Chris,
How many you need in the US and UK? I know someone who is working to
commit to 2 carriers to get coverage for both US and UK DIDs.
I been working on getting DIDs since Aug and it's a rough market with
alot of people selling the same suppliers at a wide range of pricing.
Feel free to
More of a case that in many cases the voip carrier would have to do
lookups for CNAM from either their telco or an external CNAM service.
These tend to carry an extra cost so that's why it's not wide spread
on dids via VOIP.
-- William
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http://www.bayhamsystems.com/ has a app for sending SMS with asterisk.
Don't know how their prices stack up for the UK though.
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They do have the IAXY which could be considered a single port IAX ata
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According to the small print in the bottom graphic:
http://www.sipura.com/products/spa2100.htm
The SPA 2100 would give u 2 ports + 2 RJ45 as well as 2 G729
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Groups for each trunk and check the dial plan groupcount and cycle
thru the trunks
or keep a list of trunks in a DB and just loop thru that first call
route 1 second route 2 etc.
I'll give it some more thought when I wake up but I think you would
have to track concurrent channels per trunk to
Many of these scripts are based on the from which for the most part on
this list is whoever posts a reply. When you reply it goes to the list
address but the from is infact that of the author of the current
message which causes vacation/spam/.. filters to go crazy.
For example I just got a mail
I guess if you know the channel ID you can get info on the channel and
convert the format number to the proper codec.
I'd be interested how others have addressed this too.
-- William
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1800,1866,1877,1888 are all toll free numbers in the us
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Some commerical SMS gateways can provision a # for routing inbound
messages. An example or 2 would be clickatell and ippipi
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http://www.asteriskdocs.org is a work in progress document project for
Asterisk between that and the wiki you should be ok. If that isn't
enough there is plenty of posts in the archives of this list and odds
are someone else has already had the issue you are faced with.
I've used Ipippi.com and clickatell for sms. Clickatell seems to be
quite established in the space. Both have APIs that could be used to
be intergrated into an app for asterisk
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Yes iaxcomm is an IAX softphone. I know Xten is working on improving
their linux support for their SIP based shoftphones.
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I've heard problems with the Grandstream G729 and the new digium G729
by MAC ID. Could be a compatibility issue with the implementations.
Did you ever use the Grandstream against asterisk with the old
Voiceage G729? I've heard that works just fine.
-- William
The carrier of your toll free should send you indication that it is
from a pay phone or not since some do enforce a surcharge to calls
originating from a payphone. Probably be best to contact who providers
the toll free DID to get proper clarification based on how their
system works.
-- William
http://bugs.digium.com/bug_view_page.php?bug_id=0003252
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The stable tree from cvs includes any patches since release that was
also commited for the v1-0 tag since some issues were found after the
release but not major enough for a new tar ball release.
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Astcc is mysql driven w/ perl based web ui
Areski is same concept based on postgres w/ a php frontend also tied
in w/ Areski other scripts for reports and such
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Seems to be a popular move on this list I'm sure some of those that
have taken the plunge already could be of assistance.
LiveVoip/Teliax/Netlogic are 3 that I've heard use L3 currently that
are on this list. Probably more of a -biz question though then the
general user population.
-- William
NuFone service bills in industry standard billing increments, which
are: six (6) seconds for the US48, sixty (60) seconds to Mexico and
fifteen (15) seconds to the remainder of the world.
From: http://www.nufone.net/tac.html
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I'd be interested in the patch as well
On Thu, 2004-01-22 at 13:51, Bill Hamel wrote:
Hi Chris,
This sounds what I am looking for, many thanks !
Also, I do not see an attachment, the patch that is :)
I dont know if the list strips attachments, perhaps send it to my email address
[EMAIL
Looks interesting I will check it out and see what I can do with it =)
On Sat, 2004-01-31 at 08:17, Brancaleoni Matteo wrote:
since I was annoyed this morning, I
wrote this simple php script to output
channel status from asterisk manager.
disclaimer
that's very bad written, nor commented...
more information on how I do it you can reach me at
[EMAIL PROTECTED]
-- William Suffill
On Tue, 2004-02-03 at 20:47, Joshua Colp wrote:
Hi Folks,
I recently setup an asterisk system in order to provide a telephone
phone system for my web hosting business at a very low expense. My
problem
other options would be overkill.
