Done something similar in a different application, but * should handle
it --
In my case, I took a crystalfontz LCD, type 633, and used two of the
four fan-outputs to drive two 12V relays. As a nice extra, you get
temperature capabilities thrown in, so you can monitor your set-up. The
LCD runs
Guys..
Im new to asterisk but is it possible to simulate a dialtone for example, in
other PBX when you pick up the phone you can hear a certain dialup, which is
the PBX dialtone, and when you hit 9, you can hear the PSTN dialtone, is
this possible?
Try using ALSA with asterisk. Edit your /etc/asterisk/modules.conf
And comment and uncomment lines to leave as:
; Load either OSS or ALSA, not both
; By default, load OSS only (automatically) and do not load ALSA
;
;noload = chan_alsa.so
noload = chan_oss.so
i hope this help
Ariel Pablo
hi all,
I have an SIP softphone (kphone on FreeBSD) and an IAX2 client (FireFly
on Windows) trying to call each other. When the SIP client calls IAX
client the call gets connected but the SIP client cannot hear any voice.
the IAX client can hear SIP clients voice very clearly. When the IAX
Yes, this is basically the default.
Jon.
On Sunday 20 February 2005 02:20 am, Anton Krall wrote:
Guys..
Im new to asterisk but is it possible to simulate a dialtone for example,
in other PBX when you pick up the phone you can hear a certain dialup,
which is the PBX dialtone, and when you
On Sun, 20 Feb 2005, Anton Krall wrote:
Im new to asterisk but is it possible to simulate a dialtone for example, in
other PBX when you pick up the phone you can hear a certain dialup, which is
the PBX dialtone, and when you hit 9, you can hear the PSTN dialtone, is
this possible?
I'm not
Hello,
I just started using asterisk, and have a question. I have setup two
asterisk servers, A and B. A has a Digium TDM400 11B card (1 FXO and 1
FSX modules) and is connected to the PSTN. B has same, but is NOT
connected to PSTN. I want to configure B to call A via iaxtel, and
connect to the
Thx Ariel, Ill try that.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ariel Pablo
Klein
Sent: Domingo, 20 de Febrero de 2005 02:38 a.m.
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Soundcard problems?
How do you make the page
http://hostname/cgi-bin/astcc-admin/astcc-admin.cgi
secure ? ,
so that only the person administering the calling cards can see the page
and make changes to the calling cards, I was thinking of using .htaccess
to restrict the access to the page by requiring a password,
I think it would be your last suggestion.. When I pickup the phone I hear a
tone, the sip phone box tone... Then I hit 9, no tones :) and enter the
whole phone number and it starts to ring on the other side.. So no outside
dialtone get heard ever.. I was wondering if it could be possible to make
Anton Krall wrote:
I think it would be your last suggestion.. When I pickup the phone I hear a
tone, the sip phone box tone... Then I hit 9, no tones :) and enter the
whole phone number and it starts to ring on the other side.. So no outside
dialtone get heard ever.. I was wondering if it could be
Anton Krall wrote:
I think it would be your last suggestion.. When I pickup the phone I hear a
tone, the sip phone box tone... Then I hit 9, no tones :) and enter the
whole phone number and it starts to ring on the other side.. So no outside
dialtone get heard ever.. I was wondering if it
Very friggen cool, that you very much for the information it looks like
it will do the job nicely.
What did you use in your extensions list to activate the relay?
James
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jay Milk
Sent: Sunday, 20
On Sun, 20 Feb 2005, Duane wrote:
Anton Krall wrote:
I think it would be your last suggestion.. When I pickup the phone I hear a
tone, the sip phone box tone... Then I hit 9, no tones :) and enter the
whole phone number and it starts to ring on the other side.. So no outside
dialtone get
On Sun, February 20, 2005 21:56, Peter Svensson said:
Or have the 9 dial an outside line and get the external dialtone.
Which will only work if you're actually sending the call to an outside
line...
--
Best regards,
Duane
http://www.cacert.org - Free Security Certificates
Title: Message
All,I have been trying to get CAPI4Linux
working on my machine and being frank am failing miserably! I am looking for any
help available, I am real newbie (so please be gentle) and choose to run
Mandrake 9.2 as it appears quite friendly (or so I thought!).
I have been
I am using Underwood's fax system for fax on demand and it's very cool.
I am planning to do the following and I would like to know if it's
possible before putting my hands on it.
For a specific application,
I want to dialout thousands of numbers searching for fax machines.
If somebody takes
Easy as piece of cake.
