Andrew Joakimsen wrote:
On 4/27/07, Per Jessen [EMAIL PROTECTED] wrote:
Try your local mobile phone supplier. I used a headset that came
with one of my cell phones, and it worked great w/ my SPA-941.
Not a bad idea - which make was this for? None of my phones
(Ericsson, Nokia) have a
Hi
We've got a redfone here and it's working great so far, despite all the
TDMoE bad press.
The 4-span version is slightly more expensive than a TE410P, so in the
end it's gonna be a more affordable solution as you'd need two digium
cards (plus maybe the ISDN guard).
The downside is that it
Hi all,
Can someone help me in reslving issue with priority in ACD
I am using Asterisk 1.4 and also ACD but when my agent login using priority 1
and 2 or 1 and 3 call come to both the priority which is unusual if anyone
encounter this issue please let me know also help me how to comeout of
In my * box I've configured two queues and incoming number and whenever any
one calls those number call comes to my *box and it sends call to my agents
in queue. but if no agent is available it still answer the call. Is there
any why when my agents are not available I don't want call to get
Hi List,
I'm setting up a system with one TDM400P (2*FXO + 2 * FXS) and one Junghanns
QuadBRI
on a Fedora Core 6 (Kernel 2.6.20-1.2944.fc6).
I'm using the bristuff-0.3.0-PRE-1y-e kit. It download zaptel-1.2.16,
libpri-1.2.4
and asterisk-1.2.17
When it's the time for ztcfg to do its job
Arun Kumar wrote:
In my * box I've configured two queues and incoming number and whenever
any one calls those number call comes to my *box and it sends call to my
agents in queue. but if no agent is available it still answer the call.
Is there any why when my agents are not available I don't
I would check:
Cat /proc/zaptel/
To make shure that the cards are activated in the order that you programmed
them.
Henk
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Backup
e-mail
Sent: maandag 30 april 2007 13:26
To: asterisk-users@lists.digium.com
I have dual posted this to the user and biz lists.
Has anyone ever heard of someone running an Asterisk based system, yet
abandoning SugarCRM, and opting to develop their own Visual FoxPro database/CRM?
Please don't dump on me now, this is not my idea, I am just asking for
comments, to see if
Most of the headsets at http://preview.tinyurl.com/38ow27 should work.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Per Jessen
Sent: April 28, 2007 11:02 AM
To: asterisk-users@lists.digium.com
Subject: RE: [asterisk-users] headsets for
Joe acquisto wrote:
I have dual posted this to the user and biz lists.
Has anyone ever heard of someone running an Asterisk based system, yet
abandoning SugarCRM, and opting to develop their own Visual FoxPro
database/CRM?
Please don't dump on me now, this is not my idea, I am just asking for
The solution to this issue is to edit /etc/sysconfig/zaptel and add the
following line:
MODULES=$MODULES qozap# BRISTUFF driver
Costa
Henk Dick [EMAIL PROTECTED] wrote:
v\:* {behavior:url(#default#VML);} o\:* {behavior:url(#default#VML);}
w\:*
Arun Kumar wrote:
In my * box I've configured two queues and incoming number and whenever
any one calls those number call comes to my *box and it sends call to my
agents in queue. but if no agent is available it still answer the call.
Is there any why when my agents are not available I don't
Paul [EMAIL PROTECTED] Wrote: 4/30/2007 8:53 AM:
Joe acquisto wrote:
I have dual posted this to the user and biz lists.
Has anyone ever heard of someone running an Asterisk based system, yet
Has abandoning SugarCRM, and opting to develop their own Visual FoxPro
Has database/CRM?
Please
Top of the morning all... So I reworked the pseudo IDS/Brute Force
Asterisk script for those who want to either use it, or use it as a
baseline to build a better one...
The script now does a few things... It logs those with password issues,
and blocks them as well. This was done to ensure
First - vtiger is available for those who don't like the SugarCRM
licensing.
