Michael J. Liberatore wrote:
Can you elaborate on OSLEC? I cant say I have heard of it but it sounds
very interesting considering it worked for x100p for you which was the
worst out of ALL the cards I have ever tried for echo.
Thanks
Mike
Hi Mike,
Sure...
OSLEC (Open Source Echo
On Sun, 25 Nov 2007, Rilawich Ango wrote:
It works if it specified the port exactly plugged to PSTN. I want to
clarify the dial command here.
Dial(zap/g1/1234567)
It will try channel 1, if it is busy, congested then it will try
channel 2 and so on, right?
Yes.
I wonder if I don't plug
Thanks for the tip but my question was more towards asking for return of
experience than asking technical capabilities.
For a long time now, there's a lot of buzz surrounding server consolidation
(
http://searchdatacenter.techtarget.com/sDefinition/0,,sid80_gci1070272,00.html)
along blade servers
Robert,
Do you mean T.38 passthrough ou T.38 to T.30 gateway ?
The former is said to work with Asterisk 1.4 but the latter is not ...
Cheers
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Hi Paul,
2007/11/23, Paul Hayes [EMAIL PROTECTED]:
Robert Lister wrote:
Hello,
I think I have encountered an odd bug in Siemens C460 IP/dect handsets,
which is a bit annoying, and I'm not (yet) sure how to get round it
without
lots of hacks.
Basically, on all external incoming
Hi,
I think most of what you're looking for relates to do you consider
sip.confto describe users or resources ?
If it describes resources, how do you manage other IT resources (PC,
printers, ...) ? Do you store devices passwords (BIOS passwords with Serial
numbers, ...) in an LDAP database ?
If
Olivier wrote:
Robert,
Do you mean T.38 passthrough ou T.38 to T.30 gateway ?
The former is said to work with Asterisk 1.4 but the latter is not ...
I know about what Asterisk 1.4 can do. And Asterisk 1.2 only does T.30
passthrough :) You need 'stuff' to handle fax. Stuff like spandsp,
Asterisk does not detect analog ports with no line plugged in. It does
not test for dialtone before dialing (this applies to all analog cards
except the X100P).
Rilawich Ango wrote:
It works if it specified the port exactly plugged to PSTN. I want to
clarify the dial command here.
Do the SIP-FXO gateway devices do any better?
Eric ManxPower Wieling wrote:
Asterisk does not detect analog ports with no line plugged in. It does
not test for dialtone before dialing (this applies to all analog cards
except the X100P).
Rilawich Ango wrote:
It works if it specified the
You can also take a look at the T.38 product from Attractel.
http://attractel.com/faxterisk.php
Disclaimer, I work for these guys.
Chris
On Sun, November 25, 2007 4:11 pm, Robert Moskowitz wrote:
Olivier wrote:
Robert,
Do you mean T.38 passthrough ou T.38 to T.30 gateway ?
The former is
Hi all,
Im preparing a quote for a 5 Star hotel, planning to have around 100
SIP Wifi phones for PBX operations running on 100 AccessPoints.
Network is running in ARUBA Networks - AP70 access points.
The initial recommendation is to go for Hitachi Wifiphones, but i
would like to know from the
Olivier wrote:
Thanks for the tip but my question was more towards asking for return
of experience than asking technical capabilities.
For a long time now, there's a lot of buzz surrounding server
consolidation
As SIP is not Analog FXO, my comments do not apply to them. I have no
idea if or which analog adapters might detect line voltage or dialtone.
Paul wrote:
Do the SIP-FXO gateway devices do any better?
Eric ManxPower Wieling wrote:
Asterisk does not detect analog ports with no line plugged
Hi all,
Im preparing a quote for a 5 Star hotel, planning to have around 100
SIP Wifi phones for PBX operations running on 100 AccessPoints.
Network is running in ARUBA Networks - AP70 access points.
