I'm greatly surprised when testing an Asterisk box with 802.11g. Here's the
topology:
VoIP caller --- 802.11g --- Asterisk --- 802.11g --- VoIP extension
|
FXO ___ PSTN extension
When I call a VoIP extension on that box
Does anyone have any recommendations for a phone that has easy to
understand/implement central provisioning? I've used CISCO 79XX phones,
and they're great (but too expensive). I like Grandstream phones, but
their provisioning sucks.
What is everybody else using in large environments where
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
What do you mean by mapping the 200 ?
In this example I can pickup any ringing extension:
http://www.voip-info.org/wiki/view/Asterisk+cmd+Pickup
If phone with number 42 rings you can catch the call by dialing 742. You
don't need to
On Wed, 7 Feb 2007, Mark Coccimiglio wrote:
Ok here is a real geek question,
I building my own linux kernel for my asterisk system and came across the
kernel setting for the timer frequency. I have one of 3 hardcode choices
100Hz, 250 Hz and 1000Hz. From what I understand the default Freq
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Hi,
I just want out find out how to do bill recon's when you send calls to a
provider. They send me
their CDR's, and when I compare it to my * CDR's, some calls are 1 second
off, either way.
How in general is it done by others?
Hi all,
I am looking for a solution for the following problem.
I have a little callcenter with 20 agents and 20 incomming analog lines, one
for each agent. I need to have abailable as incomming analog lines (FXO
Ports) as agents logged, not all the agents are logged all the time. It is
needed
I know Brett and Jurgen have been pretty happy with the Snom's - Brett
even wrote an auto-provision utility for the Snom's at one time.
later,
PaulH
On Thu, 2007-02-08 at 19:45 +1100, Rod Bacon wrote:
Does anyone have any recommendations for a phone that has easy to
understand/implement
Need to deploy between 50 to 300 lightweight Linux - only browser
and softphone.
You might want to consider our lightweight java softphone (Corraleta
SDK) - it can be embedded in
a web page - zero install/config in the client. The UI is in HTML and
javascript,
so you can get it _exactly_ the
I have a fairly complicated setup. Extensions (1,2 and 3). In 3 - I
execute AGI in java which plays few wav files depending on external
parameters.
Can I have a dial plan inside my AGI? If not, how do I accomodate user
who needs to reach extension 2 from my agi? I have tried stream file and
Network Configurations
Block D, Surrey Park, Barham Road, Westville, 3610
Helpdesk: (086) 163-8266
Tel: (031) 266-1563
Fax: (031) 266-4206
Hi people.
I'm hoping someone has come across this problem with version 1.2.14
In my dial plan I call various SIP phones using the following little
macro:
CB == Chris Bagnall [EMAIL PROTECTED] writes:
CB I have run a few speed tests from the sites in question (iperf to
CB the machine in the datacentre) and I'm consistently getting around
CB 380k upstream and 5.5mbit downstream, even during peak hours. Some
CB distance away from the quoted speeds,
Paul Hales wrote:
I know Brett and Jurgen have been pretty happy with the Snom's - Brett
even wrote an auto-provision utility for the Snom's at one time.
Yes, look at the latest Trixbox for the basic SNOM templates and then
off you go.
You setup a tftp server (easy), the phone looks for
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
I'm using a Thomason ST2030. Had difficulties in the beginning, but
after a firmware upgrade it works fine. And autoprovisioning works good.
Most of the parameters are described in their official (marked as
confidential) admin documentation from
Digium support cleared the issue for me, they sent me a new register
utiliy by mail and this one worked as expected. I registered my codedc
and tested my codec. If anyone needs to know, I tested the codec using a
SIPURA 3000 ATA so I can confirm this ATA works with G729.
I'd like to add:
Check from the sites in question using testmyvoip.com or whatever the
site is called.
In the UK I found that some strange things sometimes happen. At one
point I was sure that BT were perhaps misclassifying IAX packets as
P2P... However, not had a problem with SIP.
