On Thursday, March 14, 2013, Joseph wrote:
Can someone refresh my memory how to backlist caller ID in
asterisk 1.8?
I had it working in ver. 1.4 but in 1.8 it changed.
I'm still using 1.4. In that I add a number to the blacklist with
CLI database put blacklist 0123456789 1
That is to add
On Monday, January 14, 2013, Salaheddine Elharit wrote:
my problem i have a lot of calls coming from this number
(0666xx) and i want to block it.
You can either create an extension to handle it, or you can use 1.4's
blacklist feature to block calls from all unwanted numbers. Danny's
dealt
On Wednesday, January 2, 2013, Frank wrote:
Is there a way to automatically ban IP address from
attackers within asterisk ?
As others have mentioned, fail2ban does a good job. However, it may
not be enough as these attacks sometimes come from older versions of
the SipVicious hacking tool that
Hi All,
Asterisk 1.4.22.1 on CentOS 5
I've configured my dialplan to limit the maximum call length on
outgoing calls. I've done this as I get the first hour of each call
free with my bundle but I pay through the nose if the call goes over
an hour.
Family members who live overseas sometimes ask
On Sunday, December 30, 2012, Logan Bibby wrote:
I believe its actually TIMEOUT(absolute)=value. The function name is case
sensitive.
Many thanks. I've changed my dialplan accordingly.
--
Geoff
--
_
-- Bandwidth and
On Friday, August 26, 2011, linux guy wrote:
Any comments on integrating a wireless POTS system into an asterisk
system ?
All you need is an ATA channel per handset ...
FWIW, I've got three DECT analog phones in my system: two are hooked
into a Linksys PAP2 and the third is hooked into a
On Friday, August 26, 2011, linux guy wrote:
Do any of the DECT systems handle multiple incoming phone lines ?
They don't. However, that's not an issue because Asterisk does.
Incoming, I have two PSTN lines, three SIP providers, and used to have
an IAX2 provider also. Asterisk integrates them
On Thursday, April 29, 2010, David Backeberg wrote:
What do people think about both products?
Bonus points for if people have bulk deployed these, either with TFTP
and configs pushed from a server, or some other good idea.
I can't claim the bonus points. However, I did have a couple of
On Sunday, January 10, 2010, Francesco Peeters wrote:
Yes, post your question clear and consicely, include all relevant
information and snip all unneccessary history.
Note that: no reply != not wanting to help...
It *is* obviously possible people just do not KNOW the answer!... (Oh
what
On Thursday, November 12, 2009, jonas kellens wrote:
Could someone advice on a gateway that can take analogue calls and
transfer them on my local network ?!
FWIW, I've had a few recommendations for the Linksys SPA3000. However,
I haven't tried this for myself yet since I'm still in the
On Tuesday, September 29, 2009, C F wrote:
You say no reliable internet, if you can get ISDN wouldn't the
providers offer Internet thru the same ISDN Physical links?
Some countries (e.g. UK) don't offer DSL or ATM over ISDN physical
links. If a physical circuit is used for ISDN, the only way
On Tuesday, August 18, 2009, Remco Barendse wrote:
But then again, who needs Skype for business purposes anyways, i
don't think there is a huge market for it.
Me ... at least in theory! Our cellphones have built-in Skype, so a
Skype gateway should give me call forwarding and diversion to our
On Tuesday, August 18, 2009, Gordon Henderson wrote:
I was under the impression that Three (who I guess you're using)
placed a regular call over their network then Skyped it at their
HQ - rather than have the Skype client actually reside in the
handset.. (And I'm suspecting their 3G
On Wednesday, July 22, 2009, Catalin S. wrote:
I lookin' for a call in number from UK or USA. Can somebody offers
me a peering for this or specify any sip provider that offers this
thing?
There are several providers who offer UK or US regional geographical
numbers for little or no cost if you
Hi All,
I'm using PrivacyGuard to filter calls from withheld numbers. A few
percent of incoming calls from my BT landline where I know the caller
does not withhold their number. BT deny that they're not passing CLI
from all calls.
In /var/log/asterisk/messages, the following three lines preceed
On Monday, June 15, 2009, Steve Howes wrote:
On 15 Jun 2009, at 13:36, Manoj Panicker - FOES wrote:
Excuse me?
