[EMAIL PROTECTED] <> wrote:
> you can do that. But are u installing qmail and * on
> same box. i wont recommend that. i use qmail and *.
> qmail is strictly for internet email. *
> is on separate server not exposed to Internet. * box
> also has sendmail. i hv configured sendmail to use
> smart h
you can do that. But are u installing qmail and * on
same box. i wont
recommend that. i use qmail and *. qmail is strictly
for internet email. *
is on separate server not exposed to Internet. * box
also has sendmail. i hv
configured sendmail to use smart host (qmail server).
This way its safe and
On 15 Jan 2004, kemal asad wrote:
> can anyone suggest a set of equipment i could get to check and test the
> cool functionalities of Asterisk. Computer , Phones, communication
> cards.
Digium sell an "Asterisk Developer's Kit" - for about US$180 you get an
FXO card and the 1-port version of thei
On Thu, 15 Jan 2004 07:14:00 -0500, Andrew Kohlsmith wrote:
>> Is * capable to use qmail as a MTA?
>> If so, how can I set it?
>
>It shouldn't be an issue, as qmail has the standard 'sendmail' binary
>included.
>
>Regards,
>Andrew
In My * box, it has a running and working qmail (with sendmail
Or use this http://www.cam.org/~noelbou/1-step.html
bkw
On Thu, 15 Jan 2004, Troy Settle wrote:
>
> I can't reproduce this either, but I do have the gsm codec installed (though
> WMP won't play a .gsm file).
>
> I play the wav49 files in Winamp with no issue.
>
> --
> Troy Settle
> Pulaski N
Hi all,
I'm in the process of building a * box for home and ran across the
vmail.cgi script. It installs suid root in order to allow access to the
voice mail boxes. I've never been fond of suid root and was looking for a
better method.
I've patched my installation to make everything in the vm
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> [EMAIL PROTECTED]
> Sent: Friday, 16 January 2004 10:13
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Voicetronix Openline 4 + asterisk
>
> Any one has documented how-tos for making voicetroni
Create a new wav49 on your system and play it.
bkw
On Thu, 15 Jan 2004, Warwick Ward-Cox wrote:
> I'm having the same problem.
>
> Warwick
>
> - Original Message -
> From: "Jim Flagg" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Thursday, January 15, 2004 5:39 PM
> Subject: [Ast
Same here. I can't recreate the problem. I think this is a windows media
player issue.
bkw
On Thu, 15 Jan 2004, Troy Settle wrote:
>
> I can't reproduce this either, but I do have the gsm codec installed (though
> WMP won't play a .gsm file).
>
> I play the wav49 files in Winamp with no issue.
I know, but as I mentioned in the inital post, I haven't been able to get the last 2
cvs versions I've pulled to run stable enough to test.
I've seen a 0.7.0 version number mentioned. Is there newer, mostly stable version of
code I should try that just hasn't been officially released?
jesse
Hello All,
I received my Adit 600 yesterday and I have an 8 port FXO CAC FXO card
installed. That is the only module in the CB. I have the config on the *
side correct, however I am not sure.
I had the system running great for about 2 hour, and then it seemed to be
having problems. Incoming ca
On Thu, 2004-01-15 at 19:18, [EMAIL PROTECTED] wrote:
>> I do not have any zaptel hardware on the Asterisk box, I could not have
>> meetme functioning. I did modify the Makefile in zaptel directory on
>> line 168 by including ztdummy as one of the modules to compile in.
try modprobe ztdummy
This
On Thursday 15 January 2004 20:02, T. Chan wrote:
> I have a fast question, I am running a few Asterisk systems, but I
> just noticed one thing quite peculiar. After I started
> "safe_asterisk", and when I ran PS or TOP, I could see 1 PID
> "safe_asterisk" and almost 10 PIDs "asterisk -vvvg -c" eve
Dear all,
Is * capable to use qmail as a MTA?
If so, how can I set it?
I'm using asterisk v0.5, and TDM30B (FXS), Wildcard X100P(FXO), and
x-lite(SIP softphone).
