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
message, I
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 line
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 finding
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]
[mailto
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 I
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 ?
I
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
domains for
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There were rumours they had trouble with an outdated version of the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
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
://lists.digium.com/mailman/listinfo/asterisk-users
--
Martin Vít
LAM plus s.r.o.
http://www.lam.cz/
Tel.: 605 267 610
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit
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 voip
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, which in
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
certainly see
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 is
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 I
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 point
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 Business Research
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-
From: [EMAIL
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 PC
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 carrier
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]
[mailto:[EMAIL PROTECTED] On Behalf
, 3378, 4) exited non-zero on 'Zap/97-1'
-- Executing Hangup(Zap/97-1, ) in new stack
== Spawn extension (default, h, 1) exited non-zero on 'Zap/97-1'
-- Hungup 'Zap/97-1'
Any ideas as to why I can't hear anything? Thanks!
Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau
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 1.4.4.
, 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, Rochester, NY 14624
Office
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 was sent to my
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: Kevin P. Fleming
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
- Original
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 dunno.
As
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 bounced
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 Economic and Business
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 Colocation Provided by http://www.api-digital.com
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
Office: 888
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
to be included in compilation. Once you're sure it's being built,
and you can actually find the res_agi.so output file, try specifying it be
loaded in modules.conf. Check out
http://www.voip-info.org/wiki/index.php?page=Asterisk+config+modules.conf
for more on that file.
Thanks
Ronaldo.
Take care,
Martin
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 Ext. 221
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)
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 regular \n and
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: Tuesday
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 Business Research
University of Florida
(352) 392-0171 Ext. 221
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
(352
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 also
:)
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 Simon
Sent: Thursday, May 31, 2007 5:54
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 comes with
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 Discussion
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 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 preferred
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 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 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, 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 were
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 C450IP,
!). 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]
[mailto:[EMAIL PROTECTED] On Behalf Of
Lee
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
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, and
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
complains:
cdr_addon_mysql.c: mysql_cdr: Failed to insert into database: (1062) Duplicate
entry '' for key 1
I haven't found any other information regarding these errors. I am just
wondering if they are bugs. Any insight would be appreciated!
--
Jason Martin
Metrix Matrix, Inc.
785 Elmgrove Road
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
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
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 not
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
minutes is
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 our
cards for load).
Thanks!
Martin Smith, Systems
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 probably
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 asterisk...
So, I
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 integrates
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
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
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 calls.
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 there any
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. If
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
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
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, it
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
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 ability to call (be it voicemail
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,
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 handbook
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
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 would pretty much do the trick. I have no idea wether any standard
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
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 model
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
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 public
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. which
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 sipura
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 that does a
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
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 understanding
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
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 work for
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)iax2 trunk(asterisk 2)
===alaw==(pstn)
If i call from the Ext to the asterisk 2 the sound is perfect, but if
i call
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 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 turns up:
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,
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 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 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 using iax2.
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
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
Am Freitag, den 12.01.2007, 11:31 -0500 schrieb Matt:
Hi List,
I recently signed up with Voxbone to get some International DIDs. I
was just about to purchase a DID this morning... but when I went to
get it voxbone wanted the end user's address information. So I
started to put it in...
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