Sincerely,
William Suffill
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I use Saww.net and Nufone for IAX2 to PSTN at a per min basis. So if i
pushed 5 calls i'd be charge per min for each call. Granted both the
companies above cater to * quite heavily.
On Wed, 2004-02-04 at 01:40, Chris Clifton wrote:
The majority of sip to pstn gateway providers (vonage,
u probably should upgrade to 0.7.2 but as far as the caller id that
would be from your sip.conf being set improperly add to your sip.conf
callerid=Caller Name # for each sip entry and that should clear it
up.
On Sat, 2004-02-07 at 00:23, John Fraizer wrote:
I'm running Asterisk 0.5.0 and using
I've been considering deploying an IAX softphone for some remote users
that want to interface with my PBX. It seems as though IAXcomm just
prints that it was rejected if they dial an extension unassigned on the
PBX. Firefly on the other hand crashes if you dial an extension that
doesn't atleast
i search them just fine in Evolution. Filters to a different folder than
my other mailing lists and works quite well. Different pop3 acc from my
isp too =) Why use bandwidth on my colo'd boxes when I can use something
I already paid for =)
On Sat, 2004-02-07 at 10:30, Eric Wieling wrote:
On Sat,
i saw something about that on the voip-info wiki
On Mon, 2004-02-09 at 11:23, Matthew B Marlowe wrote:
The newest firmware from grandstream supports configuration by mac address.
Simply upload a file cfgmac address.txt
Does anyone know the format of a cfg.txt?
use call files there is should a sample in the asterisk src
On Mon, 2004-02-09 at 12:21, John Chambers wrote:
Newbie question coming up ...
Is it possible to use the asterisk to initiate a call to a phone?
What I'm trying to determine is ways for software to connect to a
phone and send
that would require a transfer to the centralserver and
possibly back again. Maybe someone that has worked closely with the vmail
code can comment?
-Original Message-
From: William Suffill [mailto:[EMAIL PROTECTED]
Sent: Monday, February 09, 2004 9:02 AM
To: Darren Martz
Subject: RE
A customer is looking to change to VOIP but he wants a local incoming #
where he lives. Anyone know a provider that offers them via SIP/IAX.
I'll be running Asterisk to run all the features.
Sincerely,
William Suffill
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From Posts on this list on Sat. w/ the subject Voicemail brought to
light that there is a patch for some more advanced VM features after a
message is left.
http://bugs.digium.com/bug_view_page.php?bug_id=156
On Mon, 2004-02-23 at 12:56, Walt Reed wrote:
Looking through the Wiki and mailing
why not load a client on their system they are using? There are quite a
few iax soft phones for both linux/win32
On Wed, 2004-02-25 at 13:58, [EMAIL PROTECTED] wrote:
You may be right here. I was thinking of an ActiveX plug-in. I don't expect them to
use public internet kiosks so they should be
are you on a machine that is slow or running alot of stuff? The ongoing
answer is the thread that is run by asterisk can't complete it's task
fast enough due to lack of system resources so it creates the notice
below.
On Wed, 2004-02-25 at 20:55, Carl Lougher wrote:
When I call Voicemail I get a
There are many options for remote support including Digium directly or
3rd party consultants that are on this list
On Thu, 2004-02-26 at 10:09, John Benson (Solutios Ltd) wrote:
Dear Mark
We have a customer who would like an Asterisk server setting up. Do
you provide this service,
All the digits should already be recorded so you could easily skip that
part and play back any digit from the AGI 1-9 that it was assigned.
On Sun, 2004-02-29 at 00:03, Robert Lawrence wrote:
I would be interested in the AGI Script. As for the voice prompts, I
am having Allison record some
if u add #'s to your contact list w/ @networknameinyourclient
they are connected thru that network such as firefly or others
On Sun, 2004-02-29 at 15:05, asdasd wrote:
You know what would be nice?
If Firefly could have a Network to use assigned to a contact.
I.E. I use 800 to check my
don't thank me it's documented in the app just remembered stumbling on
it in the network tab.
On Sun, 2004-02-29 at 15:46, asdasd wrote:
sweet, cheers
- Original Message -
From: William Suffill [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, February 29, 2004 8:44 PM
Subject
force all the users to a meetme extension ?
On Tue, 2004-03-02 at 11:46, Daniel ANDRE wrote:
Hello,
I am faced to a problem with call transfert with a MGCP Phone. I use
this to make a consultative call transfert:
1. send flash event
2. dial the number and speak with the other person
3.
