On Tue, Jun 17, 2008 at 12:07 AM, Jay R. Ashworth [EMAIL PROTECTED] wrote:
On Mon, Jun 16, 2008 at 11:11:00AM -0400, Steve Totaro wrote:
On Mon, Jun 16, 2008 at 10:35 AM, Jay R. Ashworth [EMAIL PROTECTED] wrote:
On Sun, Jun 15, 2008 at 01:25:18PM -0400, Alex Balashov wrote:
Is there a
On Tue, Jun 17, 2008 at 6:45 AM, Sherwood McGowan
[EMAIL PROTECTED] wrote:
Matt Florell wrote:
Hello,
I guess I am one of the lucky few to have one of these handy
screwdrivers and it saved me when my son(aged 2) somehow locked
himself in a bedroom and couldn't unlock the door. The door knob
On Tue, Jun 17, 2008 at 2:56 AM, Atis Lezdins [EMAIL PROTECTED] wrote:
On Tue, Jun 17, 2008 at 6:45 AM, Sherwood McGowan
[EMAIL PROTECTED] wrote:
Matt Florell wrote:
Hello,
I guess I am one of the lucky few to have one of these handy
screwdrivers and it saved me when my son(aged 2) somehow
I was looking for an option to weigh against a full blown asterisk system.
If I use asterisk as an expensive ata then there isn't much point in keeping
the key system is there? While I could hack a solution together (pap2 and
Wrt54gs running * on openwrt comes to mind) I'd really rather not.
On Tue, Jun 17, 2008 at 10:03 AM, Steve Totaro
[EMAIL PROTECTED] wrote:
On Tue, Jun 17, 2008 at 2:56 AM, Atis Lezdins [EMAIL PROTECTED] wrote:
On Tue, Jun 17, 2008 at 6:45 AM, Sherwood McGowan
[EMAIL PROTECTED] wrote:
Matt Florell wrote:
Hello,
I guess I am one of the lucky few to have one
On Tue, Jun 17, 2008 at 8:34 AM, Sherwood McGowan
[EMAIL PROTECTED] wrote:
Bikrish Amatya wrote:
Hi all
I am using asterisk as pbx for my company. My company has requirement
that all the incoming and outgoing calls should be recorded for all the
extensions and should be able to play recorded
Dear Sherwood,
Thanks.
Just three questions:
1. Will I be needing Apache or Asterk-stat will handle itself?
2. Are there How-tos for integerating asterisk-stat with asterisk?
3. My Recordings are being saved in the default folder i.e:
/var/spool/asterisk/monitor/ in .gsm format. When I wish
Hi:
I configured asterisk for voicemail service.My main configuration files are:
extensions.conf
[from-pstn]
exten =gt; 9711315,1,Dial(SIP/3000,30)
exten =gt; 9711315,2,VoiceMail([EMAIL PROTECTED])
exten =gt; 9711315,3,PlayBack(vm-goodbye)
exten =gt; 9711315,4,HangUp()
On Tue, 17 Jun 2008, randulo wrote:
On Tue, Jun 17, 2008 at 7:37 AM, Gordon Henderson
[EMAIL PROTECTED] wrote:
But maybe an AVM Fritz! box will work for you too...
Would anyone care to recommend a good quality, stable ATA these days
for just a single cordless phone connected to one SIP
Afternoon All,
Does anyone here have any experience with an Audiocodes Mediant 2000?
I know its a bit 'non asterisk' but i figured you guys are as likely
as any to have come across them. I'm having a few problems with one,
i.e. its not sending screening/privacy flags although it is sending
On June 17, 2008 01:45:43 am randulo wrote:
The screwdriver is reversible, it swings both ways, pull out the shank
and stick it in the other way, it becomes a Phillips. I'm tellin ya,
there Digium engineers are good!
Most every pocket screwdriver that is sold as a promotional item is like
I get that a lot since moving to 1.4.21 (from 1.4.18 or something).
[Jun 17 09:19:54] WARNING[22053]: res_config_mysql.c:360 update_mysql: MySQL
RealTime: Failed to query database. Check debug for more info.
Question 1: what debug file should I be looking at?
