that is no longer available. And the U100 is obsolete.
So my advise stay away from Sangoma
Joseph
On 1/12/21 4:17 PM, John Kiniston wrote:
> Sangoma purchased Digium.
>
> You can find Sangoma cards at https://www.sangoma.com/telephony-cards/
>
> On Tue, Jan 12, 2021 at 2:29 PM bilal
Hello All;
We were using Digium cards, now I am not able to reach for digium website that
contains the telephony cards and Asterisk website currently is taking us for
Sangoma, so what happened in Digium cards?
RegardsBilal--
_
Hello;
I have 10 Caller IDs and I need each call (each time) to use one of these
Caller IDs to be the caller id.
I know that I can use this syntax as example:
exten => _90ZXX,1,Set(CALLERID(num)=01747576)
But how I can set the callerid each time from be one of the 10 caller ids that
are
Thank you a lot for your kindly help and reply. Actually it helped me a lot.I
was using _X. in the extensions.conf at the trunkinbound context.Can you advise
me what is the difference between _X. and s? In other words, when it is better
to use s and when it is better to use _X.?
Again, I am
Hello;
I am facing a trouble with the SIP service provider, they are saying that there
is a problem related to message option 200 (the heartbeat), so what is required
to add this for the sip configuration? Below is my sip debug trace log with the
them and the sip peer configuration:
[Sep 4
Hello John;
And for GSM calls, u were using sip trunk from asterisk to these gateways?
And how you were sending sms?
> I use VoIP Innovations and ThinQ (formally SIPRoutes) and they both support
> SMS. That way it’s very easy to write it into the dial plan.
RegardsBilal--
Hello;chan_dongle can be used for sms and for gsm calls at the same time, how?
Any small example how to send gsm calls through chan_dognle and how to send sms
through chan_dongle?
> You can use a cheap 3G-USB-dongle and chan_dongle.
--
Thank you Steve.Regarding to Goip32 that you used it before: how you were
handling the received messages?In other words: if you sent a message for
someone and he replied for you, how you were able to see the reply? And was it
possible to have any action based on his reply (for example, forward
Hello;
Is it possible to send SMS from asterisk? Using DAHDI or using what is possible?
And, is there a card that can be fixed in the machine and insert the SIM card
in this card to be used for GSM calls and sending SMS through asterisk? Through
which channel? Is it DAHDI or something else?
Hello;
I do not know if the following feature is depending on the phone (can be
configured on the phone it self) or need to be configured from asterisk itself:
Is it possible to configure general SIP Phone to have one button that can be
used as following:
By pressing on it and then entering
Hello;
Is it possible to configure one button on the IP Phone (like Polycom or general
SIP Phone) to indicate (at the phone display) that the line (the line that is
connected for FXO port) is busy or not? If it is not busy, the user can press
on the button to place outside call.
Also, is it
Hello;
I need to be able to send and receive voice calls through GSM network, so do I
need GSM adaptor that will be connected to FXO port or I can use GSM card that
can be connected to PCI or PCI-E slot in the computer and asterisk can see this
card through dahdi channel?I am afraid that if I
Hello
It will be amazing if possible to do sip trunk with any of social media
providers like: whatsapp, facebook, imo, viber, ... etc.Did anyone has luck
with this? RegardsBilal
Sent from Yahoo Mail on Android--
_
-- Bandwidth
Hello;
I am thinking to use atcom card which can be shown in this link:AXE2G4AN - GSM
card - Atcom_Ip phone,IP PBX,Asterisk Cards,Voip Products Manufacturer
|
|
|
| ||
|
|
|
| |
AXE2G4AN - GSM card - Atcom_Ip phone,IP PBX,Asterisk Cards,Voip Products Ma...
ATCOM is
Hello;
Does anyone has information if possible to setup SIP trunk with whatsapp? How
can we let asterisk send and receive calls from whatsapp?