Beware that ADSL uses
Hi
Yes, I know that I am using IAX2 and not SIP for my connection to
teliax. IAX2 is the preferred protocol for connection to teliax. I
have the firewall configured to prioritorize port 4569 for IAX2.
1) 4569 is only the IAX setup port. Edit rtp.conf to limit the rtp
ports to some
Hi all,
I'm trying to send FAX with an anolog fax behind a Patton M-ATA to an
other analog fax plug on directly on the PSTN network.
I use the last stable version of Asterisk 1.4 ...
Somebody have any information why it's doesn't work a all ?
Thanks a lot,
Thomas
Hi there,
As described on voip-info here
http://www.voip-info.org/wiki/view/Asterisk+RealTime+Queue, if I use
realtime queues, alterations to the list of members don't alter until a
new call joins the queue.
Is there anything I can do about this? I've tried looking for a bug
number, but to
The part about 4569 being the IAX2 setup port, is not correct.
All traffic, including RTP, travel through this port, when you use IAX.
rtp.conf is used for SIP traffic, and possibly H232.
Jon
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ed W
Sent: 8.
Hi,
We are using following phones for large deployments using auto-provisioning.
Grandstream phones (full range)
Snom Phones (full range)
Aastra Phone (full range)
UTstarcom (Wifi phones)
~Ahsan
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of
Many thanks Stefan!
It works like a charm...
Kind regards,
Pierre
===
Write a cronjob which creates a call file. Shouldn't be a big thing.
In case you are not familiar with call files: Create a file dummy.call
with the following content.
---cut---
Hello.
To someone who have done the dCAP exam.
I would like to know about it: test and practises questions examples,
difficulty level,... I'll be very grateful if somebody sends me an
exam model.
Thanks in advance
___
--Bandwidth and Colocation
Hi
I set up call back functionally thru AMI (local channel).
The two calls are bridged and the call is established.
But when I hang up the local channel (the first extension that rang), the
other leg of the call *is not released*
Time events:
0) Socket communication(AMI)
1)extensionA
On Tue, Feb 06, 2007 at 09:41:30AM +, Tim Panton wrote:
On 5 Feb 2007, at 21:46, chester c young wrote:
Need to deploy between 50 to 300 lightweight Linux - only browser
and softphone.
You might want to consider our lightweight java softphone (Corraleta
SDK) - it can be embedded
Am 08.02.2007 um 13:02 schrieb Benito Camelas:
To someone who have done the dCAP exam.
I did the dCAP a couple of weeks ago.
I would like to know about it: test and practises questions examples,
difficulty level,... I'll be very grateful if somebody sends me an
exam model.
The practical
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Asterisk 1.2 has no support of t.38 whatsoever, the call will drop
before t.38 is ever utilised, not even pass-thru.
1.4 Adds support for T.38 pass through only and no other sort of
faxing, the endpoint must support T.38 and you must
i started asterisk my typing the command #/usr/sbin/asterisk -c but it is giving error that it couldn't establish connectiom with mysql. failed to connect database server superswitch on 192.168.1.205 unable to get our IP address , Skinny disabled. please help
Don't just search. Find. MSN
Hi,
I'm new to *,so i apologize for stupid questions.
I'm having problem with this arhitecture:
I'm calling asterisk from behind a NAT(sjphone user) with a low band so I'm
using GSM codec.
In extensions.conf I have:
exten = 337,1,Dial(SIP/99@ip_pbx2)
so when i dial 337 from sjphone Asterisk is
Lookup 'articulation'
On Feb 1, 2007, at 1:53 AM, Dovid B wrote:
Anyone know of a softphone for the Palm OS ?
Thanks.