Mick
Just an addition: that happens big time when I do a sip reload from the
CLI
I know this should help me already, but it doesn`t
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mike
Sent: Tuesday, June 17, 2008 09:23
To: 'Asterisk Users Mailing List - Non-Commercial
On Tuesday 17 June 2008 04:05:58 fateme fatah wrote:
I configured asterisk for voicemail service.My main configuration files
are:
voicemail.conf
[ff_tutorial]
555 =gt; 1234567,3000,[EMAIL PROTECTED]
But when I dial 9711315, after 30s I hear goodbye and call hangups.
in console:
--
What I have done for our office is actually built my own interface with
php and used our SQL database to store the information. Basically I keep
all the recordings in gsm format, and store them however I want. I use
MixMonitor and use DeadAGI to run a script to rename the file and move
it to
On Tuesday 17 June 2008 08:23:08 Mike wrote:
I get that a lot since moving to 1.4.21 (from 1.4.18 or something).
[Jun 17 09:19:54] WARNING[22053]: res_config_mysql.c:360 update_mysql:
MySQL RealTime: Failed to query database. Check debug for more info.
Question 1: what debug file should I be
Is it possible on a TE220p to deactivate the hardware echo canceler at
will ? (With a function in the dialpan for example)
example for fax SDA ,beeing able to shutdown the echo canceler could
give better results don't you think ?
___
-- Bandwidth and
List,
Having a little trouble with the following. Let me prefix by saying I
have blind transfers working from the following setup.
Inbound call [from-zap] (SIP/sv0071iv) answers.
Zaptel - Asterisk - SIP extension
SIP extension then blind transfers [from-sip]
---
SIP extension - Asterisk -
I always seem to figure my issues just after I post to the list. Had
to add a Wait(.5) after the hookflash.
--
[AA]
exten = s,1,Wait(.5)
exten = s,n,Background(vm-whichbox)
exten = s,n,WaitExten
exten = _5XXX,1,Playback(transfer)
exten = _5XXX,n,Flash()
exten = _5XXX,n,Wait(.5)
exten =
I've got the following setup:
PhoneA -
router -
vpn -
router-
asterisk (SIP / ISDN)
PhoneB -
asterisk (SIP / ISDN)
If phone A is connected to phone B (sip-sip), there is a noticable delay
(up to 2-3 seconds) between me speaking and the other end hearing.
If phone A calls out
On Tue, Jun 17, 2008 at 11:39 AM, Julian Lyndon-Smith [EMAIL PROTECTED] wrote:
I've got the following setup:
PhoneA -
router -
vpn -
router-
asterisk (SIP / ISDN)
PhoneB -
asterisk (SIP / ISDN)
If phone A is connected to phone B (sip-sip), there is a noticable delay
(up to
Benoit Plessis wrote:
Is it possible on a TE220p to deactivate the hardware echo canceler at
will ? (With a function in the dialpan for example)
example for fax SDA ,beeing able to shutdown the echo canceler could
give better results don't you think ?
All echo cancelers using Zaptel/DAHDI
LLCs?
On 6/16/08, Jay R. Ashworth [EMAIL PROTECTED] wrote:
On Sat, Jun 14, 2008 at 11:13:31PM -0400, C F wrote:
Happens in the commercial world all the time; it's a common way to get
cash out of the corporation -- a business's building is owned by the
corporation's owners, and rented to
Hi Steve - the vpn is a consistent as the sip-IDSN has to go through
the VPN first to get to asterisk.
i.e. to make an outside call, PhoneA goes through the vpn to the
asterisk box, and out through isdn.
Julian
Steve Totaro wrote:
On Tue, Jun 17, 2008 at 11:39 AM, Julian Lyndon-Smith [EMAIL
Dear All
I need help to implement the follwoing Senario:
1- Incoming SIP call comes to asterisk and putting caller on MOH
2- While the caller is on MOH , dialing out other party and when asterisk
recive ANSWER , MOH should be disconnected, then bridging the 2 call legs
Appreciate
You are probably confusing corporate tactics to pay less taxes vs
corporate tactics to protect assets. The first does provide some
asset protection but is mainly to pay less taxes. The second is to
basically hide assets through totally legal LLCs.
Thanks,
Steve Totaro
On Tue, Jun 17, 2008 at
Kevin P. Fleming wrote:
Benoit Plessis wrote:
Is it possible on a TE220p to deactivate the hardware echo canceler at
will ? (With a function in the dialpan for example)
example for fax SDA ,beeing able to shutdown the echo canceler could
give better results don't you think ?
