Do you know that it is coming back as FALSE, or are you assuming that from
examining the expression?
--Don
From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf
Of Dovid Bender
Sent: Thursday, February 13, 2020 1:13 PM
To: Asterisk Users Mailing List -
-12
On Thu, Feb 13, 2020 at 2:05 PM Don Kelly wrote:
Is HOUR_SELECTED a floating-point number (e.g. 11.999)? If so, you need
to account for that in your comparison.
--Don
From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf
Of Dovid Bender
Is HOUR_SELECTED a floating-point number (e.g. 11.999)? If so, you need
to account for that in your comparison.
--Don
From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf
Of Dovid Bender
Sent: Thursday, February 13, 2020 4:47 AM
To: Asterisk Users
On Tuesday 06 March 2018 at 09:05:25, Markus Weiler wrote:
> Hi Group,
>
> we're just wondering, in German we call the different types of
> phone-numbers
> (Geographic,mobile,national,VoIP...) Rufnummerngassen (phone number
> alleys
> ;-) )
> Is there an english word for this?
No.
It's just
the client to Be billed for
rendered services.
-Ursprungligt meddelande-
Från: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] För Don Kelly
Skickat: den 11 maj 2017 17:04
Till: 'Asterisk Users Mailing List - Non-Commercial Discussion'
<a
As a client, I don't want service company personnel answering my phone.
As a service company, I don't want my clients thinking that I do not trust
my employees who are at the client facility.
--Don
-Original Message-
From: asterisk-users-boun...@lists.digium.com
It's probably not practical to have them answering the client's telephone!
At a lot of sites, incoming calls would be handled by auto attendant,
diverted to answering service, etc.
--Don
-Original Message-
From: asterisk-users-boun...@lists.digium.com
You have an unusual situation--you suspect caller ID spoofing by a known
person.
Under the Truth in Caller ID Act, FCC rules prohibit any person or entity
from transmitting misleading or inaccurate caller ID information with the
intent to defraud, cause harm, or wrongly obtain anything of value.
Is this your Astribank installation? Xorcom may have an answer.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff
LaCoursiere
Sent: Friday, August 5, 2016 8:05 AM
To: asterisk-users@lists.digium.com
...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Don Kelly
Sent: Tuesday, October 07, 2014 9:38 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Asterisk Phone ( Telecom feature )
JG confirmed that it is possible
JG confirmed that it is possible, but it has not been defined.
Without knowing what kind of instruments you are using, a possible it
would be for a party to dial a 4-digit extension number to talk to someone
internally, completing a call without using the PRI trunks.
--Don
-Original
On 09/23/2014 02:17 PM, Steve Edwards wrote:
For some applications, storing recorded audio (prompts and caller
recordings) as a BLOB in MySQL has advantages.
Jeff sez:
How about a named pipe (fifo)? Of course then you might have issues with
simultaneous calls. You would have to have a pool of
On Tue, 23 Sep 2014, Steve Edwards wrote:
On 09/23/2014 02:17 PM, Steve Edwards wrote:
For some applications, storing recorded audio (prompts and caller
recordings) as a BLOB in MySQL has advantages.
On Tue, 23 Sep 2014, Don Kelly wrote:
I'm curious about what the advantages
For my NI2 PRIs I've always used 10 digits for everything and no +1
--Don
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff
LaCoursiere
Sent: Wednesday, August 20, 2014 9:41 AM
To: asterisk-users@lists.digium.com
Subject:
It's possible that Sprint is burping on the name. Try first dropping the
1. Then try dropping the name also, if necessary.
--Don
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff
LaCoursiere
Sent: Wednesday, August 20,
Doubtful that T309 or T316 are causing the problem, but you can always change
them to correspond with their defaults.
http://www.nmscommunications.com/manuals/6272-16/appendxe.htm
--Don
From: asterisk-users-boun...@lists.digium.com
The basic concept is that the original call will run a script that creates a
call file to call the paging system and play a specific audio file. It also
passes into the paging call its channel name. In the call to the paging
system, I use the SHARED function to write back to the original calls'
with the T1
interface cards that two B-channel transfers did something like this.