In zapata.conf, I put already callwaiting=yes. My PSTN doesn't not support
the callwaiting feature, so I don't expect the FXO is call wa
IF I want to play sound files,
1.) what format should it be? (*.au or *.wav)
2.) where should it reside?
3.) what syntax should I follow? Is
exten=>_.,102,Dial(SIP/[EMAIL PROTECTED],1,tHA(sound.au))
correct? I tried this and it doesn't work.
Thanks,
Brian West wrote:
Questions... happen to use webvmail?
Nope. All access is via a station dialpad. . .
This has been happening to her ever since we installed. It is really
freaky, because the higher-number messages are messages that she thought
she had deleted, and in her telling, "Then, days
At work, we just put in managed switches... one user had lots of
collisions, which is strange for a switched network... we set the
computer to full/100, and the switch to the same settings, and now it
doesnt have any more collisions...
DH
Rich Adamson wrote:
This is ifconfig on openbsd box:
f
Title: RE: [Asterisk-Users] capacity testing
ï
Hi
all, and Jesse
1. So,
you did get the experience of crashing all of a sudden with the "Disconnected
from Asterisk server" error message. I got both this and the segmentation error
when crashing. I am running the version of asterisk, libpri a
Hi,
I'm trying to bundle the powers of Asterisk and SER.
Asterisk for pabx functionalities and termination to landline/PSTN, and
SER as SIP Gateway/Proxy.
With my current configuration the SIP user just adds 0 as a prefix to a
number, and the call will go out to PSTN over Asterisk.
For this to wo
Thanx for all your help. I have been doing some research on shady dial and
also have been contacted by a few consultants, so hopefully I can have this
box up and running in the next few weeks.
thanx again
chris
From: [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: A
On Thu, 2004-01-15 at 18:27, Cameron Palmer wrote:
> http://www.intel.com/software/products/ipp/samples_table.htm#
>
> Has anyone taken a look at the value of these sample libraries for use in
> Asterisk.
Do you understand the GPL? Have you looked up the cost of those
libraries. They aren't fre
On Thu, 2004-01-15 at 17:44, Chris Albertson wrote:
> --- calvis <[EMAIL PROTECTED]> wrote:
> >
> >
> > I am real close to finalizing my hardware selection for my Asterisk
> > test
> > machine. I am going to use the following hardware:
> >
> > Dell 400SC w\Red Hat 9.0
> >
> > 1 - 4 Port TDM40
On Thu, 2004-01-15 at 19:18, [EMAIL PROTECTED] wrote:
> I do not have any zaptel hardware on the Asterisk box, I could not have
> meetme functioning. I did modify the Makefile in zaptel directory on
> line 168 by including ztdummy as one of the modules to compile in.
try modprobe ztdummy
__
Jesse Peterson wrote:
I am currently running Asterisk 0.5.0 under Redhat 9 on a single PIII 800 with 256megs RAM.
CVS UPDATE! That code is hardcore old.
Jeremy McNamara
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On Thu, 2004-01-15 at 18:27, Cameron Palmer wrote:
> http://www.intel.com/software/products/ipp/samples_table.htm#
>
> Has anyone taken a look at the value of these sample libraries for use in
> Asterisk.
They might be useful, but you still have to license the codecs from the
patent holder, th
> This is ifconfig on openbsd box:
> fxp0: flags=8843 mtu 1500
>
> I think this output shows that the fxp0 interface is on simplex mode.
>
> The voice degradation I referred was by using xlite soft phone. I open 2
> line similtaneously and dial to FWD and back to my incoming extension.
> Xlite
The freenum.org project wants to use your trunks! The freenum.org project is an ENUM
parallel tree, which has as an eventual goal the distribution of ENUM numbering in
nations or areas which due to political or other issues are not able to get secure,
inexpensive, or functional ENUM capabilit
I just moved my system over to a new server with * 0.7.1. The old machine was
using a cvs from August/Sep timeframe.
