Hi,
How can i implement VAD/DTX using zaptel with asterisk towards PSTN.
mail2web - Check your email from the web at
http://mail2web.com/ .
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Hello,
On Sun, 10 Apr 2005, Tim Connolly wrote:
I'm working on getting a new Digium TE110XP working. I no_load the chan_zap
module, otherwise * doesn't start. When I try to load it manually I see:
pbx01*CLI load chan_zap
Unable to load module chan_zap
Apr 10 22:13:23 WARNING[4349]:
Hi,
I've done some testing with asterisk and I must say I'm very impressed by
all the features. Now I want to create a production environment and am
looking into all the available ISDN cards. The cards I've found are:
1. AVM C4 (1300 euro's)
2. Eicon Diva with 4 ISDN2 ports (even more
It's my opinion that whilst asterisk indeed has some fax capability, it's
not a business-grade fax platform. If faxes are indeed as important to your
business as you suggest, I'd be inclinded to look for alternatives.
- Original Message -
From: Marc [EMAIL PROTECTED]
To: 'Asterisk
I don't know if what you're trying to do is possible, but the easiest way to
check would be to take a look at the raw packets on the ethernet interface
of your * server once a call is in progress. If indeed the RTP can be handed
off to the 2 endpoints, you should only see SIP traffic at your
On Mon, 2005-04-11 at 16:40 +1000, Rod Bacon wrote:
I don't know if what you're trying to do is possible, but the easiest way to
check would be to take a look at the raw packets on the ethernet interface
of your * server once a call is in progress. If indeed the RTP can be handed
off to the
Is it possible to use hylafax and asterisk with only the AVM C4? Or do I
need a separete fax modem?
-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Namens Rod Bacon
Verzonden: maandag 11 april 2005 8:35
Aan: Asterisk Users Mailing List - Non-Commercial
I have some troubles to use Setgroup / Checkgroup!!!
I setup a test (NoOP's are deleted): First caller should get first line,
second caller should get second line, third caller should get busy and
send an email. Note, that I used twice here to check the first line!!!
[trunkint_A]
exten =
Hello All!
I've a problem with a TDM400P digium card.
My box has no molex connectors for power supply. Simply has no any power
connector, because is not a normal PC) And I need to know if i can use a
external supply. But I've several questions:
1.- Are both circuits (PCI-power and
Marc wrote:
Is it possible to use hylafax and asterisk with only the AVM C4? Or do I
need a separete fax modem?
Works fine and dandy with a single AVM C4 here.
--
Best regards
Peer Oliver Schmidt
PGP Key ID: 0x83E1C2EA
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On Sun, Apr 10, 2005 at 09:47:36AM -0500, Andy Hamilton wrote:
On Apr 10, 2005 7:30 AM, Thomas Andrews [EMAIL PROTECTED] wrote:
Is it possible to play a different dialtone as soon as a user dials say
'0' for an outside line ? Ignorepat is an inadequate solution because
local users are
Hi
1) Don't bother considering analogue lines. Too problematic and not any
cheaper in the long term.
2) the HFC chipset ISDN cards at £13 are fine as long as you make sure
you assign each card its own IRQ in the bios.
http://www.komplett.co.uk/k/ki.asp?sku=119006cks=SPK
I have 3 of these
I'd like to make a dial plan but couldn't work it out. I'd be appreciated if
you can help me.
The client reaches asterisk by PRI and starts conferance by the SIP agent
dedicated to his number. besides, I want to add another second client who
dialed the same number to the first client's conferance
On Monday 11 April 2005 08:29, Peer Oliver Schmidt wrote:
Marc wrote:
Is it possible to use hylafax and asterisk with only the AVM C4? Or do I
need a separete fax modem?
Works fine and dandy with a single AVM C4 here.