RegardsBilal--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com
Hello List;
I am facing a trouble with a sip trunk on asterisk 1.4 and asterisk 1.8 and I
am getting the following debug, can someone advise me about the solution:
<--- SIP read from Provider_IP_Address:5083 --->INVITE
sip:22021782@Asterisk_IP_Address:5060 SIP/2.0 Via: SIP/2.0/UDP
k_l...@earthshod.co.uk> wrote:
On Wednesday 20 Jan 2016, bilal ghayyad wrote:
> Hello List;
> I am facing a trouble with a sip trunk on asterisk 1.4 and asterisk 1.8 and
> I am getting the following debug, can someone advise me about the
> solution: <--- SIP read from Provider_IP_Ad
Hello;
The the destination already have a call (talking) and someone called it, we
need the caller to hear a tone which indicate that the destination has a call
(busy) and the destination should hear a tone to indicates that someone is
calling him. How can we do this?
RegardsBilal--
Hello;
I am facing a trouble with A2Billing when using analogue lines because the
channels are not closing properly when dialing happen through A2Billing (it
seems the dialing scenario including the hangup is not handled properly through
A2Billing but I do not have control on this). But when I
Hello All;
After trying A2Billing and certainly when the trunk is analogue lines (FXO
ports), I faced a problem that the channels were not hanged up properly from
time to time which cause us to do restart for the dahdi. Without A2Billing, I
was able to handle the Dial scenario properly and no
Hello;
Instead of using Cisco Unity as voicemail for Cisco Call Manager, I need to use
asterisk to be the voicemail for the Cisco Call Manager version 7 which
supports SIP.
Did anyone try this? Was it a successful implementation?
If yes, I hope that someone gives a steps to help me.
Regards
Hello;
I have asterisk Asterisk 1.8.23.0-vici and Polycom 331 and I am able to
register from local area network and not able to register from outside the
office. Also from outside the office, I am able to register via PhonerLite
softphone and not able to register via Zoiper softphone.
So from
Hello;
Is there a method way to be able to dial the phone number through asterisk
from the outlook email contact?
Regards
Bilal--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us
...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad
Sent: Wednesday, December 18, 2013 9:46 AM
To: Asterisk Users Mailing List - Non- Commercial Discussion
Subject: [asterisk-users] Maximum number of users
Hello;
Can someone advise me what is the maximum number
Hello;
Can someone advise me what is the maximum number of users (IP Phones) that can
be supported by asterisk 1.8 or later?
Regards
Bilal--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to
Hello;
Is there Interface (web based interface) that I can login as admin, check the
emails and see the numbers that leaved voicemail and if possible to hear the
voice message, ... etc?
Regards
Bilal--
_
-- Bandwidth and
Hello;
I am looking for calls recording solution to do recording based on the network
traffic .. The solution to be competitive and appreciate if it is open source
.. Any suggested one?
Regards
Bilal--
_
-- Bandwidth and
Using Orecx, I can do search based on the extension or caller number or the
time or the agent login or the mix of these fields?
Regards
Bilal
On Tuesday, October 22, 2013 6:03 AM, Paul Belanger
paul.belan...@polybeacon.com wrote:
On 13-10-21 10:39 PM, bilal ghayyad wrote:
Hello;
I am
Hello;
I am looking for ADSL that supporting VPN so we can connect to it from our
IPhone using the VPN to be able to register at the asterisk PBX. Any
recommended one that is doing fine with voice? Also, does it support bandwidth
priority or shaping for the protocols?
Regards
Bilal--
Of bilal ghayyad
Sent: Tuesday, October 01, 2013 12:43 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] meetme conference password and time limitation
Hello;
We need to have admin page, so the administrator can create passwords to be
used to join the meetme conferences and can
Hello;
We need to have admin page, so the administrator can create passwords to be
used to join the meetme conferences and can determine the allowed time ..
Well, the admin interface can be done easy (I do not know if there is something
ready), and the password and the time limitation can be
On Wednesday, September 11, 2013 1:54 PM, longst longst...@gmail.com wrote:
I think GoIP gsm gateway also is a good choice
Sent from Shitian Long
On Sep 11, 2013, at 12:29 PM, bilal ghayyad bilmar...@yahoo.com wrote:
Hello;
I am looking for SIM adaptor to be connected with Asterisk
Hello;
I am using vicidial which is using asterisk 1.8, mean while when the extension
has voicemail, I always see the red light on the Polycom and hear the beep
sound (toot toot) in period time. Also, I can see at the LCD an option to
select it for accessing the voicemail but I am facing the
Hello;
I am looking for SIM adaptor to be connected with Asterisk to be able to send
and receive calls from the mobile operator and if possible the same adapter to
be used for SMS sending and receiving.
