On Mon, 28 Jan 2008 12:01:43 +0530, <[EMAIL PROTECTED]> wrote:
>>I have installed asterisk.When I start asterisk it starts normally and shows
>>the status running.
>> My partner also installed asterisk. I registered 1 user of her server and 1
>> user of my server in X-lite.
>> I am able to call o
Hello
I've read in the documentation that we should use the CLI
version of PHP when using it to write scripts called by AGI, because
the prepended HTML bits would confuse Asterisk when it got a reply
from the script through STDIN.
And still, the following works OK, although the CLI versio
Thanks to all.
I got the solution.
-Original Message-
From: [EMAIL PROTECTED] on behalf of [EMAIL PROTECTED]
Sent: Mon 1/28/2008 10:42 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Using x-lite -Call failed 404 not found
Hi all,
I have installed asterisk.When I star
Hi all,
I have installed asterisk.When I start asterisk it starts normally and shows
the status running. My partner also installed asterisk. I registered 1 user of
her server and 1 user of my server in X-lite. I am able to call or receive call
from the users registered in her server but not in
Abdul,
Can you explain your request more in details? From your mobile phone or
landline into your Asterisk is a a different network and will have to pay
origination to into it. For the toll free the receiver pays for it. So if
you set up the toll free that means you pay for the call. Its never fre
JR Richardson wrote:
>Hi All,
>
>After reading the sparse info and attempting to get this running, I'm
>unsuccessful and could use some guidance.
>
>I already have a MRTG server up and running serving hundreds of router
>interface graphs. I would like to add SIP/IAX channel graphs for all
>our as
Hi friends,
Is their any possibility to setup our own Toll-Free Number in Asterisk using
some PCI or FXO Card?
I have one number from my local Telecom called 123 and i would like
to setup this number in my asterisk if some one called this number from his
mobile or land line he shou
On Sun, 2008-01-27 at 20:19 +0100, Benny Amorsen wrote:
> John Millican <[EMAIL PROTECTED]> writes:
>
> > I am trying to avoid loading a soft phone since I don't want to have
> > to instruct the users on how to use one (mostly NON-technical
> > types).
>
> You can't have a USB handset without a s
On Sun, Jan 27, 2008 at 08:19:55PM +0100, Benny Amorsen wrote:
> John Millican <[EMAIL PROTECTED]> writes:
>
> > I am trying to avoid loading a soft phone since I don't want to have
> > to instruct the users on how to use one (mostly NON-technical
> > types).
>
> You can't have a USB handset with
Hi All,
After reading the sparse info and attempting to get this running, I'm
unsuccessful and could use some guidance.
I already have a MRTG server up and running serving hundreds of router
interface graphs. I would like to add SIP/IAX channel graphs for all
our asterisk servers. I'm running a
Vincent wrote:
> On Sun, 27 Jan 2008 09:09:59 -0500, Lee Jenkins <[EMAIL PROTECTED]>
> wrote:
>> Sorry, I don't have a sample for you as I write mostly in Freepascal/Lazarus
>> these days and use my own library for AGI/FastAGI. That said, did you try
>> saving the file to a fully qualified path?
The Aastra's also have a range of interested firmware bugs that
support/development just can't seem to fix. Do a search for aastra
hang/lockup and you will find what I mean. They look very nice though! For
a simple home deployment, Aastra's are probably great.
> -Original Message-
>
On Sun, 27 Jan 2008, Vincent wrote:
> If someone has an idea why it's required to use a full path for the
> file to be written, I'm interested.
If you don't specify a path, the current working directory of the
executing process is used. So, the question is, what is the CWD of your
AGI process?
John Millican <[EMAIL PROTECTED]> writes:
> I am trying to avoid loading a soft phone since I don't want to have
> to instruct the users on how to use one (mostly NON-technical
> types).
You can't have a USB handset without a soft phone. You can get some
which automatically run the phone software
On Sun, 27 Jan 2008 09:09:59 -0500, Lee Jenkins <[EMAIL PROTECTED]>
wrote:
>Sorry, I don't have a sample for you as I write mostly in Freepascal/Lazarus
>these days and use my own library for AGI/FastAGI. That said, did you try
>saving the file to a fully qualified path?
My hero! :-) That did i
Gordon Henderson wrote:
> On Sun, 27 Jan 2008, John Millican wrote:
>
>> Tzafrir Cohen wrote:
>>> On Sun, Jan 27, 2008 at 11:27:16AM -0500, John Millican wrote:
Hello All,
This may be a little OT for the list but it seems to be to be the place
to get the best answer. I have looked a
Gordon Henderson wrote:
> On Sun, 27 Jan 2008, John Millican wrote:
>
>
>> Tzafrir Cohen wrote:
>>
>>> On Sun, Jan 27, 2008 at 11:27:16AM -0500, John Millican wrote:
>>>
Hello All,
This may be a little OT for the list but it seems to be to be the place
to get the best
On Sun, 27 Jan 2008, John Millican wrote:
> Tzafrir Cohen wrote:
>> On Sun, Jan 27, 2008 at 11:27:16AM -0500, John Millican wrote:
>>> Hello All,
>>> This may be a little OT for the list but it seems to be to be the place
>>> to get the best answer. I have looked at the many Skype/Yahoo phones out
I doubt that chan_oss/chan_alsa directly support echo cancelling.