Digium has documentation on that here:
http://kb.digium.com/articles/Configuration/Two-B-Channel-Transfers
If that doesn't help, and you have a Digium card; you might call Digium tech
support to ask about it.
Don Kelly
Shouldn't the secast discussion be on the commercial list?
Note that their free version works for five simultaneous calls-then the
price goes 'way up.
--Don
(Top posting 'cause that's what's already being done.)
From: asterisk-users-boun...@lists.digium.com
On 13/3/14 6:27 pm, Eric Wieling wrote:
This is an example of why I top post. Who wrote what?
Of course, if you use a mail client that's capable of quoting correctly,
it all works beautifully.
Kevin Larson sez:
Outlook can quote correctly, but it is an all or nothing setting it would
Does he complete the call as a supervised transfer--waits for the called
party to answer before completing the transfer?
--Don
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Diehl
Sent: Monday,
Isn't it easier just to use a SIP door phone?
--Don
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ishfaq Malik
Sent: Friday, December 20, 2013 10:31 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
On Fri, 2013-12-13 at 12:48 +0500, Muhammad Usman wrote:
Hi - I have 2 Asterisk servers connected using 05 IAX2 trunks. I want
to load balance incoming calls over IAX2 trunks. If any trunk goes
down the calls traffic will be shared with other available trunks.
When it gets Up the script is
yes but I believe that least recent would ring one agent at a time? If my
understanding is incorrect please correct it. We are wanting to keep with
multiple phones ring to ensure coverage.
From what I've seen, I don't think this is possible. But maybe ask in the
#asterisk channel on
I'm making changes to an Asterisk IVR designed by someone else.
The application uses both func_odbc.conf and php agi to access an external
MySQL database.
In the php routines, I would like to use the persistent connection that is
established in the dialplan, rather than creating a new
In the php routines, I would like to use the persistent connection
that is established in the dialplan, rather than creating a new
connection each time they run. How can I do this?
You can't, they are completely separate processes and code.
Joshua Colp
Thanks--that's not the answer I
In the php routines, I would like to use the persistent connection
that is established in the dialplan, rather than creating a new
connection each time they run. How can I do this?
You can't, they are completely separate processes and code.
Joshua Colp
Thanks--that's not the answer I
Then you should analyze why it takes 5s. Opening and closing a mysql
connection should take at most a fraction of a second on a local net.
BTW, classical web sites (plain PHP and HTML) do not maintain state, so
keeping the mysql connection open may not be at all possible. I forgot
whether open
On Mon, Nov 25, 2013 at 08:25:44AM +0100, jg wrote:
So:
A calls B
B answers
B puts A on hold
B calls C
B talks to C
B ends conversation with C
B talks to A again, regardless
I this correct? Looks like a simple Hold exercise.
Hello JG,
thanks for your reply..
not exactly, it's rather
In your scenario, all the calls are from endpoints on 181 to endpoints on
183. If the endpoint devices are similar, it seems to me that there should
be no need to transcode-you can use a codec common to the endpoints. 729
would not be required.
--Don
From: asterisk-users-boun...@lists.digium.com
at 5:54 AM, Don Kelly d...@donkelly.biz wrote:
Calls on behalf of political candidates are generally legal--even to people
on the do not call lists. It doesn't seem to be possible to pass
legislation preventing them.
--Don
-Original Message-
From: asterisk-users-boun
Is there an OC-n to SIP solution that makes sense?
--Don
Hi Nick,
Going from DS1 to OC-n is a multi-step process. Typically requiring a
hardware device to handle each MUX step. But you can find hardware that
handles multiple MUX steps together.
VT1.5 is just a raw OC-n channel containing
jg
Sent: Tuesday, June 11, 2013 5:28 AM
Playing an announcement like Your call has been... to A after C has
accepted the call is probably not a good idea, because C has to wait until
the the announcement has finished. In environments where callers are
announced to C, C would typically not want to
Calls on behalf of political candidates are generally legal--even to people
on the do not call lists. It doesn't seem to be possible to pass
legislation preventing them.