You sent this message twice. Send it once, and wait for a reply.
Only received once here. My mail server is configured to remove
duplicated messages - but a different timestamp
Hi All,
I need to use partial matches on the CIDNAME family I have stored in
AstDB. For example, an organisation might have several numbers with
the same area code and the same first few digits:
1234 567890
1234 567889
1234 567824
...
I'd like to store these (e.g.) as CIDNAME/12345678*
Hi All,
Asterisk 1.4.12 on CentOS 5
Yesterday and today I got the following warnings in /var/log/asterisk/messages:
WARNING[2066] chan_sip.c: Got authentication request (401) on unknown REGISTER
to 'sip:acco...@sip.voipuser.org;tag=d8f15e1f30efddd35168b07dba9d540e.3922'
The corresponding bits
On Sunday, February 22, 2009, Mindaugas Kezys wrote:
How to determine which channel hung up first?
It doesn't seem to matter on my system since including the following
line in extension h always seems to record the channel that made the
call.
exten = h,n,Log(NOTICE,Call made via channel
Hi All,
Asterisk 1.4.12 on CentOS 5
I'm trying to increment an AstDB key with the length of the last
outgoing call. Here's what I've got for 01 UK geographical numbers:
exten = _01.,1,Dial(${UKGeographical}/${EXTEN},,g)
exten = _01.,n,Log(NOTICE,Call to ${EXTEN} lasted ${DIALEDTIME})
exten =
On Saturday, February 21, 2009, Philipp Kempgen wrote:
To be quite precise the documentation says
---cut---
g- Proceed with dialplan execution at the current extension if the
destination channel hangs up.
---cut---
So I would not expect the g option to have any effect if
Hi All,
Asterisk 1.4.12 CentOS 5
My ISP account includes nearly 500 minutes of VOIP calls per month but
the service is expensive for unbundled minutes. So I'm trying to find
a way to keep an accumulated total of calls made through that trunk so
that I can automatically switch to a lower-cost
On Wednesday, February 18, 2009, David fire wrote:
use the h exten. when someone hangup dial go to exten h. or put the
option in the dial command to go to the next priority on hangup but
there is a problem if during the call they transfer it to other
exten you dont have the next priority.
On Monday, February 16, 2009, Julian Lyndon-Smith wrote:
We also don't yet know the pricing structure of chan_skype ...
I thought it was $99 per channel for corporate licenses or $19 for a
single, personal license ... or have I got the wrong ChanSkype?
http://www.chanskype.com follow the buy
Hi All,
I'm looking for a way to filter the AstDB cidname family to show only
those entries with a specified area code in the Asterisk CLI. If this
were a SQL database it would be something like:
SELECT number, name FROM cidname WHERE number LIKE '1234%'
I've tried database show cidname 1234* and
On Monday, February 16, 2009, Jared Smith wrote:
If you have that many items in a database and want to do those types
of filters, why not stick them in a SQL database and use func_odbc
to retrieve them from your SQL database inside the dialplan?
Thanks for your suggestion. My Asterisk machine
On Monday, February 16, 2009, Jared Smith wrote:
Hopefully that helps make things a bit more clear.
It does - many thanks for your help.
--
Geoff
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On Thursday, February 5, 2009, Ralf Träskman wrote:
To dial an outside line i have to dial 0. I want to have that when
we dial local numbers, that is we are in the 08 area, I dont want
to have to dial 08, how to set this up in asterisk 1.6?
I have this in Asterisk 1.4. My local area numbers
Hi All,
Asterisk 1.4.12 on CentOS 5
Sorry for a question that I'm guessing is obvious to most of you.
I'm trying to revamp my dialplan. When I first created it, I had
something like:
exten = s,1,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})})
exten = s,2,Dial(${rgMain},${RINGTIME},t)
exten
On Thursday, February 5, 2009, Mark Michelson wrote:
Actually, jumping to priority n + 101 is a thing of the past, and
this will only occur now if you pass the 'j' option to Dial. Dial
will just go to the next priority on a timeout now, and the
DIALSTATUS channel variable will be set to
On Thursday, February 5, 2009, Philipp Kempgen wrote:
And in addition extensions.conf is a thing of the past. ;-)
extensions.ael is cleaner and easier to maintain for most purposes.
