to run Asterisk as a SMS Service Center and does it work
over a directly connected ISDN (CAPI) interface card?
Did anyone already use Asterisk for that?
Roland Zagler
mailto:[EMAIL PROTECTED]
@fog smart partners
___
On Wed, 18 Aug 2004 08:49:49 -0400, Donald Hall
[EMAIL PROTECTED] wrote:
The problem appears to be that a 7960/7940 running P003AM30, the load
shipped from the factory, cannot load a new load file that is more than
393216 bytes in size.
Easily solved. Load an older version like P0S30203 on it,
Hey,
I've got an asterisk box- P4 2.8 GHz,
800 MHz FSB, 256 MB RAM, Fedora Core II
I have an IPDialog SipTone II with an
X100P.
I experience choppiness/ticking sounds when
talking on the phone. I originally thought it was the X100P card, however,
it happens checking voicemail which is
Dear all:
Doesanybody know is it possible to use the board TE405P with only one port configured as follow, or I have to use the 4 ports at the same time ?
Thanks,
Angel
ZAPTEL
span=1,1,0,ccs,hdb3bchan=1-15dchan=16bchan=17-31
ZAPATA
[channels]context=menu-general
switchtype=euroisdn
Hello All,
Currently my setup uses Xlite and Asterisk and i found that all the RTP
voice packets are transfered via the asterisk server from one xlite to
another. Is there any possibility that we can make all the RTP Packets to be
transfered directly between the two clients once the connection is
I've found app_sms which is supposed to do that. However, I never
managed to get it work. Every phone I tried refuses to communicate with
asterisk.
This was my (very basic)config:
exten = 1,1,Answer
exten = 1,2,Wait(1)
exten = 1,3,SMS(test,as)
exten = 1,4,HangUp
This is supposed to answer the
You can use as many ports as you want. Just define how many ports
(spans) you're gonna use near the beginning of /etc/zaptel.conf.
span=1
Good luck
Bruno
Angel Diaz wrote:
Dear all:
Does anybody know is it possible to use the board TE405P with only
one port configured as follow, or I
On Wednesday 18 August 2004 10:31, Angel Diaz wrote:
Does anybody know is it possible to use the board TE405P with only one
port configured as follow, or I have to use the 4 ports at the same time ?
The four ports are independent of one another with one exception: they all
share the same
OK, I'm going nuts here trying to correctly identify null values,
specifically when callerID info is not available.
FYI, I'm running Asterisk CVS-HEAD-08/17/04-13:08:53, and Bison 1.875a
(debian Sid).
A snippit of my dialplan looks like this:
exten = s,1,SetCIDNum(${CALLERIDNUM})
exten =
[EMAIL PROTECTED] wrote:
Yes, I know this is lame, but my location limits me to using a dial-up
ISP. I am running asterisk with a T100P and a TDM400 card. I currently
have dial-on-demand setup on the same box, using diald and an external
modem. To prevent DOD from trying to dial out during
Nicola Murino a écrit :
Hi,
Hello,
I want to setup an opensource system for voip and traditional pstn calls;
I have an adm with 15 outgoing lines that is connected to al alcatel box, I
want to try asterisk and connect a linux box with asterisk to the adm instead
of alcatel box;
I have read
Michael Welter a écrit :
Occasionally my /etc/hosts file gets corrupted. The IP address and
the host name switch positions with the host name to the left.
What this happens, my 7940 phones won't register. Fixing /etc/hosts
allows the phones to register.
Do any of you Linux gurus know who is
Simone Ricci a écrit :
I've found app_sms which is supposed to do that. However, I never
managed to get it work. Every phone I tried refuses to communicate
with asterisk.
This was my (very basic)config:
exten = 1,1,Answer
exten = 1,2,Wait(1)
exten = 1,3,SMS(test,as)
exten = 1,4,HangUp
replace 1
- Original Message -
From: Stefaan [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, August 17, 2004 11:23 PM
Subject: Re: [Asterisk-Users] New $89 VOIP phone
From: Andrew Kohlsmith [EMAIL PROTECTED]
Either way a decision needs to be made. There's no magic fairy gonna
come
I dont know if this is possible from your end, but couldn't you just
put the modem on another machine, and have its phone line pluged
into a station port, that way it would dial out and be just like any
other call?
I would thing that would be the best solution
John
On Wed, 18 Aug 2004
Some things I checked when I had a similar issue:
CPU usage
IRQ sharing
Network quality
- drop packets
- jitter
- latency
- QOS/priority
A lot of the network quality stuff boils down to network volume/busy and
path (routers/switches/bridges) or basically collisions and potential
Tenorio, Leandro wrote:
Just guessing, but 've you tried the to rename Sip_4602ap1_0.ebin to
appsip.ebin
I did. The problem turned out to be with my HTTP server. I switched
HTTP servers and everything is now running fine.
___
Asterisk-Users mailing
This was recently added as README.mp3:
* Asterisk MP3 Support
==
Asterisk supports mp3 playback for music on hold via the mpg123
program, available from www.mpg123.de.
The latest release of mpg123 is mpg123 0.59r.
The latest development release of mpg123 is mpg123 pre0.59s.
Tried, doesn't work. And onestly, I've not catched this. Where's the
difference?
