i think nufone and xvoip are based on a per min basis prepaid perhaps
but no monthly fee there is probably others as well
On Mon, 2003-12-22 at 10:34, Hector Q.-datafull wrote:
> Hi,
> anybody knows a VOIP-PSTN company that doesn't charge monthly for making PSTN calls?
> thanks.
>
> ___
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
>
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.
>
>
> that's very bad written, nor commented...
> I
call in. If
you want 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 bu
rnal and personal deployments
where some 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, voicepuls
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 us
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 exi
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,
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 fo
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]> wrot
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 he
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 NU
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 PRO
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 PROTEC
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" witho
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 go
i used chan_h323 properly by Jeremey without issue and I did warn you
privately about the H323 support in Asterisk not being without issue
or configuration problems.
I'd suggest chan_h323 and follow Jeremy's docs to the letter and it should work
On Fri, 23 Jul 2004 20:21:48 -0400, Jeremy McNamar
Any more information than that? I have a copy here as well but haven't
had time to read through it.
P.S. Yes I know my name is mentioned in the book. No need to flame me
on that fact. I am a regular consumer like anyone. Author felt
inclined to put it in there.
- Original Message -
From: J
asterisk needs license to work w/ G729
10 USDs per channel. Once the box has licenses it can convert the gsm
to g729 on the fly for you for the g729 phones. Besides you wouldn't
want to record voicemails in g729 either since you want to be able to
play them back from any where.
- Original M
I should have dids in most major cities in time for the conference as
well so if the community contributes to the effort it could work.
On Thu, 29 Jul 2004 12:14:59 -0700, Mike Machado <[EMAIL PROTECTED]> wrote:
>
> > Another thing you could do is use a regular phone to call into a DID and
> > e
You will be able to purchase Cepstral voices from Digium just like you
dor for G729 already. I would guess it's 1 way to show the power of
asterisk by putting all the TTS orders thru a company such as Digium.
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Unfortunately no. Someone say the press release and bugged me about it
as well but I haven't seen anything that would indicate they plan on
doing anything more than parting with carriers with large rollouts of
these phones. That MSRP seems too good to be reality too.
-- William
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worked fine for me.
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rm -rf /usr/lib/asterisk/modules/
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You can also remove /etc/asterisk to erase the configs that were
installed but the major issue between STABLE/HEAD is the modules. The
version mismatch in the modules is what caused all the errors you got
such as Aug 14 15:04:33 WARNING[4860]:
/usr/lib/asterisk/modules/app_realtime.so: undefined sy
I'd suggest Dial(trunk/1800555,30,D(1wwww2)
That will cause it to dial that DMTF string on connect and w causes a pause. I haven't tested it just referenced
http://www.voip-info.org/wiki-Asterisk+cmd+Dial
and
http://lists.digium.com/pipermail/asterisk-users/2004-November/071853.html
Ziaxphone might fit your needs. http://www.kauss.org/Stephan/ziaxphone/
Haven't used it recently since someone broke the screen on my Zaurus =(
-- William
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it is trying to match the did in your context which it can't do
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The idea is to have people attend the conference and not primarily a
webcast. Granted due to the nature of the community it's not possible
for everyone that would like to attend to justify it for cost reasons
and distance. I think the Dev meeting is best fitting to be
broadcasted in this manner any
yes change your dial macro to use SetCallerID and SetCIDName
and it will use that instead
On Tue, 03 Aug 2004 19:50:26 -0700, Trevor Peirce <[EMAIL PROTECTED]> wrote:
> Is there any way to have asterisk set CID to Private or Unknown instead
> of "asterisk" when a call comes in that is either bloc
Give each user a voice box then use 1 of the vm broadcast patches in
the bug tracker so that 1 to all in a perticular goup can be done.