But what if anyone called this SIM card that is connected to this adapter and
no one
Hello;
I am installing asterisk and dahdi on ubuntu and I used my username bghayad to
login for ubuntu and do the installation, actually I feel my problem is related
to the username and permission but I am not able how to fix it, I am facing
now mainly the following two problems:
The first
Hello;
If our Digium Telephony Card does not support echo cancellation like
(1TDM410PLF or 1AEX410PLF), what is the best and simple way to overcome the
echo?
Regards
Bilal
--
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-- Bandwidth and Colocation Provided by
Hello
I need to deploy asterisk on production and same thing for DAHDI, which version
is recommended for this?
Regards
Bilal--
_
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New to Asterisk? Join us for
Hello;
Is it possible to configure in the sip.conf for the Phone to be auto answer?
Regards
Bilal--
_
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New to Asterisk? Join us for a live introductory webinar
But this not in the sip.conf, this in the extensions.conf, right?
Regards
Bilal
From: Yasin Suluhan ysulu...@gmail.com
To: bilal ghayyad bilmar...@yahoo.com; Asterisk Users Mailing List -
Non-Commercial Discussion asterisk-users@lists.digium.com
Sent
So it is not at asterisk configuration?
Regards
Bilal
From: A J Stiles asterisk_l...@earthshod.co.uk
To: bilal ghayyad bilmar...@yahoo.com; Asterisk Users Mailing List -
Non-Commercial Discussion asterisk-users@lists.digium.com
Sent: Wednesday, July 17, 2013
Hello;
Does asterisk support Digital Phone devices? If yes, what is the required cards
and in which channel to do the configuration? Is it dahdi or something else?
In other words, the customer does not need IP Phones.
Regards
Bilal--
Hello;
If I have load up to 220 extensions with 50 concurrent calls. Can one hardware
server carry all this load? What is the hardware server required for this?
Regards
Bilal--
_
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Hello;
We have a cisco switches but they are not PoE and we need only to have PoE
device so the cables come for it first to provide the power and then goes to
the switch (to be like batch panel), is there something like this that can be
used for the IP Phones?
Regards
Bilal--
Hello;
Anyone used PoE L2 network switches other than cisco and recommend this for us?
We need it to be stable and costly effective.
Regards
Bilal--
_
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New to
Hello;
I need to be able to send SMS messages for campaign or for specific users, also
I need to be able to receive SMS messages and do automatic reply.
Do I have to use dongle or extra channel? What is the difference?
Also, I read that there is SMS through sip, how this work and what is the
Hello;
Is there CTI module in asterisk with CTI client to login and logout and do
ready and pause?
Regards
Bilal--
_
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New to Asterisk? Join us for a live
Hello;
To let the Phone answer automatically, this can be configured from asterisk (at
the sip.conf for the phone)? Or it has to be from the IP Phone? Because, some
phones does not support auto answer, also we do not need to do it for each
Phone.
Regards
Bilal--
Hello;
When I type make menuselect and finding the channels that has the sign XXX
before it (this at the driver), how can I know the dependencies that are
causing this conflict?
Regards
Bilal
--
_
-- Bandwidth and Colocation
Hello;
Facebook and Whatsapp sort-of support XMPP, so we can use Jabber to communicate
with them. But, how much jabber channel in asterisk is stable and updated?
Regards
Bilal
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Hello;
There is no free channel to be used to have integration between asterisk and
skype? What is the software that I can use to send and receive chat messages on
skype network?
Regards
Bilal
--
_
-- Bandwidth and Colocation
Hello;
As I am using vicidial and its asterisk version which is 1.8, I need to know
the required channels to be existed so the asterisk will support fax, SMS,
gtalk, Jaber? In other words, how I can know that it is enabled in this
asterisk (actually it is 1.8.21-vici)?
Regards
Bilal--
Hello All;
Wanpipe is working only with sangoma cards so it does not work with digium
cards?
Also, who is better: to have echo canceler built in with the hardware or using
olsec?