Oh-oh ... I don't think I can keep up with the rate of change ;-)
BTW, on a related note, I'm having some
On Thursday, February 5, 2009, Tilghman Lesher wrote:
The correct string is FAILED, not FAILURE.
Thanks. For info, *TFOT says:
PrivacyManager() sets a channel variable named PRIVACYMGRSTATUS to
either SUCCESS or FAILURE. If Caller ID is received on the channel,
PrivacyManager() does nothing.
On Thursday, February 5, 2009, Mark Michelson wrote:
I've tried it and you're correct. So it looks like the docs need a
bug report - any idea how I go about that?
Thanks again,
If you're using the 2nd edition of the book, check the preface, page xix for
contact information.
Thanks -
On Wednesday, February 4, 2009, D Tucny wrote:
I use a slight variant of this...
exten =
s,n,Set(CALLERID(name)=${IF(${ISNULL(${DB(cidname/${CALLERID(num)})})}?Unknown:${DB(cidname/${CALLERID(num)})})})
exten = s,n,NoOp(Caller ID name mapped to ${CALLERID(name)})
Basically the same as
On Wednesday, February 4, 2009, Ex Vito wrote:
For a simple (but flexible) case I would consider ODBC +
func_odbc. Here is the idea (in case you aren't aware of how it
goes...)
[... snip ...]
It may be a bit more work than using the Ast DB or other means, but it
has the advantage
Hi All,
Asterisk 1.4.12 on CentOS 5
I'd like to be able to look up each incoming CLI to retrieve an
associated name, if available, and then pass that to the extensions so
that they can see both the name and number of the caller. I'm not
after LDAP or anything else maintained externally, just a
On Saturday, January 24, 2009, j...@j4computers.com wrote:
This is, hopefully, just a case of brain fade.
With zapata.conf and zaptel.conf in place, asterisk loaded, no dial
plan and all LEDS on the card lit, I get no dial tone, plugging an
analog phone into ports 1 or 2, only a buzz and
On Tuesday, January 20, 2009, Darrin Henshaw wrote:
I would use the ${DIALSTATUS} variable. In your dialplan dial the
first trunk you wish, then afterwards examine the ${DIALSTATUS}
variable. If that is not equal to ANSWER then dial your second trunk
and so on.
For example:
exten =
On Tuesday, January 20, 2009, bilal ghayyad wrote:
What is the solution for this disaster?
I live in UK, where we don't use RJ11 for telephones and so need to
use adapters, which I just leave hanging out of the FXO ports. With
the adapters in place, it's difficult to plug the phones into the
On Tuesday, January 20, 2009, bilal ghayyad wrote:
What do u mean by clibing the tang of the RJ11 plug on the end of
the BT adaptor?
On an RJ11 plug, the casing includes a springy piece that locks the
plug into an RJ11 socket. When plugged in, the end of the springy
piece sticks out of the
Hi All,
I'm confused! My Asterisk system has a Zap trunk and three SIP trunks.
I'd like to configure the dialplan to route via the first trunk in a
list and if that's not available or it's busy, fall over to the
second, then to the third, etc.
AIUI Dial(Zap/1SIP/out1SIP/out2/${EXTEN}) rings all
On Thursday, January 15, 2009, David fire wrote:
hey it is preatty easy now i understand the problem
is simple
hangup in new location
dial steal code for asterisk is just an extension and it should start an
AGI
the system search for the call in the same group
bridge the
On Friday, January 16, 2009, ddf...@gmail.com wrote:
do you program in any language? if yes just read the chapters about
agi in the asterisk book you can find it in support section in
www.asterisk.org
I'm a reasonable PHP and VBScript programmer and have dabbled since
the 1980s in a wide
On Thursday, January 15, 2009, D Tucny wrote:
It's so much nicer to use packages, in the case of CentOS, RPMs...
that way everything installed is owned by the package and removal of
the package removes most of what was installed...
Thanks for the reply.
I must be missing something, since all
Hi All,
I'd appreciate some help on how to implement call stealing. That is,
where you dial a code to redirect any call on the system to your
handset.
