I try to install astcc.
Make install gives me:
DBI connect('database=astcc;host=localhost','astcc',...) failed: Unknown
database 'astcc' at ./astcc-admin.cgi line 67
DBI connect('database=;host=','',...) failed: Access denied for user:
'[EMAIL PROTECTED]' (Using password: NO) at
Hi,
Questions keep comming up about this, so I started writing something at
http://www.soft-switch.org/foip.html . I think I covered the FAX over
VoIP issues fairly completely. T.37 is pretty simple to explain. There
is rather more to say about T.38, but at least this is a start. If
anyone
I tried to install astguiclient and it gives me for each follow page:
Object not found!
Looking into the apache log file I find:
[Sun Feb 27 16:18:30 2005] [error] [client 192.168.250.108] File does
not exist: /srv/www/htdocs/astguiclient/method=POST, referer:
HP's lawyers have contacted me about the conflict.
I suppose lawyers suffered as much as other businesses when the bubble
burst, but now they're coming back strong. Funny it took them a few
years to find your domain name :)
___
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But this doesn't work when I press any non-existent extension I get
congestion.
Can you put your 'i' extension directly in [internal] ?
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Hi, all
I have Asterisk here and SIP phone sitting at another location.
Initially, I had problems registering the phone. Now I have added 'nat=yes'
for this phone in sip.conf and phone registers.
However, I can not make calls.
SIP debug shows that phone registers with public IP address of the
- Original Message -
From: Michael Bielicki [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Saturday, February 26, 2005 5:01 PM
Subject: Re: [Asterisk-Users] Digium BRI or quad BRI
Hmm don't know about you but I
On 15/02/2005 21:05 Vledder, Hans said the following:
Hi Dan,
I've been investigating the same thing. Try to Google for Asterisk+Soekris,
Soekris is the company (http://www.soekris.com) that makes cute little 586
class fan-less single board computers that run both Linux and FreeBSD ...
i've got
On 19:45, Sun 27 Feb 05, Rudolf Ladyzhenskii wrote:
As one can see, public IP 147.10.78.157 is used at registration time, while
private IP 192.168.1.2 is used for communicating with phone.
[ext102]
type=user
nat=yes
host=dynamic
secret=ext102
context=default
[ext102]
type=peer
On Sat, 2005-02-26 at 02:59 -0600, Anton Krall wrote:
I gave a queue setup like this, but I also have it setup so that if no
agents are online, the caller cannot get in but I discovered that if that's
the case, the call hangsup on the caller:
[soportetecnico]
;Soporte Tecnico
exten =
Thanks for suggestion.
Unfortunately did not work.
What does this option do anyway?
Rudolf
- Original Message -
From: Michiel van Baak [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Sunday, February 27, 2005 8:18 PM
Subject: Re: [Asterisk-Users] NAT/Routing problem
On
On 20:52, Sun 27 Feb 05, Rudolf Ladyzhenskii wrote:
Thanks for suggestion.
Unfortunately did not work.
What does this option do anyway?
I cannot explain it as clear as the wiki.
have a look here:
http://www.voip-info.org/wiki-Asterisk+sip+canreinvite
--
Michiel van Baak
I see that to be fraught with problems.
Speaking as a radio ham, I find that non radio savvy folks have little
appreciation of the unique problems surrounding radio users.
Sure you'll be able to connect up your phone patch device to a
conference (radiophone patchSIP ATA) but your radio users
When I pick up calls on my Sipura I just dial *8# instead of *8.
The # will end the Sipura's dial plan.
If you put *8 into the dialplan, that would work too.
--On Saturday, February 26, 2005 11:39 PM -0700 Joseph
[EMAIL PROTECTED] wrote:
On Sat, 2005-02-26 at 18:22 -0700, Joseph wrote:
I can not
Rich Adamson wrote on Friday, 25 February 2005 4:18 AM:
GMRS, FRS and MURS radios may not be interconnected with the PSTN (47
CFR 95.141). There has been a lot of talk from lobbyists to clarify
this rule, but as it stands you could conceivably connect a *private*
network to GMRS or MURS
Hello list,
i have a strange problem iam using the ulaw,alaw and
g729
codecs
in sip.conf i have like this
[general]
disallow=all
disallow=g723
allow=g729
allow=alaw
allow=ulaw
even though i am disabling the g723 any UA could able
to connect to the system and then suddenly asterisk
stops
Bill Maidment a écrit :
Hi all
I'm fairly new to Asterisk, so be nice :-)
I was wondering if anyone has been able to get the Welltech K1000A USB
phone working on Linux. I see audio and HID drivers loaded when it is
plugged in to my Fedora Core 1 laptop, but that's about all that
happens. I've
On Feb 27, 2005, at 8:11, Joseph wrote:
Though, I'm not sure in what value is the time expressed.
