Am Freitag, den 10.08.2007, 11:26 -0500 schrieb Peder @ NetworkOblivion:
> That's great, now say you have 5 or 6 AA's and each one has 10 different
> parts that you want to record ("thank you for calling..." "for Steve
> press 1" "for dave press 2"). Rather than having to record a long
> messa
Am Freitag, den 10.08.2007, 09:02 +0200 schrieb Olivier:
> Hi,
>
> My question is more "what should be done" than "how should it be
> done".
> I could say :
> "If you were a teacher, teaching and preparing your courses once a
> week (as you can't be called while teaching, can you ?)
Well, yes. It
Am Donnerstag, den 09.08.2007, 20:12 -0500 schrieb David Bandel:
> Folks,
>
> I'm trying to implement a simple loop in a dialplan. The object is to
> set a counter, run through some IVR options, increment the counter,
> return to the start, then finally fall through to an operator or
> voicemail.
Am Mittwoch, den 08.08.2007, 23:55 +0900 schrieb Balgansuren Batsukh:
> Hello,
>
> I installed Asterisk on Dell Precision workstation and configured with
> sample configuration.
>
> I have two TDM400 board and one with 4xFXO and second one 4xFXS module
> installed.
>
> I made call to telephone
Have you checked out UserEvent:
http://www.voip-info.org/wiki/view/Asterisk+cmd+UserEvent
Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic and Business Research
University of Florida
(352) 392-0171 Ext. 221
> -Original Message-
> From: [EMAIL PROTECTED]
>
Am Dienstag, den 07.08.2007, 07:47 +0200 schrieb Olivier:
> So no proper logoff between logins, right ?
>
> As I will apply free sitting in school environment, chances are phones
> would then remain logged-in several hours or days between another user
> logs in.
>
> My thoughts are focused on fin
Am Dienstag, den 07.08.2007, 16:51 +0200 schrieb Olivier:
> Hi,
>
> Where can I find relevant information concerning callto:// tags ?
>
> Is it standardized or browser specific ?
> How within your browser, can you specify the software and parameters
> to used when clicking on such callto:// tags
Am Montag, den 06.08.2007, 18:09 +0200 schrieb gincantalupo:
> Hi,
> I'm trying to use a Detewe TA 33-clip but there is no rj11 connector on
> it...only a TAE connector.
> I'd like to create an adapter so I need to know which TAE pins to
> connect to RJ 11 pins.
> Is there anybody who knows where
Am Mittwoch, den 01.08.2007, 16:32 +0530 schrieb Benjamin Jacob:
> Hello good ppl,
> A couple of questions for multiple pbxes
> 1. Is it possible to support multiple pbxes in one Asterisk box(using
> contexts, etc.)?
> 2. Can we use the "domain" field in sip.conf to specify the different
> domain
Am Dienstag, den 31.07.2007, 07:39 -0500 schrieb Asterisk guy:
> 1and1 dedicated server's service has been down for a few hours ,
> unable to reach them by phone or email. do anyone know what is going
> on there ?
There were rumours they had trouble with an outdated version of the
web administr
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Martin Vít
LAM plus s.r.o.
http://www.lam.cz/
Tel.: 605 267 610
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options
Am Montag, den 30.07.2007, 14:29 -0700 schrieb Lee Howard:
> http://www.asterisk.org/node/48327
>
> I mean, really... you're kidding me, right?
It is not at all April 1st... however, I see the point in having a
simple demo app. Wether you call it helloworld or hellomarc, the
difference is not too
Am Montag, den 30.07.2007, 05:24 -0700 schrieb Vieri:
> Hi,
>
> I would like to know if one can set the outgoing
> caller ID within Asterisk when calls are going out
> through:
>
> 1) an analog POTS line (I suppose not)
> 2) a telco BRI line (I don't think so)
> 3) a telco PRI line (maybe)
> 4) a
Am Mittwoch, den 25.07.2007, 12:13 -0700 schrieb bilal ghayyad:
> Hi BaharatSamaria;
>
> Thanks for your kindly email.
>
> Are (Xlite and phoner) IAX or SIP? From where I can
> download them (Xlite and phoner)?