The initial recommendation is to go for Hitachi Wifiphones, but i
would like to know from the
On Sun, 25 Nov 2007 23:26:54 +0300, [EMAIL PROTECTED] wrote:
Hi all,
Im preparing a quote for a 5 Star hotel, planning to have around 100
SIP Wifi phones for PBX operations running on 100 AccessPoints.
Network is running in ARUBA Networks - AP70 access points.
The initial recommendation is to
My number one recommendation is be VERY VERY Careful. You could be
selling the biggest nightmare to you and the customer ever.
I have tried almost all the wifi sip phones and they are ALL sub par.
Range is terrible on most, but mainly its staying connected to the ap's
all the time and especially
Hello
I noticed something nasty with the Record() function: If the
user either hangs up during the prompt (ie. doesn't leave a message at
all), or does leave a message but forgets to hit the # key at the
end... Asterisk stops right there, so the rest of the script doesn't
run:
For campus installations such as this, you may want to look at Polycom
(Spectralink) phones. They are more expensive but are designed for tough
environments and are of better quality than any of the consumer-oriented
phones.
Asterisk should be fine for an installation of this size, no need
Vincent wrote:
Hello
I noticed something nasty with the Record() function: If the
user either hangs up during the prompt (ie. doesn't leave a message at
What exactly are you trying to do? If a user hangs up during your
Record, it'll go directly the the h extension if it exists.
On Fri, 23 Nov 2007 12:38:45 -0500, Baji Panchumarti
[EMAIL PROTECTED] wrote:
Sometimes I use press 1 to leave a msg to reduce the
number of dead air msgs from callers.
Good idea. BTW, making changes to zapata.conf did allow the Zaptel,
and hence */Record to tell if the user hung up during the
On Sun, 25 Nov 2007 19:03:41 -0500, Doug Lytle [EMAIL PROTECTED]
wrote:
What exactly are you trying to do? If a user hangs up during your
Record, it'll go directly the the h extension if it exists.
Ah, didn't know about this extension :-/ I assumed Asterisk would go
on to the next line in the
Thanks mate, this helped a lot
On Nov 24, 2007 4:40 PM, Tilghman Lesher
[EMAIL PROTECTED] wrote:
On Saturday 24 November 2007 14:48:02 Robert McNaught wrote:
Does anyone know if it is possible to use the same database and single
ODBC connection to do both CDR recording with cdr_odbc and
Hi,
I am trying to write an application which sends DTMF tones once the
called party answers the call from asterisk.
From the way I understand asterisk dialplans work - the below example
will NOT work as the dial application does not finish and move onto
the next priority once the call is
On Sunday 25 November 2007 21:55:30 Robert McNaught wrote:
Does anyone know of a way to wait x number of seconds, then send the
DTMF digits as audio once the call is answered.
Check out the M() parameter to Dial.
--
Tilghman
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Hi * Users,
What is the way from the dial-plan to get the IP address of an
incoming or outgoing SIP call? I can see the IP address of the SIP
call using 'sip show peers' from the CLI.
Thanks
Regards
--
Arpit Mehta
Graduate Student
Department of Computer Science
Columbia University
Tel:
Hi all,
I am having requirment to display the status of the users continuously
updated, for that I am having a php script which checks the status
channels every seconds using the AMI .Now for this task the manager
logs on and off every second. So, can any one help me to make manger
just log in
I need it when the SIP calls comes to me in some context in a
dialplan. The only way I can think of is right now which is kind of
messy is to do 'sip show peers' parse the output in some agi script ,
get the ip from there.
On Nov 26, 2007 1:53 AM, Johnny Tam [EMAIL PROTECTED] wrote:
if you run
hi,
I want to create connection using odbc for mysql
i have used cdr_odbc module for that.
but when asterisk insert record to my mysql database arise segfault error.
any suggetion, pls give me
tnks
Bhrugu Mehta
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Hi,
1. Is your WiFi network dedicated to VoIP or shared with data applications ?
How was it designed ?
For people using WiFi with a laptop, you propably don't need to have dense
WiFi cells as moving from one cell should be scarce.
With hand phones, those cells should overlap as it becomes very
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