Take some pics =)
On Tue, 2004-03-02 at 21:29, Matthew Marlowe wrote:
I've converted it... :) I cut, sanded and crazy glued a plastic notch
and made a whole on the handset.. Looks like it came like it. Works
perfect.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
line 1 is always default for calls when a line isn't selected prior to
dialing. Best bet would just be reverse the order you have them on the
Cisco line 1 as primary line 2 as secondary.
On Mon, 2004-06-07 at 12:57, Dragan Mickovic wrote:
I have a SIP phone (Cisco 7960) registered to 2 * pbx, is
2.4 kernel? I have a RH 9 w/ 2.4 using ztdummy just fine a bit older though.
Message seems to show that the phones have trouble reaching each
other. Did Sip to Sip between the phones work fine?
On Tue, 6 Jul 2004 09:43:18 -0700 (PDT), Jack Turer
[EMAIL PROTECTED] wrote:
Any thoughts on the
Just asking for abuse though unless it is restricted or grounds for
termination without a refund,
People prefer to set their CID to a proper call back number such as
myself but it has can be used for less positive uses.
On Wed, 07 Jul 2004 11:45:48 -0400, Jeremy McNamara [EMAIL PROTECTED]
Even to interface analog lines with asterisk you'd need hardware too
which perhaps will put
it out of the reach of your small organization.
$100 for a x100p (a analog port for asterisk)
On Wed, 07 Jul 2004 12:27:38 -0400, Mike Wagner [EMAIL PROTECTED] wrote:
That's all extremely way over my
well then lever it db driven and set the #'s in the db and update that
to the proper call order as needed
On Wed, 07 Jul 2004 13:51:10 -0300, Gelson Dias Santos
[EMAIL PROTECTED] wrote:
The problem is, there is no pattern. It´s not an open/close scenario.
This month I need to call
Normalize for Linux can tell you the levels of a wav and can be used
to adjust it according.
Been toying with using it for some of my streaming media clients since
it sucks to go from too low and having to up the volume to very loud.
On Mon, 12 Jul 2004 10:31:08 -0400, Seth Remington
[EMAIL
Using bison 1.35 here
- Original Message -
From: Fletcher Bonds [EMAIL PROTECTED]
Date: Wed, 14 Jul 2004 09:09:48 -0700
Subject: [Asterisk-Users] RE: [Asterisk-User] asterisk compile problem
To: [EMAIL PROTECTED]
Cc: [EMAIL PROTECTED]
From: Nik Martin
[EMAIL PROTECTED]
You need a cisco smartnet license to legally download the firmwares
for the phone. This would include the sip firemware
On Wed, 14 Jul 2004 20:26:27 +0200, xfastjackx [EMAIL PROTECTED] wrote:
Hi everybody,
I will receive my CISCO 7960G tomorrow. I've ordered it as a global
spare without any
voiptalk.co.uk
On Wed, 14 Jul 2004 16:36:51 -0700, Dameon D. Welch-Abernathy
[EMAIL PROTECTED] wrote:
On Wed, 2004-07-14 at 11:41, Johannes van Hulst wrote:
Can somebody help me with some names of good UK SIP providers?
I am looking for a UK number to connect to my asterisk server.
i use a p2 400 here and it has problems with the scheduling but for 1
or 2 calls that would be ok. Depending on the volume you expect at 1
time adress the hardware according. I'd suggest atleast a 1ghz or so
On Thu, 15 Jul 2004 08:11:43 +0100, Simon Chappell
[EMAIL PROTECTED] wrote:
Hello All,
Seems quite interesting. Any suggestions of where to order one and
about how much?
On 15 Jul 2004 16:54:03 -0700, Wolfgang S. Rupprecht
[EMAIL PROTECTED] wrote:
[EMAIL PROTECTED] (Tom Neville) writes:
; FXO port - Line from our office PBX.
[40]
...
secret=NOPE
Have you gotten
The ipo11's were 25 each when I ordered them + import costs since it
comes from TW.
Yet to use them w/ asterisk but it worked fine w/ their supplied
software in windows since they are Tigerjet based adapters.
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1 port so easier w/ nat + it can trunk(lowering overhead) for multiple
calls to 1 provider.
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quickest would be pattern matching and just make the reoccuring patern
of #'s so you don't have to list em one at a time.