I googled for "xlite". One of the first matches was a wiki page on
voip-info.org, whi
Am Dienstag, den 24.07.2007, 11:26 -0500 schrieb John Faubion:
> > To prevent further missunderstanding please do not refer the SI-120
> as a snom
> > phone. If you need support please contact snom India.
>
> Tim,
>
> If it is sold by snom India, and one is to contact snom India, I can
> certa
Am Montag, den 23.07.2007, 14:33 -0400 schrieb Matt:
> Hi,
> What dialplan option do I need to send a call out like this:
>
> NPA-NXX- local calls
> 1-NPA-NXX- - long distance
>
> Won't 'national' send it out NPA-NXX- no matter if it's long
> distance or not?
I do not understand your
Am Montag, den 23.07.2007, 16:21 -0400 schrieb Michael J. Liberatore:
> I noticed in 1.4.x I can no longer use n+101 ? I use this all over my
> dial plan and wouldn't even know how to replace it. Like when trying to
> call out and a channel is busy, would I need to do all if then's??? How
> can
Am Montag, den 23.07.2007, 06:44 -0700 schrieb satish patel:
> Dear all
>
>I have configure asterisk with 100 SIP PHONE ( SNOM )
> but now thing is that my boss need phonebook feature find extention
> number by Pbook so i have read about it there is a feature in asterisk
> but it i
I'd bet the emails are addressed to the list and the original sender,
both, so for the original person they appear twice, but everyone on the
list gets them a single time. I haven't seen any duplicates.
Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic and Busines
I didn't paste the actual etxensions.conf entry -- there are quotes in
the file itself.
Any other ideas?
Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic and Business Research
University of Florida
(352) 392-0171 Ext. 221
> -Original Message-
> F
- Executing Playback("Zap/97-1", "vm-goodbye") in new stack
-- Playing 'vm-goodbye' (language 'en')
== Spawn extension (default, 3378, 4) exited non-zero on 'Zap/97-1'
-- Executing Hangup("Zap/97-1", "") in new stack
and any link-level
event potentially re-hangs them.
Keep us posted if you find out anything!
Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic and Business Research
University of Florida
(352) 392-0171 Ext. 221
> -Original Message-
> From: [EMAIL PROTECTED]
> [mai
Am Donnerstag, den 12.07.2007, 16:57 -0700 schrieb Russ McBride:
>
> Newbie question(s):
>
> From what I can determine it sounds like the SMS messaging isn't as
> robust as it could be (?). I'm wondering if there's active work on
> that right now or if it's more of an issue about PSTN carri
Am Montag, den 16.07.2007, 09:44 -0500 schrieb Jeremy Mann:
> Does anyone know if X-Ten or SJPhone support multiple cordless
> handsets for multiple lines? I have an office with multiple roaming
> users(nurses) that are in and out. I’d like to provide them
> telephones, and my idea is to have a P
Am Montag, den 09.07.2007, 17:21 +0200 schrieb Matthias Huber:
> When i send more than one messages shortly after the other, my log
> (/var/spool/asterisk/sms ) looks like this
> and only two of four messages arrive.
>
> What am i doing wrong ?
>
> I am using an AVM B1 PCI with chan-capi and 1.4
ot (for example, if you have two
NICs in a Linux box and you want to make sure one card is always eth0 and the
other is always eth1 using udev), and how to make asterisk use the second
board's channels.
Thank you!
--
Jason Martin
Metrix Matrix, Inc.
785 Elmgrove Road, Building 1, Roches
Am Freitag, den 29.06.2007, 14:23 -0600 schrieb Anthony Francis:
> Andres Paglayan wrote:
> > On Jun 29, 2007, at 12:50 PM, Lenz wrote:
> >> Hello list,
> >> I am getting the list with days of delay, take for example this
> >> message:
> >> As you can see, the message was posted on June 25th and
Am Mittwoch, den 04.07.2007, 11:00 -0400 schrieb Noah Miller:
> > Is it just me? After the mail list server upgrade, the average delivery
> > time for messages to the users list is between 4 and 5 days. The Dev
> > list seems fine!
>
> I'm getting new messages within a matter of minutes. I dunn
t stand it.
If you can tell me in thre lines how to use addqueuemember doing all things
we need from callbacklogin app, then I will use it from today on.
Othervise it is a reinventing of the wheel.