When I input (6000:5999:1) as soon as I pickup the phone the time was
announced I have only 5sec. left
As the source you pasted in your original post clearly states, the
value is in ms. That's not
On Sat, 2005-02-26 at 18:22 -0700, Joseph wrote:
I can not make a call pickup to work with Sipura-3000.
I have one SIP phone and one is connected to ATA Sipura-3000
I've in all sip.conf context
callgroup=1
pickupgroup=1
in features.conf I've tired:
pickupexten = *88
Hmm, is your asterisk server behind nat with port forwarded ports? If
so, have you tried adding this to your sip.conf?
[general]
externip=xxx.xxx.xxx.xxx; your external IP, provided by your ISP
localnet=192.168.0.0/255.255.255.0 ; your LAN
...
...
[ext102]
canreinvite=no
host=dynamic
Hi Steve,
I sure hope that youre making HP pay for the domain. They should compensate
you for the inconvenience and cost of switching domains since they havent
upheld their trademark on the web untill now.
Let me know if you need a lawyer.
/Danny
On Sunday 27 February 2005 09.27, Steve
Joseph wrote on Saturday, 26 February 2005 8:38 PM:
I'm testing two options from dial command and can not make them to
work.
S N I P
exten = 21,1,Dial(${phone1},20,r,w)
exten = 21,1,Dial(${phone1},20,r,L(5[:4][:1]))
Try:
exten = 21,1,Dial(${phone1},20,rw)
Options should be
Is your asterisk server behind nat with port forwarded ports? If
so, have you tried adding this to your sip.conf?
[general]
externip=xxx.xxx.xxx.xxx; your external IP, provided by your ISP
localnet=192.168.0.0/255.255.255.0 ; your LAN
...
...
[ext102]
canreinvite=no
host=dynamic
nat=yes
Good day list,
I have a problem with ATA Handy Tone 286. It has been unsuccessfully
downgraded via HTTP. Seems like during downgrade there was a problem
with connection, because now it's not responding at all. There is no way
to get to it's voice menu via phone (by pressing button on it). The
Steve,
Excellent summary thus far!!! Will the summary stay on that url for a
lengthy period of time, or would it be possible to copy it to the wiki
when complete?
One item you might add to the T.38 discussion is a relatively short
paragraph that describes/relates the current analog fax
I fiddled around for way too long with the RedHat distributed perl, and
then gave up, installed the active state perl binary RPM, and used it's
executable and it worked great. Thanks you all for your help =).
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I'm having an issue with my current configuration. I have a single
PSTN line connected to an X100P and a couple IAX trunks to NuFone and
VoipJet. When I make an outbound call it doesn't properly detect if
my PSTN line is in use with another call and then overflow to my
outbound IAX connections.
Wow,
The mediatrix configuration tru the unit manager software is like going
tru hell.My mediatrix unit is failing when it come to registering it
self with the asterisk box. I have set the realm to asterisk but still
nothing.
Does any one know what proper MIB variables to set and their full path
Hello all!
Has anyone expirienced the following:?
With an IAXclient softphone (like diax/iaxcomm/etc) Dialing to the PSTN
(zap) or a SIP device has no problems .. but when I make calls between 2
softphones I have weird problems
in about 4 out of 10 IAX-2-IAX softphone calls I get a big
Hi,
- Original Message -
From: [EMAIL PROTECTED]
Has anyone expirienced the following:?
With an IAXclient softphone (like diax/iaxcomm/etc) Dialing to the PSTN
(zap) or a SIP device has no problems .. but when I make calls between 2
softphones I have weird problems
in about 4 out of
I'm having an issue with my current configuration. I have a single
PSTN line connected to an X100P and a couple IAX trunks to NuFone and
VoipJet. When I make an outbound call it doesn't properly detect if
my PSTN line is in use with another call and then overflow to my
outbound IAX
The mediatrix configuration tru the unit manager software is like going
tru hell.My mediatrix unit is failing when it come to registering it
self with the asterisk box. I have set the realm to asterisk but still
nothing.