--Don
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Guys and gals - these are all excellent answers - I am not being clear, I
think.
Let me see if I can illustrate it.
If you cannot see my diagramme, let me know and I will make a word-type
chart.
So, the Ip device at the top is a SIP phone
Asterisk Server
Gateway /IP
* This
If inbound reliability is important, you may be able to accomplish what you
want with redundant servers, multiple sip providers and toll-free numbers
that can be readily switched between the sip providers.
--Don
From: asterisk-users-boun...@lists.digium.com
The idea is to get your toll-free numbers from a top-tier carrier, pointed
to ring-to numbers on the sip provider of your choice. You can then quickly
(hopefully automatically) switch the toll-free numbers to other ring-to
numbers on your backup sip provider.
--Don
From:
On Tue, 26 Feb 2013, Eric Wieling wrote:
For me, PHP with its C-like syntax...
Steve Edward said:
For me, C with it's C-like syntax...
So that brings up the question I have. Shouldn't a daemon be a compiled
process?
--Don
--
Press # after entering the number to see if it's an extension pattern
matching delay
--Don
-Original Message-
On Behalf Of Steve Edwards
Sent: Monday, February 11, 2013 4:16 PM
Extension pattern matching (waiting for the digit timeout) can also induce
perceived dialing delays.
--
-Original Message-
snip
What I had in mind is to use someone's cellphone as a presence detector.
Let me explain:
- as the first thing you take along when leaving a room or location,
is your own cellphone, why not use chan_mobile and a bluetooth dongle
on your on PC (as you're not
But with max silence at 2 seconds, couldn't someone leave a 30-second
message, pause for a couple seconds to gather their thoughts or dig up a
phone number, and get hung up on?
I'd think that the 3-second and 10-second settings are sensible. Saving a
message that is known to be total silence
I get direct replies when people reply to my posts. I thought that was just
'cause they wanted to make sure I saw their replies!
--Don
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Pete Mundy
Sent:
If you dial 2001# does it complete the call immediately?
Your dial plan may be ambiguous about numbers starting with 2, so it waits
a few seconds to see if you're going to dial a longer number.
--Don
From: asterisk-users-boun...@lists.digium.com
Sounds like a conference with all attendees permanently muted (except the
moderator).
The moderator uses whisper to communicate with individuals.
--Don
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Yves A.
Sent: Wednesday,
Jai,
It should not be necessary for me to remove my email address from your list.
It should not be on there to start with-we do not have, and have never had,
a relationship that justified you sending me email.
--Don
Don Kelly
PCF Corp
People Come First
651 842-1000
651 842-1001 fax
From
Your carrier is apparently using a Japanese switch (at least it's a Japanese
standard).
If you don't get a good answer from the list, you might send an email to
louis.lecl...@denphone.com. Denphone is a company installing Asterisk
systems in Tokyo.
--Don
-Original Message-
From:
Doesn’t the OP wish to page all phones? So it’s not an issue of dumping dozens
of call files all at once.
Does paging work?
http://www.voip-info.org/wiki/view/Asterisk+cmd+Page
http://www.voip-info.org/wiki/view/Asterisk+Paging+and+Intercom
Overhead paging might also be something to
I have the same requirement, but it's important that the caller ID
information from the original caller is presented to the destination and we
announce the call before the transfer is complete. The carrier requires a
diversion header if the ANI is not one of our DIDs. Does someone have
experience
I'm the opposite. I'm likely not to scroll down 10 pages to see the
comments at the end.
Wouldn't need to if people trimmed their posts properly.
Precisely (e.g., see above)! Indeed, my sense is that top-posting
*discourages* properly trimming email and that's my main reason against it.
It is not hard to follow the rules . If the nice folks at Digium took the
time to post rules we should at least TRY to follow them. If you do not like
the rules you can always petition Digium to change them but, taking up
bandwidth on the list in this all to frequent pissing match is a futile
On 1/2/2013 Don Kelly wrote:
... what product/procedure/whatever would
enable me to follow and participate in bottom-posted discussions as it
doesn't appear that Outlook or gmail are very effective.