Just wanted to chip in to say that Eicon's Diva Server 4BRI-8M is
In article [EMAIL PROTECTED],
Robert Keller [EMAIL PROTECTED] wrote:
Thanks Rich, I wasn't sure where to find that context. I found the outbound
context in the extensions_additional.conf and added w's in the following
manner:
[outrt-001-Out1]
include = outrt-001-Out1-custom
exten =
Hi,
Weird issue, 2 asterisk boxes running 1.0.7, when I call iax from box 1
to box 2 it works fine, when I dial from box 2 to box 1 I get a
On Box 1
Apr 11 17:25:39 WARNING[8969]: chan_iax2.c:5553 socket_read: Call
rejected by 192.168.69.1: No authority found
On Box 2
Apr 11 17:26:07
Hi Mat,
Did the following:
1. Upgraded to new CVS HEAD version CVS-NHEAD-04/11/05-16:08:03
On the Makefile, enabled the ff:
# Optional debugging parameters
DEBUG_THREADS = -DDEBUG_THREADS -DDO_CRASH -DDETECT_DEADLOCKS
Hi list!
I'm working to replace a PBX with group ring indication.
On the current PBX each phone has 3 buttons with a light to identify an
incoming call ringing for a certain group. For example if the phone is
ringing at sales a led lights up to indicate a call coming in on that
group (but that
Hi,
Got 2 asterisk boxes, Box 1 has SIP users, and Box 2 has a E1 card
connected to a TDA200, when a sip user from box 1 calls someone on the
tda200 there is no ringing noise just dead silence until the person on
the TDA picks up there extensions.
Is there a way in thse situations to supply a
Good day all
I got the latest cvs asterisk
But when making a call out threw the voicetronix openline4 card the dtmf
doens not work
I got this in vpb.conf
ecsuppthres = 4096
indication = 1
dtmfidd = 3000
ast-dtmf-det=1
relaxdtmf=1
break-for-dtmf=yes
Please help
Thanks
Altus
I saw in the dump captured by Ethereal that X-Lite received 200(OK)
from asterisk after sending INVITE. So I guessed X-Lite registered
well. But I got null reply when I ran sip show peer in asterisk
console.
What is your opinion about that?On Apr 8, 2005 8:43 PM, Rich Adamson [EMAIL PROTECTED]
If the answer is yes:
a) how can I do that?
b) how can I restart an asterisk console?
Best regards,
Abe
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Then,
I realised a spent lot of time thinkin about this solution. Other option is that
you put a prepaid calling card platform in Russia. I saw in CEBIT some russian
companies selling prepaid calling cards.
In order to give access to your customers without them to know where is the
platform, you
James Bean wrote:
Hi,
Weird issue, 2 asterisk boxes running 1.0.7, when I call iax from box 1
to box 2 it works fine, when I dial from box 2 to box 1 I get a
On Box 1
Apr 11 17:25:39 WARNING[8969]: chan_iax2.c:5553 socket_read: Call
rejected by 192.168.69.1: No authority found
On Box 2
Apr 11
Hullo :)
I've been trying to use a stable 1.0.7 codebase against the patches at
http://www.lusyn.com/asterisk/patches.html - but am having no joy. Even if I
copy-paste the instructions on that site verbatim, everything compiles
perfectly, but simply no incoming number is received.
If I then
Title: Message
My understanding (by no means
definitive):
You need a solution to the NAT problem for the
audio stream. STUN will help with non symmetric NAT but not with symmetric NAT
so it's not a complete solution. If you have UAs behind symmetric NAT you will
need Asterisk or an RTP
Hello,
on http://www.voip-info.org/wiki-Aculab it has been said, that there is
a Aculab card, which works with Asterisk. Two questions:
1. Which card is this?
2. How do I configure it with Asterisk / Linux?
If anybody has any experiences regarding this, I would very much
appreciate to get some
Hi,
I wantto use
the cdr to record the call log to my Microsoft SQL Server using unixodbc and
freetds
but when I compile,
I've got this message
Does anyone have the
same problem and/or know how to solve it ?
