On Sun, 3 Jul 2005, Josef Seger wrote:
I have one other Dect phone connected to Digiums Card(TDM400P), an
Ericsson DT 260. The Ericsson phone only supports true swedish standard
CallerID (DTMF signalling before the first ring), and CallerID does not
work for this phone:(
I have measured
I'd like to these three things about asterisk:
1. How the asterisk program can be configured to run as a different user from
root.
2. what directories and files it must have read and right access to
3. Setup an asterisk group, which also has some of the rights the asterisk user
has rights to,
We are running * V1.0.9 on a demo box.
We have set up everything in our dialplan and we have a directory where we store
individual extension settings. That directory is called extensions-phones.d
and it contains a number of .conf files.
In my extensions.conf file I have put a
#include
Just a thought, but I seem to recall that in the dialplan, inlcude
and other similar statements are not prefixed by the hash character
(#). Try include = .
-Bryce
On Jul 4, 2005, at 00:05, [EMAIL PROTECTED] wrote:
We are running * V1.0.9 on a demo box.
We have set up everything in our
On Mon, Jul 04, 2005 at 06:55:42AM +, 8 hours after posting exactly
the same message, Obelix wrote:
I'd like to these three things about asterisk:
1. How the asterisk program can be configured to run as a different user from
root.
man asterisk. There is a switch -U
2. what
On Jul 4, 2005, at 12:05 AM, [EMAIL PROTECTED] wrote:
We are running * V1.0.9 on a demo box.
We have set up everything in our dialplan and we have a directory
where we store
individual extension settings. That directory is called
extensions-phones.d
and it contains a number of .conf files.
On Jul 4, 2005, at 12:17 AM, Bryce Chidester wrote:
Just a thought, but I seem to recall that in the dialplan, inlcude and
other similar statements are not prefixed by the hash character (#).
Try include = .
-Bryce
You're thinking of contextual includes, not filesystem includes --
which
On Mon, Jul 04, 2005 at 02:05:17AM -0500, [EMAIL PROTECTED] wrote:
We are running * V1.0.9 on a demo box.
We have set up everything in our dialplan and we have a directory where we
store
individual extension settings. That directory is called extensions-phones.d
and it contains a number of
I think it might have. My understanding is BV does not require username and secret for incoming calls from BV to Asterisk, so a patch was written and then included in Asterisk to fix this.
I have been testing for 4 weeks now, and have been shot down once in the digium bug tracker and pretty much
Hello
AstLinux seems quite suited for my use.
Can you configure more incoming port via a web interface?
I'd like to install it to a normal hdd. Can that cause any problems?
BR
Amund Nygaard
-Opprinnelig melding-
Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne av Kristian
Tzafrir,
Do you have patch description file which explains what the different patches do?
Thanks,
Brent
Quoting Tzafrir Cohen [EMAIL PROTECTED]:
On Mon, Jul 04, 2005 at 02:05:17AM -0500, [EMAIL PROTECTED] wrote:
We are running * V1.0.9 on a demo box.
We have set up everything in our
There's a tiny bit of new info available:
asterisk only strips off the trailing digit of calls coming from
ANALOG lines; calling from e.g. a (fully digital) gsm mobile, I get
the 3 digit extension as it should be.
does this ring a bell for anyone?
On 7/3/05, no name [EMAIL PROTECTED] wrote:
so
On Mon, Jul 04, 2005 at 03:13:26AM -0500, [EMAIL PROTECTED] wrote:
Tzafrir,
Do you have patch description file which explains what the different patches
do?
Extrat the asterisk_*.diff.gz using
zcat that_file.diff.gz | patch -p1
in an empty directory. This will create a subdirectory
Hello anyone who can help
I have two Asterisk boxes with identical hardware (Dev Production). I
recently rebuilt the DEV box using Fedora Core 3 and the latest CVS Head.
The hardware is an Intel CA810e, onboard everything with a PIII processor.