Hope there will be a alternate application in newer versions of asterisk.
Thanks
Martin
thre lines how to use addqueuemember doing all things
we need from callbacklogin app, then I will use it from today on.
Othervise it is a reinventing of the wheel.
Hope there will be a alternate application in newer versions of asterisk.
Thanks
Martin
- Original Message -
From: "K
Am Donnerstag, den 28.06.2007, 07:07 +0200 schrieb Adam KOSA:
> Hi guys,
>
> sorry for the long e-mail, i'm only trying to give as much information
> as i think is relevant to my problem (console log, sip.conf and
> extension.conf parts). I've sent this e-mail a couple of days ago, but
> it boun
You're including a context in your dialplan that doesn't exist. Given
that it has been prefixed with AEL, I'd check extensions.ael for the
Asterisk Extension Language sample file. I bet it does some including.
Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Econom
else
interesting about this or non-ringing auto-dialing, I'd be curious to
hear as well!
Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic and Business Research
University of Florida
(352) 392-0171 Ext. 221
___
--Bandwidth and Colocatio
to do this? The complaint we are getting now is the call
rep doesn't want their phone to ring when making a call. Can the manager
interface give a phone number to dial on an off hook Zap line?
Thanks!
--
Jason Martin
Metrix Matrix, Inc.
785 Elmgrove Road, Building 1, Rochester, NY 14624
O
Am Samstag, den 23.06.2007, 09:52 -0300 schrieb Ronaldo Z. Afonso:
> Hi all,
>
> Does anybody know any USB phone that I can use as an IAX Client?
The USB "phones" I saw on the market just behave like an additional
sound card, with some control buttons perhaps, and those will not work
without a so
that file.
> Thanks
> Ronaldo.
Take care,
Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic and Business Research
University of Florida
(352) 392-0171 Ext. 221
___
--Bandwidth and Colocation provided by Easynews.com --
aste
ed
this a different way.
Has anyone been able to do this, via caller ID, messaging, the
mini-browser in those phones, or some other way?
Thanks!
Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic and Business Research
University of Florida
(352) 392-0171
Am Dienstag, den 12.06.2007, 09:57 -0400 schrieb Shad Mortazavi:
> Dear Group,
>
> I have a scenario where I would like to change the caller ID based on
> the number dialled;
>
> For example;
>
> ;Outbound UK and London Calls
> exten=>_8.,1,Set(CALLERIDNAME=0207100)
> exten=>_8.,2,Dial(SIP/$
Am Donnerstag, den 07.06.2007, 01:15 +0200 schrieb Patrick Zwahlen:
> Hi everyone,
>
> How do you send multiline SMSs using smsq or .call files ?
>
> smsq --motx-channel="mISDN/g:bri/" 078 "line1 line2"
>
> How can I have a carriage return between line1 and line2 ? I have tried
> the reg
case it helps :)
Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic and Business Research
University of Florida
(352) 392-0171 Ext. 221
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dean
Collins
Sent: Tu
Hi Mike,
I believe Polycom has directed resellers to supply firmware updates
directly to buyers. I'd recommend you speak with whomever you purchased
the phone from.
Best,
Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic and Business Research
University of Florida
nferencing a call
I plan on teaching users how to operate a softphone (haven't decided
which yet) and a hard phone (we have Polycom 430s and 501s) as well.
I'd welcome any advice or materials! Thanks!
Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic and Bus
Am Samstag, den 02.06.2007, 11:34 +0200 schrieb [EMAIL PROTECTED]:
> Hi,
>
> Problem is:
> I have a Dell 1950 server with 6 NIC's ( 1 for Voice / Asterisk rest of
> them for other functions).
>
> The Voice LAN is on the 172.16.3.0 (255.255.0.0) subnet. One the other
> NICS there are different but
rs, please jump in if I'm way wrong :)
Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic and Business Research
University of Florida
(352) 392-0171 Ext. 221
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adi Simo
Am Samstag, den 26.05.2007, 02:45 -0700 schrieb Crazy Boy:
> Hi Friends,
>
> I am planning to buy "IMate PDAL" mobile phone. This contains "Wi-Fi
> 802.11b/g" feature. So, Is it possible to get internet using my
> wireless router in my office?