I'm getting rid of my BRI service and I'm trying to replace the
functionality of my existing ISDN2e PBX (Cybergear Gold) with VOIP and
Asterisk.
On Thursday, January 15, 2009, David fire wrote:
and if you use the trasnfer app whit the features chann?
Thanks for the suggestion. I'll see if I can find it in the docs.
--
Geoff
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On Thursday, January 15, 2009, Danny Nicholas wrote:
Why not use call-conferencing? If you transferred your call into a
conference room, you could join the conference from any extension on
your *. When the caller hangs up, just end the conference.
Thanks for the reply.
AIUI, you need to set
On Thursday, January 15, 2009, Drew Gibson wrote:
Would SLA (Shared Line Appearance) work for this?
Put call on hold, press button beside flashing light on second handset?
Thanks for the reply.
I don't think it would work with my hardware. I've got two Nortel 355
analog handsets, one plugged
On Thursday, January 15, 2009, Jeff LaCoursiere wrote:
Cordless phones?
Sorry, couldn't resist :)
I've got some but the range isn't good enough to cover my entire
house. Besides which it's bad enough playing find the phone when a
cordless handset gets eaten by the settee or wanders off to the
On Thursday, January 15, 2009, Jeff LaCoursiere wrote:
I'm a bit confused as to how your old system exactly worked. When
you initially answer the phone (on presumably the wrong
extension), what did you do with that handset before getting up and
going to the right extension to steal it?
On Thursday, January 15, 2009, David Gibbons wrote:
I'm confused as to why you think leaving a phone off the hook is
better than parking the call and hanging up the phone.
Simply that you don't have to remember to park the call. With call
parking, if you forget to park the call before moving
On Thursday, January 15, 2009, Danny Nicholas wrote:
What about Chanspy()?
Thanks for the reply, but I suspect it won't do what I want.
AIUI, ChanSpy() doesn't transfer the call - it just lets another
extension listen in (and join in the conversation in whisper mode). So
(AFAICT) the call will
Hi All,
I'm running Asterisk 1.4.22.1 on a CentOS 5 machine fitted with a
TDM400P. When I upgraded from Asterisk 1.2.12.1, Zap stopped working.
Doing zap show channels etc from the Asterisk CLI results in an
error saying there's no such command.
The machine has Zaptel 1.2.9.1, which I've tried
On Wednesday, January 14, 2009, Jose P. Espinal wrote:
Have you tried recompiling/installing the new zaptel source before
Asterisk?
Thanks for the reply.
It's the old Zaptel source that was working with Asterisk 1.2.12.1 and
so was already compiled and installed prior to upgrading Asterisk.
On Wednesday, January 14, 2009, Carlos Chavez wrote:
Zaptel 1.2.9.1 will not work with Asterisk 1.4.22. I would recommend
you install Zaptel 1.4.12.1 or go to DAHDI.
Thanks for the reply. Uninstalling DAHDI and switching to Zap 1.4 did
the trick. I can now make calls to and from the PSTN and
Hi Everyone,
Got a setup as follows:
Telco Siemens HiCom 300E Asterisk1 IAX2 Trunk
Asterisk2 Siemens HiPath 4xxx
The solution works except for one problem. Incoming calls from the telco get
redirected to the Asterisk1 box with the correct extention, only if there is
a
I was wondering if anyone knows why when I register a user
agent like XLite with Asterisk I am noticing that the
port assignment on the sip show peers command shows the port to
be different than any of the other user agents. The other user agents are
logging in from different networks from
to do with asterisk.
lane
P.S. Anybody that can construct an if-then statement, whatever the
language, is a programmer. If you don't believe that then you must feel very
alone in your ivory tower.
On Thursday 19 May 2005 16:27, Preston Garrison wrote:
Again it all depends what you want to do
of
freeze and stop consuming cpu (I hope).
Is there a way (within asterisk) to flush these processes completely? Or
should I just run a cron job?
Thanks,
lane
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On Friday 13 May 2005 22:52, Adam Goryachev wrote:
On Fri, 2005-05-13 at 15:12 -0500, Lane wrote:
Hi,
KMail crashed because I saved too many asterisk-users messages ... now I
have to ask directly, instead of searching in my private archive :)
Here's the problem: I'm using MusicOnHold
On Friday 13 May 2005 22:52, Adam Goryachev wrote:
On Fri, 2005-05-13 at 15:12 -0500, Lane wrote:
Hi,
KMail crashed because I saved too many asterisk-users messages ... now I
have to ask directly, instead of searching in my private archive :)
Here's the problem: I'm using MusicOnHold
during the
Wait(300).