Thanks
Baste
regards
David
Masure
make[1]: Entering directory
Hi !
What is wrong with my dial plan?
I can't get my call forwarding and parking to work.
Do I need to edit more config files?
Thore
extensions.conf :
[general]
static=yes
writeprotect=no
[macro-dialout]
; ${ARG1} CIDNAME
; ${ARG2} Device
; ${ARG3} Num
; ${ARG4} SIP EXT
exten =
Username and secret in sip.conf are the credentials for the sip user. Any
sip UA can then connect to Asterisk using those details and will ring when
extension 176 is dialled.
Look at sip.conf on the wiki.
Regards
Cameron
- Original Message -
From: Don Murray [EMAIL PROTECTED]
To:
Jochen,
Recently I contact Aculab in UK about that and
They asked me to call Digium Sales.
I called Digium Sales and they told me that nothing is confirmed yet
about a deal between Aculab and Digium.
Maybe something changed
Isamar
On Mon, 11 Apr 2005, Jochen Witte wrote:
Hello,
on
Look up the answer command on the wiki.
Regards
Cameron
- Original Message -
From: Min Hwan Chang [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Friday, April 08, 2005 10:20 AM
Subject: [Asterisk-Users] How to turn off automatic pick up for
Incomingcalls [EMAIL PROTECTED]
Install Asterisk at home which includes AMP. This will allow you to
configure SIP and voicemail using a web browser. Couldn't be easier.
Regards
Cameron
- Original Message -
From: Michael D Schelin [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
On Monday 11 April 2005 10:06, Gavin Hamill wrote:
Hullo :)
Bah, I got bitten by my own hacks. Things are now working much better than
before :)
I'd forgotten that I'd commented out the line:
if (p-use_callerid p-cid_start == CID_START_USEHIST)
in my previous CVS version, and this made CID
To exit from the console type quit
To restart the console:
type asterisk -vcr (that's several v's in front of the cr.
The more v's you put, the greater ther verbosity. I think the max is
10)
On Apr 11, 2005 3:30 AM, Abraham WEI [EMAIL PROTECTED] wrote:
If the answer is yes:
a) how can I
Does anyone know if it is possible to connect say Grandstream ip phone
directly to LiveVoip?
How to setup the phone?
Any help will be appreciated!
--
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.308 / Virus Database: 266.9.5 - Release Date: 07/04/2005
Hi, I have a problem to register with my provider, because my username
is myphonenumber@provider's domain.
Thus my registry line contains a double @ sign and everything is parsed
incorrectly.
How can I quote the username to ignore the first @ ?
cu chrisb.
Thanks Rich, I wasn't sure where to find that context. I found the outbound
context in the extensions_additional.conf and added w's in the following
manner:
[outrt-001-Out1]
include = outrt-001-Out1-custom
exten = _1NXXNXX,1,Macro(dialout-trunk,1,w${EXTEN})
exten =
Whooppss after research for several hours before posting, another
asterisk user passed on the answer to me.
Add ,r to the Dial string over the E1 to hear the ringing on the line.
James
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of James Bean
Sent:
Sorry again sorted it out, the [definition] has to be the same as the
username or it doesn't work, well for me anyway.
:-)
Gotta reasearch a few extra hours and play a bit more before I post I
think.
Sorry guys and girls.
James
-Original Message-
From: [EMAIL PROTECTED]
I currently use another PBX system which takes care of VM. Is there a
way to prevent [EMAIL PROTECTED] v0.6 from picking up Incoming calls?
I'd still like to dial out from Asterisk (I have IAX trunking on). Is
there a way to do this? My knowledge of the Extensions.conf is
limited.
Go in
I've a problem with a TDM400P digium card.