All echo
Hi,
Did someone try to package new releases for ubuntu version like
gutsy/hardy ?
thanks
--
Cyril SCETBON
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You can use the m flag of dial on the incoming sip channel, such as:
exten = s,n,dial(Local/[EMAIL PROTECTED]|60|m)
Fred Posner
On Jun 17, 2008, at 12:04 PM, Mohammad Mirzaee wrote:
Dear All
I need help to implement the follwoing Senario:
1- Incoming SIP call comes to asterisk and
On Tue, Jun 17, 2008 at 12:00:18PM -0400, C F wrote:
On 6/16/08, Jay R. Ashworth [EMAIL PROTECTED] wrote:
On Sat, Jun 14, 2008 at 11:13:31PM -0400, C F wrote:
Happens in the commercial world all the time; it's a common way to get
cash out of the corporation -- a business's building is
On Tue, Jun 17, 2008 at 1:02 PM, Jay R. Ashworth [EMAIL PROTECTED] wrote:
On Tue, Jun 17, 2008 at 12:00:18PM -0400, C F wrote:
On 6/16/08, Jay R. Ashworth [EMAIL PROTECTED] wrote:
On Sat, Jun 14, 2008 at 11:13:31PM -0400, C F wrote:
Happens in the commercial world all the time; it's a
On Tue, Jun 17, 2008 at 01:05:59PM -0400, Steve Totaro wrote:
On Tue, Jun 17, 2008 at 1:02 PM, Jay R. Ashworth [EMAIL PROTECTED] wrote:
On Tue, Jun 17, 2008 at 12:00:18PM -0400, C F wrote:
On 6/16/08, Jay R. Ashworth [EMAIL PROTECTED] wrote:
On Sat, Jun 14, 2008 at 11:13:31PM -0400, C F
The asset protection entities are completely legal. There's nothing
wrong ipso facto with doing it.
The question is only whether they will succeed in protecting your assets
when your assets are actually challenged. It depends on the size and
scope of the judgment, the circumstances in which
Michael,
I agree. Here we use e1s(which have even more channels). Some clients
just don't want to change some if their old infrastructure.
Thanks
Michael Graves wrote:
I just hafta ask, why does one face down a requirement for 48 FXOs?
Would it not be more practical to have 2 T-1s dropped
Some customers are locked into two year contracts.
That was the answer I got when adding four POTS lines to a system with
four BRIs...
Thanks,
Steve Totaro
On Tue, Jun 17, 2008 at 1:39 PM, James Mutuku [EMAIL PROTECTED] wrote:
Michael,
I agree. Here we use e1s(which have even more
Hi list,
we upgraded to v1.6 and have a problem understanding the queue() behaveour
of the v1.6 in queues.
we try to set the queue up to not hangup if an agent answeres the call but
then hangs up again.
we would then like the queue to go on in the dialplan. But the queue does
not want to go
Can the PAP2 be set up such that a second call will ring the second line
when the first is busy but only register once with the SIP provider? A beep
tone on the same line to denote another incoming call just will not do, The
second port needs to act like a seperate line tied to the same DID in a
On Tue, Jun 17, 2008 at 06:45:30PM +0200, Cyril SCETBON wrote:
Hi,
Did someone try to package new releases for ubuntu version like
gutsy/hardy ?
The Ubuntu packages are based on the Debian ones and basically packaged
from the same repository.
http://pkg-voip.alioth.debian.org/
You can
Hi all
I appreciate the help that you have given me on call recording. I would
like to share how i achieve the way i wanted. I used monitor and soxmix
for this. First i used monitor to record the calls and made use of
system command to create directory of each extension and inside each
Hi all
In my company there is oracle database which has the information about
the client. Now my requirement is... when my clients calls to our
company .. they should be able to get information about them when they
call to our pbx. I mean how can reterive information from oracle
database and
How can they even set such 1234567890 callerIDs anyway?
For example, our inter/intra state calling depends a lot on the callerIDs.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt Florell
Sent: June 13, 2008 8:20 AM
To: Asterisk Users Mailing List -
http://www.msnbc.msn.com/id/25119259/
Anyone know if this was built using Asterisk? Seems like a perfect
vehicle for it's deployment.
Regards,
Dean Collins
[EMAIL PROTECTED]
+1-212-203-4357 (Direct)
+1-917-207-3420 (Mobile)
+61-2-9016-5642 (Sydney in-dial)
http://www.Cognation.net
2008/6/16 Syed Nasruddin [EMAIL PROTECTED]:
Thanks for the link. I think I will be using this product.