Does any one know what proper MIB variables to set and their full path
I see I am using a quite old version of DIAX (I am more an iaxcomm user
where I DO use the newest version of :-)
I will make some tests with the newest version of DIAX today or tomorrow
and get back on this!
Regards, Niels
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Title: [EMAIL PROTECTED]
Is there a forum for [EMAIL PROTECTED] Its a great Asterisk option but I have some questions and this forum doesnt seem the right place to ask.
Bill Seddon
Lyquidity Solutions Limited
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Title: [EMAIL PROTECTED]
There is a forum type thing on its
sourceforge page http://sourceforge.net/projects/asteriskathome/
under the forum tab.
Hope it helps.
C
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bill Seddon
Sent: 27 February 2005 14:45
To:
In article [EMAIL PROTECTED],
Clay Reiche [EMAIL PROTECTED] wrote:
I didn't realize that the stable branch was never added to... So it will
NEVER have any more features than it currently has???
That is true for the 1.0 stable branch. But it doesn't mean there will
never be another stable
Title: [EMAIL PROTECTED]
Its a pretty active forum too,
though if you have AMP related questions then sometimes better to ask them in
the AMP sourceforge forum
https://sourceforge.net/forum/?group_id=121515
Cheers,
Dean
From:
[EMAIL PROTECTED]
[mailto:[EMAIL
Hi, thanks for the reply,
I finally got asterisk to register my mediatrix, I am now able to dial
an analog phone (connected to the mediatrix, which in turn is connected
to *) from a soft phone (X-lite).I made a typo when creating the
corresponding user on my asterisk box, after i corrected that,
I've added an introduction article about the ARA on my web site
http://www.voip-forum.com/
The same text is now also added to CVS head as README.realtime.
On the same site, you will also find the news item about how we used
Asterisk for a call from an airline jet above Greenland to Stockholm,
On Sun, Feb 27, 2005 at 04:14:46PM +0800, Steve Underwood wrote:
Questions keep comming up about this, so I started writing something at
http://www.soft-switch.org/foip.html . I think I covered the FAX over
VoIP issues fairly completely. T.37 is pretty simple to explain. There
is rather
Hello Steve.
It's an excellent read.
In two places you mention that V.34-Fax is 28,800 bps. Actually,
V.34-Fax has speeds ranging from 2400 baud to 33600 baud all using
V.34. And, while most V.34 connections are going to not probably be
more than 28,800 bps, I have seen sustained analog V.34
On 2005.02.27 04:26 Rich Adamson wrote:
Back in the olden days, I recall several modem vendors bundling PC fax
software with their products. All of those old Win v3.1 apps created
a fax file (eg, pdf or otherwise) that could be distributed via email.
Well, I don't remember any of them doing PDFs.
Title: DISA and a long delay; ideas?
Hi,
I have just setup a DISA setup whereby people can dial in, authenticate, are given a dialtone and can then call out.
Everything works however there is a 10 second delay after the user enters the number and presses #, until the system does anything.
On 2005.02.27 08:34 Martijn van Oosterhout wrote:
Hi,
I read it and found it very enlightening. I do have one question
regarding Modems don't like relativity. It says modems need a
constant delay; is there a limit to what it can handle. For example,
would it be possible to configure a jitterbuffer
[EMAIL PROTECTED] wrote:
I didn't realize that the stable branch was never added to...
So it will NEVER have any more features than it currently has???
1.0 STABLE will never have any more features.
1.2 STABLE will be released in the next 3 to 6 months, and it will
include all features that
Title: DISA and a long delay; ideas?
Hi,
I have just setup a DISA setup whereby people can dial in, authenticate,
are given a dialtone and can then call out.
Everything works however there
is a 10 second delay after the user enters the number and presses #, until the
system does
On Sun, Feb 27, 2005 at 09:10:48AM -0800, Lee Howard wrote:
Fax cannot handle a one-second delay. As Steve mentions in the
article, per-spec fax has some timings (particularly silence in
direction switching) set at 75 ms +/- 20 ms. So if the delay gets
much larger than 75 ms, then there's
[EMAIL PROTECTED] wrote:
I've added an introduction article about the ARA on my web
site http://www.voip-forum.com/
The same text is now also added to CVS head as
README.realtime. On the same site, you will also find the
news item about how we used
Asterisk for a call from an airline jet
Hello All,
My test server has a dedicated public IP and was set up using the article
from ONLamp:
http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html
except that I fixed the mailbox entries in the sip.conf.
Can some one please tell me what this means?