The config is pure VOIP using IAX2 ilBC with
We just released a new version of the idefisk iax2 softphone, version
1.21 beta, available for download at
http://www.asteriskguru.com/tools/idefisk_beta.php
Some bugs were fixed, some new bugs might have been introduced :) - The
problem with delays is finally gone!!!
(one of the bugs was a
Would just like to warn everybody for Wengo.fr
Once you sign up there is no possibility to remove your credit card and
even though you send them resignation letters they keep charging your
credit card.
Now I understand what they mean when they say `unlimited
subscription'.
Robert Goodyear wrote:
I am confused about one of my installed server
The dial plan seems to be ok, but sometimes NOTHING happens if I
try to dial an extension (from X-Lite), next time it is done.
X-Lite does not have a tone, nothing and does also have no time
out. It seems it is not
Hi Yair,
Thanks for your email.
Unfortunately no reply or response from anyone yet.
Please let me know if you hear anything - I'm also battling to resolve the
problem.
Kindest regards
David Wilson
___
D c D a t a
Tel +27 33 342 7003
Fax +27 33 345 4155
Cell +27 82
Hi,
Sorry to re-post, but I'm still having hassles with ztdummy. I'm using
kernel 2.6.11.4-21.7-smp and Asterisk 1.0.8 on SuSE 9.3
The first 3 makes (see below) for zaptel work out ok - but the
ztdummy.ko (etc) files *are* created even though I haven't yet
uncommented ztdummy in Makefile.
Hello,
they are successful to start asterisk, task that the error that I had
previously
had had to a configuration problem.
Start asterisk in modality consol and when two softphone speaks is not felt
well, and I have the following error:
-- Registered SIP '1000' at 10.0.0.7 port 5060 expires
There is going to be another great Paris lunch with Mark this Friday.
The restaurant will probably be in the southern part of Paris in the
14th arrdt. like last time.
Please contact me off list if you are able to attend.
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Bernie Ott wrote:
There's a tiny bit of new info available:
asterisk only strips off the trailing digit of calls coming from
ANALOG lines; calling from e.g. a (fully digital) gsm mobile, I get
the 3 digit extension as it should be.
Try an extension with four digits and one with two. You
hi,
I have dabian sarge with asterisk 1.0.7 and chan_capi 0.3.5 with AVM
fritz card.
I would like use detecion of fax, but it don't work.
So, i would like know if it's possible to work fax detection with this
card? And if it's possible how??
Thanks you for your help
Hi;
Is it possible for me to use my cisco7920 with Asterisk in
any way?
Cheers
Betul
Onemli not : Bu e-mail iletisi, sadece adreste belirtilen kisi veya kurulusun kullanimini hedeflemekte olup, mesajda yer alan bilgiler kisiye ozel ve gizli olabilir, yasalar ya da anlasmalar
Sure!
http://www.voip-info.org/tiki-index.php?page=SCCP-HOWTO2
regards, roland
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Betül Gözlükoglu
Sent: Monday, July 04, 2005 12:19 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Once you sign up there is no possibility to remove your credit card and
even though you send them resignation letters they keep charging your
credit card.
Now I understand what they mean when they say `unlimited
subscription'.
That's been true of every cellphone and Internet company I've
Hi Ronald,
* chopping after 10 digits is fine - our number is 12345673 digit
ext though so there's a total of 9 digits.
On 7/4/05, Ronald_Wiplinger [EMAIL PROTECTED] wrote:
Bernie Ott wrote:
There's a tiny bit of new info available:
asterisk only strips off the trailing digit of calls
Hello
I am at extension 200 and I know there is a
voicemail message waiting. I dial *97 and am prompted for the
password. I enter 1234 which I have set as my voicemail password.
What can I do to troubleshoot?
Angus ComberItel Office Software Ltd5
Enmore GardensLondon, SW14 8RFTel: 020
Thanks for link roland. The document mentions about firmware for
Cisco 7920 ? Is it SIP firmware ? I asked the cisco seller for
Sip firmware but they said sip firmware is unavailable for 7920?