It's very, very good. You can hook it up to MySQL instead of sqlite if
needed, just e-mail support.
--
http://www.suretecsystems.com/services/openldap/
And there are people like me who still can't get PRI's for less than
$1100/month. (Granted, I doubt I'll ever need a pri for the business I
am with now, but I was with an ISP for a long time that still supported
dial-up and we had 8 PRI's with a bulk discount that got them for us at
Johann Steinwendtner wrote:
I thought the ec gets disabled only by the ec disable tone and not the CED
tone.
The CED tone *is* the echo canceler disable tone.
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - The Genuine Asterisk Experience (TM)
I like Grandstream 286s No, seriously
Thanks,
Steve T
On Tue, Jun 17, 2008 at 2:07 PM, Eric Fort [EMAIL PROTECTED] wrote:
Can the PAP2 be set up such that a second call will ring the second line
when the first is busy but only register once with the SIP provider? A beep
tone on the
Dean
One of the firms involved, playareacode.comhas a division called Big
Games, and they used Asterisk as a platform to create an interactive
mystery game.
http://itp.nyu.edu/blogblender/2007/10/15/the-mystery-of-the-beautiful-c
igar-girl-location-plotting-and-ia/
I'd place my bets on
Here are some other interesting applications they built off Asterisk
http://itp.nyu.edu/blogblender/2007/11/15/voice-recognition-with-lumenvo
x/
http://www.prophecyboy.com/itp/redial/booty-dialer-update/
http://www.prophecyboy.com/category/itp/redial/
On Tue, Jun 17, 2008 at 1:48 PM, Andrew Kohlsmith (lists)
[EMAIL PROTECTED] wrote:
Most every pocket screwdriver that is sold as a promotional item is like that.
It's not always good; I cut my hand pretty badly when the phillips end slid
clean through the screwdriver and into my hand once.
Some
and being only a single line device how exactly would I get 2 lines out of
it?
Eric
On Tue, Jun 17, 2008 at 12:22 PM, Steve Totaro
[EMAIL PROTECTED] wrote:
I like Grandstream 286s No, seriously
Thanks,
Steve T
On Tue, Jun 17, 2008 at 2:07 PM, Eric Fort [EMAIL PROTECTED] wrote:
Buy two..
On Tue, Jun 17, 2008 at 3:46 PM, Eric Fort [EMAIL PROTECTED] wrote:
and being only a single line device how exactly would I get 2 lines out of
it?
Eric
On Tue, Jun 17, 2008 at 12:22 PM, Steve Totaro
[EMAIL PROTECTED] wrote:
I like Grandstream 286s No, seriously
On Tue, Jun 17, 2008 at 3:46 PM, randulo [EMAIL PROTECTED] wrote:
On Tue, Jun 17, 2008 at 1:48 PM, Andrew Kohlsmith (lists)
[EMAIL PROTECTED] wrote:
Most every pocket screwdriver that is sold as a promotional item is like
that.
It's not always good; I cut my hand pretty badly when the
On Tue, Jun 17, 2008 at 12:32 PM, Gordon Henderson
[EMAIL PROTECTED] wrote:
This might depends on your country (re. availability), but I've had a lot
of good results with the Siemens DECT range... (eg. S450IP) The
base-station has a built in ATA, so 2 sockets, one PSTN, one Ethernet...
No
On Tue, Jun 17, 2008 at 9:03 AM, Steve Totaro
[EMAIL PROTECTED] wrote:
It seems you get these goodies at Astricon events.
Thanks,
Steve T
Digium also gives away the best mouse pad ever and I've gotten dozens
of these from every trade show. Theirs is the only one my wife and i
fight over
On Tue, Jun 17, 2008 at 3:51 PM, randulo [EMAIL PROTECTED] wrote:
On Tue, Jun 17, 2008 at 12:32 PM, Gordon Henderson
[EMAIL PROTECTED] wrote:
This might depends on your country (re. availability), but I've had a lot
of good results with the Siemens DECT range... (eg. S450IP) The
base-station
They're real cheap where you could get 2 of them. I got one and
actually have no complaints. Call quality was really good and it's
very, very small and portable. Looks cheap, but you get over that. I
have a SPA2102 2 liner which works fine but gets really hot.
Fred Posner
How do 2 of them register only once?
-Eric
On Tue, Jun 17, 2008 at 12:52 PM, Steve Totaro
[EMAIL PROTECTED] wrote:
Buy two..
