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 stu
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 PROTEC
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
>
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:[EMA
I just posted my rough code that I use for this to the bugtracker.
http://bugs.digium.com/bug_view_page.php?bug_id=0001160
On Thu, 2004-03-04 at 12:12, WipeOut wrote:
> Jason Miller wrote:
>
> >I have a question about the capabilities on a user who wants to roam around and
> >keep the same exten
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 t
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
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, bu
If you change the extension to the follow
exten => *55,1,VoiceMailMain(${CALLERIDNUM})
The voicemail will now user their caller id for the mailbox
>
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They modified iax to include the presence packet but only works on their
customized firefly network. I was thinking along the lines of a software
app for those of us who use hardware phones but still want to keep TXT
chat and presence and perhaps integrated into 1 of the iax soft phones
as well to
so I would lean toward integration to that standard as
well.
On Wed, 2004-04-07 at 21:05, Duane wrote:
> William Suffill wrote:
> > They modified iax to include the presence packet but only works on their
> > customized firefly network. I was thinking along the lines of a software
>
u suggest for 2 business
cell phones?
-- William Suffill
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in the context of the incoming DID assuming their Caller ID is equal to
the mailbox for their voicemail aka DID #
exten * => 1,VoicemailMain(${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 y
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
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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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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 cont
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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or create a file in another dir. Change the time on the file then put
it in the call spool. It should be covered on the WIKI as well. Or you
could write your own app to use the manager api to originate the calls
depending on the needs you have.
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FWD sends all the 411 calls to TellMe.com which also provides
professional VoiceXML services and development resources
(studio.tellme.com)
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I'm #11 but I have notice of late a few problems but nothing major
given the price differences assuming you don't have the volume to
commit to another carrier directly for the destinations you are
after.
-- William
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Asteri
I concur with Ed. The web orders get put in a massive queue and are
harder to follow up on. When you use a rep then they are there to help
you with your sales concerns so you use them for your other needs they
can fill.
-- William
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What codec are you using to asterisk and what codec to VPC? Also does
this occur if you test the service with another ITSP
(nufone/voipjet/teliax)
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Point to Point connectivity if they are close enough. Only use
DSL/Cable if you have to since results may vary depnding on
location/route/utilization/ISP.
On 5/19/05, Andrew Latham <[EMAIL PROTECTED]> wrote:
> yes
>
> On 5/19/05, David Sampson <[EMAIL PROTECTED]> wrote:
> >
> >
> >
> > Hello –
>
Which of their services are you refering to? Conference Bridge worked
fine in my tests but I haven't used them for anything else to date.
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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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[EMA
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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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 U
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 UN
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 others
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 sqli
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 TX
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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there should be 1 addons for mysql and anthm wrote res_sqlite which
would add the same functionality but use sqlite to backend it
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Bonus:
Sipura SPA-3000s purchased from Voxilla include the following:
* One free month, with all activation fees waived, of any
Broadvoice's unlimited plan, including "Unlimited World Plus";
* Up to 100 free calling minutes through iConnectHere;
* One free month, with activation fee wa
With a 7960 you could easily just dial a music on hold extension using
1 of the 6 lines then you could put that on speaker when you aren't
using your phone.
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 m
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 lik
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 Wash
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 t
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 fro
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
> INTERFA
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 m
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
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