it is trying to match the did in your context which it can't do
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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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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
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
http://www.ohse.de/uwe/software/utftpd.html
worked fine for me.
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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
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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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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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
I am
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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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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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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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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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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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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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
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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I prefer to use a numerical exten. but same result.
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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
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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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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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
___
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
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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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
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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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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http://bugs.digium.com/bug_view_page.php?bug_id=0003252
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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
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
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 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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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.
Some commerical SMS gateways can provision a # for routing inbound
messages. An example or 2 would be clickatell and ippipi
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1800,1866,1877,1888 are all toll free numbers in the us
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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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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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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
1 port so easier w/ nat + it can trunk(lowering overhead) for multiple
calls to 1 provider.
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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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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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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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[EMAIL
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
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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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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Great job Jeff. Lets hope the dbscret can be patched up soon too but
this is a great leap forward.
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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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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
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
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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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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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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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.
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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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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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
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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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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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
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
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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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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:
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
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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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
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
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'
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,
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
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
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 lot
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
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 imagined,
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
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
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
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:
Cool! I
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 caller
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
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)
-Original
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