Can this be done?
Thanks,
lane
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up the sip
phone. If no one picks it up, it stops ringing after 15 seconds. The
bridged answering machine does its thing whenever it wants to.
Thanks.
Man it seems the more difficult the problem, the simpler the solution!
lane
___
Asterisk-Users
place the call or receive the call I cannot hear the remote user!!
They can hear me, though. (I'm inside the firewall).
My remote users are on XLite, if that makes a difference.
Anybody got an idea why the firewall is blocking traffic for these SIP phones,
but not for voipjet?
Thanks
lane
incoming calls from the PSTN. If I dial my
home phone from my cell phone asterisk does not even recognize the ring.
Below is (I think) the relevant config files. My question is: How do I make
asterisk recognize when the phone is ringing?
Thanks in advance
lane
;zaptel.conf
fxsks=4
fxoks=1
is ringing?
Thanks in advance
lane
;zaptel.conf
fxsks=4
fxoks=1
loadzone = us
defaultzone=us
;
;zapata.conf
[channels]
language=en
rxwink=300
usecallerid=yes
cidsignalling=bell
hidecallerid=no
callwaiting=yes
usecallingpres=yes
threewaycalling=yes
! I want this PBX to work so bad that I can almost taste it!
Please advise.
lane
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notwithstanding) , and I can deal with sound
quality later. But I don't get why she don't actually dial out, but instead
just looks at the phone line until it hollers at 'er.
And why she no answer the incoming call?
TIA!
lane
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On Wed, 29 Dec 2004 16:32:40 -0600, Lane [EMAIL PROTECTED] wrote:
I've been working on the local side of asterisk for several days, and I
have the in-house dial plan pretty well corn fingered to my satisfaction.
Today I began working on the other side to make asterisk do things like
place
On Wednesday 29 December 2004 17:22, Adam Goryachev wrote:
On Thu, 2004-12-30 at 09:32, Lane wrote:
I've been working on the local side of asterisk for several days, and I
have the in-house dial plan pretty well corn fingered to my satisfaction.
Today I began working on the other side
of that call? How could I discreetly begin a
recording of that call?
Thanks,
lane
P.S. I don't gotta ex-wife, I'm just saying what if?
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Hi,
Is it possible with asterisk to deliver a dialtone to a software phone, such
as kphone?
I'm able to dial, but the silence seems to confuse my users :)
thanks,
lane
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phone.
I can even hear the demo, but it is wa choppy. So I figure that the
choppiness will diminish once I can get the FXS module to load.
So ... how do I get wcfxs to load?
Thanks!
lane
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Hi,
I am new to asterisk and I am trying to get things set up so I can
prove to the boss it works and get the budget to do a full
implementation. Does anyone have an ata adaptor or an ip phone laying
around they would be willing to sell me for around 30-50 dollars, I will
need 2 of them.
correct it?
Lane Hoskins, MCP
Network Engineer
(540) 767-7600 main
(540) 767-7626 direct
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up that way in the dialplan) I can put other caller on hold and
answer line 8 simply by pressing the button.
Is this an easy thing to do that I'm simply not seeing?
Thanks,
Lane Hoskins, MCP
Network Engineer
Automated Horizons Inc.
Direct - 540.767.7626
Main - 540.767.7600
quality to called parties outside our system is intermittently
horrible: static filled and raspy where we have to ask people to repeat
themselves many times. Could this be related to powerline noise or
something like that?
Lane Hoskins, MCP
Network Engineer
Automated Horizons Inc.
Direct
: Lane Hoskins [mailto:[EMAIL PROTECTED]
Sent: Friday, January 30, 2004 4:22 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Call quality questions
Our basic system is as follows:
P4 3.0 Ghz w/ HT, 1GB PC3200 RAM, 120 GB HDD, RH
over. If
you have the files other than the POS30600.bin which I know is licensed
could you please send them to me so I can figure out if its my
files or my phone?? I really would appreciate any possible help with this.