My box has no molex connectors for power supply. Simply has no any power
connector, because is not a normal PC) And I need to know if i can use a
external supply. But I've several questions:
1.- Are both circuits (PCI-power and
In article [EMAIL PROTECTED],
David Masure [EMAIL PROTECTED] wrote:
I want to use the cdr to record the call log to my Microsoft SQL Server
using unixodbc and freetds
but when I compile, I've got this message
Does anyone have the same problem and/or know how to solve it ?
I saw in the dump captured by Ethereal that X-Lite received 200(OK) from
asterisk after
sending INVITE. So I guessed
X-Lite registered well. But I got null reply when I ran sip show peer in
asterisk console.
What is your opinion about that?
If sip show peers does not show your xlite
This has already been answered...but I can't find it...
Has anyone set up multiple fax lines in asterisk...
Fax Extension #1 goes to email1
Fax Extension #2 goes to email2
ETC...
In other words, I want to be able to give numerous users each
a virtual fax machine..
Bill
; Assumes
On Sun, 10 Apr 2005, Eric Wieling wrote:
No. r instructs Asterisk to provide a fake ringback tone. If you
need r then something is seriously wrong. Asterisk will always
provide rinback tones when it thinks it should.
For PRI channels you may need it if the equipment at the oher end does
I own elmeg C46e PBX (ISDN PBX with 6 a/b ports). I
connected two of these ports to a TDM02B installed on a Slackware 9.1/
Asterisk 1.0.2 and I bought a Cisco ATA186 to connect two analog
telephone sets two floors down. Each a/b port is assigned only to one
ATA port. The problem is that after the
Hi all,
We were willing to try the SIP Attended/Supervised transfer with * realease
1.0-7. From the wiki´s feature.conf config page we found that a special
section called featuremap had to be added to the config:
[featuremap]
blindxfer = #1; Blind transfer
disconnect = *0
MWI works just fine.
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Version 0.79 - 11. April 2005.
* Norvegian language added - thanks to Kåre Sundland
* German language updated - thanks to Marco Walz
* Russian language updated - thanks to dnz63
* Caller ID name added to call buttons when on call
FREE Download: http://ipswitchboard.thorben.dk
Would you like to
Hello,
I have Asterisk 1.0.6 - I try to setup Sangoma A101 T1 board together with
the Rhino fxs chanelbank.
Things done:
- T1 cross cable = I have carrier, signalling and framnig leds on
the channelbank green.
- channelbank configuration:
t1 - Proto: LOOP
exit and asterisk -r
- Original Message -
From:
Abraham
WEI
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Monday, April 11, 2005 3:30
AM
Subject: [Asterisk-Users] Can I exit from
asterisk console without stoppingasterisk?
If the answer is
Giles thank you for getting back so quickly, dmesg
doesnt output anything, but even if it did, I am not sure that I could
recompile the kernel.
The server I am using is in a virtual dedicated hosting environment, I
do not have access to recompile the kernel, nor can I replace it. The
Has anyone worked out how to get the Shared call appearances working on
a SPA841 with Asterisk. Googling found a few people asking the same
question last year, but alas no answers. Just wondering if anybody has
made the breakthrough in the meantime.
craig
Title: Re: [Asterisk-Users] Zaptel Compile on a virtual dedicated host.
Hi,
Do you happen to know what VPS system your host uses (e.g. UML, Virtuozzo, VMWare, FreeVPS, etc.)? It could make a lot of difference, as some platforms will allow changes that others will not.
-- Henry Owens.
On
On Apr 6, 2005 10:34 PM, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Anyone have any ideas on where I can find the right kernel
source? I have look at rpmfind.net and
google'd with no avail!Hi,
You're never going to find the kernel source. The reason for this is
that your VPS
John Breeden wrote:
Been there, done that - no joy :-)
It appears the modifier only excepts a numeric, anyone know if/how you
can feed it adecimal/hex for ascii #?