Regards
Bilal
--
_
-- Bandwidth and
Hello;
To be able to send and receive faxes through asterisk and to be able to have
trunk with google voice and to be able to have integration with those that
support Jaber .. What are the channels and libs that I have to be sure that
they are existed?
Regards
Bilal
--
Hello;
What is the best method to let the voice quality through Dahdi channels to be
clear and no echo? Is it the wanpipe or it is working only with sangoma?
Regards
Bilal
--
_
-- Bandwidth and Colocation Provided by
Hello;
I need two scenarios:
1) If someone sent SMS message, then we need to query information from the
database based on information sent by the SMS (like the name or the mobile
number), after querying from the database, we need to reply by the SMS. Can
asterisk do this? To which level?
2)
Hello;
As I am using vicidial and still vicidial is using asterisk 1.4, so how is the
SMS module with asterisk 1.4? Is it stable?
Also, I am looking to integrate with social medial like whatsapp and facebook,
so how is asterisk 1.4 with jaber channel?
Regards
Bilal
--
Hello;
How I can compare between Asterisk 1.8 and 11 with reference to the following
points:
1) SMS.
2) gtalk and other social media.
3) GUI.
4) Any main difference?
Regards
Bilal
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Hello;
I have a SIP trunk with a service provider, the caller from landline or mobile
is hearing us very well, but the agent who is sitting on the handset is not
hearing well, the voice at the agent is not crystal (like he is talking from
well or far deep place). Although the IP Phones are
Hello;
Is there any modules or channels or integration between asterisk and any of the
following:
whatsapp, facebook, viber, yahoo and hotmail messanger?
Regards
Bilal
--
_
-- Bandwidth and Colocation Provided by
Hello;
I am facing a problem to let Cisco IP Phone 7942G register on Asterisk. The
firmware has been downloaded from the TFTP successfully and currently I am
running this load SIP42.9-3-1SR2-1S*
I feel that there is a problem in the SEPMAC.cnf.xml but really I do not know
which one to be used
Hello;
The phones are registering now. I found a SEPMAC.cnf.xml file and I used sip
firmware version 8.3 and I configured nat=no at sip.conf and nat to be false in
xml file.
But I am facing a time problem, I am in Kuwait country and the time that is
appearing at the Phones screen is delayed
am using chan_mobile for call termination, you can use it
but you need
to tweak chan_mobile.c it is broken from a long time.
let me know if you want give it a try.
On Mon, Mar 11, 2013 at 6:22 PM, bilal ghayyad bilmar...@yahoo.com
wrote:
-
What are the elements
-
What are the elements of this solution? Is it only: 3G
dongles and chan_dongle only? Or there are something else?
Bash and perl programing, asterisk and chan_dongle.
* Bash and perl programing to do what? It is going to use AMI instead of
sending the messages from the
I am not mixing. I need this for LAB testing.
How? This PCI passthrough, how to enable it on virualbox?
---
Hi All;
How to let the virualbox (ubuntu OS) to be able to see
the digium card? Because when I install elastix or asterisk
with dahdi, it is not able to see the digium card
something
by the way did you install Elastix in the virtual box
?
Sent from Shitian Long
On Mar 8, 2013, at 10:21 PM, bilal ghayyad bilmar...@yahoo.com
wrote:
Hi All;
How to let the virualbox (ubuntu OS) to be able to see the
digium card?
Because when I install elastix or asterisk
wrote:
Yes, you can check solutions from sangoma and khomp.
Saludos/Regards
--
Ing. Gerardo Barajas Puente
Proyectos Especiales/Preventa | www.neocenter.com
http://www.neocenter.com
T:+52 (55) 8590-9000 x 7003
On Fri, Mar 8, 2013 at 6:32 PM, bilal ghayyad bilmar
Hello Gertjan;
I've heard a lot about it but I'm running Asterisk on ESXi5 Dell boxes without
problems
* How your ESXi saw the digium? Is it using PCI Passthru?
Regards
Bilal
-
It's called PCI Passthru and from what I've tried, the
timing is horrible in a
Hi All;
How to let the virualbox (ubuntu OS) to be able to see the digium card? Because
when I install elastix or asterisk with dahdi, it is not able to see the digium
card if the installation though the virualbox .. What is the solution?