On the new machine I did an make samples but then ovewrote with tar files of the
production configs in the
/etc/asterisk
/var/spool/asterisk
/var/lib/asterisk
folders.
Now the s
Hi, all !
I have a fast question, I am running a few Asterisk systems, but I just
noticed one thing quite peculiar. After I started "safe_asterisk", and when
I ran PS or TOP, I could see 1 PID "safe_asterisk" and almost 10 PIDs
"asterisk -vvvg -c" even when there was no call. However, for the othe
I know nothing about telephony ip phone etc.. however i have a few $$
that i am willing to spend to learn Asterisk . and i am very very
curious, i believe in learning by doing( but with some hand holding) so
i am looking for equipment suggestion . can anyone suggest a set of
equipment i could get t
Questions... happen to use webvmail?
bkw
On Thu, 15 Jan 2004, Brian Capouch wrote:
> I have a user, running CVS a/o 11/23/03, who has complained about
> "phantom" messages showing up days or even weeks after she has deleted them.
>
> So I asked her to let me know when it happened again, and she
The hardware drivers do seem to be migrating to 2.6. On the
other hand ztdummy has seen no love in awhile. For our environment
it is a better choice (limited slots, 3.3v instead of 5v, etc).
I'm no kernel programmer, but I've been working on at least clearing
the compile errors and loading of zt
--- "[EMAIL PROTECTED]" <[EMAIL PROTECTED]> wrote:
> Referring to my previous post about degradation of voice quality when
>
> having more than 2 connection.
>
> The actual route is:
>
> pc xlite -> local asterisk box -> iaxtel -> local asterisk
>
> I have tried out a different situation:
>
>
Sorry for the malformed mail. My responses are marked with '***' below.
jesse
==
Hi,
I am a newbie in Asterisk as well, intending to use it in a similar way as
you are, communicating with AS5300 as well as other gateways including
MAXTNT.
I have had similar, but yet different experiences th
I do not have any zaptel hardware on the Asterisk box, I could not have
meetme functioning. I did modify the Makefile in zaptel directory on
line 168 by including ztdummy as one of the modules to compile in.
The error message from the concole:
-- Executing MeetMe("SIP/1002-e9ca", "4700") in n
At one point I had Asterisk running on a Fedora Core 1 based embedded
system using a Soekris embedded device. Once the OS is running, the only
hard part is finding a source of timing for the MOH and conference calling.
However, I think the new Soekris units have a timing source on them (USB).
-
I did initially, but I was having problems (possibly just in thinking it through)
getting the provided h323 driver to either
a) register as a gateway with my gatekeeper - that just does not seem to be and option
(please correct me if I'm wrong!!!)
or
b) setup a 'variable' extension (yes, extensi
On Fri, 16 Jan 2004, [EMAIL PROTECTED] wrote:
> This is ifconfig on openbsd box:
> fxp0: flags=8843 mtu 1500
>
> I think this output shows that the fxp0 interface is on simplex mode.
Yes its in simplex mode, but this parameter is NOT related to half/full
duplex on the port.
Check this output fr
> # ifconfig xl0
>
> xl0: flags=8843 mtu 1500
>
> address: 00:01:02:78:11:e8
> media: Ethernet autoselect (10baseT)
> status: active
> inet 203.219.167.126 netmask 0xfffc broadcast 203.219.167.127
> inet
> I am
> rather
> curious as to why I seem to be using up all memory although I am not
> running
> any unnecessary processes, or should I actually disable all modules,
> other
> than really necessary ones to support VOIP?
Do you mean that Asterisk is using up all of your memory
or that all of you
I have a Dell 400sc sever on order. It will be shipped on the 27th. It is a
2.4GHz P4 with a 533 MHz front side bus, a 40GB disk, 128MB of memory, sound
card, ethernet, and year of on-site next day maintenance.
It is $318 delivered after rebates. Yes, $318.
This is a real server, by the way, n
http://www.intel.com/software/products/ipp/samples_table.htm#
Has anyone taken a look at the value of these sample libraries for use in
Asterisk.
cameron.