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We are looking at the Polycom IP300 or the Sipura SPA-841 for low end
type client needs at this point. We didn't feel comfortable with the
GS to our type of customers but if it fits your needs that's an option
as well.
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between asterisk boxes and fixed line SMS I believe but never was 100%
sure on this either.
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If each account has an account code it should spawn off a CSV CDR or
you can just do a mass select from SQL by account code.
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Should be an account code field in the DB table that can be used in
queries to just pull 1 accounts records
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7. How Much Does It Cost?
Sign up today for a RISK-FREE 30-day trial of CallWave! Keep it, and
you'll pay a special, introductory rate of only $3.95 per month.
Cancel any time before your trial ends and you pay nothing.
Hmm seems they aren't exactly sure what to expect. TOS didn't seem to
have
Give the FAX SIP device a different account and force it to Ulaw. For
example if the user was account you could create F for fax
and V for voice and have sperate allow/deny codecs
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curl could also be used. Since people asked I'm going to write it up
tonight since I use a GS as well until my Cisco shows up.
On Sat, 2004-04-03 at 09:52, Duane wrote:
Walker Haddock wrote:
I know that you can reboot the GS phones by hitting the rs.htm URL on the phone.
But, you have to
Would it be possible to use an IAX softphone in your situation?
I know iaxcomm is available for both Windows and Linux and can handle
multiple accounts.
On Tue, 2004-04-06 at 10:26, WipeOut wrote:
Martin Mielke wrote:
Hi Markus,
Markus Miertschink wrote:
The one I know of is
wrote:
William Suffill wrote:
Would it be possible to use an IAX softphone in your situation?
I know iaxcomm is available for both Windows and Linux and can handle
multiple accounts.
yes, iaxComm works for both Linux and Windows, but the sound quality is
poor compared to SIP softphones
They modified iax to include the presence packet but only works on their
customized firefly network. I was thinking along the lines of a software
app for those of us who use hardware phones but still want to keep TXT
chat and presence and perhaps integrated into 1 of the iax soft phones
as well to
would lean toward integration to that standard as
well.
On Wed, 2004-04-07 at 21:05, Duane wrote:
William Suffill wrote:
They modified iax to include the presence packet but only works on their
customized firefly network. I was thinking along the lines of a software
app for those of us who
cell phones?
-- William Suffill
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in the context of the incoming DID assuming their Caller ID is equal to
the mailbox for their voicemail aka DID #
exten * = 1,VoicemailMain(${CALLERIDNUM})
You might want to improve this though like so:
Add all assigned DIDs to an Asterisk DB
On * check if callerid is a valid did u assigned
if
Thinking about it further you could set the 6th line to autoanswer and
have the pbx call you and play MOH when none of your lines on the
asterisk box are in use.
On Thu, 2004-05-13 at 10:57, Joseph wrote:
Is there any way to play background music on a sip phone
while the phone is not in use like
Sure you could even use the examples posted here and the wiki to use the
outgoing spool to make calls. Just use a crontab to place a call file in
the outgoing spool every x # of days and problem should be solved.
On Thu, 2004-05-13 at 14:41, Mark Phillips wrote:
Those of you whom have a free
Billy,
Attachment seems to be due to a GNUPG sig file
-- William
On Sun, 2004-05-09 at 12:00, Billy Huddleston wrote:
Mark,
Would you please re-config or use a different mail client as to not send
your replies back as attachments??
It's VERY kludgy, and, I'm just going to stop reading
check the caller id in your incoming extension before you pass to to a
end user. Reset $calleridname to unavaliable if no number is given
On Tue, 2004-05-18 at 15:18, Roger wrote:
I have a question - if a user calls up w/ blocked caller id I get the
following on my phone
Incoming call from
ztdummy will suffice. A Zaptel interface is used as a timing device for
the conference.
On Thu, 2004-05-27 at 11:58, pesb wrote:
Hi there,
I need to implement a SIP Conference Server. I've saw that
asterisk has an application called meetme. But, it says that A ZAPTEL
INTERFACE
I just downloaded it today and the config menus just have for Firefly no
SIP or IAX2
On Thu, 2004-05-27 at 12:14, Tony Mountifield wrote:
In article [EMAIL PROTECTED],
brian [EMAIL PROTECTED] wrote:
Just an FYI FireFly no longer works with anything but the FireFly network.