Dovid
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To UNSUBSCRIBE or update options
Set a variable that you can then use GotoIf in the dialplan to branch to the
required exten
Jon Farmer
Telford, Shropshire, UK
- Original Message
From: prasanth [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday, 8 February, 2007 10:06:07 AM
Subject:
On Thu, 2007-02-08 at 13:55 +0100, Tomislav Parčina wrote:
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Asterisk 1.2 has no support of t.38 whatsoever, the call will drop
before t.38 is ever utilised, not even pass-thru.
1.4 Adds support for T.38 pass through only and no other
Which H.323 channel driver are you using, and could you post a log or debug
of a session.
Craig
- Original Message -
From: Andrei U [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday, February 08, 2007 2:41 AM
Subject: [asterisk-users] H323 to SIP - One way voice
Yuan Liu wrote:
My multiple postings to this list this morning got garbled in
http://lists.digium.com/pipermail/asterisk-users/, and don't come back from
list. (e.g.,
http://lists.digium.com/pipermail/asterisk-users/2007-February/179315.html) I
thought it was Hotmail, so I saved one
That's what I would have thought. I set the timeout to be 300 secs, but
the phone never seems to re-register. We could do a group dial, but like
you said, there would be a lot of errors in the log, which we are trying
to avoid. Has anyone been able to get a polycom 501 to re-register
itself
I must clarify my original message. Maybe
confusion is due to my poor english. So I'll make a list of statements:
- Each ISDN line in Italy can be splitted in two analog lines
- You can use those analog lines as normal analog lines
- I have already invested in analog hardware (my
fault of
Is it possible to use Digium (or Sagnoma, or Beronet) cards with Asterisk on
Vmware?
Has anyone done it?
--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)270248
Mob.: +385(91)1212148
SIP: [EMAIL PROTECTED]
e-mail: tparcina#lama.hr
http://www.lama.hr
It's not that Digium don't want fax or t.38 support, it's just that it is
not very likely for Steve Underwood to provide it for Asterisk. I'm sure
that Digium are very keen for someone to write and contribute t.38 code for
Asterisk, it's just that there aren't very many people with the
I'm tempted to rebuild my asterisk network with AsteriskNow - my
question is, can you ADD anything to it? i.e. cdr_mysql logging? I thought
I saw it didn't have that
And how does it handle the hardware? I don't use digium cards in all of my
servers because of country issues (Junghanns in
On Thu, 2007-02-08 at 10:32 -0200, Paulo Vicentini wrote:
Hi
I set up call back functionally thru AMI (local channel).
The two calls are bridged and the call is established.
But when I hang up the local channel (the first extension that rang),
the other leg of the call *is not
We have a similar situation and we do a realtime lookup in an external
db, works like a champ
Steve Murphy wrote:
On Wed, 2007-02-07 at 22:21 -0500, Lee Jenkins wrote:
Eric Germann wrote:
We're beginning to test MultiTech's CallFinder CDMA Units, one for Sprint
PCS and one for
Asterisk is getting red alarms on my T1, sometimes once or twice a
day, but today it happened 5 times. Even once is too many. Every
call in progress is dropped.
Red alarm means that the hardware is not seeing the T1 signal coming in.
This most likely is a cable or wiring or perhaps a
This is a solution if your provider is using IAX, but we are stuck with SIP.
I find it surprising that txfax and rxfax not compiling under 1.4, but oh well.
Warm Regards,
Remzi Turer
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ardjan Zwartjes
Sent:
On Thursday 08 February 2007 07:32, [EMAIL PROTECTED]
wrote:
Does anyone have any recommendations for a phone that has easy to
understand/implement central provisioning? I've used CISCO 79XX phones,
and they're great (but too expensive). I like Grandstream phones, but
their provisioning
This worked. Great and thanks
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Carlos Chavez
Sent: Wednesday, February 07, 2007 5:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] After upgrade to 1.4 transfers
our Polycoms reregister almost immediately. I think the problem your
running into is that when the softphone is registered the polycom gets
some kind of error from asterisk which prevents it from reregistering
Rob Schall wrote:
That's what I would have thought. I set the timeout to be 300
Chris, (or others), do you have any negative experience with Thomson
2030? it looks very promising!