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On Thu, 15 Jan 2004, Peter Pauly wrote:
> Are there any cheap SIP phones (like the Grandstream
> for example) that support power over ethernet?
SNOM-105, SNOM-200, and all Cisco phones should support PoE.
> What is necessary to support SIP phones in a
> Cisco Call Manager environment?
easiest s
--- calvis <[EMAIL PROTECTED]> wrote:
>
>
> I am real close to finalizing my hardware selection for my Asterisk
> test
> machine. I am going to use the following hardware:
>
> Dell 400SC w\Red Hat 9.0
>
> 1 - 4 Port TDM40B Card (FXS)
> 3 - Wildcard X100P Cards (FXO)
It does not matter if th
Steven Critchfield wrote:
On Wed, 2004-01-14 at 13:26, Jorge Mendoza wrote:
Hi,
A customer has an old PBX, which accept only T1 (not PRI) trunks. The
local telco only provides Euro PRI. Could the following config works?:
[telco] <-- E1 PRI --> [Asterisk] <-- T1 --> [PBX]
Many thanks for your t
On Tue, 13 Jan 2004, Christopher Arnold wrote:
> i have a setup with chatrooms, several MeetMe conferences wich users can
> change inbetween. 10 users maximum in each room.
>
> It seems like when i have more than 40-45 users on the system at the same
> time asterisk drops abt 20 and continnues b
THat's not bad 20 calls through a 800Mhz P3. I new 3Ghz P4
could likely handle 60 then. Not bad.
But don't beleive "top". First off if acverages. Think for
a minute. We all kow a CPU can never by "20% in use" it is either
in an idle loop (at 0%) or doing real work (100%) it can't be
in an in-
On Tue, 13 Jan 2004, Areski wrote:
> Sorry Chris, actually, I cannot help you regarding your problem!
> But I would like to know how allow an user to change of conferences (go
> to an other room) !?!
>
When a user presses "#" he exits the conference. Then you just direct hiim
to another.
Dan Austin wrote:
The only short term issue I see with * for this is we are
standardized on platform where Digium cards are not an option, and
ztdummy and zaprtc cannot be loaded (2.6 kernels).
Mark has taken a stab at fixing zaptel to be compatible with the 2.6
kernel. Why don't you test i
Any one has documented how-tos for making voicetronix openline 4 to work
with Asterisk.
I have been contacting Australian Digium resellers and Digium cards are
not approved in Australia. So I suppose Australian users are interested
into putting Voicetronix in use.
Any expereience to share will
Hi,
I'm interested in participating on the embedded side. One of our R&D
labs is working on a number of embedded server solutions, including
servers that are built around a 3" square PCB, linked to a 2" square
PCB with a compact flash interface. It's robust, and up to military
standards (but i
I would just like to follow-up on the ringback
problem I'm getting from *. As I've said in my previous post, I am not
hearing the "real ringback" from the Cisco gateway terminating my call. I
don't want to provide false ringback from * (r option of dial), because it'll
still give me ringba
Referring to my previous post about degradation of voice quality when
having more than 2 connection.
The actual route is:
pc xlite -> local asterisk box -> iaxtel -> local asterisk
I have tried out a different situation:
pc xlite -> local asterisk box -> iaxtel
and the second connection
pc xl
Hi,
I am a newbie in Asterisk as well, intending to use it in a similar way as
you are, communicating with AS5300 as well as other gateways including
MAXTNT.
I have had similar, but yet different experiences than yours.
1. Asterisk does crash with the number of calls, but in my case, about or
le
On 15/01/04 19:39, Jesse Peterson wrote:
#0 ast_smoother_feed (s=0xcbf90080, f=0x5de5c4a8) at frame.c:72
#1 0x41eb00b1 in oh323_write (c=0x8214488, f=0x5de5c4a8) at chan_oh323.c:1504
Do you experience the same problems when you use the other (bundled)
h323 driver? (asterisk/channels/h323/README
That's why I stoped using app_festival and instead use the Festival
text2wav program to generate a .WAV file and use app_playback to stream
the audio to the user.