Remove ignorepat=9
add:
exten = 9,1,DISA(no-password|my_outbound_context)
[my_outbound_context]
exten = NXX, 1, blah-blah-blah
All the Best!
Sergey.
Peter Svensson wrote:
On Sun, 20 Feb 2005, Anton Krall wrote:
Im new to asterisk but is it possible to simulate a
Hello,
On Sun, 20 Feb 2005, Isamar Maia wrote:
For a specific application,
I want to dialout thousands of numbers searching for fax machines.
If somebody takes the call(voice), I would flag that number as bad in the
DB. If it's a voice only answer machine, I would flag that number also as
Title: Message
So
true.
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ed
BradySent: Saturday, February 19, 2005 10:50 PMTo:
Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re:
[Asterisk-Users] This is
Actually, it was requested to me to build a fax number database.
The real purpose is unknown. I am an IT guy, not marketing guy.
Isamar
On Sun, 20 Feb 2005, Torsten Krueger wrote:
Hello,
On Sun, 20 Feb 2005, Isamar Maia wrote:
For a specific application,
I want to dialout thousands of
On February 20, 2005 08:30 am, Isamar Maia wrote:
I want to dialout thousands of numbers searching for fax machines.
You are an evil, evil man. Worse than the goddamned telemarketers, IMO.
-A.
___
Asterisk-Users mailing list
On February 20, 2005 01:41 am, Michael Giagnocavo wrote:
Well, sure, if you want to spend 8x the amount, yea, it's going to be a
much nicer setup.
Show me a TDM404P for $100. Now show me a system with two of them working
reliably and repeatably.
-A.
Ok. I will be burned in fire.. :-)
Or better.. I won't go to the heaven...
Isamar
On Sun, 20 Feb 2005, Andrew Kohlsmith wrote:
On February 20, 2005 08:30 am, Isamar Maia wrote:
I want to dialout thousands of numbers searching for fax machines.
You are an evil, evil man. Worse than the
Sorry, I understood the O.P. already had the hardware bought and installed
and simply wanted to throw on an extra line.
-Michael
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Kohlsmith
Sent: Sunday, February 20, 2005 8:14 AM
To: 'Asterisk Users
On February 20, 2005 09:25 am, Michael Giagnocavo wrote:
Sorry, I understood the O.P. already had the hardware bought and installed
and simply wanted to throw on an extra line.
You understood correctly; But again even a TDM401P is $133 on Digium's site.
Considering you could probably get 60%
On February 20, 2005 09:16 am, Isamar Maia wrote:
Ok. I will be burned in fire.. :-)
Or better.. I won't go to the heaven...
heh. Either way I'm pretty sure you'll be on your own to write this kind of
app. Personally I think you'd be FAR better off taking an electronic
phonebook and
-Original Message-
From: Andrew Kohlsmith
On February 20, 2005 09:25 am, Michael Giagnocavo wrote:
Sorry, I understood the O.P. already had the hardware bought and
installed
and simply wanted to throw on an extra line.
You understood correctly; But again even a TDM401P is $133 on
Some of my clients of hosted PBX service want to use it for callback
when they cannot use the ATA.
This is the scenario
1. Asterisk calls Party A at numA.
2. When A picks up the phone, he hears the announcement to enter the
destination number, numB. He enters numB
3. Asterisk Dials numB and
This is what I tryied on last Tuesday. It ran fine until yesterday (4
days) then asterisk stopped re-registering again. A sip reload fixed
the problem and asterisk now re-registers happily again. I'm just unsure
for how long ...
Stefan Gofferje wrote:
Stefan Gofferje schrieb:
Hi folks,
I'm
I just finnished a setup where I had the follwoing:
35 SIP phones.
2 TDM400 Cards with 8 FXO modules.
An account by a VOIP provider for incoming/outgoing calls.
Using private lines we connected 4 locations using DSL and T1.
In each location we have a Bogen PCM 2000 Paging module
A previous poster mentioned the same thing, with no response:
http://lists.digium.com/pipermail/asterisk-users/2004-
December/080161.html
Fresh asterisk 1.0.5 install on FC3, started with make samples,
nothing fancy. It's so bland, I'm surprised the list isn't full of
people having the same
[EMAIL PROTECTED] wrote:
If you have a TDM card already, buying a T1, channelbank,
etc. to add a few lines is the stupidest thing I've heard of today.
Not necessarily stupid, but certainly expensive.