I hesitate between thomson and linksys spa 922/942,
I'm not sure, what is better for bussines use :-\
snoms are probably also good, but functionality/price ratio is, imho,
better for thomson or
On 2/8/07, Remzi Semsettin Turer [EMAIL PROTECTED] wrote:
This is a solution if your provider is using IAX, but we are stuck with
SIP.
Huh? What do the two have to do with each other?
___
--Bandwidth and Colocation provided by Easynews.com --
We Use both Grandstream and Aastra phones
Some simple scripts improve the Grandstream configuration ease of use
but the Aastras use a straight text file and are much better documented.
regards,
Drew
Ahsan Masood wrote:
Hi,
We are using following phones for large deployments using
That's not nessecerily true, if you install iaxmodem and hylafax on your
asterisk machine you'll use IAX for the internal communication, but
faxes can go out and come in on SIP or whatever you like.
One thing that's important to mention here: We get unpredictable results
if the fax is transmitted
Hi all,
Me again ... for a new question! Again
Here the scenario:
A call B ( A -- B)
B transfer to C ( A -- C)
In this case, how can I have the B caller id number and the A caller id
number?
Thanks a lot for your help
Thomas
Is there some way to test this, or to cause the polycom to ignore the
errors, and try back later (unlimited times). The fear I had or
re-registering, was that the softphone would be in use, and the hard
phone would take the number back over. That shouldn't happen until the
number is available
We have considered working on this. T38 is a short term solution, though.
Justin Newman
--
From: Tomislav Par?ina [EMAIL PROTECTED]
Subject: [asterisk-users] Re: Asterisk Faxing Support
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Asterisk 1.2 has no
There are two types of ISDN line, Primary Rate Access (PRI) and Basic Rate
Access (BRI). PRI has 30 (+ 1) channels, BRI has 2 (+1) channels. You are
talking about BRI which consists of two 64 kbit/s data channels and 1
signalling channel. In telephony, the two data channels are decoded and used
Craig Guy wrote:
it wouldn't make business sense for Digium to have code in the free
distribution that can't be in their commercial distribution.
Yes, I do suspect that Digium sees things this way.
Maybe I'm too much of a free-thinker - too believing in the open-source
philosophy, but I
ha ok, I understand now
1) I don't think that Asterisk has any support for meter pulse detection
on analogue cards.
2) If you already have an ISDN line, why do you not spend the eur 20 on
a BRI card and do the job properly? The way you propose you are going
from ISDN -- Analogue --
Ardjan Zwartjes wrote:
One thing that's important to mention here: We get unpredictable results
if the fax is transmitted entirely over VOIP, if the fax passes a
regular telephony channel once it works fine but if it's purely VOIP,
transmission errors occur. This is probably a timing problem,
I liked polycom a lot.
- Original Message -
From: Rod Bacon
To: asterisk-users@lists.digium.com
Sent: Thursday, February 08, 2007 10:45 AM
Subject: [asterisk-users] Best phone for easy provisioning
Does anyone have any recommendations for a phone that has easy to
your asterisk box has to do audio conversion, its getting bogged down
Yuan LIU wrote:
I'm greatly surprised when testing an Asterisk box with 802.11g.
Here's the topology:
VoIP caller --- 802.11g --- Asterisk --- 802.11g --- VoIP extension
|
I am running some Polycom phones and have Auto Answer setup(*51
initiates that when you call an extension)
With an attended transfer you can take a call, hit transfer,
*51extension, announce the call and if the person wants it, complete
the transfer, the call is now on speaker at the end.
Best and easiest provisioning I´ve found imho is Snom, great web
interfase , followed by Polycom (web interfase used to be poor and slow,
but once you set it up, works very well)
Dovid B escribió:
I liked polycom a lot.