On Thu, 05 Aug 2004 15:57:36 +0200, Altus Snyman <[EMAIL PROTECTED]> wrote:
> Good day all
>
> I got my voicemail message working,thanks but now,keep in mind I'm
i've only used chan_h323 which suggests u download the tars and
extract them in /root. Takes a while to build but I did manage to get
it working
On Thu, 5 Aug 2004 13:47:29 +0200, Asmine Ouloube
<[EMAIL PROTECTED]> wrote:
> This is what I've done:
> Take asterisk, libpri and zaptel with cvs
> Afte
add noload = app_txtcidname.so to your modules.conf would be a temp
fix. I would cvsup and rebuild it if you need txtcidname
On Fri, 6 Aug 2004 21:03:57 -0500, Lyle Giese <[EMAIL PROTECTED]> wrote:
> I am putting together my first *. I had it running with two other pc's
> running xlite and setu
use 1 of the broadcast mail patches on bugs.digium.com so when a msg
for a shared box comes in it is copied to all the priv boxes
associated w/ that group so the mwi on all those phones goes on as
well
On Sat, 07 Aug 2004 13:32:30 -0400, Don Hughes
<[EMAIL PROTECTED]> wrote:
> The message waiting
In sort no.
Depending how many concurrent calls you do on that system at once you
will hit cpu issues. Also if you do any transcoding between codecs
would will have a performance hit.
And why 25 sip accounts at your provider? Why not 1 or 2 that can
handle concurrent calls.
- Original Messa
It was attached to the e-mails to the list atleast here. Perhaps it
should be put in the wiki for others to reference instead of posting
all revisions to the lists where many won't even notice the script at
all.
On Sun, 8 Aug 2004 14:51:39 -0700, hank <[EMAIL PROTECTED]> wrote:
> where can we get
the mirrors of rc1 are also listed in the wiki as well
On Tue, 10 Aug 2004 10:43:36 -0400, Seth Remington
<[EMAIL PROTECTED]> wrote:
> On Tue, 2004-08-10 at 09:19, [EMAIL PROTECTED] wrote:
> > > On Tue, 2004-08-10 at 07:35, [EMAIL PROTECTED] wrote:
> > >> Hello!
> > >> Is there a asterisk mirror?
CVS has them
- Original Message -
From: Wiley E. Siler <[EMAIL PROTECTED]>
Date: Sat, 14 Aug 2004 16:50:43 -0700
Subject: [Asterisk-Users] Free MOH MP3
To: [EMAIL PROTECTED]
Hello All,
Sorry to rehash a question I am sure has shown several time but I
cannot google up the answe
That could right don't really use MOH much but I noticed there was in
CVS. Although why would it be in CVS of asterisk if not used for MOH
though?
On Sun, 15 Aug 2004 18:57:39 +0100, Kevin Walsh <[EMAIL PROTECTED]> wrote:
> William Suffill [EMAIL PROTECTED] top-posted:
> > CVS
you should be able to transfer using the manager interface from 1
user's phone to another
- Original Message -
From: Ben Merrills <[EMAIL PROTECTED]>
Date: Mon, 16 Aug 2004 11:29:44 +0100
Subject: [Asterisk-Users] Call stealing
To: [EMAIL PROTECTED]
Hi,
How can I (through the
Depending on the application the Grandstream is decent but for
prolonged use I've found it's better to not pinch the pennies and go
with something a bit more expensive but with less problems. For a
simple SOHO deployment I'm just putting a Sipura SPA 3000 on their
cordless base station (Panasonic).
If you are going to do hylafax why not just do it seperate from
asterisk on a regular modem and just email o ut the results. Don't see
the big bonus to using a FXS and the adding cost and point of
failures.
On Mon, 23 Aug 2004 11:13:25 +1200, Carlos Hernandez
<[EMAIL PROTECTED]> wrote:
> About f
I had this issue with a grandstream as well a week or so ago and have
yet to solve the issue. Until I get my Budgetone here physically again
I won't be able to mess with it hands on. What did you use for
codec/signaling and did your asterisk box see any warnings or errors?