Regards
Bilal
--
Hi;
If my landline service provider does not provide the ability to send the SMS,
and I need to send SMS from asterisk, then what is the required? How?
Is it possible to send SMS from asterisk using SIM card to be connected via GSM
adaptor connected to FXS ports? Or HOW?
From the other side,
Hi;
I used vicidial for call center and I would like to try elastix. Can someone
advise about the advantages?
Does Elastix has a screen for the agent to login/logout from their PC and deal
with the inbound/outbound calls and Integrated with the CRM?
Regards
Bilal
--
Dears;
I am facing a problem in disconnecting the calls, it is related to the
rtptimeout (disconnecting if there is no RTP packets from both sides).
My Asterisk Box is behind NAT but there is a static real IP address at the ADSL
router. We call from the Mobile to the PSTN analogue numbers
Dear;
Using Cisco IP Phones: How I can assign a button for a function. For example,
if we pressed on this button, then we need to pickup the call from the group.
Another thing:
If the button pressed, then the call forward function to be enabled (and it
should appear on the phone that it is
and create messages. You might have to go a
little higher level
like C or Perl, but it sounds very doable.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com]
On Behalf Of bilal ghayyad
Sent: Tuesday, January 22, 2013 4:27
Both: SPA and 7900. let us say 7942. How?
Regards
Bilal
Dear;
Using Cisco IP Phones: How I can assign a button for a
function. For
example, if we pressed on this button, then we need to
pickup the call from
the group.
Which model line? The SPA series, or the 7900 and
Dears;
Can someone advise me where to find a technology (open source) that let us able
to integrate with social media like whatsapp and facebook? And use this in call
center (queuing the messages and routing it for agent)?
Anyone give me a light to start?
Regards
Bilal
--
Dears;
I am using asterisk for call center and I used also VICIDIAL. But they are fine
for voice, I need the agents to be able to handle email and web chat messages
as long with the voice calls, in addition to be integrated with the social
media like Facebook and twitter.
Where I can find
that you create. The
format is very
specific.
On Tue, 1 Jan 2013, bilal ghayyad wrote:
* How can I know this format? Because I need to know
what should I place
in this file so it will execute Paging for this group
of Phones?
This may help:
http://www.voip-info.org/wiki
Hi;
How can I know the duration that the DAHDI channel is still used? I need to
know its status and since when it is in this status, how?
Also, is it possible to hangup the channel if it has been openned more than 90
minute? Other than using the timeout in the Dial command (because this I know
the
speaker (without pickup the handset).
By using AMI, then I can build PHP script that will use the AMI to do the Page?
Thanks and Regards
Bilal
A call file is a text file that you create. The
format is very
specific.
On Tue, 1 Jan 2013, bilal ghayyad wrote:
* How can I know
How many customers will be receiving these reminders?
* It is required that all the employers at the company to hear this on their IP
Phones.
What religion is this targeted to?
* Islam.
A call file is a text file that you create. The format is
very specific.
* How can I know this
question:
What was u meaning by call file and why it is required to place them in the
'astspooldir.'?
Regards
Bilal
Please don't top-post.
On Thu, Dec 27, 2012 at 1:29 PM, bilal ghayyad bilmar...@yahoo.com
wrote:
How can I have Paging on Asterisk to call for pray?
The pray is 5 times
On Thu, Dec 27, 2012 at 1:29 PM, bilal ghayyad bilmar...@yahoo.com
wrote:
Hello;
How can I have Paging on Asterisk to call for pray?
The pray is 5 times in a day and there is a timing for pray
(actually it can
be existed in a text file or database for the next 2 or 5
years).
My
Hello;
How can I have Paging on Asterisk to call for pray?
The pray is 5 times in a day and there is a timing for pray (actually it can be
existed in a text file or database for the next 2 or 5 years).
My question is compound from two parts:
How can I have Automatic Page?
The automatic page
Hi All;
How I can acheive the following:
From sip client softphone (from the iPhone for example), if I dialed a number
that I need to call it, then a call to be initated to a specific number
through DAHDI channel and another call to be initiated for the destination
number (the number that I
Hello;
I remember that I saw at asterisk website (this was maybe before 1 year or
around) some pages are talking about having SDK and APIs for asterisk that will
be used to build softphone for mobile and will be used to build some
applications for asterisk, also it was mentioned in this page
Hi;
How I can make my configuration to allow the sip phones only from specific IP
addresses range (for example from 192.168.10.1 - 192.168.10.50) to be allowed
to connect for asterisk?