Have you looked into buying some cheap multiport ATAs? 2 port
SIP/IAX2 ATA should be around
On 20 Feb 2005, at 16:22, Jim Van Meggelen wrote:
[EMAIL PROTECTED] wrote:
If you have a TDM card already, buying a T1, channelbank,
etc. to add a few lines is the stupidest thing I've heard of today.
Not necessarily stupid, but certainly expensive.
Have you looked into buying some cheap multiport
[EMAIL PROTECTED] wrote:
On February 20, 2005 09:25 am, Michael Giagnocavo wrote:
Sorry, I understood the O.P. already had the hardware bought and
installed and simply wanted to throw on an extra line.
You understood correctly;
Uh, nope. I've been unclear. It was a purely hypothetical
[EMAIL PROTECTED] wrote:
On 20 Feb 2005, at 16:22, Jim Van Meggelen wrote:
[EMAIL PROTECTED] wrote:
If you have a TDM card already, buying a T1, channelbank, etc. to
add a few lines is the stupidest thing I've heard of today.
Not necessarily stupid, but certainly expensive.
Have you
I've never set up an mgcp device before. I have an Adtran IAD with the
MGCP firmware on it. I have it configured in mgcp.conf like this:
[general]
port = 2427
bindaddr = 0.0.0.0
[adtran]
host = 192.168.2.2
context = default
canreinvite = no
line = aaln/1
line = aaln/2
The device is configured
Well, I appreciate everyone's input, and I'll give the matter some more
thought.
Just so no one stays up at night worrying, this is not a situation I am
facing, it is simply a hypothetical scenario.
As with so many things, there is always a trade-off between economy and
functionality. The Adit
Jim Van Meggelen wrote:
Yep, that's a possibility, but it's rather more kludgy than I'd like.
(heck, the channel bank and T1 is more kludgy than I'd like - an
integrated card would be my preference).
I haven't followed this thread closely but have you looked into the
Voicetronix OpenSwitch cards?
Hi,
I'm trying to set up a fresh system for use with Asterisk. I've never
installed or used Asterisk before, so I do not know much about it.
I'm using Slackware Linux 10.1 and followed this guide:
http://www.automated.it/guidetoasterisk.htm
When I try to run asterisk though, at the point the
I haven't used it in a while, but I had to put subscribecontext=sip
for the phone's (in your case the snom) sip entry.
This seems like it has been removed from the wiki. Has it changed or
is this incorrect?
http://www.voip-info.org/tiki-pagehistory.php?page=Asterisk+phone+snomdiff=7
On Sat,
On February 20, 2005 11:34 am, Jim Van Meggelen wrote:
Is there a place to buy brand new Adit600's with 5+ FXOs and a T1 card
for $800? (I've looked on eBay, but that's not a reliable supply chain,
and I have yet to see such a price for new equipment). Seems to me one
is looking at more like
On February 20, 2005 11:44 am, Jim Van Meggelen wrote:
I like the thinking; the challenge is often where in the world you are,
and how much competition there is. Here in Ontario, T1's were generally
priced such that fractional T1s hardly saved anything. There is more
competition now, so prices
On February 20, 2005 11:47 am, Jim Van Meggelen wrote:
As with so many things, there is always a trade-off between economy and
functionality. The Adit 600 and T1 integration is certainly quality, but
I have not been able find an economical way to do this (purchasing used
equipment on eBay is
That app_disa is new to me... Is there a list of available apps? Im still
quite new to asterisk but I guess you can find out which apps you have by
using a show applications but my question would be more of how to make new
apps or download/get new ones, is this possible?
Also, is there a list of
from the console, show modules
-
Yeah, we rocked the vote all right. Those little
bastards betrayed us again.
- Hunter S. Thompson on the 2004 election.
On Sun, 20 Feb 2005, Anton Krall wrote:
That app_disa is new to me... Is there a list of
And go here:
http://www.voip-info.org/wiki-Asterisk+-+documentation+of+application+commands
-
Yeah, we rocked the vote all right. Those little
bastards betrayed us again.
- Hunter S. Thompson on the 2004 election.
On Sun, 20 Feb 2005, Anton Krall
On Sun, 20 Feb 2005, Anton Krall wrote:
That app_disa is new to me... Is there a list of available apps? Im still
quite new to asterisk but I guess you can find out which apps you have by
using a show applications but my question would be more of how to make new
apps or download/get new ones,
I have a weird problem... very puzzling..
Yesterday I had sound problems with the voice prompts, I couldnt hear them,
so I rebooted the system and voila, I was able to hear everything.. so I
went to bad.. and I just woke up and tried the system again and its back!!!
