If I recall correctly Fast Busy basically means the destination number
is not busy (regular busy) but your provider most likely is either over
loaded, or has some other issues.
I've been getting busy signals with Nufone pretty regularly over the
last few days, and there email support is not
Olle E. Johansson wrote:
During the developer's conference call yesterday evening,
it was decided that we finally should release the much-awaited
Asterisk 2.0 Stable release, also called codename AAFJ.
AAFJ as in Asterisk April Fool's Joke?
Nice :)
___
Greetings *`s,
When initiating a call to an outside line (in this case a cellphone), *
starts playing the sound file before the call is answered, so when the
called party picks up, the message is already halfway thru, or
completely played out.
I have tried a few things to get around this, read
to dear martijn,
i made every possible change i can make
i have a TDM400P Zap card...
i had connected PSTN line to FXO Kewlstart at channel 1.
and analog phone to FXS Kewlstart at Channel 4.
i can hear continous ring tone when i hook up the receiver.
plz have a look at my confs.
my
Good one guys, for a minute you actually had me there. The give away is:
Rumours has it that one developer actually ported the
Erlang runtime and executed an Ericsson AXE switch within
Asterisk.
:-)
Nir S
On Fri, 2005-04-01 at 09:40 +0200, Olle E. Johansson wrote:
During the
I've got a couple of Fritz! chan_capi installs under my belt here in
Australia. I've elected to use the mISDN capi drivers over the AVM ones and
it works quite well except for broken DID support, and of course all the
limitations of using non Zaptel drivers.
Craig
- Original Message -
On Fri, 2005-04-01 at 09:40 +0200, Olle E. Johansson wrote:
During the developer's conference call yesterday evening,
it was decided that we finally should release the much-awaited
Asterisk 2.0 Stable release, also called codename AAFJ.
This relaese is based on the hidden cvs that has been
I don't know following has debated here or not, but is there in this world
following stuff:
A small, physically small box, like cable/DSL router, which comes with:
1) Ethernet port, 2) Console port, 3) CompactFlash or USB port, 4) memory
module port, like SODIMM
Box has built-in flash (256MB
Irakli Natsvlishvili wrote:
I don't know following has debated here or not, but is there in this
world following stuff:
I think you want a Soekris.
Cheers,
--
Ykoz Un Max - La VoIP en pr-pay!
Essayez gratuitement - 5 crdits offerts.
--- http://ykoz.net/voip/max ---
Hi, I have searched around as much as I can and can't find any good info to
help me try this problem. I have added a FXO card to my server and from
everything I can see, I have configured it right. *Obviously not*
Below is my config, any ideas on troubleshooting this ?
regards
Mark
*CLI
Olle E. Johansson wrote:
* New source code structure - C# and .net
Asterisk 2.0 was moved to a Microsoft platform due to the
demand for higher stability and a more secure foundation.
Therefore, the code was quickly moved to C# on the
.net
Thanks to every body for the solution.
It works fine!! :D
El Viernes, 1 de Abril de 2005 06:02, MF Hulber escribió:
The way it works with my provider is that although both numbers enter
the same context, each number will match its own extension. If I have
two numbers: 11 and
hi list
i know i am asking question out of the scope of this
list. actualy i cant find any place to ask question
like this. may be someone using ODBC with asterik.
actualling i want to make ODBC connection for asterisk
on my new fedora core 2. i have tried every thing.
tried rpms. compiled
Matthew Boehm [EMAIL PROTECTED] writes:
sipfriends is deprecated. You should have seen the warning. This tells
me that you did not infact read the wiki.
Just wanted to mention that this Wiki page
http://www.voip-info.org/wiki-Asterisk+RealTime
says the following:
RealTime
On Fri, 1 Apr 2005, Chris Hills wrote:
Olle E. Johansson wrote:
* New source code structure - C# and .net
Asterisk 2.0 was moved to a Microsoft platform due to the
demand for higher stability and a more secure foundation.
Therefore, the code was
Hi,
I've never used fxs/fxo modules, only E1 cards so I'm not entirely
sure. However, this log:
*CLI -- Starting simple switch on 'Zap/1-1'
-- Executing Dial(Zap/1-1, Zap/1/6998256) in new stack
-- Called 1/6998256
-- Zap/1/6998256-busy-1013475805 is busy
-- Hungup
On Mar 31, 2005 3:32 PM, Mimmus [EMAIL PROTECTED] wrote:
Hi,
I'm trying to compile channel_capi with current Asterisk CVS.