Would Creating files mentioned on the document be enough for configuration?
Thanks again
I have found that if I dial from another extension *98 and select extn 200
and enter password 1234 it works. So is it something to do with
configuration on my IP Phone? It is a Grandstream GXP2000 running:
Software Version: Program-- 1.0.0.3Bootloader-- 1.0.0.3
Anyone got any ideas?
Hi
I've a mgcp fon (swissvoice IP10s) behind a NAT router. Configured is
NAT for both in/out going on port 2427. Now I got the following mgcp
debug messages when i try mgcp audit endpoint endpoint
--
from 172.16.98.57:2427
Verb: 'RSIP', Identifier: '5346',
On Mon, Jul 04, 2005 at 05:19:39PM +0800, Ronald_Wiplinger wrote:
Robert Goodyear wrote:
I am confused about one of my installed server
The dial plan seems to be ok, but sometimes NOTHING happens if I
try to dial an extension (from X-Lite), next time it is done.
X-Lite does not have a
On 7/4/05, Ronald_Wiplinger [EMAIL PROTECTED] wrote:
Robert Goodyear wrote:
I am confused about one of my installed server
The dial plan seems to be ok, but sometimes NOTHING happens if I
try to dial an extension (from X-Lite), next time it is done.
X-Lite does not have a tone,
Hello,
I would like to use Intel Blade machine for running Asterisk.
Is there anyone who already use Intel Blade server for running Asterisk? Can
you please explain, how perform Asterisk with Intel Blade machine?
I would appreciate for giving me feedback regarding this
issue.
The one that looks identical is selling at $180 from
www.broad-tel.com/index_en.php
On 7/1/05, Richard Malcolm-Smith [EMAIL PROTECTED] wrote:
If it does materialize, im up for 3 or 4 of them at that price.
Huddleston, Robert wrote:
Well poo - if I can use that word I'm one of those poor
Don't know about your asterisk box, but i've made
Max talk to an IPOffice (Avaya) using H323.
in brief, it was a case of going to modules in the
Max and adding a 'virtual trunk module' , and in the IPOffice create an IP
Trunk, most of the fields in both systems are self evident -
Nahid Hossain wrote:
I would like to use Intel Blade machine for running Asterisk. Is
there anyone who already use Intel Blade server for running
Asterisk? Can you please explain, how perform Asterisk with Intel
Blade machine?
We've had Asterisk running on a blade for some time. Blades as
Hi all.
I've got an Auerswald 4410USB (
http://www.auerswald.de/int/products/c4410usb.htm ) which I connected
to my 2nd ZAP interface (s0 - Zap) via Crossoverr ISDN cable (which
I crimped myself, I guess that's not the source of my trouble).
Now what is annoying however, there is a very loud and
On Mon, Jul 04, 2005 at 02:15:36PM +0200, Andreas Sikkema wrote:
Nahid Hossain wrote:
I would like to use Intel Blade machine for running Asterisk. Is
there anyone who already use Intel Blade server for running
Asterisk? Can you please explain, how perform Asterisk with Intel
Blade
Betül Gözlükoğlu wrote:
Thanks for link roland. The document mentions about firmware for
Cisco 7920 ? Is it SIP firmware ? I asked the cisco seller for
Sip firmware but they said sip firmware is unavailable for 7920?
Would Creating files mentioned on the document be enough for configuration?
[EMAIL PROTECTED] wrote:
We've had Asterisk running on a blade for some time. Blades as
such can be used but with a couple of restrictions:
- There's probably no room for PCI cards, so no zap hardware
- Check the kind of USB supported on the board (UHCI vs OHCI,
for ztdummy support, see
hello
i am trying to work with radiusclient form portaone.
but i have some problems in installation. when i am
trying to use example from
http://www.voip-info.org/tiki-index.php?page=PortaOne+Radius+auth
error sip debug
Can't locate Asterisk/AGI.pm in @INC (@INC contains:
I think it might have. My understanding is BV does not require username and
secret for
incoming calls from BV to Asterisk, so
a patch was written and then included in Asterisk to fix this.