Rich Adamson wrote:
Is there anyway to append the '#' symbol to a dial string? -
hex/octal whatever? I'm surprised that I can't
[EMAIL PROTECTED] wrote:
Hi,
How can i implement VAD/DTX using zaptel with asterisk towards PSTN.
TDM (PSTN/telcos) do not support VAD. The entire idea of VAD is not
even a valid idea.
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David Masure wrote:
Hi,
I want to use the cdr to record the call log to my Microsoft SQL Server
using unixodbc and freetds
but when I compile, I've got this message
Does anyone have the same problem and/or know how to solve it ?
Update of /usr/cvsroot/asterisk/doc
In directory
Title: Re: [Asterisk-Users] Zaptel Compile on a virtual dedicated host.
It appears to be Virtuozzo
-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Henry
Sent: Monday, April 11, 2005 9:34
AM
To: Asterisk Users Mailing List -
Non-Commercial
On Tue, Apr 05, 2005 at 09:06:33PM +0200, Peter Svensson wrote:
A yellow alarm means the remote end is sensing some error condition. Try
looking for an error message at the remote end. It may be as easy as a
broken cable (where the Hipath does not hear the Asterisk box).
The problem is,
Tony, I don't see ${EXTEN} anywhere in the [macro-dialout-trunk] context.
Am I missing something?
Robert Andrew Keller
Ferndale School District #502
[EMAIL PROTECTED]
360-383-9228 PH.
360-383-9218 FAX
Paving the way for tomorrows genius.
From: [EMAIL PROTECTED] (Tony Mountifield)
Organization:
Honestly, the best script I've ever found is the wondershaper script (
google it ). I tried the correct one posted in this thread, tried
modifying it, but in the end I just used wondershaper.
Does a great job. My only fear is it doesn't specifically target IAX2
traffic as high priority, but
Hi all
I installed asterisk on a dual PIII 700 with two NICs. I then
proceeded to configure both NICs with bonding enable (bonding
miimon=100 mode=1). I know certain features (like load balancing) under
a bonded configuration is not understood by some switches, so I
configured it using mode=1
I agree that Wondershaper is a great script; prior to using it in an office
where I set up asterisk, there were some major problems with call quality,
but it seems to have helped hugely (the same DSL line is used for both VoIP
and everyday 'net usage for seven people - not ideal, but I didn't set
I used the one posted to this list and for a test did a
speedtest.dslreports.com bandwidth test duringa call, no loss in
quality.
I set ports 1-11024 to RTP in rtp.conf, I dont need 10k ports for
that as I have few calls being processed. I also added sip to the queue
although that prolly
Keep on bugging the Sangoma guys, I know they are working on several RBS T1
issues right now(They called me Friday to go over a few things) They just
need help from users like you and I to find the bugs in their drivers.
Have you tried any other signalling types other than LOOP?
MATT---
On April 11, 2005 10:08 am, Sean Kennedy wrote:
Honestly, the best script I've ever found is the wondershaper script (
google it ). I tried the correct one posted in this thread, tried
modifying it, but in the end I just used wondershaper.
:-) I started out with wshaper and just didn't like
Previously I have posted the same mail but no one answered me...Sorryfor resending the mail.I have bought a Wildcard X101P for my Asterisk PBX. Now I can placeand get calls through the lines/channel. Everything is okay but theproblem is when I call outside through our PSTN line, after fewminutes
On Mon, 2005-04-11 at 10:32 -0400, Andrew Kohlsmith wrote:
On April 11, 2005 10:08 am, Sean Kennedy wrote:
Honestly, the best script I've ever found is the wondershaper script (
google it ). I tried the correct one posted in this thread, tried
modifying it, but in the end I just used
I've got a X100P in a compaq proliant 3000. My system stops taking calls
and making calls. I had been getting the FXO PCI Master abort before
updating, I am now running a cvs head checkout from a week or so ago. Now
I still have the problem but get more error messages:
Found a Wildcard FXO:
Good morning all..