No more SIP, No
I wouldn't trust it to do any real detection. I use the press # mod in
6 sec mod to be able to fwd to other phone #s without risking hitting
the answering machine or wrong person. I don't believe there is any
real way to detect what you are after as far as if the call is picked
up. You would get
why not use ztdummy which doesn't require USB on 2.6.x? Uncomment it
in the zaptel make file and away you go =)
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their permission might be a good idea too =) Don't want anyone to get
hostile when you show the pics to the community.
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Good idea Matt. Tad far for you unfortunately and too costly for me at
this time but hearing all the latest and greatest news would be
supper.
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To
the dev conf is friday from 9am - 4pm EST as far as i know
Any more info would be cool. I think an outline of the topics are on
astericon's site
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To
If anyone who got the 1.0 tar's would be able to get them to me I'd be
more than willing to donate traffic toward the effort by mirroring it
on some bandwidth.
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Glad it was mirrored. I will contribute a mirror as well when I return
to the office. No reason Nacs should be the only one taking the
burdon.
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To
Probably should just create a page like SF that would round robin the
HTTP links and as 1's are removed and added the users wouldn't need to
find a different url.
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There used to be an NPA NXX sql on 1 of the asterisk site's.
http://www.fnords.org/~eric/asterisk/
I doubt you will find a nice complete 1 for free unless you parse the
npana data yourself which you could do. I did it recently not exactly
fun. Still might not be 100% though.
-- William
Agreed. It's a big accomplishment and wouldn't be possible with
Mark/Digium starting it as well as those of the community that give
whatever time they can besides their normal jobs to help other users.
We all started at the beginning one time or another why not give back
where we can to help those
Interesting. I think either the phonelabs adapter or cellsocket might
be an interesting idea. We are moving to a biz mobile package I use
iax2 term to fwd to a nextel since it's free inbound but having a cell
on the asterisk box is probably a better fit. Besides on a biz plan w/
tmobile and
Cirelle did you delete the .version file in the src tree on your box?
I doubt cvs is 2 wks behind since I got cvs commit emails this
morning. I believe make update will remove the .verision for you too
which will fix that issue.
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Anyone here have any pointers of where to get 1 of the PAP2-NA. Given
all the talk about it I'd be curious as to testing one myself .
-- William
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Depending on your needs I don't know if you will find 1 that used IAX2
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Sorry about that cut off . Like I was saying I'm not sure if you will
find once advanced enough using IAX2 currently. Firefly was the most
evolved when I too was looking but their oem terms weren't exactly
what I wanted to spend given the fact that I probably would be going
hardphones eventually.
In short yes. You put users in a context and only allow certain
features in that context. As far as the limit you probably wish to
write an agi or app to handle the tracking of the mins used per day
and disconnect the user in need be. It could be all done in extensions
with dbput and dbget or
Ntop.org probably could fit you needs from the console.
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Ya good question. Looks like a nice phone with 2 lines for $100. Maybe
one of the places that carries sipura stuff will get them in and start
pushing them. It says they should be available to the public in Nov. I
guess we just wait and see.
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Hooked a 4-line vtech phone up to 2 PAP2-NAs and basically had created
a 4
line ATA for $100.
2 ATA's w/ 2 Ports each I think.
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Scott,
I use an AMD 2400 hosted in The Planet (www.theplanet.com) to host my
asterisk box currently. They don't directly offer AMDs but a provider
that colocates there does. $60/mnth. SeverMatrix.com is the low end
dedicated biz of The Planet directly. It is only 60ms from my home in
NJ even in
Why not just create a context that plays static msgs whenever someone
is transfered thereThank you for calling Monthly special etc
...
then transfer them back when the person at the biz picks up
On Sun, 24 Oct 2004 14:23:04 -0400, Emilio Panighetti [EMAIL PROTECTED] wrote:
Looks like
Wouldn't http://www.areski.net/asterisk-meetme/about.php?s=0 already
provider the webbased/db frontend to manage something like the above
request? I haven't used it myself but I came across it when looking
for other asterisk related scripts.
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Great job Jeff. Lets hope the dbscret can be patched up soon too but
this is a great leap forward.
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Could be a case of routing from you to them and the various links
inbetween. Hard to really pinpoint given the numerous factors that
could cause such issues
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Sounds more like a requirement for custom development since I'm sure
your needs will vary from some others that are also using astcc as a
starting point for their prepaid cards
-- William
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What codec and signalling is being used?
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