Umm, what about positioning the cursor below the previous post before
writing your reply
:
On 01/02/2013 12:16 PM, Don Kelly wrote:
I don't think Outlook does what I'd like, so I'm not limiting my
options. I can use different email to keep track of the Asterisk
lists.
Thunderbird (by default) bottom posts. And it does the nice indenting
and allows you to turn off that HTML
Deleting everything would really confuse me
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
Tongue firmly in both cheeks? How do you do that?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steven Howes
Sent: Monday, December 31, 2012 3:26 AM
To: Asterisk Users Mailing List - Non-Commercial
As I did two years ago, I'm posting a new thread with the Top Posting
subject rather than hijacking the Paging for Praying thread.
Two questions:
1. Steve K: What do you mean by /coat?
2. How do we change rule #5?
--Don
Don Kelly
PCF Corp
People Come First
651 842-1000
651
This thread confuses me. I've not worked with the Asterisk MySQL CDR, but
have worked with SQL for years.
Every table should have a primary key. Is no column identified as a PK?
If there is a PK, you will not be able to designate another column as PK.
If there is a PK, you don't need
Is this happening for all callers, or just iPhone callers?
--Don
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vik Killa
Sent: Wednesday, October 10, 2012 11:29 AM
To: Asterisk Users Mailing List -
--Don
Don Kelly
PCF Corp
People Come First
651 842-1000
888 Don Kell(y)
651 842-1001 fax
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Daniel
Seagraves
Sent: Wednesday, June 06, 2012 2:40 PM
To: asterisk
(or
customer complaints)
On Sat, May 26, 2012 at 8:46 PM, Don Kelly d...@donkelly.biz wrote:
I don't think it's possible to suggest a ratio without knowing what your
actual application similar to calling card services is.
--Don
Don Kelly
PCF Corp
People Come First
651 842-1000 tel:651
I don't think it's possible to suggest a ratio without knowing what your
actual application similar to calling card services is.
--Don
Don Kelly
PCF Corp
People Come First
651 842-1000
651 842-1001 fax
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun
What flavor are flashphoner minties?
--Don
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Balashov
Sent: Friday, April 27, 2012 12:37 PM
To: asterisk-users@lists.digium.com
Subject: Re:
Although I do feel that 100+ Euros/month is more than most of us could
manage, I don't think a one-time list is of much value. I would be
interested in establishing a database if there was interest from enough
users for a modest subscription price.
--Don
Don Kelly
PCF Corp
People Come First
651
If we had reports of every call, we could downgrade status of routes that
had frequency calls not completed that were outside the norm.
--Don
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Markus
Sent:
How about a central coop that manages the “normalized” rate sheet and
distributes it with “unknown” call quality metrics for each route. Coop members
report call quality for all calls/routes so the call quality metrics can be
updated in the rate sheet and distributed to members.
--Don
Don
or in lieu of
passing them along to the legacy system.
--Don
Don Kelly
PCF Corp
People Come First
651 842-1000
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C F
Sent: Sunday, January 08, 2012 10:01 AM
inbound-only, or
will you be making outbound calls? Or will you be redirecting calls to
outside agents? What is there about the SIP providers that you find
unsatisfactory?
--Don
Don Kelly
PCF Corp
People Come First
651 842-1000
651 842-1001 fax
-Original Message-
From: asterisk-users-boun
September 22, 2011 11:20 AM Kevin P. Fleming wrote:
For many people, with modern CPUs, current versions of DAHDI and Asterisk,
and appropriate configuration (using the faxbuffers option in
chan_dahdi.conf, for example), such a system can be setup to work very, very
close to 100% of the time.
(Top posting 'cause that's what others did--and I like it that way, anyway.)
Not so obvious that a single mic can't be shared--if one call is muted, it
would work great.
This split-ear feature would be handy when, while on hold/in queue on call
A, you want to answer call B. You want to talk to
that what should be simple TDM FXS to
PRI does not work?