I was following a discussion on this list about the
TDM400P revisions. It is my understanding that the current
revision that one should have is the Rev. H and not the
E/F. I have not yet been able to verify the rev stamped on
the board, but zaptel is reporting that I have
I use a headset w/out any problems, except for if my cell phone is close by
and rings. Otherwise, volume is ok and no humming. Could it be your headset?
At 01:56 PM 4/10/2005, you wrote:
Just make sure you don't have a cordless or cell phone near by or the
headset jack will receive a
Joe S [EMAIL PROTECTED] writes:
Hi,
I am new with asterisk. I was wondering if there is a way to call a
OH323 user or SIP user using Netmeeting/SJPhone with H323 as the
default protocol without having a gatekeeper.
I can make a call from SIP to OH323 by specifying it in the
I'm having similar issues using an X100P Ambient Chipset Clone Card
any ideas?
Regards,
Sahil Gupta
VoiceValley
On Mon, 11 Apr 2005, Dave Weis wrote:
I've got a X100P in a compaq proliant 3000. My system stops taking calls and
making calls. I had been getting the FXO PCI Master abort before
Sorry for the initial no subject line. Was in a hurry when
I typed this and somehow missed putting it in.
Please accept my apologies
On Mon, 11 Apr 2005 10:54:30 -0400
Robert Webb [EMAIL PROTECTED] wrote:
Good morning all..
I was following a discussion on this list about the
TDM400P
Hello list,
I have been successful in setting up my first * box with a
pair of x100ps, Cisco 7960, and a Digium iAXy.
I would like to incorporate an Aastra
480e using my iAXy and ADSI. I want to be able to
answer phone calls with my 7960 in the back of the house and park the call,
Hi,
Is there anyway to manipulate the asterisk internal database from the
manager (the one you can telnet to)? And if so.. how does one do it?
(ie for enabling call forwarding, etc)
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Hi,
I always thought that if there is no called extension in context, then 's'
extension is started (I use latest bristuffed Asterisk)
I have context 'from-isdn' :
[from-isdn]
exten = s,1,Wait,2
exten = s,2,NoOp(ISDN call from outside ${CALLERID}: Name: ${CALLERIDNAME},
Number:
I need to make a time loop in the Extensions.conf. I want it to play a
file every 5 minutes on a call. If I can't use wait because it ignores all
audio. Anyone have any suggestions?
Regards,
Chris___
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I was following a discussion on this list about the
TDM400P revisions. It is my understanding that the current
revision that one should have is the Rev. H and not the
E/F. I have not yet been able to verify the rev stamped on
the board, but zaptel is reporting that I have the Rev.
E/F.
I have a home user for asterisk that is not ready to let asterisk manage the
entire dialplan ... he's still got an answering machine on the outside line
and has this in the [incoming] context for that line:
exten = s,1,Wait(300)
exten = s,2,Answer
exten = s,3,DigitTimeout,5
exten =
Hi, I want to download the CVS HEAD version. Any one can show how to get
this version ?
Is the version from: http://www.asterisk.org/index.php?menu=download the
CVS Head version?
How I can check if my version is CVS HEAD or not?
Best Regards,
--
Guillermo Salas M.
Telconet S.A. Manta
Calle
I think it is i you want, s is the start for a context, meaning anything
coming in through that context will start there, i is invalid, and fires
if an invalid extension is keyed in that context.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Robert
Rozman
If You do a checkout in CVS without specifying a version (as shown on
the referenced site) You will allways get the HEAD (means the most
recent) branch.