I have been testing for 4 weeks now, and have been shot down once in the
digium bug tracker
and
I have a remote installation that connects via IAX from my office pbx.
When I call an extension on the remote pbx, after the dial period, the
call is terminated. Nothing I do in configuration of that extension
seems to matter:
-- Executing NoOp(IAX2/[EMAIL PROTECTED]:4569-5, Dial
710) in
I am trying the setup the TDM01B card. 1 FXO. I
connected it to a regular home line. in the
/etc/zaptel.conf, I have
fxsls=4
In the /etc/asterisk/zapata.conf
I have:
signaling=fxs_ls
language=en
group=1
context=default
channel = 4
When I start asterisk, I get this error:
Joseph [EMAIL PROTECTED] :
Would Creating files mentioned on the document be enough for configuration?
You will need to get the sccp channel software here:
http://chan-sccp.berlios.de/
Download it and follow the instructions included.
Please keep in mind that is under development.
The
I have a remote installation that connects via IAX from my office pbx.
When I call an extension on the remote pbx, after the dial period, the
call is terminated. Nothing I do in configuration of that extension
seems to matter:
-- Executing NoOp(IAX2/[EMAIL PROTECTED]:4569-5, Dial
Talk to BT about getting an ISDN30 line put in... you'll get some sort
of guaranteed quality and it'll be much better than a pure SIP solution.
Talk to anyone APART from BT, their pricing is much more than OLOs etc.
Who else can I order an ISDN30 line from in the UK?? I am looking for one at
Chris Mason (Lists) wrote:
-- Executing NoOp(IAX2/[EMAIL PROTECTED]:4569-5, Dial
710) in new stack
-- Executing Dial(IAX2/[EMAIL PROTECTED]:4569-5,
SIP/710|30|tr) in new stack
-- Called 710
-- SIP/710-4841 is ringing
== Spawn extension (office, 710, 2) exited non-zero on
Am Mo, den 04.07.2005 schrieb Mathias Röhl um 13:40:
ok, *done*, my fault, error in NAT configuration...
regards
mathias roehl
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To
Hi,
I'd like to understand what should i do to use G729 codec in a legal way,
how do I order licences ? to whom ? how do I install them on asterisk etc ?
thanks in advance ,
jl
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find it here:
http://www.digium.com/index.php?menu=product_detailcategory=extrasprod
uct=G729
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jean-Louis
curty
Sent: Monday, July 04, 2005 3:45 PM
To: Asterisk Users Mailing List - Non-Commercial
Dear All :
We are using [EMAIL PROTECTED]
We did install [EMAIL PROTECTED] on a Virtual Machine ..
All the SIP Calls are working fine ..
But - We noticed that the codec on MeetMe Application is not
working probably How can we solve this problem ???
Thanks ,,
Mohamed Farid ,,
Jean-Louis curty wrote:
Hi,
I'd like to understand what should i do to use G729 codec in a legal way,
how do I order licences ?
On Digium's website.
http://www.digium.com/index.php?menu=product_categorycategory=extras
to whom ?
Digium.
how do I install them on asterisk etc ?
did you use the zaptel drivers? you need a timer interface for meetme
application! use ztdummy!
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mohamed
Farid
Sent: Monday, July 04, 2005 3:59 PM
To: Asterisk Users Mailing List - Non-Commercial
On 7/1/05, Mark Charlton [EMAIL PROTECTED] wrote:
On 7/1/05, Wilson Pickett [EMAIL PROTECTED] wrote:
I have been trying for a while to find a way to get an SMS send when I
receive a voicemail into my asterisk system. I don't want to send an
SMS if the caller doesn't leave a message. I
On Jul 2, 2005, at 11:35 PM, Jay Milk wrote:
That's all doable. How many residents are you talking about? -- could
take quite a while to call them all.