Are you suggesting this is an Asterisk problem or a Digium hardware problem?
Is this really everyone's experience?
--Don
Don Kelly
PCF Corp
People Come First
651 842-1000
888 Don Kell(y
, if
someone answers on a phone, then the PBX won't answer.
If you want to be certain that the Asterisk system won't interfere with an
active call, you can install an exclusion device between the PSTN and the
FXO card.
Google telephone exclusion device.
--Don
Don Kelly
PCF Corp
People Come First
651 842
a parallel device on a line goes off hook. As it
has not been implemented in Asterisk, it can be handled by an inexpensive
device. This will enable you to do as you planned--test your implementation
step-by-step, starting with the answering machine.
--Don
Don Kelly
PCF Corp
People Come First
651 842
phone.
If it is a proprietary phone, you need to try a different approach.
--Don
Don Kelly
PCF Corp
People Come First
651 842-1000
651 842-1001 fax
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com
to a unique
number (received as DNIS by the other server) that would identify the call
as transferred from the first server and, perhaps, the reason for the
transfer.
It looks like TBCT may not have been implemented in Asterisk for EuroISDN.
--Don
Don Kelly
PCF Corp
People Come First
651 842-1000
Continuing top posting...
The same argument could be made for any commercial solution. Why use
Asterisk when we could throw $4,000 at our problem for a commercial
solution?
I'd like to have a solution that would have the features you suggest for
$400.
--Don
On Behalf Of C F
Sent: Monday,
end answers
Question #2
Don't think so since you're asking Asterisk to detect on hold from outside
(this might be do-able in a SIP environment, but DAHDI tends to be copper).
Hope this is correct/helps.
[Don Kelly]
Looks like the call is to the DAHDI Channel from an outside caller, so
on-line and haven't figured out what I should be doing to
fix it.
I'd really appreciate some help.
--Don
Don Kelly
PCF Corp
People Come First
651 842-1000
651 842-1001 fax
--
_
-- Bandwidth and Colocation Provided by http
Zeeshan Zakaria
Sent: Friday, January 21, 2011 6:11 AM
For a client I am setting up a system which will use T1 PRI from Primus,
who offer only NI-1 and NI-2 protocols for D-Channels. Previousely I have
only
used switchtypes euroISDN and National. Although the documentation says
Asterisk does
--
Don Kelly
PCF Corp
People Come First
651 842-1000
888 Don Kell(y)
651 842-1001 fax
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs
On 01/19/2011 12:18 AM, randulo wrote:
Although there's no requisite mention of ${Horrible_Dictator}, can't
we pretend there was, call a Godwin and kill this subject?
11:39 Parker said
That would fall under Quirk's Exception: Intentionally invoking Godwin's
Law to attempt to kill a
There was a typo in the res_fax documentation. Application_SendeFax
should be the correct documentation. I don't know where Application_SendFax
is coming from - it's probably old. When the next import happens,
Application_SendFax should be replaced by the correct version (then I'll try
to
I also agree this is a pointless discussion because, clearly, nobody is
willing to budge, and it has nothing to do with Asterisk.
Amen :)
It may yet have a point - another few hundred (thousand) of these and the
board will blacklist items with the words top post and bottom post :)
And maybe If
I'm top-posting this simply to be consistent with the previous couple posts.
I agree that top-posting is preferable for the reason that Andrew pointed
out and I prefer no trimming (other than signatures--especially legal
disclaimers, etc.) so I can delete every message except the most recent and
snip
Although I put my e-mail in /etc/hylifax/Dispatch I can't receive.
Flavio Roberto Miranda
It may be different for your Hylafax version, etc., but you may want your
email in
/var/spool/hylafax/etc/FaxDispatch
And you probably want to post your questions to the Hylafax list
On Tuesday 18 Jan 2011, Don Kelly wrote:
PLONK is retro--like bottom-posting :)
--Don
boun...@lists.digium.com] On Behalf Of A J Stiles
Retro? For those of us who actually know what PLONK means, it's
hilarious.