Jochen
Am Montag, den 11.04.2005, 10:27 -0500 schrieb Guillermo Salas M:
Hi, I want to download the CVS HEAD version. Any one can show how
I have a question about dial tone as well, when calling a company typically
they have an answering system, i.e. press 4 for Bill etc. I am using a diax
soft-phone and have been unable to get the receiving system to forward me
on. Is there a feature change in Asterisk that needs to be enabled or is
I have a home user for asterisk that is not ready to let asterisk manage the
entire dialplan ... he's still got an answering machine on the outside line
and has this in the [incoming] context for that line:
exten = s,1,Wait(300)
exten = s,2,Answer
exten = s,3,DigitTimeout,5
exten =
On 09:17, Mon 11 Apr 05, Gonchi Mateos wrote:
Hi all,
We were willing to try the SIP Attended/Supervised transfer with * realease
1.0-7. From the wiki?s feature.conf config page we found that a special
section called featuremap had to be added to the config:
[featuremap]
blindxfer =
Some questions:
What country are you in?
Is there anything else connected to the line from the PSTN? It sounds
like you have a marginal condition, such as insufficient loop current
perhaps.
Do have any features, such as call waiting, on the line?
Do you know how far you are from the central
In article [EMAIL PROTECTED],
Robert Keller [EMAIL PROTECTED] wrote:
Tony, I don't see ${EXTEN} anywhere in the [macro-dialout-trunk] context.
Am I missing something?
It's probably ${ARG2}.
When you call Macro(name,1,${EXTEN}), say for extension 1234, then the
macro [macro-name] gets called
As you are a new Linux and asterisk user you best path is to use a Linux
Distribution that is easy to install and setup.
I have heard that Mandrake is very good, but for me I like Fedora 2/3
from Red hat.
You will need an OS that has clear documentation in the form of books
and a well supported
Here's an excerpt from that page. Obviously, the hyperlinks are
missing for some things, but I would reccommend rereading the page,
specifically where it says, ...download a tarball of the released
sources... These are release versions.
If you want the CVS Head version, perhaps where it says, To
This might seem really dumb but tack enough silence onto the back of
your file to make it five minutes long. Then the message play for 5
minutes and repeats.
Race The Tyrant Vanderdecken
This was a dumb idea.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
hi,
When I reboot my Fedora 3 box with Asterisk (latest version) I see
Asterisk is failing to shutdown properly. All other processes shutdown
and show success but Asterisk shows failed. What could be causing this.
Thanks
___
Asterisk-Users mailing
Folks,
I have * running well but latency is too high (seems to be about 300-500
msec). This is on a lightly loaded Covad ADSL line running IAX to
teliax, voipjet and voicepulse. Ping times to the teliax server are
consistently in the 51-53 msec range. The others are similar.
I am looking for
Brian Capouch wrote:
Roy Sigurd Karlsbakk wrote:
1.Does Asterisk support SS7 and ISDN?
ISDN is supported out of the box. SS7 support is (or will soon be?)
supported by a commercial version of Asterisk. Search the list
archives or
post to asterisk-biz.
Steve Underwood (here on the list) has a
Hi,
Can anyone recommend a very low cost box that could support Asterisk and
at least one (preferably two) TDM400p cards and cost less that $150
(preferably under $100). The box should be able to run without a
keyboard/mouse or CDROM. It also needs at least one Ethernet port. I
know about
Paul P. Pongco wrote:
Hi Mat,
It's Matthew :)
I skip enabling pg and continue with make clean and make valgrind.
That's fine. I found that out too.
gdb backtrace still gives vague output:
snip
Still not clear, any pointers to make the backtrace more verbose?
you probably need
Hi,
I have been reading about some of the problems encountered with the
TDM400p cards needing a reboot. I am still testing my system so I have
not seen this yet. Also my card is showing as a version 2 when I run
'modprobe wtcdm' - is this problem in this card as well. If so can
anyone
Thanks!
Right syntax - wrong box :-) (inter-iax between to *s - needed to apply
the suffix to the box talking directly TO the zap channel ... duhhh .)
Caught yet again by my own wrong assumtion
Eric Wieling wrote:
John Breeden wrote:
Been there, done that - no joy :-)
It appears the
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