Tell me about it -- we're doing it manually now!
Considering you have outlay in
hardware, phone-cost, utilities (a 100W computer draws
Hello all,
First of all, let me apologize about the length of this message, but I suppose
it was necessary to include the details.
I've spent quite some time already trying to get the call transfer function to
work on my Asterisk installation. Let me first describe the general situation
of
Hi,
How could I change the
defaultpermissions for voicemails?
When I try to installthe patch
mentionedat http://www.voip-info.org/tiki-index.php?page=Asterisk+gui+vmail.cgi,
I get the following response:
patch
voicemail.patch
patching file app_voicemail.cHunk #1 FAILED at
39.Hunk #2
Does anyone know how to join two .wav audio files via the command line
in Linux for playback with Asterisk?
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To UNSUBSCRIBE or update
On 15:15:19 July 04, 2005 [EMAIL PROTECTED] wrote:
Joseph [EMAIL PROTECTED] :
Would Creating files mentioned on the document be enough for
configuration?
You will need to get the sccp channel software here:
http://chan-sccp.berlios.de/
Download it and follow the instructions
On Monday 04 July 2005 16:31, Kevin Kiely wrote:
Does anyone know how to join two .wav audio files via the command line
in Linux for playback with Asterisk?
Kevin, you might want to try Sox, see http://sox.sf.net for more information.
I'm not sure it can join or concatenate audio files, but I
Frank Schoep wrote:
Hello all,
First of all, let me apologize about the length of this message, but I suppose
it was necessary to include the details.
I've spent quite some time already trying to get the call transfer function to
work on my Asterisk installation. Let me first describe the
On Mon, Jul 04, 2005 at 10:31:13AM -0400, Kevin Kiely wrote:
Does anyone know how to join two .wav audio files via the command line
in Linux for playback with Asterisk?
install the package sox (should be part of most distros).
sox infile1 [...] outfile1
e.g: sox in1.wav in2.wav in3.wav
Is the X100P FXO PCI Card capable of detecting a fax, answering the
call, and then emailing the fax content to an email address?
Thanks.
Paul.
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Not sure why I see *97 and *98 here, but I would check your dtmfmode= line
in sip.conf. Often times, using rfc2833 works when inband or sip-info
doesn't.
See http://www.voip-info.org/tiki-index.php?page=Asterisk+sip+dtmfmode
Chris Coulthurst
[EMAIL PROTECTED]
|-Original Message-
Paul Goodyear wrote:
Is the X100P FXO PCI Card capable of detecting a fax, answering the
call, and then emailing the fax content to an email address?
Thanks.
Paul.
Yeah I recall there is a module for asterisk to do this. Search the list
archives.
Is the X100P FXO PCI Card capable of detecting a fax, answering the
call, and then emailing the fax content to an email address?
I haven't tested the digium x100p for several months, but I believe
it has the same issue as the TDM card relating to missed data frames
across the pci bus.
On 7/4/05, Rich Adamson [EMAIL PROTECTED] wrote:
Is the X100P FXO PCI Card capable of detecting a fax, answering the
call, and then emailing the fax content to an email address?
I haven't tested the digium x100p for several months, but I believe
it has the same issue as the TDM card
Is the X100P FXO PCI Card capable of detecting a fax, answering the
call, and then emailing the fax content to an email address?
Yes, using spandsp. My own experience has beent his works on receive
80% of the time. Some machines never are able to sync with it somehow.
That's just my experience.
Would it be possible to see your fax sections in your Extensions.conf
file to see what you have there?
This is a good place to start:
http://scottstuff.net/blog/articles/category/Asterisk?page=4
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On Sun, 2005-07-03 at 12:31 -0400, Leo Burd wrote:
Hello there,
I'm trying to configure my voicemail system and I have a couple of
questions:
* Is real-time voicemail already working? If so, where is it that I
should specify the database name, user and password? Where can I get
more
Hey there folks --
I have been continuing development on the multi-threaded, c-based Asterisk
Manager Proxy program, AstManProxy.