On Tue, Jun 17, 2008 at 3:46 PM, Eric Fort [EMAIL PROTECTED] wrote:
and being only a single line device how exactly would I get 2 lines out
of
it?
Eric
On Tue, Jun 17, 2008 at 3:53 PM, randulo [EMAIL PROTECTED] wrote:
On Tue, Jun 17, 2008 at 9:03 AM, Steve Totaro
[EMAIL PROTECTED] wrote:
It seems you get these goodies at Astricon events.
Thanks,
Steve T
Digium also gives away the best mouse pad ever and I've gotten dozens
of these
On Tue, Jun 17, 2008 at 9:56 PM, Steve Totaro
[EMAIL PROTECTED] wrote:
DECT can make you and others around you feel very ill. No long term
research exists but if immediate effects are feeling ill, it may
possibly lead to long term effects.
That may or may not be true, but if we go down that
On Tue, Jun 17, 2008 at 9:51 PM, Steve Totaro
[EMAIL PROTECTED] wrote:
Digium is so bigtime they should be giving away re-branded (acid
etched so it doesn't rub off) leatherman tools to everyone who has
ever ordered from them ;-)
You get those for every order of 10 or more ABE!
I use them for analog Polycom Soundstation EXes and other analog
conference phones. Great quality and that is in boardrooms where
complaints would fly if there was even one little issue.
I generally don't mess with Grandstream but the ATAs aren't bad.
Thanks,
Steve T
On Tue, Jun 17, 2008 at
On Tue, Jun 17, 2008 at 4:06 PM, randulo [EMAIL PROTECTED] wrote:
On Tue, Jun 17, 2008 at 9:56 PM, Steve Totaro
[EMAIL PROTECTED] wrote:
DECT can make you and others around you feel very ill. No long term
research exists but if immediate effects are feeling ill, it may
possibly lead to long
anyone has used or bough one?
would appreciate comments.
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Hello,
If you have a PRI-T1 in the USA, then you can set outgoing CallerID
with just about any carrier.
MATT---
On 6/17/08, Mark Hamilton [EMAIL PROTECTED] wrote:
How can they even set such 1234567890 callerIDs anyway?
For example, our inter/intra state calling depends a lot on the
I can set to anything on my Qwest circuit. All zeros or whatever,
just has to be ten digits. I have seen some that will send less than
ten like a four digit extension number on a misconfigured system.
Thanks,
Steve T
On Tue, Jun 17, 2008 at 4:38 PM, Matt Florell [EMAIL PROTECTED] wrote:
My questions was to the fact that JRA mentioned he knows at least 3
owners. to which I asked if it was LLCs or other type of
corporations, since LLCs have different rules. What I mentioned about
it being illegal is for non LLC type of corporations, but for most of
the other types of
On Tue, Jun 17, 2008 at 10:56:06AM -0500, Kevin P. Fleming wrote:
Benoit Plessis wrote:
Is it possible on a TE220p to deactivate the hardware echo canceler at
will ? (With a function in the dialpan for example)
example for fax SDA ,beeing able to shutdown the echo canceler could
give
Yeah, but what do you get billed as? I understand if your callerID and the
called party is from within a state, it's interstate routing. If between
states, then it's intrastate, etc
The billing depends on the callerID you send.
So, if you send a 000-000- clid to a 917 area code, what would
If it shows up as the BTN on the CDRs then technically you should be
billed at the highest possible tariff. Whether your provider will do
that or not depends what they are charged. In general the provider/s
shouldn't use CID as the BTN and therefore you shouldn't be over or
under charged. Even in
Dear,
I'm having problems with the configuration of this gateway(GrandStream GXW
4108), I used the instructions from GrandStream but it doesn't work. Someone
has a good configuration for this gateway?
Thanks in advance,
Nelson
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Hi,
I'm installing zaptel-source_1.4.10.1~dfsg-1_all.deb (from Debian SID)
into my ubuntu 8.04 box with:
dpkg -i zaptel-source_1.4.10.1~dfsg-1_all.deb
ECHO_CAN_NAME=OSLEC m-a -t a-i zaptel
Loading the wcfxo module and/or zaptel:
[EMAIL PROTECTED]:~# modprobe wcfxo
WARNING: Error inserting
Guillermo Salas M. wrote:
[ 830.118287] zaptel: Unknown symbol oslec_echo_can_identify
Make sure you get the latest version of OSLEC from SVN - the downloadable
tarball has a bug in it which prevents it from compiling properly (although
it acts like it worked just fine); which then prevents
Hi,
See comments in-line
On Tue, Jun 17, 2008 at 04:56:53PM -0500, Guillermo Salas M. wrote:
Hi,
I'm installing zaptel-source_1.4.10.1~dfsg-1_all.deb (from Debian SID)
into my ubuntu 8.04 box with:
dpkg -i zaptel-source_1.4.10.1~dfsg-1_all.deb
ECHO_CAN_NAME=OSLEC m-a -t a-i zaptel
Syed Nasruddin wrote:
Dear Sherwood,
Thanks.