It's not a licensing complaint. At least that has not surfaced. It is more
that the
programmer does not seem to be comfortable with SugarCRM, MySQL and php.
Biggest compliant about sugar is - hard to configure,
Time Bandit wrote:
First - vtiger is available for those who don't like the SugarCRM
licensing.
It's not a licensing complaint. At least that has not surfaced. It
is more that the programmer does not seem to be comfortable with
SugarCRM, MySQL and php. Biggest compliant about sugar is -
Joe acquisto wrote:
Paul [EMAIL PROTECTED] Wrote: 4/30/2007 8:53 AM:
Joe acquisto wrote:
I have dual posted this to the user and biz lists.
Has anyone ever heard of someone running an Asterisk based system, yet
Has abandoning SugarCRM, and opting to develop their own Visual FoxPro
I love these :)
- Original Message -
From: C F [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, April 27, 2007 7:54 PM
Subject: Re: [asterisk-users] Test
Failed
On 4/26/07, gc [EMAIL PROTECTED] wrote:
I have heard of people rejecting Sugar for their existing CRM/ERP product
based on VS Foxpro. I'm not a huge fan of Foxpro myself, but if the system
already exist then a lot of people see little advantage in changing.
On 4/30/07, Paul [EMAIL PROTECTED] wrote:
Joe acquisto wrote:
Paul [EMAIL
Hi,
This does not work with early audio (the use of Progress() on a Zap channel
before Dial(,20,m)).
The caller will not need to pay anything before anyone answers(). But I want
to play music or audio, while the call is progressing.
Håkon
_
Fra: [EMAIL PROTECTED]
I fully understand that but the OP says the programmer does not seem to
be comfortable with SugarCRM, MySQL and php. That is quite different
from It's easier to build on the code I already have. If I had
resisted growth and change over the years, I might be looking for ways
to integrate FORTRAN
HI All;
I want to use Asterisk for just Voicemail Server and I need a Dynamic creation
of Mailboxes.
My users 's Mailboxes are same as Extensions but I donot want to add
mailboxes in
Voicemail.conf
Is there any way to create mailbox from Asterisk dial-plan ?
Appreciate any suggestions
Noah Miller wrote:
At the time I set this up, MySQL replication was really designed for
one-way replication. Two way replication was possible, but required
somewhat unorthodox methods. (Maybe this has changed, I don't know).
Configuration is also a little tricky. It's not too bad to set it
Paul wrote:
Joe acquisto wrote:
Paul [EMAIL PROTECTED] Wrote: 4/30/2007 8:53 AM:
Joe acquisto wrote:
I have dual posted this to the user and biz lists.
Has anyone ever heard of someone running an Asterisk based system, yet
Has abandoning SugarCRM, and opting to
Hi:
I can try and answer some of your questions.
[EMAIL PROTECTED] wrote:
Hello All,
We have been doing Asterisk and CME implementations recently but we
almost always exlusively bring in analog lines and or PRI for PSTN
access to our systems. I have known about providers providing SIP
Justin Hamade wrote:
The 501 is more weird then that. The cat5 cable with the built in
power injector is cool but to use it with a PoE (802.3af) switch you
need a special cable (the pairs are just different you can probably
look it up and make your own).
Is this true? I read earlier on the
Richard Lyman wrote:
Paul wrote:
Joe acquisto wrote:
Paul [EMAIL PROTECTED] Wrote: 4/30/2007 8:53 AM:
Joe acquisto wrote:
I have dual posted this to the user and biz lists.
Has anyone ever heard of someone running an Asterisk based system,
yet Has abandoning
Klaverstyn, David C wrote:
All,
I have a Polycom 650 phone, when turned on displays “Checking application”.
Can any give me some information as to what is wrong? I have copied the
CFG files from a 601 phone to work with this 650.