Thanks,
Lane Hoskins, MCP
Network Engineer
540.767.7626
without changing anything)
#make dep
then in /usr/src/zaptel
make clean install
Hope this helps
Lane Hoskins, MCP
Network Engineer
540.767.7626
-Original Message-
From: Franz Edler [mailto:[EMAIL PROTECTED]
Sent: Thursday, January 15, 2004 9:19 AM
To: [EMAIL PROTECTED]
Subject
I'd be happy to give my docs to the project. I just noticed that it was
in progress after I posted but I'd be happy to help.
Lane Hoskins, MCP
Network Engineer
540.767.7626
-Original Message-
From: Jared Smith [mailto:[EMAIL PROTECTED]
Sent: Monday, January 12, 2004 1:56 PM
To: [EMAIL
dials for staff here) and then to
go to voicemail or fwd to a cellphone after that if the extension is not answered.
Has anyone done this that could provide an example for me or point me to better
documentation? We have searched extensively and not found anything yet.
Lane Hoskins, MCP
Network
Thanks David,
That is exactly what we had to do. We got some help from Digium as well
and have it taken care of.
Lane Hoskins, MCP
Network Engineer
540.767.7626
-Original Message-
From: David Gomillion [mailto:[EMAIL PROTECTED]
Sent: Tuesday, January 13, 2004 10:33 AM
To: [EMAIL
whatever you're
comfortable with.
Lane Hoskins, MCP
Network Engineer
540.767.7626
-Original Message-
From: Jean-Christophe Heger [mailto:[EMAIL PROTECTED]
Sent: Tuesday, January 13, 2004 10:58 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Best Linux Distribution
Il personally
Thanks we just figure it out a bit ago. It's amazing how simple some
things are when you just ask - and then realized that you were making it
too hard to begin with!! :-)
Lane Hoskins, MCP
Network Engineer
540.767.7626
-Original Message-
From: Jared Smith [mailto:[EMAIL PROTECTED
(like us) get up and running smoothly and
pointing to the correct places for help.
Again, Thanks to the entire community and I hope that our
documentation will be of help.
Lane Hoskins, MCP
Network Engineer
540.767.7626
image001.gif
?
Thank in advance.
Steve Lane
Unless you use a Valcom or Belkin solution. They make all different
types of amplifiers and zoning solutions for paging. Asterisk can work
with what ever you want hardware wise and the actual paging will sound
better than your typical phone system with paging.
Steve Lane
-Original Message
Nufone won't answer their phones. I am very interested in finding out
pricing from them as Jeremy stated they are very good with their rates.
Steve
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam Roach
Sent: Thursday, August 21, 2003 10:23 AM
To:
Hit me up off-line Jeremy. I want to know what your wholesale rates are.
Steve
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jeremy
McNamara
Sent: Wednesday, August 20, 2003 10:32 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] VoIP dialtone?
I am having problems trying to run
asterisk from a telnet session. I am able to su to root and the command
asterisk does not work. Any ideas why this may be occurring? I thought Asterisk
could be configured remotely as well as run remotely?
Thanks in
advance
Steve Lane
a bit when I leave out simple details.
Highest regards,
Steve Lane
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adams, Gavin
Sent: Friday, August 15, 2003 12:43 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Can I runAsterisk remotely from
with this?
Thanks
Steve Lane
. Can anyone guide me along to accomplish this?
I am trying to prove a point to the carriers owner. Thanks in advance.
Steve Lane
Vision Communications
I am trying to do the same thing you are doing. I am new to asterisk and
a friend of mine owns a carrier. They are using vocal data as the
platform, which is sip capable and uses sip phones. What I was trying to
do as well is register * with the redirect/registers with the carrier so
that they can
Would the firewall pose a problem? I thought Asterisk had the solution
for working behind a firewall?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nathan
Littlepage
Sent: Tuesday, August 12, 2003 8:19 AM
To: [EMAIL PROTECTED]
Subject: RE:
So in other words... Asterisk can do what the Avaya Conversant can do if
you have a full understanding of it? Please pray tell how to do these
things. I am all ears.
Steve Lane
Vision Communications
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[mailto:[EMAIL PROTECTED] On Behalf Of Jeremy
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