Just three questions:
1. Will I be needing Apache or Asterk-stat will handle itself?
2. Are there How-tos for integerating asterisk-stat with asterisk?
3. My Recordings are being saved in the default folder i.e:
/var/spool/asterisk/monitor/
IP670 was just released...about 30% more than the IP650.
http://polycom.com/usa/en/products/voice/desktop/soundpoint_ip/soundpoint_ip670.html
-Matt
On Tue, Apr 29, 2008 at 1:02 AM, Patrick
[EMAIL PROTECTED] wrote:
On Mon, 2008-04-28 at 14:49 -1000, Matt Darnell wrote:
Anyone seen anything on
El mié, 18-06-2008 a las 01:37 +0300, Tzafrir Cohen escribió:
That's a strange place. Is there
/lib/modules/2.6.24-16-server/misc/oslec/oslec.ko ?
find /lib/modules/2.6.24-16-server/ -name oslec.ko
I suspect there's an older and incompatible copy of oslec.ko around.
You are right:
El mar, 17-06-2008 a las 18:01 -0500, Guillermo Salas M. escribió:
find /lib/modules/2.6.24-16-server/ -name oslec.ko
/lib/modules/2.6.24-16-server/oslec.ko
/lib/modules/2.6.24-16-server/misc/oslec/oslec.ko
I will be deleting all oslec.ko references, modules/zaptel directory
and
start
Try this. It WFM:
localnet=10.0.0.0/255.255.255.0
nat = yes
stunaddr = stun.ekiga.net ; or some other stun server, e.g.: foo.stun.com:3478
externrefresh = 15
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asterisk-users
Right now the issue I see is you are using overlapping extensions
so maybe that's not working as expected?
you have in context sipura line exten 201, exten 201 included from
context spa and also exten 2xx included from context spa.
What you want to do with sending calls elsewhere if they are
On Tue, Jun 17, 2008 at 11:39 AM, Julian Lyndon-Smith [EMAIL PROTECTED] wrote:
I've got the following setup:
PhoneA -
router -
vpn -
router-
asterisk (SIP / ISDN)
PhoneB -
asterisk (SIP / ISDN)
If phone A is connected to phone B (sip-sip), there is a noticable delay
(up to
Bzzzt I'm not an accountant, and don't play one on tv but you are wrong.
This only relates to the classification of the income as passive and has
nothing to do with can a director of a business shield himself.
Go pay someone $250 an hour and they'll tell you how it affects you and
stop wasting
LinkedIn
Vinicius Bossle Fagundes requested to add you as a connection on LinkedIn:
--
Ricardo,
I'd like to add you to my professional network on LinkedIn.
-Vinicius
View invitation from Vinicius Bossle Fagundes
So my SIP Provider states they do not offer the service to list my numbers w/
the Whitepages.
We phoned the Whitepages and they said we can't do it, the SIP Provider must?
Either one/both of them is/are useless or I must switch SIP providers to one
that can get this done.
Anyone familiar with
On Tue, Jun 17, 2008 at 7:57 PM, Dean Collins [EMAIL PROTECTED] wrote:
Bzzzt I'm not an accountant, and don't play one on tv but you are wrong.
This only relates to the classification of the income as passive and has
nothing to do with can a director of a business shield himself.
Go pay
Joseph L. Casale wrote:
So my SIP Provider states they do not offer the service to list my numbers w/
the Whitepages.
We phoned the Whitepages and they said we can't do it, the SIP Provider must?
Either one/both of them is/are useless or I must switch SIP providers to one
that can get
Joseph L. Casale wrote:
So my SIP Provider states they do not offer the service to list my numbers w/
the Whitepages.
We phoned the Whitepages and they said we can't do it, the SIP Provider must?
Either one/both of them is/are useless or I must switch SIP providers to one
that can get
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