I have seen some work on various IAX activex phones but not really sure if
they can do what I want or if there is something that can.
I just want a simple link on a webpage that dials an extension
automagically. I do not need a nice phone displayed, the ability to dial
any numbers, just autodial
Trond G. Andersen wrote:
Trond Andersen wrote:
Thank you, but I have tried that... Then the To is:
Can you do a NoOp(${ARG1}) and then show us the result?
--
Cheers,
Matt Riddell
Thank you for taking the time to help me out Matt !
-- Executing NoOp(SIP/trond-c7f0, ARG1=20170) in
Can Hylafax be made to produce ccitt G4 instead of ccitt G3 encoded images?
The G4 tiffs are smaller than G3 and are much more efficient to convert to
pdf. I was able to patch spandsp to produce G4 encoded tiffs and was
wondering if Hylafax could be made to do the same as I'd really like ECM
Hello everyone,
Can anyone explain to me what the streamplayer util is for? If
possible, it would be great if someone could also send me an example of
how it's used in practice. It looks very interesting...
Thanks so much,
Leo
___
--Bandwidth
Damian Minkov wrote:
http://sip-communicator.org/
LOL, have you tried it with Asterisk?
--
Cheers,
Matt Riddell
___
http://www.sineapps.com/news.php (Daily Asterisk News - html)
http://freevoip.gedameurope.com (Free Asterisk Voip Community)
ram wrote:
http://www.sokol-associates.com/IaxPhone.htm
i dont see this is working
any one have idea
any other place i can get this
Zoa just posted:
http://iaxclient.sourceforge.net/
--
Cheers,
Matt Riddell
___
Marc Storck wrote:
Hello,
I would like to know if there is a way in IAX2 and SIP to tell a client
to register at a different server.
For example:
Client tries to register at server B but server B answers with some sort
of redirect to tell the client to register at server C. The client
Kevin P. Fleming wrote:
Leo Burd wrote:
Any ideas about what is going on?
Yes. You didn't read the warnings prominently displayed at the end of
'make install' about removing old modules from /usr/lib/asterisk/modules.
LOL Kevin, I think you're going to have to redo the message in
Craig Guy wrote:
Can Hylafax be made to produce ccitt G4 instead of ccitt G3 encoded
images? The G4 tiffs are smaller than G3 and are much more efficient
to convert to pdf. I was able to patch spandsp to produce G4 encoded
tiffs and was wondering if Hylafax could be made to do the same as
Obelix wrote:
I want to play a sound on the connection of a call using the LIMIT_CONNECT_FILE
option but can't find any examples.
Does anyone have any examples? Examples of the usage of the other LIMIT_xx
options would also be appreciated.
Obelix
At the top of extensions.conf you can put:
Title: IAXmodem
I know you can send faxes using a hylafax client for
windows and sending thru hylafax and iaxmodem out from asterisk but I was
wondering, how do you receive faxes? can you received them as tiff and then
convert to pdf and send via email or something?
From: [EMAIL
On Nov 18, 2005, at 11:08 PM, Anton Krall wrote: I know you can send faxes using a hylafax client for windows and sending thru hylafax and iaxmodem out from asterisk but I was wondering, how do you receive faxes? can you received them as tiff and then convert to pdf and send via email or
300 seconds is a mighty long time to keep state on a udp connection. Our
firewalls time out udp states out in 2 seconds of inactivity. But your
point is valid and taken...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of tim panton
Sent: Friday,
Joseph wrote:
VOIPJET - one of their servers was down, so switching the server IP
solved the problem.
Is it possible to specify two or three alternate server in iax.conf just
in case once goes down so the system will try alternate server
connection?
Yes you can. The logic for cascading
Hi. I'm a new user of Asterisk. My question is:
I want to log outbound calls in a database ( postgres ). Everything is OK
except that asterisk always marks calls to my FXO iface ( Zap/4 ) as
answered as soon as it accepts to dial the specified number and the
purpose of logging calls is to know the
[snip]
Thanks for offering,
I was trying to reach area code: 011-63-917-xxx.xxx during the day and
the phone wouldn't even ring, one time the connection was congested, but
now it goes through via VOIPJET.
It would seem to me poor line connection with the Philippines. They
never had a good
I'm trying to retrieve multiple variables for a database into their
respective variable names.
i.e.
select technology,username,password,servername from mytable where type=1
then I want to capture those values for those columns into a variable name
by the name of the column.
i.e.
if
Hi
Actually, I was hoping not use any phone at all. My idea was to
install a softphone on the laptop and use a standard BT headset
without using any cell phone at all.
This is the way I use it too..:-)
My laptop doesn't have a decent quality speaker or mic, that's where
the headset comes
Jonathan k. Creasy wrote:
What context are your phones in? (context= in sip or iax config)
If your phones are in the local-users context, they will be able to dial
numbers found in local-users, extensions and local.
If your phones are in the long-users context, they will be able to dial
[EMAIL PROTECTED] wrote:
[EMAIL PROTECTED] wrote on 11/16/2005 09:46:17 PM:
Hi,
Yes, I'm using wav for my recording and the file is quite
large.
I too am using WAV files because of the volume issue: WAV files are
shifted two bits louder than any other format. (slight details
Rafael R. GV wrote:
Hi
I have this 3 warnings running a2billling with asterisk new version:
a2billing.php|2:
-- AGI Script Executing Application: (SetLanguage) Options: (en)
Nov 18 12:06:19 WARNING[17440]: pbx.c:5435 pbx_builtin_setlanguage:
SetLanguage is deprecated, please use
Hi all,
Is sending text to a conference supported by asterisk-1.2, ie one member
of the conference sends text, it is received by all other members of the
conference (provided their channel supports text of course) ?
I made a quick test with IAX softphones, and it seems that text isn't
sent to
I think that delay in answering is due to caller ID detection.
I have no idea about rest of your question :)
regards,
Umair
On 11/18/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hi. I'm a new user of Asterisk. My question is:I want to log outbound calls in a database ( postgres ). Everything
I posted the following a couple of days ago. My problem was inbound,
but the workaround might be worth a try:
==
Bug or feature?
I'm fairly sure this behavior changed between CVS HEAD Nov 1 and RC2.
In the earlier release, I could successfully execute the Read()
application immediately
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