On Tue, 24 Aug 2004 15:5
Post some pictures when you are all done. Looks like an interesting
task just no need to dive into it myself at this time.
On Tue, 24 Aug 2004 18:08:28 -0500, Jay Milk <[EMAIL PROTECTED]> wrote:
> Yep, great idea, that's what's next -- and I have two extra extensions
> (Sipura)
>
>
>
> > -O
Andrew,
Sounds like it could be a good fit for your needs. Although that
raises many questions as to how exactly you should deploy it. If you
have a good Internet connection to the office in question you could
perhaps use VOIP termination for your outbound calls instead of the
current 4 PSTN lines
just store the cids of your high paying accs and give them vip
treatment or a different did to call in =)
On Thu, 26 Aug 2004 12:39:49 +1200, [EMAIL PROTECTED]
<[EMAIL PROTECTED]> wrote:
> On 25 Aug 2004 at 21:34, Nicolas Gudino wrote:
>
> > On Wed, 2004-08-25 at 20:38, Chris Shaw wrote:
> > > Co
astdb dbget for the cid would probably be cleaner and if doesn't
return a result then unknown cid but good idea
On Thu, 26 Aug 2004 22:22:33 -0400, Mark Woods <[EMAIL PROTECTED]> wrote:
> David,
>
> Yes I have, and also with call through direct for friends.
>
> What I've done is implemented a ca
1) should be more than enuf for 1 channel. I use a P2 400 here for
testing and it worked ok for transcoding besides the schedule notices.
2) Depending how much timing you need to do X100P or ztdummy could
even work just fine.
3. -head
4. i'd rebuild it from src and just copy your configs and an
The wiki allows everyone to post pages on whatever they wish. This
means a company can
post settings in reference to their company or anyone else could for
that matter.
On Wed, 1 Sep 2004 21:56:30 -0400, Michael Workman
<[EMAIL PROTECTED]> wrote:
>
> On your web you have a link
>
> http://www.vo
star38.com .25 connection .07-.13 per min What a bargin
On Thu, 02 Sep 2004 16:16:26 +0200, Stefan de Konink <[EMAIL PROTECTED]> wrote:
> Brian Capouch wrote:
> > FYI. Reading is free; if you don't have an account it is trivial to
> > sign up, and they're very politically correct, as might be ima
I know someone who was looking into it but they decided not to make
the investment at this time versus other options they had available.
Prices did look decent though.
On Thu, 2 Sep 2004 11:10:37 +0100, David Gurr
<[EMAIL PROTECTED]> wrote:
> Has anyone out there used the PipeMedia/PipeCall PSTN
In theory yes. Depending on if you wish to use your own PRIs or remote
termination for asterisk as well as the phones you choose it could be
done quite economicly versus the other options. Although due to the
nature of asterisk you have alot of decisions as to how to go about
it. Since you haven't
Digitnetworks is profiting off the cards so they should support them.
If it wasn't for Digium there wouldn't be Asterisk anyway. So doesn't
that make it better to support the primary company for software that
many of you use every day at home and work?
On Fri, 3 Sep 2004 08:40:59 +0100, Kannaiyan
Best bet for such a CoOp would be a give and take relationship. If
they also give you access to something of theirs they are more likely
to treat your stuff with care as well.
But it is risky.
On Sat, 4 Sep 2004 22:11:37 +0100, Kevin Walsh <[EMAIL PROTECTED]> wrote:
> Marconi Rivello [EMAIL PR
check voip-info.org for call recording. There is a dial plan example
using Monitor for that
- Original Message -
From: William C. Lohr Jr. <[EMAIL PROTECTED]>
Date: Sun, 5 Sep 2004 00:35:57 -0400
Subject: [Asterisk-Users] Call recording
To: [EMAIL PROTECTED]
Newbie here. Learning a lo
why not use a tcp socket and use the manager api and avoid the
permission issues all together
enable it in manger.conf and you connect over tcp log in and execute
the command nice and cleanly in your application. There should be
decent examples on voip-info.org
On Sun, 5 Sep 2004 23:52:13 +0200, R
Roger,
I haven't had any problems doing confs w/ g729. My guess is the Sipura
is asking for ulaw first. Try adjusting the codec priority on the
sipura side. IF you still have problems I an get my spa-3000 out and
trying and solve it for you.