On Thu, 2004-01-15 at 13:41, Iain Stevenson wrote:
> app_festival currently seems to chop the start of sound it plays back -
> probab
I have a user, running CVS a/o 11/23/03, who has complained about
"phantom" messages showing up days or even weeks after she has deleted them.
So I asked her to let me know when it happened again, and she called a
few minutes ago.
The directory listing below shows a listing of the
/var/spool/a
This is ifconfig on openbsd box:
fxp0: flags=8843 mtu 1500
I think this output shows that the fxp0 interface is on simplex mode.
The voice degradation I referred was by using xlite soft phone. I open 2
line similtaneously and dial to FWD and back to my incoming extension.
Xlite is runnning on a
I just got an email from SIPphone advising that there have been problems
with the above firmware and advising to reload from their server. This
does in fact reload 1.0.4.35 into the phone. And now voicemail has gone
AWOL again.
--
Dave Cotton <[EMAIL PROTECTED]>
__
Fran Boon wrote:
I have been compiling information on this configuration onto the Wiki:
http://voip-info.org/wiki-Asterisk+cisco+FXO
I can call out to the PSTN just fine, but inbound calls all appear in my
default [bogon-calls] context, not in [pstn-incoming]
Hmmm.
As I understand it, the Cisco i
Andrew Kohlsmith wrote:
http://www.imagestream.com/PCI_720.html
Regards,
Andrew
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Sipura recommended disabling the echo cancellation on the SPA-2000 for
modem pass-through. It does help although still not 100% success rate.
Stephen
> -Original Message-
> From: Christopher J. Wolff [mailto:[EMAIL PROTECTED]
> Sent: Thursday, January 15, 2004 12:14 PM
> To: [EMAIL PROT
are you running safe_asterisk ?
If so try to modify safe_asterisk ... CONSOLE=yes to CONSOLE=no
or if not
list all the asteirsk threads 'ps -axum | grep asterisk'
find the thread that takes the most CPU and connect with gdb
gdb /usr/sbin/asterisk pid
and do 'bt'
and post the last few lines back
I am real close to finalizing my hardware selection for my Asterisk test
machine. I am going to use the following hardware:
Dell 400SC w\Red Hat 9.0
1 - 4 Port TDM40B Card (FXS)
3 - Wildcard X100P Cards (FXO)
Are there any known conflicts using this setup in this machine? I will be
occupy
Are there any cheap SIP phones (like the Grandstream
for example) that support power over ethernet?
What is necessary to support SIP phones in a
Cisco Call Manager environment?
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Oooops... A little jump ahead. It asked for sign on etc... Got it now,
mucho thanks and understanding my slow brain... :-}
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David Mynatt
Sent: Thursday, January 15, 2004 12:39 PM
To: [EMAIL PROTECTED]
Subj
This is not directly * related, but could be. My company is
using a VoIP conferencing solution that is suffering from
developer neglect.
I've considered trying to leverage *, and our internal developers
can build the management interfaces. If that plan is not accepted
by management, I need to fi
Has anyone played around with QoS or TOS relative to * and sip phones?
I was just doing a little real-time research and noticed our C7960's
mark IP packets with "low delay" and "high throughput" (presumably due
to tos_media: 5 in the SIPDefault config file), and rtp packets flowing
"from" asteris
It's spelled MCI, WorldCom, Sprint, T-Mobile... All the same except for
the billing and the twists and turns of the contract. Whatever happened
to POTS (i.e., Bell System.)