I dial the voicemail system
Why not just kill yourself, fucking wannabe spammer? DIE DIE DIE
I am using Underwood's fax system for fax on demand and it's very cool.
I am planning to do the following and I would like to know if it's
possible before putting my hands on it.
For a specific application,
I want to dialout
Ouch,
Do you know how to use gdb, the Gnu Debugger?
That will give you a clue as to where the segmentation fault is coming
from.
Good, then let me move on to the insults and ranting.
1. Why are you running on Slackware?
Are you trying to prove a point or just enjoy being frustrated?
Race Vanderdecken wrote:
Good, then let me move on to the insults and ranting.
1. Why are you running on Slackware?
Are you trying to prove a point or just enjoy being frustrated?
Open Source is like Broad Spectrum Pesticide, it works but
your results may vary and you may end up killing your
On Sun, 20 Feb 2005 23:16:00 +0900 (JST), Isamar Maia
[EMAIL PROTECTED] wrote:
Ok. I will be burned in fire.. :-)
Or better.. I won't go to the heaven...
You are probably right. But in the the mean time, while you are here
on earth, you will probably spend some time in the legal system too.
On February 20, 2005 01:11 pm, Race Vanderdecken wrote:
1. Why are you running on Slackware?
Are you trying to prove a point or just enjoy being frustrated?
Open Source is like Broad Spectrum Pesticide, it works but
your results may vary and you may end up killing your lawn.
Got a problem
Because I am more civilized?
By the way, it was Samuel Clemens's fool... quote, who stole it from
Mr. Lincoln, who stole if from Confucius (another educational Tyrant.)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian
Capouch
Sent: Sunday, February
Hello,
I bought a TDM400P, and intended to use it with my analog
phone, which is RJ11 ofcourse. So, the question now, how do I plug in
my RJ11 phone to the TDM400P card, which has an RJ45 interface? Also,
since it's an 11B card, I also intend to bring in an analog line into
the RJ45, so i am still
Brian Capouch wrote:
Race Vanderdecken wrote:
Good, then let me move on to the insults and ranting.
1. Why are you running on Slackware? Are you trying to prove a
point or just enjoy being frustrated?
Open Source is like Broad Spectrum Pesticide, it works but
your results may vary and you
Sorry, I didn't say I was using it with * -- just on a PC with a
different app. I don't think it would be difficult to use something
like lcdproc or even their test-app --
http://www.crystalfontz.com/software/633_WinTest/index.html (link to
linux source at the bottom), and use agi to call the
This is getting hard.
So what do we have?
1. The Asterisk server and the phones are using good CODECS.
2. Sound is moving from phone to phone.
3. Sound from the prompts is not playing back to the phones.
So let's go with the first principle: Eliminate the variables.
Do this:
Extensions.conf
On Sun, 20 Feb 2005 [EMAIL PROTECTED] wrote:
Hello,
I bought a TDM400P, and intended to use it with my analog phone, which is
RJ11 ofcourse. So, the question now, how do I plug in
my RJ11 phone to the TDM400P card, which has an RJ45 interface? Also, since
it's an 11B card, I also
Push it with enough force, it will come in.
On Sun, 20 Feb 2005 11:51:05 -0700, [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
Hello,
I bought a TDM400P, and intended to use it with my analog phone, which is
RJ11 ofcourse. So, the question now, how do I plug in my RJ11 phone to the
TDM400P
I have no problem with Slackware,
But when you are learning to drive a car you should first try a Chevy
with an automatic transmission first before strapping on a 6 speed
Ferrari.
Humor helps in teaching and getting a person to step out of a rut they
are having a problem in and gives them a
Thx Sergey!! Ill give it a try
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sergey
Kuznetsov
Sent: Domingo, 20 de Febrero de 2005 07:34 a.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Simulated
Ok... I added the extension and here are the results:
-- Executing Wait(SIP/intruder-phone1-8613, 2) in new stack
-- Executing Answer(SIP/intruder-phone1-8613, ) in new stack
-- Executing Playback(SIP/intruder-phone1-8613, vm-isunavail) in new
stack
-- Playing 'vm-isunavail'
On Sun, 2005-02-20 at 13:51 -0500, Paul wrote:
Or maybe a double fool because he also disrespected Debian GNU/Linux in
his reply.