I've incorporated several ideas I received at the recent Astricon Europe,
including:
- Supports proxying of multiple Asterisk servers at once
- Abstracted,
HI,
usually those errors arise when you try to use a patch with a different
version. It happened to me when I tried to patch Asterisk 1.0.7 with a
different version patch.
Giorgio Incantalupo
Victor Alvarez wrote:
Hi,
How could I change the default permissions for voicemails?
When I
On Sun, 3 Jul 2005, Subhi S Hashwa wrote:
Telephony is a critical system to a business, if your phone system is down
your
business is as good as dead. If it costs me £600 for OS with support for 3
years, it's a price worth paying in the grand scale of things. You're buying
Xeon server,
Yair,
When you have an older version you can try to use DBput/DBget (if still
working, because set will replace it in CVS)
Set(DB(CFIM/${CALLERIDNUM})=u${CALLERIDNUM})
will be;
DBput(CFIM/${CALLERIDNUM}=u${CALLERIDNUM})
Set(CFIM=${DB(CFIM/${ARG1})})
will be;
DBGet(${DB(CFIM/${ARG1})
but not used
make[1]: *** [cdr_tds.o] Error 1
make[1]: Leaving directory `/usr/src/asterisk/cdr'
make: *** [subdirs] Error 1
I checked the version of FreeTDS, it is freetds v0.64.dev.20050704. (upon
seeing on another post checking version of FreeTDS, I updated it to the most
recent one)
I checked
Hello,
I was again asked to try to add support for presence
(SUBSCRIBE/NOTIFY) and IM using SIMPLE. I have few questions:
a.) are there any, at least partial projects, patches, anything,
that at least partly implements presence and/or IM to chan_sip? I
don't care about presence on other
If I recall correctly (and I'm not a BV user), incoming sip calls to asterisk do not use all the parameters as one would expect. I thinkOlle posted something in the last month or two that such incomingcalls attempt to find the first context that matches IP Address (orsomething like that)
Using a vanilla 2.6.13-rc1 kernel zaptel compiles but then when
modprobing dmesg gives:-
zaptel: Unknown symbol class_simple_device_add
zaptel: Unknown symbol class_simple_destroy
zaptel: Unknown symbol class_simple_device_remove
zaptel: Unknown symbol class_simple_create
Loads OK with 2.6.12.2
On Mon, 4 Jul 2005, sylvain garcia wrote:
hi,
I have dabian sarge with asterisk 1.0.7 and chan_capi 0.3.5 with AVM
fritz card.
I would like use detecion of fax, but it don't work.
So, i would like know if it's possible to work fax detection with this
card? And if it's possible how??
Yes, I know all of that, the problem is asterisk is NOT trying to
match anything except the IP from Broadvoiice. So all calls from BV
will be cached to the phone number and context of the first BV call.
Asterisk will not look at the phone number or context of the other BV
number when they
on spamdsp page found thisIt seems possible for the libtiff library to fall over when handling some
bad TIFF files. If spandsp is being used with Asterisk, this might bring the
entire PBX down. So far only one person has reported this. Recent security
update patches for libtiff 3.5.7, 3.6.0, and
The /var/log/messages lists:
kernel: Module 0: Not installed
kernel: Module 1: Not installed
kernel: Module 2: Not installed Jul 3 22:21:10
kernel: Module 3: Installed -- AUTO FXO (FCC mode)
kernel: Found a Wildcard TDM: Wildcard TDM400P REV I
(4 modules)
the ztcfg -vv:
Zaptel Configuration
I am confused about one of my installed server
The dial plan seems to be ok, but sometimes NOTHING happens if I
try to dial an extension (from X-Lite), next time it is done.
X-Lite does not have a tone, nothing and does also have no time
out. It seems it is not connected to the server.
Hello Everyone,
This is my first post, and this is my problem :-).