Asterisk compiled successfully but channel_capi (patched with all patches
needed, as suggested from some nice people on IRC #Asterisk) compilation
fails with:
Hi *,
we successfully integrated the eicon diva 4 bri card in our Asterisk system.
I can dial in to system and route to sip peers.
I tried to dial out with following configuratin without any luck:
extensions.conf:
exten = _5.,1,Dial(CAPI/@301:b${EXTEN})
capi.conf:
[general]
mode=immediate
In article [EMAIL PROTECTED],
Bruno Hertz [EMAIL PROTECTED] wrote:
Andrew Kohlsmith [EMAIL PROTECTED] writes:
Call it archaic if you like but mailing lists get the job done faster,
better and without all the bullshit that forums bring to the table.
It's not archaic but reasonable.
In article [EMAIL PROTECTED],
Tim Bass [EMAIL PROTECTED] wrote:
In addition, the lag time between posting a message to this list and
having it delivered is a joke. I posted this message below at 2:35 and
it was delivered to me, a new subscriber, an hour later.
My postings normally come back
In article [EMAIL PROTECTED],
Tim Bass [EMAIL PROTECTED] wrote:
I just joined this list yesterday,
And already you are telling the rest of us we're doing it all wrong.
Great.
Cheers
Tony
--
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] -
Asterisk 2.0 was moved to a Microsoft platform due to the
demand for higher stability and a more secure foundation.
It wasn't until I read this line that I knew it was a joke. I mean,
seriously, who associates Microsoft with stability and security? A fool,
that's who.
-Matthew
While this doesn't make sipfriends look deprecated, the link on that
If you are using a recent enough CVS version, it will tell you they are
deprecated when you start asterisk.
-Matthew
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I wrote:
In article [EMAIL PROTECTED],
Tim Bass [EMAIL PROTECTED] wrote:
In addition, the lag time between posting a message to this list and
having it delivered is a joke. I posted this message below at 2:35 and
it was delivered to me, a new subscriber, an hour later.
My postings
Eric Wieling aka ManxPower wrote:
Does anyone know what I need to do to use patlooptest? I have what I
think is a T-1 loopback plug in the card (1-port, TE110P), but I still
see a red alarm. Is this normal? I don't even know where to start for
this.
From Digium Support:
You will need to
On Thu, Mar 31, 2005 at 05:14:50PM -0500, Tim Bass wrote:
I use procmail and know very well how to manage email. All asterisk mail
goes to a folder,etc.
Your point...because a few people don't understand how to manage e-mail is
nonsense and shows why this list should be moderated.
Hello, Olle!
OEJAsterisk 2.0 was moved to a Microsoft platform due to the
OEJdemand for higher stability and a more secure foundation.
Nice...
I remember that about 10 years ago, when I was working in a daily newspaper
we wrote and article on April 1st on a first page about scientific
Thanks to Yves's commitment I was able to configure oh323 channel, cleared
the codec issue, registered to Gatekeeper, placed a call, but receive this
message on the console. What might be the problem?
Asterisk Ready.
*CLI -- Registered with gatekeeper '[EMAIL PROTECTED]'.
-- Executing
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part
de Kamran Ahmad
Envoyé : vendredi 1 avril 2005 11:08
À : asterisk-users@lists.digium.com
Objet : [Asterisk-Users] using unixODBC
hi list
i know i am asking question out of the scope of this list.
On Fri, April 1, 2005 11:31, Tony Mountifield said:
In article [EMAIL PROTECTED],
Bruno Hertz [EMAIL PROTECTED] wrote:
Andrew Kohlsmith [EMAIL PROTECTED] writes:
I totally agree. I run a local INN server and all the mailing lists I
subscribe to get turned locally into newsgroup postings in
On Fri, April 1, 2005 11:53, Tony Mountifield said:
That posting was back in my POP3 mailbox 2.5 minutes after I posted it.
I didn't see it for another 2 minutes because it arrived 5 seconds after
I polled the mailbox.
That's plenty fast enough for me. Perhaps there is a problem with your
Hi,
problem solved, I found somethind in this mailing list!
extensions.conf:
exten = 0237482,1,Dial,CAPI/@301:0237482,5,tr
Regards,
Kib
---BeginMessage---
Hi *,
we successfully integrated the eicon diva 4 bri card in our Asterisk system.
I can dial in to system and route to sip peers.
I tried to
Hi,
Has anyone got this card working with Asterisk? If so what kernel are
you using?