1. You need at least SIP 2.0.1 (2.1.0 recommended minimum,
Paul wrote:
Third - I have enough exposure to Visual FoxPro to quickly rule it out
as a choice for anything new. The fact that somebody is proposing to
use it might give you the idea that they don't know what they are
talking about at all. BTW - my exposure to it did include things like
Hi all,
This is a simple concept, however I'm not entirely comfortable with
available applications and functions available to me to make this happen.
I have a simple dialout macro such as the following:
[macro-dialout];
arg1 = callerid number;
arg2 = phone numberl
exten =
Paul wrote:
Richard Lyman wrote:
Paul wrote:
Joe acquisto wrote:
Paul [EMAIL PROTECTED] Wrote: 4/30/2007 8:53 AM:
Joe acquisto wrote:
I have dual posted this to the user and biz lists.
Has anyone ever heard of someone running an
We are using VXI headsets with our Asterisk PBX as well as our legacy
PBX's. A nice feature of these headsets is that you can use the same
headset in either a USB port with their DSP translator cord or in a
traditional rj-11 port with another cord. This adds some redundancy to
your system if you
Stephen Bosch wrote:
Paul wrote:
Third - I have enough exposure to Visual FoxPro to quickly rule it out
as a choice for anything new. The fact that somebody is proposing to
use it might give you the idea that they don't know what they are
talking about at all. BTW - my exposure to it did
That's the way we want to go, but have been unable to divine the correct
settings for getting it working with MS Exchange.
CP
Tim Panton wrote:
If I were starting a project now, I'd
take a look at the (newish) support for IMAP storage for voicemail.
Richard Lyman wrote:
Paul wrote:
Richard Lyman wrote:
Paul wrote:
Joe acquisto wrote:
Paul [EMAIL PROTECTED] Wrote: 4/30/2007 8:53 AM:
Joe acquisto wrote:
I have dual posted this to the user and biz lists.
Has anyone ever
Hello to all
I need send a data to sofphones screen when I use a Dial () .
Thanks a lot
Regards
Andres Gomez
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
Hi all,
I am using a TDM400P card with 2 FXS and 2 FXO modules. Everything
seems nice, but i'm not able to make calls nor to receive any. When I
try to make a call, I keep receiven the all circuits are busy now
message, and when I receive calls, asterisk doesn't seems to care
(don't get anything
Andres Gomez wrote:
I need send a data to sofphones screen when I use a Dial () .
SendText()?
Regards,
Philipp
--
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Let's use IT to solve problems and not to create new ones.
Asterisk? -
I would like to know if anyone here knows the answer to the following question
I need to implement the following conferencing feature for my agents.
1. Agent receives call from caller
2. Agent conferences a verification service
3. After finishing the verification, agent
I am looking for a way to automatically close a meetme conference
when either a user hangs up or through an agi call?
Some method that would automatically terminate the meetme.
Is there a way to do that?
Jerry
___
--Bandwidth and Colocation provided
Andres Gomez wrote:
Hello to all
I need send a data to sofphones screen when I use a Dial () .
There is the applications SendText, SendImage or SendURL. Also, for SIP
phones you could possibly use SipAddHeader...
Thanks a lot
Regards
Andres Gomez
Anthony Rodgers wrote:
That's the way we want to go, but have been unable to divine the correct
settings for getting it working with MS Exchange.
Just for laughs...
what sort of problem do you have?
(Stinky, stinky MS Exchange... worst IMAP support -- but hell, maybe we
can find a solution)
Stephen Bosch wrote:
Justin Hamade wrote:
The 501 is more weird then that. The cat5 cable with the built in
power injector is cool but to use it with a PoE (802.3af) switch you
need a special cable (the pairs are just different you can probably
look it up and make your own).
Is this true? I
On 2007-03-26 01:46:40 -0700, Salvatore Giudice
[EMAIL PROTECTED] said:
This is a multi-part message in MIME format.
I opened up a ticket with them, but I'm not holding my breath. I think it's
time to start moving my DID's before the inbound stops working.