-- William
- Original Message -
From: box100
It is but you need to modify your dial plan to make it work.
I do it like such
[inbound] ; context that takes inbound calls and matches em and routes according
exten => 91808,1,Macro(stdexten,101,SIP/101) ; fwd
exten => 55,1,Goto(all-exten,101,1)
; fwd goes start to my stdexten to 101 whi
Good call Daniel I didn't even notice that.
As far as number of license it really depends on how many concurrent
calls you will be doing and if asterisk needs to transcode at all. If
you call from g729 device to g729 you are fine but g729 to vm would be
1 license etc.
On Mon, 06 Sep 2004 04:51:2
On Mon, 06 Sep 2004 13:22:51 +0300, Vladyslav <[EMAIL PROTECTED]> wrote:
> Today morning cvs server checkout problem:
>
> cvs server: Updating asterisk-addons/format_mp3
> cvs server: failed to create lock directory for
> `/usr/cvsroot/asterisk-addons/format_mp3'
> (/usr/cvsroot/asterisk-addons/fo
Sounds like it be best as a custom app or AGI depending how many calls
you will be taking and how bad the performance hit of using an AGI vs
Compiled app is for your needs
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I don't really see how that's possible with the current Queue setup. I
don't see why you couldn't use AGI or an app to query a callback table
and orginate the call back and connect it to an available agent.
I'm curious on this as well so feel free to contact me offlist. I'm
going to add it to 1 of
ztdummy should be able to work natively on the 2.6 kernel w/o need of
usb. I use it on a fc2 box single processor w/ a 2.6 kerenl w/o issue
and a rh 9 w/ 2.4 w/ usb
- Original Message -
From: Chad Brown <[EMAIL PROTECTED]>
Date: Wed, 15 Sep 2004 18:59:38 -0700
Subject: RE: [Asterisk-User
Could you read the post and reply off the list like it was requested?
I agree that the -biz list is a better place for it as well though.
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someone pack a wrt54gs and create your own wifi =)
On Fri, 17 Sep 2004 16:12:00 -0500, Kristian Kielhofner <[EMAIL PROTECTED]> wrote:
>
>
> Michael Welter wrote:
>
> > Does anyone know if the Marriott has Wi-Fi? LAN connection in the room?
> >
> > Mike
> >
> >
Dial has the D flag for doing just that Not sure how you would do it
for the call spool though
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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 st
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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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 bala
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 bo
I prefer to use a numerical exten. but same result.
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Seems what we all want but since it's new there are always problems
especially since we as a whole complain when they charge too much.
There will be a happy medium eventually but for now it's probably best
not to having too important dependent on voip origination since
unlike termination you can't
FWD is availabe via iax as well as sip. Easiest solution would be to
sign up for a FWD acc and enable iax. You could even use sip plenty of
examples between the list and voip-info.org that should get you going.
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Why not DBGet from the SIP.Registry in ASTDB? Wouldn't that be cleaner
since it would only return the 1 you want instead of parsing what
could be a load of sip peers?
-- William
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Yes same provess you did to register the license in the first place.
You can rereg the license I think 3 times or so before you have to
call Digum and have them manually change what your license is tied to.
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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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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 a
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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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
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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
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http://bugs.digium.com/bug_view_page.php?bug_id=0003252
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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 cfg.txt
>
> Does anyone know the format of a cfg.txt? â
>
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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
> Again where centraloffice is identified in the IAX config file. To me this
> would allow a separate box to handle all the voicemail calls.
>
> Internally I suppose that would require a transfer to the centralserver and
> possibly back again. Maybe someone that has worked closely with th
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 b
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