Currently I have installed Fedora Core 3 with the 2.6.10-1.770_FC
kernel
Chan Capi 0.3.5
Asterisk 1.0.7
The diva card is detected by linux and the chan capi is installed in
asterisk
When asterisk
Cenk Yabas wrote:
Thanks to Yves's commitment I was able to configure oh323 channel, cleared
the codec issue, registered to Gatekeeper, placed a call, but receive this
message on the console. What might be the problem?
Asterisk Ready.
*CLI -- Registered with gatekeeper '[EMAIL PROTECTED]'.
On Thu, 2005-03-31 at 11:05 -0500, Tim Bass wrote:
We installed one Digium TE410P in the PCIX slot and put the power cable back
on. The machine tried to come up, but the TE410P card flashed red lights
in all four ports and there was no video output, no motherboard beeps or
anything. This
I tried to dial out with following configuratin without any luck:
extensions.conf:
Can you help me or give me tips?
from the asterisk cli console
asterisk -r
type capi debug
place a call
and post your capi debug log
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Hello all,
There are any way for the queue agents in asterisk that they do not
need to login in the queue to begin recibing calls?
I want to use this queue for our recepcionist, with only one agent.
All that I want is,
1. The recepcionist do not need to make a login in the queue.
2. The
Hi,
try this:
mknod /dev/capi20 c 68 0
chmod 660 /dev/capi20
I have same configuration as you. It worked for me since yesterday.
Regards,
Kib
---BeginMessage---
Hi *,
we successfully integrated the eicon diva 4 bri card in our Asterisk system.
I can dial in to system and route to sip
Hi,
try this:
mknod /dev/capi20 c 68 0
chmod 660 /dev/capi20
I have same configuration as you. It worked for me since yesterday.
Regards,
Kib
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Asterisk-Users@lists.digium.com
Thanks, problem solved, I found somethind in this mailing list! Wrong
extensions.conf entry.
extensions.conf:
exten = 0237482,1,Dial,CAPI/@301:0237482,5,tr
?? But, what does ,5,tr mean ??
Regards,
Kib
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Kib Eki wrote:
Thanks, problem solved, I found somethind in this mailing list! Wrong
extensions.conf entry.
extensions.conf:
exten = 0237482,1,Dial,CAPI/@301:0237482,5,tr
?? But, what does ,5,tr mean ??
5 tells Asterisk to hang up if the call is not answered in 5 seconds.
t tells Asterisk to use
I do not claim/pretend to speak for everybody on this list, but I *do*
think that others that promote web forums should not do so either...
Hear hear!!
Let's let it die, folks; there are more pressing issues to deal with.
It's true that as long as the Digiumites hang out here, it's
On March 31, 2005 11:28 pm, Tim Bass wrote:
The discussion should not be laced with profanity, you should treat this
list and others like there are women on the list and try to be polite so
everyone is comfortable. Most professionals discuss matters in a way where
everyone is comfortable to
Hi list,
I try to explore making use of the variable ${DIALSTATUS} for
auto-answering purposes an similar.
On my asterisk box this does not work because ${DIALSTATUS} always
returns empty :(
Didn't find much in the web on this issue.
Can someone help?
regards Manuel
Folks,
I know that this is off-topic... but... I'm fulfilling a longtime dream
today... launching my on-line tea business. Teas grown /locally/, right
here in the midwest. I'm so excited that I'm telling everybody. :-)
(Naturally, we are using * and voip for our phone system.)
Please visit
I am looking to roll out an Asterisk VoIP implementation to our 200
employees.
So far I have hooked up the Asterisk box to our Elmeg PBX via a PRI
interface card and have that working, plus about 30 test users on Xlite
softphones.
Up til now all the configuration has been done by hand editing
I run asterisk on a soekris 4801, it works great. If you needed more
horsepower a via epia mini-itx would work too. I can't say enough how
much I like the soekris boxes though...
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Asterisk-Users@lists.digium.com
Francesco Peeters [EMAIL PROTECTED] writes:
On the other hand imagine a forum with subtopics like sipura, softphones,
zap or whatever. Wouldn't that maybe help to put some load off at least
the casual reader and poster seeking or giving advice for topics he/she
specialized in, and maybe even
Steve,
Looks much better now, although it didn't end the cpu load surges: they just
arrive less frequently (period of several minutes). There are some reports
about cpu spikes hitting your machine every few hours - when using G711.