-
Ok all here is a
strange one..
I have a TDM400P
with 3 fxo modules. I can very rarely make an outbound call to the
PSTNabout once every 10 tries. However if I use a analog phone pluged
into the same phone line as one of the tdm channels say channel 4, and I place
the analog phone
Put a 'w'ait in your dial command. * is
probably dialing too quickly after going off-hook.
Lyle
- Original Message -
From:
Guy C. Guckenberger
To: asterisk-users@lists.digium.com
Sent: Sunday, February 27, 2005 12:00
PM
Subject: [Asterisk-Users] Outbound
[snip]
Try:
exten = 21,1,Dial(${phone1},20,rw)
Options should be combined.
and
exten = 21,1,Dial(${phone1},20,rwL(30:24:6))
This limits the call to five minutes, warning every 60 seconds when four
minutes are remaining. Keep in mind that time is specified in
I found an old Dialogic card in an abandoned PC, that I think is a
Dialogic D/41EPCI based on some googling.. The lspci output says:
00:09.0 Bridge: PLX Technology, Inc. PCI - IOBus Bridge (rev 01)
Subsystem: Dialogic Corp: Unknown device 0529
I'm just getting started with Asterisk to
All the stuff on features.conf doesn't work for me too... I press *1 or # to
transfer a call and I get nothing...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joseph
Sent: Domingo, 27 de Febrero de 2005 12:21 p.m.
To: Asterisk Users Mailing List -
Any computer fax application that I know of will write the image to
file, modern ones usually use TIFF. HylaFAX, mgetty+sendfax, efax,
spandsp, etc. Then you just write the glue to make it deliver that fax
image by e-mail. HylaFAX already has that built-in, and just requires
some minor
Hello,
WavePad worked perfectly in the free version. Thank you.
l.
In data Sat, 26 Feb 2005 11:47:56 -0500, mattf [EMAIL PROTECTED]
ha scritto:
The free utility WavePad for Win32 will play and edit GSM files as well:
http://www.nch.com.au/wavepad/
To convert to/from GSM on Win32 you can use
Guy C. Guckenberger wrote:
Ok all here is a strange one..
I have a TDM400P with 3 fxo modules. I can very rarely make an
outbound call to the PSTNabout once every 10 tries. However if I
use a analog phone pluged into the same phone line as one of the tdm
channels say channel 4, and
Guys.. which free softphone is the best,grandest,most recommended one out
there? based on your own experiences..
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Hello All,
I have a macro and want to jump to another macro if a conditition is true or
false.
Asterisk is jumping to the next macro, but then the {ARG1} variable is not
working anymore.
part of config:
[macro-default]
exten = s,1,DBGet(do-not-disturb=DND/${ARG1})
exten =
I guess you should do:
[macro-default]
exten = s,1,DBGet(do-not-disturb=DND/${ARG1})
exten = s,2,GotoIf($[${do-not-disturb} = YES]?200)
...
exten = s,200,Macro(do_not_disturb,$ARG1) ; Call the macro, do not
jump directly like if it was a context
[macro-do_not_disturb]
exten = s,1,Wait(2)
exten =
You need to tell asterisk to stay in the media path. You must add 't'
to the Dial options:
exten = 21,1,Dial(${phone1},20,trwL(30:24:6))
Or set canreinvite=no in your sip peer definition.
Julian J. M.
but even adding it and commenting out automon = *1 didn't work.
and of course I
Yeah Im sure they are FXO. They came from Digium install in 2,3,4.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Richard
Lyman
Sent: Sunday, February 27, 2005 1:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
You wouldn't happen to know how to do this would you?
I currently have a box with both hylafax and asterisk installed.
asterisk handles the dedicated voice lines over a t100p and
hylafax handles the dedicated fax lines over a 4port serial card
with external modems.
It would be really nice if I
Hi,
how can I see which codec a channel is currently using? I havn't found any
command to show this.
regards
Jens
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Hi
how do i set an SIP users to make outgoing calls
that is worth only $5. if they exceed $5 they can't make any calls. what i need
is not a calling card, but to limit outgoing calls for SIP users depedning
on a value i give.
I use realtime asterisk.
Thank You
Kanishka
Well, I thought I had my problem solved, but it is acting up again.
Hopefully this time I can provide enough information.
What I have is an * box setup with one X100P and TDM400 with one FXO and
one FXS. For my regular setup with interfacing with my PSTN and my
entire house with analog phones,
On Mon, 2005-02-28 at 05:58, Riphagen, Ferdy wrote:
Hello All,
I have a macro and want to jump to another macro if a conditition is true or
false.