In other words, in addition to be authenticated based on the username and
password, it is required that the
Dears;
It seems my service provider is requesting a complicated settings to allow me
to send from behind NAT.
What they said:
It shouldn't matter as long as you are handling the NAT correctly your end. We
do not fix NAT so if you're sending internal addresses in your INVITEs or SDP
then
Dears;
What Jian said is the right and it worked.
But I have the following questions:
Why 192.168.10.2 is wrong and I have to use 192.168.10.0? Also, do I have to
set the localnet or it is enough to set the externip?
From the other side, I am using Asterisk 1.8.12.0 and when I was searching
Dears;
I am facing the following problem:
Already we requested from the service provider to enable the auto jumping
service for our analoge telephone lines, so because we have 4 telephone lines
from the service provider, then if you called line # 1 and it was busy, then
the call will be sent
Actually I am not talking on how to handle it in the extensions.conf because I
am doing same as you wrote. But even so, I am facing a problem that some calls
are captured and some calls are not captured.
Currently, I set the callwaiting=no in the chan_dahdi.conf, it seems it is
working fine.
Dear AJS;
I have fedora core 14 and I did yum install libsrtp-devel and it is already
existed, the only thing happened that it updated it.
Currently:
libsrtp-1.4.4-1.20101004cvs.fc14.i686
libsrtp-devel-1.4.4-1.20101004cvs.fc14.i686
Again, I did make menuselect and the same problem, I am not
Hi;
It seems the SNOM Phones are requesting to have SRTP but I do not have the
module res_srtp.
I tried to compile it with asterisk 1.8, make menuselect, but I found that it
can not be used (I am not able to select it) with the following details:
Secure RTP SRTP
Depends on: srtp E
Can use:
Hi All;
Is there a module (addon or already built in) that enable us to receive the fax
on the telephony card and save it as image (or any other format) and sent it to
email?
Regards
Bilal
--
_
-- Bandwidth and Colocation
to do with RPID. Then you need to set allowrpid=yes in
the sip peer
settings of A party and B party. I did that on CISCO 79X0
phones and it
worked perfectly,
Regards,
Sammy
On Tue, Aug 7, 2012 at 3:43 AM, bilal ghayyad bilmar...@yahoo.com
wrote:
Hi All;
Asterisk 1.8.11-cert1
I
Dears;
I discover that I have to place the wave files in the
/var/lib/asterisk/sounds/custom/
So, can I understand that the only solution I have is to copy the files that
are existed in the path /var/lib/asterisk/sounds/en/ to the path
/var/lib/asterisk/sounds/custom? Or there is any other
Hi All;
Asterisk 1.8.11-cert1
I need to do the following, how?
If my extension is 500 and I need to call the extension 501, so when dialing
501, then I need to be able to see the name of the 501 (for example, the name
was: Mike, so I need to see at my IP Phone that I am calling Mike which is
Hello;
What is the difference between using the Background Playback in Asterisk 1.8
without cert and Asterisk 1.8 cert?
I surprised that in cert version, I do not hear the sound ! And it is not
working properly, but in the normal version, it is working.
So what is the new?
Is it the version?
Hi All;
Really it is miserable.
I bring 8 Digium Phone D40 and I used them with a customer, the voice quality
is bad internally (between the extension), there is no clearance at all ! We
are hearing the voice like another person.
The used codec is ulaw.
The firmware version is: 1_1_0_0_48178
Dears;
Is it possible with Asteisk to have IPTV (ability to show the TV channels using
the video over IP, but to be live). In other words, to use Asterisk to watch
the TV Channels.
Which open source that can do this, so we can install it on the same asterisk
machine?
Also, is it possible to
Selby wcse...@selbytech.com
wrote:
On Wed, Jul 11, 2012 at 4:56 PM, bilal ghayyad bilmar...@yahoo.comwrote:
Fine, did you read the question well and understand
about what I am
asking?
Perhaps I did not understand what you were
asking. I thought you were
wanting to do something
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