Most probably yes. The device runs windows, so it com
Am Donnerstag, den 24.05.2007, 10:44 +0200 schrieb dima:
> Hello, everyone.
> I'm having a strange problem with my asterisk. After dialing and before
> the other side picks up the phone I should hear the tones (I'm not sure
> what are they called: p---pii) and in almost
Am Donnerstag, den 24.05.2007, 08:23 +0300 schrieb Cosmin Prund:
> > -Original Message-
> > From: [EMAIL PROTECTED] [mailto:asterisk-users-
> > [EMAIL PROTECTED] On Behalf Of Remco Post
> > Sent: Wednesday, May 23, 2007 10:47 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussio
Am Dienstag, den 22.05.2007, 20:37 -0500 schrieb Eric "ManxPower"
Wieling:
> David Florella wrote:
> > Thank you knox. Finally, I have chosen this solution : find
> > /var/spool/asterisk/voicemail/default/*/Old/ -atime -7|xargs rm –f, executed
> > every night by the CRON. However, I would have pref
Am Dienstag, den 22.05.2007, 21:49 +0300 schrieb Cosmin Prund:
> Googling arround I found a number of pocket pc softphones. Of those I was
> only able to install SJ-something (removed it).
>
> Is there one (pocket pc softphone) that works?
When I searched for one, about half a year ago, there we
Am Dienstag, den 22.05.2007, 17:35 +0300 schrieb Jonson Player:
> Thank you for reply. Can you send me some working configs? I'm still
> confusing about this sms option.
Just to get you started, try this:
Find out which user asterisk runs as. Get a shell for that user.
Run (all in one line)
smsq
Am Dienstag, den 22.05.2007, 13:21 +0300 schrieb Jonson Player:
> Hello,
> i just want to activate SMS service between my asterisk local sip
> accounts and between asterisk and local sip accounts. How can i do
> this thin? Also i tried smsq to an account but all i obtained is a
> error message:
>
Am Montag, den 21.05.2007, 23:16 -0500 schrieb Mike Hammett:
> If it is easy, could you enlighten me? I have another thread on caller ID
> matching, but I haven't received any positive responses.
In the context where your internal calls usually are handled, like this
(my internal phones have SIP
Am Donnerstag, den 17.05.2007, 10:40 +0200 schrieb [EMAIL PROTECTED]:
> Hi all.
> We have Snom phones which do have a defined key in order to drop incoming
> call WITHOUT answering.
>
> Pressing that key, a "SIP/2.0 486 Busy Here" message is sent back.
>
> We have other phones (I.E. DECT Siemens
the word proxy!). Figured I'd send this out in case
someone hadn't seen it.
Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic and Business Research
University of Florida
(352) 392-0171 Ext. 221
> -Original Message-
> From: [EMAIL PROTECTED]
> [mai
If you use edit the config files on a trixbox system like you would on
an * box, any time you reboot or hit the red update bar, it will reset
the files to what the gui has. The only files you can edit on a trixbox
system are the _custom.conf files. This may be the issue with the time out
Martin D
On May 14, 2007, at 12:34 PM, Tim Panton wrote:
On 14 May 2007, at 17:50, Martin Joseph wrote:
Hello again gurus.
I have been using Asterisk with great results going on a couple of
years now.
My primary box is running asterisk 1.42 built from a tar ball on
Mac OSX 10.4.9.
I have a
6:MMI-Y:200705081051010077',
'uniqueid' '51010077',
'userfield' '',
'MMI_field' 'not found'
Issue #2: When a call is not answered, a record of that call is written to the
database, but uniqueid is left blank. The next time a call isn
Hello again gurus.
I have been using Asterisk with great results going on a couple of
years now.
My primary box is running asterisk 1.42 built from a tar ball on Mac
OSX 10.4.9.
I have a very odd issue that I cannot seem to nail down, which is
related to my Nokia E60 SIP phone.
I use
Am Freitag, den 11.05.2007, 18:44 -0400 schrieb Jon Pounder:
> just out of curiousity - anyone ever hijack pairs and get away with it ?
> (do your own cross connects on the street and utilize some crossconnect
> all within one branch of F1 cable out of the CO ?)