*And* recommended Fedora, which makes it triple. I just dumped FC3 and
replaced it with Debian because Fedora's kernels constantly gave me
issues, e.g. with
I dont know if it has something to do but I see 2 mpg123 processes running:
3552 pts/1S 0:00 mpg123 -q -s --mono -r 8000 -b 2048 -f 8192
fpm-calm-river.mp3 fpm-sunshine.mp3 fpm-world-mix.mp3
3553 pts/1S 0:00 mpg123 -q -s --mono -r 8000 -b 2048 -f 8192
fpm-calm-river.mp3
Ok Noload modems, alsa and oss... No errors... Is this ok?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall
Sent: Domingo, 20 de Febrero de 2005 01:18 p.m.
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE:
On Sun, 2005-02-20 at 11:47 -0500, Jim Van Meggelen wrote:
Well, I appreciate everyone's input, and I'll give the matter some more
thought.
Just so no one stays up at night worrying, this is not a situation I am
facing, it is simply a hypothetical scenario.
As with so many things, there
Yes, running extra code/libraries/.so means more variables, which we are
trying to eliminate.
Really Anton I am stumped.
Does anyone know? Do you have to have the gsm codec to hear the .gsm
sound files.
Is there an [EMAIL PROTECTED] mailing list?
I found these:
From: James Bean [EMAIL PROTECTED]
Has anyone every setup an external open/close relay, off say a serial
interface, and have an extension trigger the relay?
The following will do the trick. Just add a 5vdc solid state relay
('cause you can't sink too much current out of the RS232C port).
Hi all,
I have call recording enabled via the Monitor command and it seems,
the call stops being recorded after the call is transferred. Is this
normal behavior? If so how can I continue recording of calls after
they have been trasnferred
___
Hi,
I mistakenly posted this to Dev list
I wonder if I use quadbri or octobri cards to insert Asterisk between ISDN
PBX and ISDN line - if power of Asterisks fails - will those card connect
PBX directly to ISDN line ? If not are there any other simple switching
devices, that would do this
Hi,
I wonder if I use quadbri or octobri cards to insert Asterisk between ISDN
PBX and ISDN line - if power of Asterisks fails - will those card connect
PBX directly to ISDN line ?
No, you need a isdn failover switch
If not are there any other simple switching
devices, that would do this
[EMAIL PROTECTED] wrote:
Jim Van Meggelen wrote:
Yep, that's a possibility, but it's rather more kludgy than I'd like.
(heck, the channel bank and T1 is more kludgy than I'd like - an
integrated card would be my preference).
I haven't followed this thread closely but have you looked into
I've got a strange situation that started yesterday -- I have a ton
of calls listed in the log for number = 18883629704
It initially looked like I was getting an incoming call on Zap/4 (LD
trunk) from 18883629704, which was going to an extension at Zap/2,
and then trying to dial out again
[EMAIL PROTECTED] wrote:
On Sun, 2005-02-20 at 11:47 -0500, Jim Van Meggelen wrote:
Well, I appreciate everyone's input, and I'll give the matter some
more thought.
Just so no one stays up at night worrying, this is not a situation I
am facing, it is simply a hypothetical scenario.
As
Title: Message
Just plug
it in. The RJ11 is narrower than the RJ48, but has the exact same connection
mechanism. it'll fit perfectly (the centre two pins are the
contacts)
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL
Dave Weis wrote:
I've never set up an mgcp device before. I have an Adtran IAD with the
MGCP firmware on it. I have it configured in mgcp.conf like this:
[general]
port = 2427
bindaddr = 0.0.0.0
[adtran]
host = 192.168.2.2
context = default
canreinvite = no
line = aaln/1
line = aaln/2
Check
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Jon Radon
Sent: Monday, 21 February 2005 2:55 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Snom phone hint exten question
I haven't used it in a
[EMAIL PROTECTED] wrote:
I have no problem with Slackware,
Me neither. I learned Linux with Slack. Found it to be extremely
friendly. And that was 10 years ago. Last time I chacked, it was still
friendly (and not at all GUI, unless you want it served that way)
But when you are learning to
Title: Message
just reinstalled
@home and i have a one of those 100 cards, anyways when i call from the pstn the
box picks up but i hear nothing, then it clicks a couple times, then nothing
again, i am trying to get the digital receptionist to work but it won't save my
wav file to the @home
Yesterday, I've checked tariffs from Bell Canada, For Full voice T1 it
was costs around $1000 + tax.
$216 - is access fee, $34 per channel.
You can get the PRIs from Allstream with 3 years commitment ~$600 per
month.