- Original Message -
*From:* Rod Bacon mailto:[EMAIL
Dovid B wrote:
I liked polycom a lot.
- Original Message -
*From:* Rod Bacon mailto:[EMAIL PROTECTED]
*To:* asterisk-users@lists.digium.com
mailto:asterisk-users@lists.digium.com
*Sent:* Thursday, February 08, 2007 10:45 AM
*Subject:* [asterisk-users] Best phone
I finally got my X100P working and now I have a question.
I have several Skutch phone line simulators. My X100P works as expected with
both a POTS line and an analog PBX port, but when I use a phone line
simulator it doesn't answer the line. The phone line simulator doesn't power
the line until
config problem . what pbx does ip_pb2 runs ? ( is it asterisk ? ) in peer
definition try allowing all codecs .. ( gsm , ulaw,alaw,ilbc )
On 08/02/07, Florea Igor [EMAIL PROTECTED] wrote:
Hi,
I'm new to *,so i apologize for stupid questions.
I'm having problem with this arhitecture:
I'm calling
You can easily recompile asterisk with mysql logging enabled also use all
add-ons u can use on debian and any other distro ..
On 08/02/07, Chris Earle [EMAIL PROTECTED] wrote:
I'm tempted to rebuild my asterisk network with AsteriskNow - my
question is, can you ADD anything to it? i.e.
great
i join you Thomson ST is a good choice, also you can see linksys
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
really you can't make a difference between them, i like thomson
2007/2/8, Pavel Jezek [EMAIL PROTECTED]:
Chris, (or others), do you have any negative experience with Thomson
2030? it looks very promising!
I hesitate between thomson and linksys spa 922/942,
I'm not sure, what is better for
did you test to call from a soft phone using pstn, if you get a bad
sound that s mean that the zaptel param must be changed if not try to
call from a soft phone your wirless phones and test
2007/2/8, Yuan LIU [EMAIL PROTECTED]:
I'm greatly surprised when testing an Asterisk box with 802.11g.
for massive deployment phone provisioning/fw updating through web
interface is not optimal,
best way is via config files/templates periodicaly downloaded from
central tftp/http server...
PJ
MF wrote:
Best and easiest provisioning I´ve found imho is Snom, great web
interfase , followed by
Does anyone have some method, or AGI scripts that will automatically
call a list of numbers from a database and play a pre-recorded
message?
For example, you have a database of
FirstName LastName PhoneNumber
Jon
--
***
Forrest Beck
IAXTEL: 17002871718
[EMAIL PROTECTED]
Does anyone have some method, or AGI scripts that will automatically
call a list of numbers from a database and play a pre-recorded
message?
Just for example, you have a database of
FirstName, LastName, PhoneNumber
Jon, Beck, 9194713175
So it would pull each record with phone number, dial the
David Ruggles wrote:
I finally got my X100P working and now I have a question.
I have several Skutch phone line simulators. My X100P works as expected with
both a POTS line and an analog PBX port, but when I use a phone line
simulator it doesn't answer the line. The phone line simulator
when i compile zaptel
make linux26
make install
i got these errors:
make[1]: Leaving directory `/usr/src/zaptel-1.4/wct4xxp'
make -C datamods clean
make[1]: Entering directory `/usr/src/zaptel-1.4/datamods'
make -C /lib/modules/2.4.27-3-386/build
SUBDIRS=/usr/src/zaptel-1.4/datamods clean
We used Aastra's for a good while, but gave up on them (and switched to
Cisco). Aastra's seem cheaper up front (hardware costs), but the time
wasted chasing firmware bugs, lack of documentation, and poor support
quickly eat up any savings. (unless your needs are very basic).
MD
_
How many disk do you need? I'll burn you one and mail it to you if you want.