That seems like it was probably
If it supports the old Cisco POE, you might be able to try this:
568b
1 OrWh
2 Or
3 GrWh
4 Bl
5 WhBl
6 Gr
7 BrWh
8 Br
Phone Side
1 OrWh
2 Or
3 GrWh
4 BrWh
5 Br
6 Gr
7 Bl
8 WhBl
(From voip-info.org wiki, Cisco POE)
That config has allowed me to run 7940g's on a standard Dell POE switch.
On Mon, Apr 30, 2007 at 12:25:07PM -0500, Diego Quintana Cruz wrote:
Hi all,
I am using a TDM400P card with 2 FXS and 2 FXO modules. Everything
seems nice, but i'm not able to make calls nor to receive any. When I
try to make a call, I keep receiven the all circuits are busy now
message, and
Does anyone know of an (E)AGI or program to develop a IVR dial-plan
which will take a list of words and then do something when a unique
branch has been found.
i.e.
Say there's 3 words
demon
deacon
bishop
On a phone they'd be represented as
33666
332266
247467
So if the user enters 2 we know
Jeff Davis wrote:
The IP 501 supports both Cisco and 802.11af with different cables. While
there are pin assignments differences, there are also electrical
differences in the discovery protocols. The special cable is an artifact
of this.
I don't know of anyone who was able to make the phone
Hi Guys,
I am having an issue that I have been able to replicate and I want to know if
anyone else has this.
Extension 100 dials an external number. He speaks for 5 minutes and then
transfers the call to extension 200. Extension 200 speaks for 1 hour. When we
go through the call logs we see the
Hi Steve -
[macro-dialout];
arg1 = callerid number;
arg2 = phone numberl
exten = s,1,Set(CALLERID(number)=${ARG1})
exten = s,2,GotoIf($[${LEN(${ARG2})} = 10]?3:4)
exten = s,3,Set(ARG2=1${ARG2})
exten = s,4,Dial(${TRUNK}/${ARG2},,m)
exten = s,5,Congestion()
exten = s,105,Busy()
This macro
2007/4/30, Tzafrir Cohen [EMAIL PROTECTED]:
On Mon, Apr 30, 2007 at 12:25:07PM -0500, Diego Quintana Cruz wrote:
Hi all,
I am using a TDM400P card with 2 FXS and 2 FXO modules. Everything
seems nice, but i'm not able to make calls nor to receive any. When I
try to make a call, I keep
Andrew Kohlsmith wrote:
On Saturday 28 April 2007 11:22 am, Chris Bagnall wrote:
Thanks to all who replied to my thread a few days ago SIP devices with
packet loss tolerance. One of the suggestions that came out of that thread
was to replace routers at users' premises with ones that support
Howdy Noah,
I just re-read my original inquiry and noticed my original purpose for
mailing the list was not simple to dig out of the message.
Ultimately, the dialout macro works fabulous. My issue is that I'd like
to be able to override one particular SIP endpoint with its own unique
callerID
On Mon, Apr 30, 2007 at 12:25:07PM -0500, Diego Quintana Cruz wrote:
Hi all,
I am using a TDM400P card with 2 FXS and 2 FXO modules. Everything
seems nice, but i'm not able to make calls nor to receive any. When I
try to make a call, I keep receiven the all circuits are busy now
message, and
On Monday 30 April 2007 4:14 pm, bails wrote:
I'm still looking for a miniPCI ADSL chipset that Linux can use, or an
actual raw ADSL non-PCI chipset that I can design into an embedded
system. If anyone has any leads, please don't hesitate to contact me!