Andrew Kohlsmith wrote:
On February 20, 2005 11:44 am, Jim Van Meggelen
Disable the call waiting feature in the phone, so it will signal 486
- Busy here to additionally incoming calls.
Is it possible to test if a call to SIP/xxx is in place before dialling
out? This could help a lot to centralize administation of whether or
not to use call waiting instead of
Hello,
I have asterisk setup with Broadvoice.
It works great as PBX and Outgoing calling server for all local clients
withing 192.168.1.0 network. My asterisk is running over NAT.
I use linksys router.
Now, I am trying to connect from outside to my asterisk server.
I use Diax as iax client.
For
Thanks a lot for your message.
Race Vanderdecken schreef:
Ouch,
Do you know how to use gdb, the Gnu Debugger?
That will give you a clue as to where the segmentation fault is coming
from.
No, I once used it being instructed exactly by a developer to solve a
problem in Dosemu, but I never did
It is weird.. I did a full asterisk reinstall... (no asterisk at home
now)... And well Problem persists but this is weird, when it happens, I
reboot the machine, starts working again and sometimes sound stops,
sometimes it doesnt... This machine seems to have an attitude :)
Last reboot one
On Sun, 2005-02-20 at 22:59 +0100, Julius Schwartzenberg wrote:
I'm using a pretty old system and I have good experiences with Slackware
on other systems. Here are the specs of the system I'm using:
IBM/Cyrix PR-200 (@150MHz), 64MB RAM, two HDs which are combined ~2GB.
Here is your
Okay, now you are getting off track.
Hold old is the motherboard?
How big is the case?
How big is the power supply?
If it is a smaller case and server then sometimes heat can be an issue
when you are on the threshold of the temperature limit. Things will work
mysteriously and then not work.
Roy Sigurd Karlsbakk wrote:
Is it possible to test if a call to SIP/xxx is in place before dialling
out? This could help a lot to centralize administation of whether or not
to use call waiting instead of configuring the ATAs.
app_groupcount can be used to provide call counting in any fashion you
I was studying the asterisk-users list archives
to learn if anyone has had success with an X100P on a
sparc. I noticed some postings on the subject. I am
wondering if anyone has learned anything new?
I have an Ultra-60 running Gentoo with 2.6.10 and
udev. I built * 1.0.5 and have been enjoying
Title: Message
Can you work through a process of
elimination if you record the file using an internal extension by dialing *77
and seeing if that works?
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kurt Fankhauser
Sent: Sunday, February 20, 2005
7:42 PM
To:
On Sun, 20 Feb 2005 22:45:42 +0100, Roy Sigurd Karlsbakk
[EMAIL PROTECTED] wrote:
It seems to me wiki downtime is somehow regular.
Is this the fact?
If so, should it be moved?
Just to add some balance to this threadJim and colleagues, thanks
for hosting the Wiki. You should take it as a
Kurt Fankhauser wrote:
just reinstalled @home and i have a one of those 100 cards, anyways when
i call from the pstn the box picks up but i hear nothing, then it clicks
a couple times, then nothing again, i am trying to get the digital
receptionist to work but it won't save my wav file to the
It seems so simple, but I'm having no luck installing a HFC-s ISDN BRI card
on [EMAIL PROTECTED] 0.5.
I probably have to install BRI-stuff from Junghanns.net but that also
downloads and installs another copy of * from Digium.
I'm not sure if zaphfc has to be installed *before* Asterisk or if it's
Just to be sure.. I checked the cards...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Race
Vanderdecken
Sent: Domingo, 20 de Febrero de 2005 04:18 p.m.
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] No
Title: Message
I'll
buy a IP phone tomarrow so i can do that
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of dean
collinsSent: Sunday, February 20, 2005 2:40 PMTo:
Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE:
I think the box is answering calls but I don't think the digital
receptionist is working properly.
Kurt
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Thompson
Sent: Sunday, February 20, 2005 3:05 PM
To: Asterisk Users Mailing List -
Message also do you have to use a ip phone to record your greeting because
this wav file stuff isn't working.
I didn't try uploading. You can just setup a SIP softphone and dial *77 when
looking at the menu you want to record in the GUI.
Regards,
Erwin
Message I'll buy a IP phone tomarrow so i can do that
No need:
http://www.xten.net/index.php?menu=productssmenu=download
Regards,
Erwin
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Title: Message
Just download a free softphone and do it
that way eg xten
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kurt Fankhauser
Sent: Sunday, February 20, 2005
9:18 PM
To: 'Asterisk
Users Mailing List - Non-Commercial Discussion'
Subject: RE:
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