Maybe these spikes are the same ones I see. When I change from
On Fri, April 1, 2005 15:10, Bruno Hertz said:
Francesco Peeters [EMAIL PROTECTED] writes:
On the other hand imagine a forum with subtopics like sipura,
softphones,
zap or whatever. Wouldn't that maybe help to put some load off at least
the casual reader and poster seeking or giving advice
On Mar 31, 2005 8:55 PM, Bruno Hertz [EMAIL PROTECTED] wrote:
Brian Capouch [EMAIL PROTECTED] writes:
Hmmm. I just got the latest beta build, which identifies itself as 1105d.
The keypad functionality is perfect.
Hmmm. Good for you. We were talking about sjphone, though :)
Regards,
I'm using Sipura SPA-841 and SPA-2000 phones and ATAs... Yes.. it's a
good assumption that I'm using asterisk, since I posted to this
list Umm *70 is there to turn call waiting on/off in the asterisk
database.
On Mar 31, 2005 7:57 PM, C F [EMAIL PROTECTED] wrote:
What phones? are you using
I read in the archives a number of discussions about livevoip, DID,
and DTMF not working.
However, no resolutions.
I just setup a livevoip DID and indeed the DTMF does not work.
The same asterisk context works via broadvoice and via
direct dialing in to the asterisk server via SIP.
Tim Bass [EMAIL PROTECTED] writes:
the excellent movie Vanilla Sky)...
Ahem. . . .
B#2.
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Hi Eric -
I'm having a problem with my Polycom phones and hoping someone else
has experienced the same thing: Outbound calls are fine, and inbound
calls originating from another SIP phone are fine, but inbound calls
to the Polycom phone from an IAX channel sound like you're talking to
a robot.
Hi
Was hoping someone could point me in the right
direction.
using asterisk cvs in various VOIP configurations
On a call when the loudness of transmit receive then
all receiving is null.
In practical terms this causes background noise (from
the other end)to stop when you are talking and
I am starting to use livevoip but when I configure they way they suggest, I
see errors.
[livevoip]
exten =_51NXXNXX,1,Dial(IAX2/myusername:[EMAIL PROTECTED]/${EXTEN:1})
snip
Heres the error message:
-- Executing Dial(IAX2/[EMAIL PROTECTED]:4569-6, 1000|15) in new stack
Mar
[EMAIL PROTECTED] (Tony Mountifield) writes:
I totally agree. I run a local INN server and all the mailing lists I
subscribe to get turned locally into newsgroup postings in moderated
groups. When I make a posting, it gets mailed out through a filter to the
moderator address, which is just
Take a look at the voicemail.conf.sample that comes with asterisk.
Inside you will see how to change the voicemail email message that is
cerated and add the phone number (and remove the name) for callerid.
-Matt
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Hi,
When I Dial(SIP/1SIP/2SIP/3) if any of these phones has a forward to
another destination (302: moved response) then the simultaneous ring
stops immediately and the incoming call goes to wherever the forward
points to.
We are using simultaneous ringing as a fallback when the receptionist
The lag time on SMTP list depends on three factors:
(1) The volume of the traffic;
(2) When you registered (if you registered two years ago, for example, you
receive mail in a large list before someone, say, who registered a month
ago);
(3) Various points of network congestion and delays.
I registered 1 week ago, and this message took 3 minutes to reach me.
Granted I'm in the UK so there is bound to be some strange effect causing
the speeding up of the message.
This topic is like a bad penny, it just won't go away.
Mark
-Original Message-
From: [EMAIL PROTECTED]
Wow!
Only 3 minutes delivery. That is much better than the one hour yesterday!
I am glad to see the list working a bit faster today :) One hour lag
yesterday was painfully slow.
I stand corrected on the lag time issue.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
On April 1, 2005 09:14 am, Tim Bass wrote:
(2) When you registered (if you registered two years ago, for example, you
receive mail in a large list before someone, say, who registered a month
ago);
You really have very little understanding of mailing list technology. Please,
do some basic
[EMAIL PROTECTED] wrote on 04/01/2005 12:36:07 AM:
I'm new to the VOIP world and need some advice. I currently have a
premium/ full functioned Panasonic PBX installed in my house/ small
office... and have some extra unused telco lines available on the
PBX. I'd like to use one of these
Jack,
Several voip clients can optionally suppress silent packets. If no voice is
detected, rtp packets are kept back. This saves bandwith but can disturb a
conversation (Hello John, still there?). Softphone XLite has this option
active by default.