Any chance we can get to see this as
Hello everyone,
After several years of using Asterisk I have always been frustrated
by the support for DNS. I have seen all kinds of strange behavior
when Asterisk is used on a system with iffy DNS servers:
- no failover to other DNS servers in /etc/resolv.conf (might be a C
library thing)
-
On Mon, Apr 30, 2007 at 11:37:22PM +0300, Tzafrir Cohen wrote:
The logs show:
Apr 30 14:50:53 DEBUG[6003] chan_zap.c: Message status for 401 changed from
-1 to 0 on 1
Apr 30 14:51:00 NOTICE[6003] chan_zap.c: Got ZT_EVENT_REMOVED. Destroying
channel 3
Apr 30 14:51:16 DEBUG[6004]
Kristian Kielhofner wrote:
Hello everyone,
After several years of using Asterisk I have always been frustrated
by the support for DNS. I have seen all kinds of strange behavior
when Asterisk is used on a system with iffy DNS servers:
- no failover to other DNS servers in
Jerry Geis ha scritto:
I am looking for a way to automatically close a meetme conference
when either a user hangs up or through an agi call?
Look at MeetMe docs.
http://www.voip-info.org/wiki-Asterisk+cmd+MeetMe
create the MeetMe with the 'x' flag and then put inside it some marked
users
Hi Ed
Ed Nuñez ha scritto:
I would like to know if anyone here knows the answer to the following
question
I need to implement the following conferencing feature for my agents.
1. Agent receives call from caller
2. Agent conferences a verification service
No problem since
This is a pretty common setup. Just make sure you have ACL's restricting
traffic between your data and voice vlan's. Generally, we recommend more
than two VLAN's for QoS and security. Usually customers setup the following:
1.) Voice VLAN's for Phones
2.) Data VLAN's for workstations
3.) Voice
Hi All,
I have an issue with the ODBC voicemail storage option with asterisk. All
appears to work fine, however, I get several sql execute warnings. I was
wondering if anyone out there could help me get to the bottom of what is
causing this and how I could possibly go about rectifying it.
The
Evening,
My latest asterisk box is having a difficult problem. It is
configured with one TE210P and TDM400P with four FXO modules. I'm
running FC6.
The TE210P only has a single PRI.
When the system boots, it is completely random what order the zaptel
modules will get loaded in. Sometimes
mitch,
not that I can answer your problem but is this ver 1.4.1? I had a
similiar problem in that zapscan was updating the zaptel.conf and
nothing would work until I mucked with zaptel.conf.zapscan... I might
have the filename wrong as I have multiple files now :(... it has
zapscan in the
Try the intertex gateways http://www.intertex.se/
Here their page outlining the their QoS settings:
http://www.intertex.se/products/page.asp?iPageID=143
They have models with ADSL models and wireless access point components.
--
Salvatore Giudice
It's generally not recommended to put an analog and digital card in the
same box, however that being said Try this.
Write a little hack in /etc/rc.local
/sbin/modprobe wct4xxp
sleep 5
/sbin/modprobe wct4xxp
sleep 5
/sbin/ztcfg
sleep 5
/sbin/modprobe wctdm
sleep 5
/sbin/ztcfg
On Mon, Apr 30, 2007 at 07:14:22PM -0500, Mitch Jackson wrote:
Evening,
My latest asterisk box is having a difficult problem. It is
configured with one TE210P and TDM400P with four FXO modules. I'm
running FC6.
The TE210P only has a single PRI.
You have exactly two modules. either
We have the same problem, and it also showed up when clients made three+
way calls. The CDRs would show them making the same call three times
simultaneously as the destination field for the second and third calls
was still showing the first number they had dialed. One of our clients
caught it and
Your experience with database replication is not unique. I have seen
this happen with many flavours of database, not just MySQL. At the
critical sites where I've worked, database replication is not even on
the table as an option for precisely the reasons you state above: I have
yet to meet
Hi,
I have upgraded two production Asterisk servers to 1.2.18
from 1.2.17 in response to the security alert on April the 25th. Since
the upgrade one server has seg faulted 4 times and the other 2, nothing
else has changed on the two servers in the recent past. I use the
safe_asterisk script so
JR Richardson wrote:
Your experience with database replication is not unique. I have seen
this happen with many flavours of database, not just MySQL. At the
critical sites where I've worked, database replication is not even on
the table as an option for precisely the reasons you state above: I
71 matches
Mail list logo