On 2/6/07, Tom Poe [EMAIL PROTECTED] wrote:
I wonder if there are CDs available for purchase. I don't have any way
to burn one from a downloaded iso image. Any help appreciated.
Tom
-- Forwarded message --
From: younss azzayani [EMAIL PROTECTED]
Date: 8 févr. 2007 17:58
Subject: error when compiling zaptel-1.4
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
when i compile zaptel
make linux26
make install
i got
I've looked around and couldn't find much on this, but using two different
TFTP servers (linux / windows), my Cisco 7960s won't load the RINGLIST.DAT
and dialplan.xml files. On both the TFTP servers and the phone, I get TFTP
Timeout Errors.
The SIP configuration files load fine.
Any ideas?
This device can solve many problems, and is a must for most
applications where asterisk is connected using FXO ports and the host
PBX deosn't give CPC.
http://www.sandman.com/wizard.html#CPCGenerator
On 2/6/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
Yuan LIU wrote:
After reading
I am looking for some Linksys and GrandStream ATAs in Canada. I am
looking for places that ship from Canada so I don't have to deal
with the clearing of customs and tax remittance.
Any suggestion?
--
Thanks
___
--Bandwidth and Colocation provided by
Am Mittwoch, den 07.02.2007, 21:57 -0800 schrieb Jason Kim:
Hi,
This is the configuration I want.
Hard Video phone---video---Soft Video Phone(PC)
^
|
audio
|
V
Audio Only Phone
Any idea?
You could see wether having a second call that does a
From: David Ruggles [EMAIL PROTECTED]
Date: Thu, 8 Feb 2007 11:57:41 -0500
I finally got my X100P working and now I have a question.
I have several Skutch phone line simulators. My X100P works as expected
with
both a POTS line and an analog PBX port, but when I use a phone line
simulator it
From: Jason Fuermann [EMAIL PROTECTED]
Date: Thu, 08 Feb 2007 10:33:26 -0600
your asterisk box has to do audio conversion, its getting bogged down
Thanks for your reply, Jason. Two further questions:
1) I thought all networking would be done in the card, not taxing CPU much?
2) I get
Hi Stefano,
I have a question, how would you go about using the billing pulses to
generate an invoice/bill. Also can you provide an ascii drawing of the
layout of the equipment as you intend to use it, they say a picture is
worth a thousand words:)
db
On Thu, 2007-02-08 at 15:13 +0100,
Hello Asteriskers, :-)
We're trying to set up an outbound notification calling for system
alerts with Asterisk 1.4.0. We generate a call file in
/var/spool/asterisk/outgoing and the outbound call is originated through
Zap/1 (Sangoma A200D to a Canadian POTS line). The problem is that
From: younss azzayani [EMAIL PROTECTED]
Date: Thu, 8 Feb 2007 17:20:30 +
did you test to call from a soft phone using pstn, if you get a bad
sound that s mean that the zaptel param must be changed if not try to call
from a soft phone your wirless phones and test
I've tested Zaptel with
From: younss azzayani [EMAIL PROTECTED]
Date: Thu, 8 Feb 2007 17:58:08 +
when i compile zaptel
make linux26
make install
i got these errors:
make[1]: Leaving directory `/usr/src/zaptel-1.4/wct4xxp'
make -C datamods clean
make[1]: Entering directory `/usr/src/zaptel-1.4/datamods'
make -C
I can only speak for Aastra phones.
Central provisioning is very easy. All you need is one simple text file on
a TFTP, FTP, or HTTP server which all the phones point to. To customize
individual phones you add a second text file for each phone you want
customized. The custom text file is given
On Thu, Feb 08, 2007 at 05:58:08PM +, younss azzayani wrote:
when i compile zaptel
make linux26
make install
i got these errors:
make[1]: Leaving directory `/usr/src/zaptel-1.4/wct4xxp'
make -C datamods clean
make[1]: Entering directory `/usr/src/zaptel-1.4/datamods'
make -C
This should do what you asked:
http://voip-info.org/wiki/view/Asterisk+auto-dial+out+deliver+message
bp
On 2/8/07, Forrest Beck [EMAIL PROTECTED] wrote:
Does anyone have some method, or AGI scripts that will automatically
call a list of numbers from a database and play a pre-recorded
message?