An Old AT&T 4A/ETS and ESS/S7 Craft
==
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Hi To ALL
i have made an application for billing a traffic but i have strange
problem
with free message from Telco provider because when dial the number and Telco
reply "The customer have change number..." i dont receive a connect so i cant
listen nothing... Yes is right from PRI dont r
> > fxp0: flags=8843 mtu 1500
> > address: 00:02:55:30:54:28
> > media: Ethernet autoselect (100baseTX full-duplex)
> > status: active
> > inet 192.168.1.1 netmask 0xff00 broadcast 192.168.1.255
> > inet6 fe80::202:55ff:fe30:5428%fxp0 prefixlen 64 s
Hi All,
I have a e100p that is not receiving any interrupts. My /proc/interrupts
look like
CPU0
0: 87288 XT-PIC timer
1:104 XT-PIC keyboard
2: 0 XT-PIC cascade
8: 1 XT-PIC rtc
10: 814092 XT-PIC
On 15/01/04 13:12, Roy Sigurd Karlsbakk wrote:
hi all
for new users, finding asterisk info is unneccesary troublesome. the
asterisk.org page has very little information about the product and
using google for 'asterisk' is like using google for 'linux'. you get
all too many hits that has nothing to
I think that it will be greate to include * inside of a router like ix66
from intertex... 1 GB usb removable flash to record voice mail.and prompts
in the computer..2 fxo...real internal sip server ...internal dns
server..good user interface.. all nat / firewall nightmare ended, no
computers to w
app_festival currently seems to chop the start of sound it plays back -
probably something to do with rtp and maybe the same problem that was
present in voicemail prompt plauback.
Iain
--On Thursday, January 15, 2004 11:16 am -0600 Steven Critchfield
<[EMAIL PROTECTED]> wrote:
On Thu, 2004-0
- Original Message -
From: "Zhang Peihao" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, January 14, 2004 8:51 PM
Subject: [Asterisk-Users] Cooperate with SIP ITSP
> Hi All,
>
> When I want use Asterisk as a PBX to cooperate SIP ITSP,
> I can not set the caller ID, so SIP IT
Hi,
I'm relativle new to *, so I may be wrong. I build up * from cvs today
(show version: CVS-01/15/04-16:27:36). In an test I use 2 SIP phones
(linphone) to connect to eachother.
The phones are called via the extensions 100 (user 'kwe') and 200 (user
'phone').
I can call from one to another and
Hello all. I'm new to asterisk and have been using and testing it for about a week
now. My initial hope has been to use it as a sip<->h323 gateway to tie SIP & H323
based ip phones together with my Cisco AS5300 and Lucent MaxTNT/MVAM networks.
I am currently running Asterisk 0.5.0 under Redhat 9
Hi friends,
Could some one recommand a good cdr processing software out there for post
paid billing (invoicing, web-based payment processing) etc.,
Thanks a bunch.
SW
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> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Chris Albertson
> Sent: Thursday, January 15, 2004 12:40 PM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] re hardware requirement - asterisk
>
>
>
> I don't think 10BaseT can run full du
Don Pobanz wrote:
On Wednesday, January 14, 2004 5:48 AM, Daniel Bichara
[SMTP:[EMAIL PROTECTED]] wrote:
Hi,
Once a day, * drops all calls (E100P board). Yesterday, I updated *
version to CVS but I got the problem again today. Monitoring log
files,
I found this messages just bef
well, it does say SIMPLEX in the fxp0 flags section. I don't honestly
know if this means it's negotiated half duplex, or something beyond
that 10baseT is capable of running full duplex, although this
requires a NIC capable of is, as well as a switch that can do FD. And
regarding the 1% comme
On Thu, 2004-01-15 at 14:31, Chris Albertson wrote:
> I'm looking to do about the same thing, build very low cost
> systems. (I'm looking at putting Asterisk at some
> non-profit organizations.) but one thing you can't make
> a compromise on is reliabilty. It has to work and keep working
> for
Hi To ALL
i have made an application for billing a traffic but i have strange problem
with free message from Telco provider because when dial the number and Telco
reply "The customer have change number..." i dont receive a connect so i cant
listen nothing... Yes is right from PRI dont re
Actually he found it in the dumpster after the police threw it out
following a bust! Does anyone want to send a dollar to Mr. Happy?!
-Original Message-
From: C. Maj [mailto:[EMAIL PROTECTED]
Sent: Thursday, January 15, 2004 12:40 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] *
Are you wanting to make a pre-built * box, with hardware to connect a
single dial line and one traditional phone, or.. ?