>
> I've been tempted in the past,
Am Sonntag, den 06.05.2007, 00:48 -0400 schrieb Salvatore Giudice:
> Just forward them to 1-800-big-dick or some other 800 toll free phone sex
> line. They can't tell they've been forwarded. They'll figure it out
> eventually.
Whoa, that was _my_ coffee that's now on the screen.
I will urgently h
Am Freitag, den 04.05.2007, 00:48 -0400 schrieb Doug Crompton:
> Can anyone suggest a source for a free robot dialer or examples? I need to
> do some non-commercial auto dialing using Asterisk. Simple phone numbers
> in a file, line by line format.
>
> I found one called AstAutoDiaker but I was no
Am Mittwoch, den 02.05.2007, 20:04 +0100 schrieb Goke Aruna:
> Hello all,
>
> I have a set up that answer my customer. and its working well,
>
> however, the number of call to technical dept is what i want to reduce.
>
> I want all call to get to voice prompt except that that enter when
> minute
machine and our
primary, active one. We can't really give up the PRIs without some
downtime, so we're specifically interested in solutions that allow a
primary machine to remain in operation while testing a secondary, and
without using up the PRI circuits for testing (but we want to test ou
On 2007-03-26 01:46:40 -0700, "Salvatore Giudice"
<[EMAIL PROTECTED]> said:
This is a multi-part message in MIME format.
I opened up a ticket with them, but I'm not holding my breath. I think it's
time to start moving my DID's before the inbound stops working.
That seems like it was probab
Am Mittwoch, den 18.04.2007, 13:18 +0200 schrieb Knud Müller:
> Dinesh Nair wrote:
> >On Wed, 18 Apr 2007 09:04:22 +0200, Knud Müller wrote:
> >
> >>I
> >>think it can be done by using the dialplan and the database to store the
> >>statistical information but maybe there is an easier way that int
On 2007-04-17 00:53:56 -0700, Dinesh Nair <[EMAIL PROTECTED]> said:
On Mon, 16 Apr 2007 20:14:40 -0700, Martin Joseph wrote:
The phone no longer registers with asterisk, although it displays the
little icon as though it has, and it doesn't even seem to try to pass
calls to aster
Just a warning for you all that are using Nokia series E phones for SIP
function.
I updated my phones firmware today using the Nokia Updater, and now
the SIP functionality, which previously worked pretty well is
completely broken.
The phone no longer registers with asterisk, although it dis
I am having problems with my zaptel channels on my fresh install of Asterisk
1.4.2 on Fedora core 6.
I have a new Digium TDM400P with 2 FXO modules.
Both dmesg and ztcfg -vvv show the FXO modules loading correctly:
-
Zaptel Version: 1.4.1
Echo Canceller: MG2
Configuration
Am Freitag, den 06.04.2007, 18:23 -0700 schrieb Am Turnip:
> When I listen to voicemail from my Google Talk buddy, the envelope says,
> "from an unknown caller". But the voicemail correctly records the caller
> ID of calls that arrive via Zapata into the same context that receives
> Google Talk ca
Am Mittwoch, den 28.03.2007, 12:32 -0400 schrieb Brian Capouch:
> Jordan Novak wrote:
> > Okay, I get it. I still have a problem though. I have no way to wire 30%
> > of these end-points. P{hysically impossible. They do have cat3 twisted
> > pair to each phone. But of course they want IP. Are the
Am Donnerstag, den 29.03.2007, 15:04 +0300 schrieb Khaled Chehab:
>
>
> How to configure cisco 7902 with asterisk ,if you please can send me
> step by step configuration steps .
Khaled,
you already have a 7905 and a 7960, your older posts suggest that. Try
to configure the 7902 the same way. I
Am Samstag, den 24.03.2007, 11:43 -0400 schrieb Steve Totaro:
> You will probably want some sort or script to reboot the phone regularly
> (everyday) or it will just stop working (lose registration with *). The
> speaker phones really do stink on these but for a simple doorphone
> application,
On 2007-03-23 14:37:18 -0700, "Tom Lynn" <[EMAIL PROTECTED]> said:
Now I know where they've been spending my remaining balance...
I still use Sellvoip as my primary terminator, and have found the call
quality to be superior to any other ITSP from my location (Seattle).