The error lies here:
make[2]: Entering directory `/usr/src/kernel-headers-2.4.27-3-386'
make: *** arch/i386/boot: No such file or directory. Stop.
do you have the kernel-headers installed? (e.g.
glibc-kernheaders-2.4-9.1.87.i386.rpm for Fedora)
Alyed
I've made a very simple one time ago I
could share with you, it's made on bash, takes as input a CSV file,
places the calls using the /var/spool/asterisk/outbound directory, and
restricts the number of calls to a given number at a time (say 10)
I can share it with you only if you
Forrest Beck wrote:
Does anyone have some method, or AGI scripts that will automatically
call a list of numbers from a database and play a pre-recorded
message?
Just for example, you have a database of
FirstName, LastName, PhoneNumber
Jon, Beck, 9194713175
I'm currently working on an
Am 08.02.2007 um 18:39 schrieb Forrest Beck:
Does anyone have some method, or AGI scripts that will automatically
call a list of numbers from a database and play a pre-recorded
message?
Just for example, you have a database of
FirstName, LastName, PhoneNumber
Jon, Beck, 9194713175
So it would
When using a SIP phone with Asterisk, hitting the # key (pound or hash
depending on where in the world you happen to be) tells Asterisk that there
are no more digits coming, and to put the call through immediately based on
the digits already entered. This is the same functionality as the PSTN
On Thu, Feb 08, 2007 at 01:38:30PM -0500, C F wrote:
This device can solve many problems, and is a must for most
applications where asterisk is connected using FXO ports and the host
PBX deosn't give CPC.
http://www.sandman.com/wizard.html#CPCGenerator
How does it compare to busydetect of
Yuan LIU wrote:
From: younss azzayani [EMAIL PROTECTED]
Date: Thu, 8 Feb 2007 17:58:08 +
when i compile zaptel
make linux26
make install
i got these errors:
make[1]: Leaving directory `/usr/src/zaptel-1.4/wct4xxp'
make -C datamods clean
make[1]: Entering directory
Had the same issue time ago, but Eric shed good light on it,
have a look at:
http://lists.digium.com/pipermail/asterisk-users/2006-November/172079.html
Summary: sorry, no nice work around.
Alyed
Return-Path: [EMAIL PROTECTED]
On Thu, Feb 08, 2007 at 11:55:24AM -0800, Yuan LIU wrote:
From: younss azzayani [EMAIL PROTECTED]
Date: Thu, 8 Feb 2007 17:58:08 +
when i compile zaptel
make linux26
With 1.4: just 'make'
make install
i got these errors:
make[1]: Leaving directory `/usr/src/zaptel-1.4/wct4xxp'
Alyed Tzompa wrote:
The error lies here:
make[2]: Entering directory `/usr/src/kernel-headers-2.4.27-3-386'
make: *** arch/i386/boot: No such file or directory. Stop.
do you have the kernel-headers installed? (e.g.
glibc-kernheaders-2.4-9.1.87.i386.rpm for Fedora)
Alyed
On Thu, 2007-02-08 at 13:27 -0500, Brian M. Arlinghaus wrote:
I've looked around and couldn't find much on this, but using two different
TFTP servers (linux / windows), my Cisco 7960s won't load the RINGLIST.DAT
and dialplan.xml files. On both the TFTP servers and the phone, I get TFTP
Hi Guys and Girls, Freaks and Geeks,
I know you all had a blast at least years Astricon and are looking
forward to this years as well.however that's not why I'm writing
to you today. I know most of you are familiar with the www.Barcamp.org
http://www.barcamp.org/ events I'm writing to
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