I have a [EMAIL PROTECTED], work excellent (only internal users), but i need
outbound calls. One person give me an access to his Cisco 5300 Media
Gateway, he give me a dial rule and the router ip address.
I've created
David Wilson wrote:
I have a weird thing happening sometimes with users calling from a
GrandStream phone through Asterisk onto a PSTN.
Sometimes after a user hangs up a call on a GrandStream phone the phone
starts ringing after a couple seconds.
When the call is answered there is no one there.
Hi all,
I need some help/guidance on writing the specs needed on a project
that will be scaling up to 10,000 users.
I will have some T1's to provide PSTN connectivity, and all the users
will be SIP and/or H323 phones. Services offered will include
conferences, voicemail (20 megs per users), etc
I have a Gizmo account working perfectly in my Xten Eyebeam, so there
should be no problem using it for Asterisk. You already have the
username (1747...etc) and your password, the proxy is
proxy01.sipphone.com (or you can sniff packets to see where SIP
messages are being sent to).
On 6/30/05,
when i dial an extension and the time on ring expiry how to know
if called party is bussy or not answer.
thanks-- David Romero##
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Hi the list,
ztmonitor 3 -v start ztmonitor in graphical mode on Zaptel device #3.
What is the correct syntax for dBm or voltage ?
TIA
Best Regards,
Francois BERGERET,
France.
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${DIALSTATUS} will tell you, also rtfm that will help you a lot.
The wiki is at: www.voip-info.org
Google is at: www.google.com
Browse this list: lists.digium.com
If you want to search the list with google, then type in
site:lists.digium.com when you enter your search terms on google.
On 7/4/05,
El lun, 04-07-2005 a las 09:53, Paul Goodyear escribió:
Is the X100P FXO PCI Card capable of detecting a fax, answering the
call, and then emailing the fax content to an email address?
For me work fine this card, the spanDSP and the
Follow these steps:
/etc/asterisk/zapata.conf
Hi list!
I have several boxes running asterisk as I want, no problems there but the
one thing I haven't sorted out is how to properly start asterisk on boot
time.
This is probably a n00b class question but how do I properly set this up
(I didn't find any docs on this).
The zaptel script
Hi, I'm using asterisk with a digium TDM400P: I can't send/recive faxes
with a Philips fax machine.
It seems that the fax machine doesn't recognize the carrier.
How can I see the spandsp logs? I've enabled debug on the asterisk CLI,
but I can't see any output while the txfax/rxfax application
Dave Cotton wrote:
Using a vanilla 2.6.13-rc1 kernel zaptel compiles but then when
modprobing dmesg gives:-
zaptel: Unknown symbol class_simple_device_add
zaptel: Unknown symbol class_simple_destroy
zaptel: Unknown symbol class_simple_device_remove
zaptel: Unknown symbol class_simple_create
You start to not make any sense, you posted a question like this:
i try to do the following:
1) call comes in on extension 999
2) caller should hear music (NOT MoH!!!)
3) a call should be initiated to SIP Phone 100
4) when SIP Phone 100 is answered, a sound file should be played to the
user at
hi ferdy,
again, thanks for all your help. I will try this and report back.
as for your questions:
1. my version is from stable, Asterisk CVS-v1-0-01/22/05-12:27:04
2. the line used that gets this database result is:
exten = 154,2,DBPut(DB(CFIM/${CALLERIDNUM})=${CALLERIDNUM})
which is, of
I think they were hoping that the client would connect to Asterisk,
which makes it kinda useless, really.. But connecting Asterisk to the
Gizmo network is handy.
--
Dana
On 7/4/05, Adrian A [EMAIL PROTECTED] wrote:
I have a Gizmo account working perfectly in my Xten Eyebeam, so there
should
there in the
future.
I have decided to update the installation to 1.0.9. However, during make,
I receive:
[...various errors...]
I checked the version of FreeTDS, it is freetds v0.64.dev.20050704. (upon
seeing on another post checking version of FreeTDS, I updated it to the most
recent
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