Now, here is a link
http://www.youtube.com/watch?v=R1JXYgwwDeY
I've been working with computers for over 40 years and don't have the
foggiest notion how the Green Day--Wake Me Up When September Ends video
applies to Top Posting.
It's a reference to the Everlasting September in 1993. AOL added
usenet access to its service, unleashing a horde of dirty,
Paul Belanger wrote:
It is not a matter of preference, it is actually a rule [1]. Top-posting
is also an annoying practice [2] and NOT the general accepted way to
reply.
[1] http://www.asterisk.org/community/rules
[2] http://linux.sgms-centre.com/misc/netiquette.php#toppost
Thanks for
otherwise. This gives everyone fair warning to delete my posts
before reading them.
--Don
Don Kelly
PCF Corp
People Come First
651 842-1000
888 Don Kell(y)
651 842-1001 fax
--
_
-- Bandwidth and Colocation Provided by http
.
On 01/14/2011 07:42 PM, Don Kelly wrote:
Bruce et al
Im posting a new thread with the Top Posting subject so I wont draw
complaints about hijacking the 4-port thread.
snip
When I post (which is rarely, as I have little to offer the list), I top
post and explain that its my
, 2011 9:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Top Posting
On Jan 14, 2011, at 8:52 PM, Don Kelly wrote:
I have nothing to add to the nascent flame war that I thought we had so
narrowly avoided when I sent my last message. However:
What did
45K GBP would probably cover breakfast in South London. It's about 70 USD.
--Don
Don Kelly
PCF Corp
People Come First
651 842-1000
888 Don Kell(y)
651 842-1001 fax
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C
come up with the right
combination of stuff.
Thanks for any help you can give,
--Don
Don Kelly
PCF Corp
People Come First
651 842-1000
888 Don Kell(y)
651 842-1001 fax
--
_
-- Bandwidth and Colocation Provided by http
.
--Don
Don Kelly
PCF Corp
People Come First
651 842-1000
888 Don Kell(y)
651 842-1001 fax
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Abdul Basit
Sent: Saturday, September 25, 2010 4:43 PM
To: Asterisk Users Mailing List
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David
Backeberg
Sent: Friday, September 24, 2010 11:28 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Record() Cmd
Backeberg
Sent: Friday, September 24, 2010 12:56 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Record() Cmd and My SQL
On Fri, Sep 24, 2010 at 1:32 PM, Don Kelly d...@donkelly.biz wrote:
Don sez: I don't know how to make Outlook indent. I usually top
He's fortunate that the hotel insists he stay there until his situation
improves.
--Don
Rough area. Consider yourself lucky you haven't been ripped apart :P
Pete wrote:
I hope someone has helped poor Rob, I would as I am just over the bridge
in Bristol, UK but some evil internet scammer
He is looking for competitive information...what are prospects paying for
Avaya when they could be saving lots of money with Asterisk systems.
Probably a better question for the biz list, but he doesn't deserve the
responses he's getting.
--Don
Don Kelly
PCF Corp
People Come First
651 842-1000
It could be that I'm entirely confused, but I think he asked what people are
paying for Avaya solutions--so he'd know what competitive pricing would be
for the open source solution he's prepared to offer.
When someone replied with open-source suggestions, he pointed out that that
was not the
I'm looking for a way to use our implementation of HylaFax on Asterisk with
Cardiff (an old installation of Cardiff document stuff).
Is someone doing that?
If no one has direct experience, is there a HylaFax client that emulates
WinFax print-to-fax?
--Don
Don Kelly
PCF Corp
People Come
...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff
LaCoursiere
Sent: Monday, August 23, 2010 3:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk, HylaFax and Cardiff
On Mon, 23 Aug 2010, Don Kelly wrote:
I?m
...@lists.digium.com] On Behalf Of Doug Lytle
Sent: Monday, August 23, 2010 3:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk, HylaFax and Cardiff
Don Kelly wrote:
I looked at http://www.hylafax.org/content/Desktop_Client_Software and
visited several
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