I agree completely
On 2007-03-24 01:53:16 -0700, Edoardo Serra
<[EMAIL PROTECTED]> said:
Hi Francois,
[EMAIL PROTECTED] ha scritto:
Hi men,
I have already encountered some issue like this with few switches (very
known great brand) which doesn't like VoIP traffic !
I also have switches of a very known gre
Am Freitag, den 23.03.2007, 17:09 +0800 schrieb Christopher Chan:
> Anselm Martin Hoffmeister wrote:
> > Am Donnerstag, den 22.03.2007, 22:17 -0700 schrieb shadowym:
> > Let us see the facts: Telephone systems with more than a handful
> > telephones and more than just the a
Am Donnerstag, den 22.03.2007, 22:17 -0700 schrieb shadowym:
> As far as I can tell, the phone system does not run on a Desktop/Server OS
> on a standard PC. Just the config clients run on the desktop.
>
> Then again they are using Dlink as one of the 3 manufacturers of the Phone
> Server so I wo
Am Dienstag, den 06.03.2007, 05:18 -0400 schrieb Chris Mason (Lists):
> Of course, it would be highly unlikely anyone on the list would want
> to report Rehan...but in case anyone does:
I have been told that unsolicited commercial e-mail (I do not imply that
Rehan's post fulfills the criteria, ju
Am Montag, den 05.03.2007, 09:01 -0300 schrieb Assis, Eduardo:
> We installed na Asterisk System whith 400 Softphone users (Eyebeam 1.5
> from Counterpath).
>
> As far as we know, Asterisk don't support yet IM (Instante Message)
> feature,instead Eyebeam have this feature.
I cross-read their hand
On 2007-02-22 04:22:20 -0800, "Frederico Madeira" <[EMAIL PROTECTED]> said:
Hi guys,
My asterisk is show me some errors on line registration.
This message appear on console: Request to schedule in the past?!?!
What it mean ?
Thanks.
I see this message all the time on my lowely powerPC mac (
On 2007-02-14 22:12:23 -0800, "jameson asterisk" <[EMAIL PROTECTED]> said:
I'm currently looking to deploy an Asterisk server using an FXO media
gateway to connect to the PSTN and was looking for any user experiences that
may aid in selecting a gateway. Specifically i'm looking for a 4-port mo
Am Dienstag, den 20.02.2007, 16:33 -0700 schrieb Natambu Obleton:
> I would guess that registration would be by the telco for the blocks
> just like with reverse dns today, so then each telco would have a
> local server to manage their 'reverse' cnam lookup and the people
> in charge would be N
can be pretty sure that freshly
assigned numbers do not have dangling cache records, assuming the 3
months "gap" before assigning the same number again.
Assuming one could add an additional TXT record to enum, say
name.0.6.0.7.x.x.x.enum.info. TXT "Hoffmeister, Anselm Martin"
this w
Am Montag, den 19.02.2007, 12:39 -0700 schrieb Robert Norton - SophMedia
LLC:
> Hey Guys,
> I’m curious if there’s an interest in a free, CallerID database? For
> those of you in the same spot we are, our current provider only
> provides us with the CND, excluding CNAM.
>
> Would creating a publi
Hi list,
I bought two UTStarcom F1000 phones, pre-equipped with the latest
firmware, including WPA support. Those are configured to register to an
asterisk server on the internet (not LAN), and registration works.
Calling and being called also, with transfer and all bells and whistles.
After a fe
Am Dienstag, den 13.02.2007, 21:41 + schrieb Razza:
> Hi all, is it possible to to dumb down a "FRITZ!Box Fon
> ata" (http://www.avm.de/en/Produkte/FRITZBox/FRITZ_Box_Fon_ata/index.html##)
> and have the two FXS ports AND the ISDN interface register with Asterisk. In
> much the same way a sip
Am Mittwoch, den 14.02.2007, 07:17 +0800 schrieb Ronald Wiplinger:
> Where can I get a starting point for setting up sms via VoIP and via web.
>
> I want to send SMS from VoIP or web to VoIP phones and GSM phones.
>
> 1. how to set-up?
> 2. which smsc should I use? (what is the price?)
> 3. whi
Am Mittwoch, den 07.02.2007, 21:57 -0800 schrieb Jason Kim:
> Hi,
>
> This is the configuration I want.