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Thursday, January 15, 2004 11:08 AM
To: [EMAIL PROTECTED
I get 'Access Denied'... Can it be downloaded zip or tar ball?
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Philipp von
Klitzing
Sent: Thursday, January 15, 2004 11:01 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] asterisk.org webp
- Original Message -
From: "Iain Stevenson" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, January 15, 2004 11:41 AM
Subject: [Asterisk-Users] People detected as fax machines
>
> A caller to me was this afternoon detected as a fax machine:
>
> Jan 15 15:31:17 NOTICE[41997]: F
how do you spell Teleecooomm again?
[EMAIL PROTECTED] wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Gary Franczyk
Sent: Thursday, January 15, 2004 10:37 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Basic Asterisk capabilities question
Hello,
I have a Hypercom T7P swipe card terminal sitting on a dedicated G711ulaw
port. The Hypercom operates at either 1200 or 2400bps. I get about a 50%
success rate when I try to authorize cards. On this same G711ulaw port, I
have a fax machine with a 100% success rate operating at 9600bps.
I have been compiling information on this configuration onto the Wiki:
http://voip-info.org/wiki-Asterisk+cisco+FXO
I can call out to the PSTN just fine, but inbound calls all appear in my
default [bogon-calls] context, not in [pstn-incoming]
Can anyone help me locate why?
(Config files are on th
It's not what I would want to depend on day in and day out. I know that
you can buy Dell PowerEdge SC400 servers for $299 with HDD, memory, and
either a celeron or p4, depending on what day of the week it is. I'd put
my name on the Dell based solution before the white box solution for the
same mone
If you don't have a fax connected to * then create and exten:
exten => fax,1,Goto(day,s,1)
I had the same today... :/
Andy
*** REPLY SEPARATOR ***
On 15/01/2004 at 16:41 Iain Stevenson wrote:
>A caller to me was this afternoon detected as a fax machine:
>
>Jan 15 15:31:17 NO
On Thu, 15 Jan 2004, mattf waxed:
8<'s
> There is a group of Asterisk users that decided to modify the code of
> Asterisk to try to make it a predictive dialer, called shady_dial I believe,
> but I haven't heard anything about it lately.
http://shadydial.sourceforge.net/
Lots of recent updates
I think this is a MAROR bug in the new dsp.c routines, recompile using the
old dsp stuff by changing the makefile and set OLD_DSP_ROUTINES
- Original Message -
From: "Iain Stevenson" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, January 15, 2004 8:41 AM
Subject: [Asterisk-User
On Thursday, January 15, 2004 10:42 AM, Iain Stevenson
[SMTP:[EMAIL PROTECTED] wrote:
...
>Is there any way to stop * even considering an
> incoming
> call on a line as a fax call?
>
Sure, just don't have
exten => fax.
in the same context (or included context).
> Iain
>
--
Don Pobanz
_
--- "[EMAIL PROTECTED]" <[EMAIL PROTECTED]> wrote:
> I have just checked the Openbsd box on the if interface.
>
>
> fxp0: flags=8843 mtu 1500
> address: 00:02:55:30:54:28
> media: Ethernet autoselect (100baseTX full-duplex)
> status: active
> inet 192.168.1.1
On Thu, 15 Jan 2004, Steve waxed:
> On Thursday 15 January 2004 12:03 am, [EMAIL PROTECTED] wrote:
> > sounds like one of those pesky auto dialers the simpsons make fun of.
>
> It sure does...
The AT-5000 was Prof. Frink's first patent, and it was
"designed to alert children of snow days and suc
Hi,
Any * users in sweden, particularly in the Malmo or Lund areas? Mail me
off-list, i have some questions :)
Faiz
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I'm looking to do about the same thing, build very low cost
systems. (I'm looking at putting Asterisk at some
non-profit organizations.) but one thing you can't make
a compromise on is reliabilty. It has to work and keep working
for years to come. I was able to keep the price of a new PC
to
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