Search for options like silence
Francesco Peeters [EMAIL PROTECTED] writes:
I think you took my Nah a itsy bit out of context there... ;-)
Hehe, I guess context is what your neurons link to - which, as you look at
them, might account for the itsyness :)
Totally OT:
I have been looking at this as a plugin for my own (non
Greetings *`s,
I have set up a call which constantly loops a pre-recorded message
waiting for the user to press a digit on their phone. At this point the
call is sent elsewhere in the dialplan.
But if the called party doesn`t press any buttons and hangs up, the
message carries on playing...the
Hi,
In response to:
http://lists.digium.com/pipermail/asterisk-users/2005-March/098214.html
quote
How about simply doing a Q.931 to SIGTRAN conversion module would
that not be simpler to implement?
/quote
Implementing this idea would help Asterisk become integrated with SS7 gateways
in a
When building an on-line community with robust software such as vBulletin,
it is easy to find someone who will create a customized hack that will do
as you suggest.
For example, posters who want to receive the full email message could, by
checking a box, get the entire message emailed to them.
On Friday 01 April 2005 02:40, Olle E. Johansson wrote:
During the developer's conference call yesterday evening,
it was decided that we finally should release the much-awaited
Asterisk 2.0 Stable release, also called codename AAFJ.
Olle, you better take a break!
For the rest of you, good
You missed:
(4) the server overload caused by people who don't like e-mail lists
telling the people who are perfectly happy with them they are fools.
Wait a moment. I've got it. All these pro-web-forum messages are 1st
April posts, aren't they? :-)
Regards,
Steve
Tim Bass wrote:
The lag time
[EMAIL PROTECTED] wrote on 04/01/2005 09:04:38 AM:
Take a look at the voicemail.conf.sample that comes with asterisk.
Inside you will see how to change the voicemail email message that is
cerated and add the phone number (and remove the name) for callerid.
Thanks. Once I found that it was
Dana Olson [EMAIL PROTECTED] writes:
I'm pretty sure that I used SJphone to check my VM. I'll test again.
But there is a new beta out on their site (and it's newer than the
Windows build). Maybe they added a dialpad?
Thanks, Dana, I know keypad dtmf worked with sjphone at some stage,
but at
what's the load on a soekris? how much can it handle?
On Apr 1, 2005 4:09 PM, Matt Ryanczak [EMAIL PROTECTED] wrote:
I run asterisk on a soekris 4801, it works great. If you needed more
horsepower a via epia mini-itx would work too. I can't say enough how
much I like the soekris boxes
Hi Matt -
how can I get all the phones to enable call waiting by default
instead
of having to dial *70 on each one to activate call waiting?
What phones? are you using Avaya, or Toshiba? Since you are posting to
this list I will guess you are using Asterisk, in which case I have no
clue why *70
Hi,
Thanks for the help I think it gives me a starting point. Also I do not
know to many nurses who can spoof a CID. A client of mine is trying to
find an easy way for nurses to record their time and just about anyone
can use a telephone. The client is not really interesting in getting
Mr. Underwood,
You might have noticed that I did not start this thread and simply am
agreeing with the original poster.
You might have noticed that I will not be shouted down and insulted to stop
agreeing with the original poster. In fact, if you don't like the thread,
do not respond to it.
I'm new to the VOIP world and need some advice. I currently have a premium/
full functioned Panasonic PBX installed in my house/ small office... and
have some extra unused telco lines available on the PBX. I'd like to use
one of these extra lines for VOIP into the PBX/ phone arrangement.
Hi,
I really regret bringing the subject up... I guess I hit some nerves so
please accept my apology. I have adapted to using the mailing list
(Mozilla Thunderbird with filters directing traffic a specific folder,
and threading) and it works, not ideally, but it works. The search of
goggle
Fine but don't mix up Swedish Danish beer ...
-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED] namens steve szmidt
Verzonden: vr 1/04/2005 16:39
Aan: Asterisk Users Mailing List - Non-Commercial Discussion
CC:
Onderwerp: Re:
Hey Bass,
Tim Bass wrote:
Mr. Underwood,
You might have noticed that I did not start this thread and simply am
agreeing with the original poster.
You might have noticed that I will not be shouted down and insulted to stop
agreeing with the original poster. In fact, if you don't like the thread,
First, trim your posts. Why include extra copies of the footer? Does it
help this discussion?
On Fri, Apr 01, 2005 at 02:17:52AM -0500, Tim Bass said:
I'm saying that as a long as long as Digium supports this dinosaur
technology in support of their community that is exactly what the
On Fri, 2005-04-01 at 09:39 -0500, steve szmidt wrote:
For the rest of you, good luck! You'll need it. I think finally the Danish
Elephant beer that is so strong has gone to Olle's head.