>
> Hard Video phone<---video--->Soft Video Phone(PC)
>^
>|
> audio
>|
>V
>Audio Only Phone
>
> Any idea?
You could see wether having a second call th
Am Donnerstag, den 01.02.2007, 16:15 -0600 schrieb Larry Alkoff:
> I wish to have my Grandstream GXP-2000 phones make a different
> distinctive ring for internal calls ( Internal ) or if the incoming call
> has no caller id 'NOCID'.
>
> The Grandstream phones calls allow 3 distinctive rings depe
On 2007-01-28 08:37:43 -0800, "Eric Germann" <[EMAIL PROTECTED]> said:
We LOVE Teliax. We're on a Time Warner business class fiber connection and
avg 25ms latency from Ohio to Denver CO.
With that connection I would love Teliax also.
Marty
___
--
Am Montag, den 29.01.2007, 11:58 +0100 schrieb Thomas Winter:
> Hi,
> If I develope an dialplan, some AGI and AMI functions for Asterisk and ship
> it
> as an complete product to an coustomer, do I have to put my developed code or
> the complete product under the GPL?
IANAL, but in my understan
Facundo, the company that I work for use Crossfone,
www.crossfone.com.ar
Best Regards,
Martín
On 1/26/07, Facundo Ameal <[EMAIL PROTECTED]> wrote:
Hello everyone!
I 've looking for carriers which can terminate my international calls.
They must accept payments from Argentina and give me inter
Am Dienstag, den 23.01.2007, 05:41 -0200 schrieb Barzilai Spinak:
> I've seen several examples that use extensions such as;
> s-BUSY
> s-NOANSWER
>
> etc...
>
> It's more or less evident what they do, but I've searched for some
> FORMAL documentation everywhere and have found nothing.
> Do they
On 2007-01-17 10:29:43 -0800, Yelson Vivas <[EMAIL PROTECTED]> said:
Hi Guys
I'm conecting 2 astersk servers using this arquitecture
(Ext softphone)<==sip==>(asterisk 1)(asterisk 2)
<===alaw==>(pstn)
If i call from the Ext to the asterisk 2 the sound is perfect, but if
On 2007-01-14 22:01:44 -0800, Tomer Horn <[EMAIL PROTECTED]> said:
Hello,
I am looking to purchase a new quad-band cellphone and I'm looking for
one with WiFi and enough CPU power for stable SIP calls. I was
wondering if anyone here can share his experience and recommend on a
good cellphone.
Am Dienstag, den 16.01.2007, 15:04 -0800 schrieb chester c young:
> the answer sucks, but is apparently correct.
If your application involves the caller (e.g. an employee of your
company) to rate the call he just did, or to enter any data to a mysql
database over the phone right after the call, yo
Am Mittwoch, den 17.01.2007, 07:38 +0800 schrieb Leo Ann Boon:
> Andrew Joakimsen wrote:
> > I have some Audiocodes units which appear to be running Linux,
> > according to the unit's own "System Log"
> >
> > kern.warn Linux version 2.4.21openrg-rmk1 #2 Wed Aug 30 17:05:29 IDT 2006
> >
> Googling t
Am Dienstag, den 16.01.2007, 12:01 -0800 schrieb Alejandro Duplat:
> Hi,
>
> Someone knows an Open Source solution that can handle Outbound IVR for
> asterisk?. What I'm looking it a pre-preprogrammed a telephone call that
> reach a Person and start making an Interview over the telephone.
>
>
Am Montag, den 15.01.2007, 14:22 -0800 schrieb chester c young:
> > Silly question: how are the calls going out? If they're going out
> > through an analog line without the ability to detect hang-ups, then,
> > that's the problem.
> >
>
> calls are coming in and out thru an Asterisk server usin
Am Montag, den 15.01.2007, 13:38 -0700 schrieb Andrew Niemantsverdriet:
> If you would bother to read my post you will see that what I am
> wanting to do is not the asterisk directory cmd. I don't want them to
> be able to search or anything fancy like that. I want an app that will
> go through and
Am Sonntag, den 14.01.2007, 17:13 -0800 schrieb chester c young:
> cannot make Dial(...,,g) work correctly, although Dial(...,,gh) works
> just fine. (to make matters worse, it does seem to work sometimes,
> although once working or not working between changes it either works or
> doesn't work all
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