Asterisk is jumping to the next macro, but then the {ARG1} variable is not
working anymore.
Try SetVar(SAVEARG=${ARG1}) in one macro then
On Fri, Feb 25, 2005 at 09:15:19PM +0100, Martijn van Oosterhout wrote:
You misunderstand. Ofcourse I need to run the register program on the
machine itself. The point is I build them from images and every now and
then I roll out a new image. My question is, what do I need to preserve
from the
asterisk_on_oelf wrote:
Hi,
how can I see which codec a channel is currently using? I havn't found any
command to show this.
regards
Jens
sip show channels
iax2 show channels
--
Kristian Kielhofner
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Hi,
I just tried your config setup out, seemed to work great.
I guess the reload scripts etc are a work in progress :p
I could edit files just fine though, threw the scrips in my /www dir and
didn't tweak anything else, I guess its all done due to my phpconfig
installation.
Drop us a line if
On Sun, 27 Feb 2005, C. Tomlinson wrote:
I have just setup a DISA setup whereby people can dial in, authenticate, are
given a dialtone and can then call out.
Everything works however there is a 10 second delay after the user enters
the number and presses #, until the system does anything.
I agree. The following commands may also be of use...
.
.
exten = s,3,DigitTimeout,5 ; Set Digit Timeout to 5 seconds
exten = s,4,ResponseTimeout,10 ; Set Response Timeout to 10 seconds
.
.
- Original Message -
From: Greg Hill [EMAIL PROTECTED]
To: Asterisk Users Mailing List -
On Sun, 2005-02-27 at 19:23 +, Julian J. M. wrote:
You need to tell asterisk to stay in the media path. You must add 't'
to the Dial options:
exten = 21,1,Dial(${phone1},20,trwL(30:24:6))
Or set canreinvite=no in your sip peer definition.
Still didn't work.
I had can
I've been playing with a variety of them over the past month.
The 'candidates' I've got down to are X-Lite, Firefly and SJphone. They all
have strengths and weaknesses, and all behave differently behind my firewall
(STUN client differences?). So far, I am happiest with the performance of
Joseph wrote:
Note: I added all this section manually, when I compiled * 1.0.5 this
section wasn't there (I don't know why).
[featuremap]
;blindxfer = #1; Blind transfer
;disconnect = *0 ; Disconnect
;automon = *1 ; One Touch Record
;atxfer = *2
Hello,
The astGUiclient suite has it's own mailing list for questions like this:
https://lists.sourceforge.net/lists/listinfo/astguiclient-users
The easy fix is for you to set PHP globals to on and see if it works like
that first, also you could try making that directory writable.
MATT---
On Sun, 2005-02-27 at 15:53 -0600, Eric Wieling wrote:
Joseph wrote:
Note: I added all this section manually, when I compiled * 1.0.5 this
section wasn't there (I don't know why).
[featuremap]
;blindxfer = #1; Blind transfer
;disconnect = *0 ;
On Sun, 27 Feb 2005 12:57:37 -0600, Anton Krall
[EMAIL PROTECTED] wrote:
Guys.. which free softphone is the best,grandest,most recommended one out
there? based on your own experiences..
Shameless plug : http://www.marccharbonneau.com/asterisk/mediaxphone.php ;)
But it only works on Windows for
Joseph wrote:
On Sun, 2005-02-27 at 15:53 -0600, Eric Wieling wrote:
Joseph wrote:
Note: I added all this section manually, when I compiled * 1.0.5 this
section wasn't there (I don't know why).
[featuremap]
;blindxfer = #1; Blind transfer
;disconnect = *0 ;
I am trying to install a TE405P which is to be connected to E1 trunks.
I have set the jumpers to E1 (closed), but modprobe wct4xxp still fails
with the message: ZT_CHANCONFIG failed for channel 97 - which obviously
means the card is still in T1 mode. I have tried everything I can think
of, any
On Mon, February 28, 2005 8:33, Rod Bacon said:
I agree. The following commands may also be of use...
Actually I disagree, I'm running 2 different asterisk servers, one with
1.0.5 and the other with CVS and I noticed this last night, the cvs
version attempts to send within a reasonable time,
ok so I put the wait in and still have the same
results.