Oh yes Elephant beer, 30 years ago I drove from Stockholm to
Nortalia(sp?) after 3 or 4 bottles of that,
Hi,
Kib thanks for this - still no luck for me - can you send me more
details of what your setup is?
D.
__
e: [EMAIL PROTECTED]
t: +44 207 397 8451
m: +44 7966 926694
w: www.bright-talk.com
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-Original Message-
From: Scott Nelson [mailto:[EMAIL PROTECTED]
Perhaps you have an earlier hardware revision than I do; I also have
never rebooted the system. I have two TDM04Bs.
If so, they must have sold me old stock. I bought the cards less than two
months ago.
Hi !
I cant get my Zyxel Prestige 2002 (ATA) to answer the phone.
Outgoing calls i working perfect, but i get no incoming calls.
Everything sems normal on Asterix
This is my setup for P2002 (sip.conf):
[203]
type=friend
username=203
secret=302
callerid=Office 203 203
host=dynamic
context=dialout
Hello,
When performing a blind transfer to another extension
i.e.
originating extension = 103
transfer extension = 101
# 101
as soon as the extension rings, the handset initiating
(103) the transfer gives a busy tone (or congestion) once
the transfer extension rings
asterisk returns:
SIP/101-71ec
I've got an issue on the snoms, and I'm wondering if anyone has some
recent experience with it; I've contacted the one specific reference I
found to it in the list archives, and the person in question didn't
seem to find an answer (and snom doesn't appear to be finished moving
their offices
Hi,
we have released a brand new release of chan_misdn!
Here's a list of some new features:
* NT and TE mode
* PP and PMP mode
* BRI and PRI (with BNE1 and BN2E1 Cards)
* DTMF Detection in HW+mISDNdsp (much better than asterisks internal!)
* Display Messages to Phones (which support display msg)
*
It leaves the IQ level too low, and I don't mean this to be insulting,
but browsing through the Unix for Advanced and Expert Users I came
across one question about how to use tar, and the other advanced users
got confused that he was extracting it from a tape device, and not a
file, one about
David,
Zoa helped me, but were not working together. What's more, I cannot focus on
load tests too much: the setup I work on must be ready in may and starts small
scale. This system must be functional and reliable first and should scale well
later. The scaling part determines how long I am
Is there a way to specify the codec in the dial plan for an outbound
call using IAX?
Thanks,
Linn
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I have been using Asterisk for a couple of years now. I recently
upgraded to CVS HEAD (March 9, 2005). Independently (and perhaps this
is the problem) I also upgraded from Postfix 2.0.16 to 2.2.1.
Anyway, I just realized this morning that I have not been getting
emails when someone leaves me
See subject:
Does [EMAIL PROTECTED] support Dual-Processor installations? I didn't see
anything on the sourceforge page clarifying that. I suppose they could
leave out the SMP version of the Linux kernel to save space on the .iso?
I had trouble some time ago installing version .5 of [EMAIL
Hi Josh -
I've got an issue on the snoms, and I'm wondering if anyone has some
recent experience with it; I've contacted the one specific reference I
found to it in the list archives, and the person in question didn't
seem to find an answer (and snom doesn't appear to be finished moving
their
Manuel Schroeder wrote:
Hi list,
I try to explore making use of the variable ${DIALSTATUS} for
auto-answering purposes an similar.
On my asterisk box this does not work because ${DIALSTATUS} always
returns empty :(
Didn't find much in the web on this issue.
Can someone help?
regards Manuel
On Fri, 2005-04-01 at 17:53 +0300, Loucas Gatzoulis wrote:
what's the load on a soekris? how much can it handle?
A Soekris 4801 can easily support 20 - 25 SIP clients if they are all
running the same codec (I use ulaw), I know of others that have 20+ sip
clients and a t-1 card in the soekris for
You might try adding:
exten = h,1,Hangup
Chris Blake wrote:
Greetings *`s,
I have set up a call which constantly loops a pre-recorded message
waiting for the user to press a digit on their phone. At this point the
call is sent elsewhere in the dialplan.
But if the called party doesn`t press any
On Fri, 1 Apr 2005, Obihuan wrote:
Hello all,
There are any way for the queue agents in asterisk that they do not
need to login in the queue to begin recibing calls?
I want to use this queue for our recepcionist, with only one agent.
All that I want is,
1. The recepcionist do not need
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