Cheers,
Simone.
administrator tootai ha scritto:
Simone Ricci a écrit :
I've found app_sms which is supposed to do that. However, I never
managed to get it work. Every phone I tried refuses to communicate
with
Hi all,
I registered with sipgate.de and tried to put together a working config
on my * box (CVS-HEAD-08/16/04-14:02:58). My * box is on a LAN w/
private IP behind an Alcatel SpeedTouch 5.1 ADSL modem (so I am using
NAT). Does anyone have some pointers how to configure sip.conf to get *
to
Hey guys, I have run into one last issue before I do my full *
conversion this evening. I can't seem to get paging to work. I have the
chan_oss module loaded as per the wiki, and I have the following in my
dial plan
;here is our intercom
exten = 6000,1,Dial,console/dsp
when I dial it here is the
Another n00b question..
Realizing they will be all the same ext.
What is the maximum qty of phones one TDM400P FXS module will support
Or what would be the max REN alowable on that module
Again assuming north american usage etc...
Thanks
John B.
From: Chris Shaw
From: Andrew Kohlsmith [EMAIL PROTECTED]
2 ethernet connections by using all 4 pairs of the cable.
Put one at your desk, and one at your switch, et voila; 2 independent ethernet
connections
over one cable.
You could also do this without those splitters by
On Wed, 18 Aug 2004, Chris Shaw wrote:
You could also do this without those splitters by splitting 2 pairs of
wires
to 2 connectors on both side of the cable.
You're kidding right?
There's a reason why category 5 cable is twisted the way it is... to
eliminate or greatly reduce RF
1)
Who bought Pingtel's phone line?
2)
Anyone seen this chinese-made VoIP phone that supports 8 different
protocols?
http://www.telecom.globalsources.com/GeneralManager?language=endesign=cleanaction="">
Mike
:)
Andrew Kohlsmith wrote:
With PRI you can do this, and that is more or less exactly why it's allowed.
Just don't Answer() the call, just issue Playback() and Hangup().
Playback will implicitly answer the line unless you have a noanswer flag
as an argument
*CLI show application Playback
Jeremy
Hello Patrick,
Beside the correct NAT configuration on your Router you need the following
in the sip.conf:
register=sipgateid:password:@sipgate.de
[sipgateid]
disallow=all; Disallow all codecs
allow=alaw
allow=ulaw ; Allow codecs in order of preference
I can't say 100%, but I'm almost postive that I was running that version of software
on mine before I upgraded. I now testing 7.1 on it with no issues.
I took a network trace to figure out what it was doing. One other thing I did was
make a set of binaries without a complete filename, eg:
do a 'ps -ef | more' or 'ps -aux | more' and look at the processes that are listed to
see if there is something running that might be doing it.
Otherwise, I'd approach it by going through each of the startup scripts (rc#.d, etc.)
and then each application's startup scripts. A bit tedious,
Hello Jason,
No, in this case I only needed to remove the old source code completely and
make a new checkout. After that compiling works fine without changing the
make file.
Thanks,
Markus
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Im Auftrag von
Why not try 'lsof' to see what processes might have it open or might be
writing to it...
- Original Message -
From: Mark Woods [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, August 18, 2004 10:09 AM
Subject: Re: [Asterisk-Users] [OT] What's changing /etc/hosts?
do a 'ps -ef |
On Wednesday 18 August 2004 13:09, Mark Woods wrote:
do a 'ps -ef | more' or 'ps -aux | more' and look at the processes that are
listed to see if there is something running that might be doing it.
Otherwise, I'd approach it by going through each of the startup scripts
(rc#.d, etc.) and then
Sharing 2 100mbps ethernet with 1 Cat 5 cable, I do it left and right, I
wouldn't recomend it on a cable over 200 ft(60 mts), I cabled the building
where I live (family thing) where I have about 6 such cables all working
fine.
On cable length, I heard there's people sucessfully using 900ft
defiance wrote:
Hey guys, I have run into one last issue before I do my full *
conversion this evening. I can't seem to get paging to work. I have the
chan_oss module loaded as per the wiki, and I have the following in my
dial plan
;here is our intercom
exten = 6000,1,Dial,console/dsp
when I dial
On 11:15 AM 8/17/2004, Chris Modesitt wrote:
I have
a question about how Asterisk Parses the Dial Plan. To create a
hunt-group which would be the appropriate dial plan:
[CompanyABC]
exten = 722,1,Dial(SIP/801722,60,r)
exten = 722,102,Dial(SIP/8014361234,60,r)
exten =
Hi!
I have a problem compiling the zaphfc driver for my HFC-PCI cards. I use
Asterisks latest CVS and bri-stuff.0.1.0-RC4.
The install.sh compiles zaptel and libpri without problems. But when it
tries to compile qozap and zaphfc it show the following errors:
qozap.c:206: error: structure has
That should work on a 100% full duplex switched network.
With good enough quality cable, the atinuation should be ok.
(no need for amplifiers)
The 300 foot limit was more about issues with late collisions.
In environments that you have collisions (half duplex), first bit
transmitted must reach
Does anyone know of an alternate source for spandsp? opencall.org is
down and all the links returned by Google just point to the dead site.
Thanks
David Filion
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
1 - 100 of 160 matches
Mail list logo