Extensions.conf
exten = s,5,SetCallerID(${OUTCID})
exten = s,6,Wait(2) -I
added this
exten = s,7,Dial(${OUT}/${ARG1})
exten = s,8,Congestion
exten = s,107,Macro(outisbusy)
Im still only getting out every
few
(responder para [EMAIL PROTECTED], caso contrario não
receberei)
Boa tarde,
Gentileza estaremos fazendo manutençao do servidor
e esqueci que a lista é baseada em IP já que no seu arquivo de zonas do
asteriskbrasil.org listas.asteriskbrasil.org esta apontando para um IP,
para
I just had problems getting through to this list so please accept this
test.
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From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Guy C.
Guckenberger
Sent: Sunday, February 27, 2005 5:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Outbound call on TDM400P
ok so I put the wait in
Many thanks, that was the problem.
I didn't paste the context that forwards the call into the DISA context; it
had this in:
...DigitTimeout,5
..ResponseTimeout,10
Doh!
It works great with the mobile number, as I can pattern match 10 digits:
-Original Message-
From: [EMAIL PROTECTED]
Guy,
I
think what Lyle meant was to put a wait as in dial -- wait ---
number.
Therefore the line is seized and then after a wait the number is
dialled.
Dave
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Guy C.
GuckenbergerSent: 27
Hello,
Does anyone know of a web
based client that can be used with g723, g729 codec that may integrate with
Asterisk at all?
I would consider commercial solutions as I understand g723 never
comes free
Thanks for any help
___
Jeez, I need to work out the shortcut to send an email which I keep pressing
by accident!!
-Original Message-
From: C. Tomlinson [mailto:[EMAIL PROTECTED]
Sent: 27 February 2005 22:48
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] DISA and a
hi,
could you please provide your zaptel.conf, it's probably a configuration
problem.
BSDR a crit :
I am trying to install a TE405P which is to be connected to E1 trunks.
I have set the jumpers to E1 (closed), but modprobe wct4xxp still fails
with the message: ZT_CHANCONFIG failed for channel
Hi,
I am running Asterisk 1.0.5 Stable, and changing the pattern matching e.g to
10 digits made it call out instantly :-)
Not sure what your problem is.
C
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Duane
Sent: 27 February 2005 22:18
To: Asterisk
This is getting VERY annoying.
Is there anyone in here that has access to the list administration to
delete the user below???
This is an automatically generated Delivery Status Notification.
Delivery to the following recipients failed.
[EMAIL PROTECTED]
This is an automatically
thanks for replying, here it the zaptel.conf:
span=1,1,0,ccs,hdb3
span=2,0,0,ccs,hdb3
span=3,0,0,ccs,hdb3
span=4,0,0,ccs,hdb3
bchan=1-15,17-31
dchan=16
bchan=32-46,48-62
dchan=47
bchan=63-77,79-93
dchan=78
bchan=94-108,110-124
dchan=109
loadzone= nl
defaultzone = nl
hi,
could you
Hi,
I guess I'd need to run Beronet quad and octo bri cards under bristuff to
get zaptel features (echo canceling, timing source) Am I right or could
I achieve this also with chan_misdn - their native driver ?
Running bristuff on Beronet cards is unsupported. Has anyone succesfully run
On Sun, 2005-02-27 at 19:39 +, Kanishka Somaratne wrote:
Hi
how do i set an SIP users to make outgoing calls that is worth only
$5. if they exceed $5 they can't make any calls. what i need is not a
calling card, but to limit outgoing calls for SIP users depedning on
a value i give.
I
1.0.5 is considered 1.0.x. Meaning, v1.0.x means CVS-HEAD at the
moment.
Thank you for explanation; now it is clear to me.
--
#Joseph
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Hello my name is Ilija Poznic and I have a problem.
My configuration is
1. Digium TDM4000P with one FXS.
2. AVM Fritz ISDN adapter (configured with capi).
When I connect to my ISP and then start *. Asterisks is registering me to SIP
provider iconnect. After that
at least 4 me.
Anyone knows what are the variables in an inbound IAX2 call who reflect the
actual codec and DNID, DNIS, original peer description, I'm only able to see
it during an iax debug
Timestamp: 3ms SCall: 1 DCall: 0 [66.98.146.34:5036]
VERSION : 2
CALLED
I have a recording studio in Brooklyn NY and I am seeking help in some
lessons and setting up. I have been learning linux and asterisk very
well thanks to lurking in the group and info on the voip-info. I would
like some one to come over and give me a tutorial, check my box out,
make sure
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