However depending on exactly how you are using the inputs and outputs on
a sound card, you could very well need echo cancellation.
90% of the time, echo is generated at the junction between a channel
that separates received and tra
Iirc, there used to be such an adaptor in the digium dev kit years ago.
Maybe somebody here remembers what it was exactly ?
Zoa
John Millican wrote:
> Hello All,
> This may be a little OT for the list but it seems to be to be the
> place to get the best answer. I have looked at the many Skype/
Tzafrir Cohen wrote:
> On Sun, Jan 27, 2008 at 11:27:16AM -0500, John Millican wrote:
>> Hello All,
>> This may be a little OT for the list but it seems to be to be the place
>> to get the best answer. I have looked at the many Skype/Yahoo phones out
>> there and none seem to be what I am looking
"Chris Bagnall" <[EMAIL PROTECTED]> writes:
>> I'm looking for recommendations (good/bad!) on what people use for a
>> reception console type of phone. So-far I've used Grandstreams and Snoms,
>> both have good and bad points.. Not after anything too fancy -
>> programmable "extension" buttons wou
On Sun, Jan 27, 2008 at 11:27:16AM -0500, John Millican wrote:
> Hello All,
> This may be a little OT for the list but it seems to be to be the place
> to get the best answer. I have looked at the many Skype/Yahoo phones out
> there and none seem to be what I am looking for.
> I have a need for a
Hello All,
This may be a little OT for the list but it seems to be to be the place
to get the best answer. I have looked at the many Skype/Yahoo phones out
there and none seem to be what I am looking for.
I have a need for a USB handset that I can use with an Asterisk server.
This will be on t
That's surprising.. When I looked at pricing, the Snom 370 was about $50 more
expensive than a 57i for us (the 57i was $205). Also, configuration wasn't too
bad on the Aastra, but that may just be me.
BTW, it also looks like the Snom has support for an electronic headset "lifter"
on some GN Net
> I'm looking for recommendations (good/bad!) on what people use for a
> reception console type of phone. So-far I've used Grandstreams and Snoms,
> both have good and bad points.. Not after anything too fancy -
> programmable "extension" buttons would be nice, good display essential...
We usually
Ronald Wiplinger wrote:
> [Jan 27 16:03:32] -- Executing RxFAX("SIP/88621001-00728610",
> "/var/spool/asterisk-fax/3000/1201421004.8.tif") in new stack
> vpbx*CLI>
> Disconnected from Asterisk server
>
>
> I have no idea why it disconnects and hope somebody can help me to get
> to work.
>
>
Vincent wrote:
> Hello
>
> I'm pretty much a newbie when it comes to C, but I have to use
> this language to write a couple of AGI proggies because I need them to
> be statically compiled.
>
> Strangely enough, Google didn't return much when looking for the
> "Hello, world!" of AGI in C.
>
I'm looking for recommendations (good/bad!) on what people use for a
reception console type of phone. So-far I've used Grandstreams and Snoms,
both have good and bad points.. Not after anything too fancy -
programmable "extension" buttons would be nice, good display essential...
(Brief backgro
Indeed, through the dialplan configuration, Asterisk allows you to
place calls from whatever channel type (SIP, H323, ISDN, ..) to
GoogleTalk clients.
The revert call configuration is trickier though, as the GoogleTalk
client user interface does not allow you to dial numbers over DTMF for
example.
I used this as a manual for blukprovisioning..
http://voipspeak.net/index.php?option=com_content&task=view&id=73&Itemid=28
Rob Hillis schreef:
Hi Eric,
You may want to contact me off-list - the company I work for offers a
product which aims to be a zero configuration service fo
Hi Eric,
You may want to contact me off-list - the company I work for offers a
product which aims to be a zero configuration service for Asterisk. The
Linksys 942 and 962 phones /are/ supported.
Erick Perez wrote:
> Hi there,
> We have plans to install an office (not call center) with the follow
On 04:41, Sun 27 Jan 08, Vincent wrote:
> Hello
>
> I'm curious about what can be done when using Jabber with Asterisk.
> What are good examples of this combination?
What I do is make asterisk send a message to one of my
jabber accounts when a call comes in. Also, because asterisk
is in my buddy
Erick,
I'm not aware of any precompiled package with such capabilities.
However, I've implemented this capability myself by creating a
python script that generates automatically tftp profiles
for linksys phones (941, 942, 922, etc.), as well as entries
for sip.conf, extensions.conf, voicemail.conf
Below is my extensions.conf for the fax part
[incoming_28345474]
;
;
; BEGIN - Inbound call handlers
;
;
exten => 8862100,1,NoOp(${CALLERID(num)})
exten =>
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