Actually it's $15/channel.
Mark
On Sat, 14 Jun 2003, Matthew John Darnell wrote:
It should work, but there is a fee of $30 per channel for the software.
Check the archives
- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, June 14, 2003 12:44 PM
I'd stick to strace and see if you can see what directory it opens:
strace /usr/sbin/asterisk -vvvg 21 | grep ^open
You should see mentions of /usr/lib/asterisk/modules and at the very least
see it loading modules.conf
Mark
On Sat, 14 Jun 2003, Moshe Yudkowsky wrote:
I checked that; the
Hi,
The D41/E does not support any sort of duplex audio path operation. That
seems a major limitation with Asterisk. What functionality can it
actuakky support with Asterisk?
Regards,
Steve
Matthew John Darnell wrote:
It should work, but there is a fee of $30 per channel for the software.
Dave,
In my (limited) experiance with Asterisk, your header in sip.conf must be
identical to the 'username' property. Note the example:
[phone1]
type=friend
username=phone1
secret=phone1
host=dynamic
defaultip=192.168.1.28
dtmfmode=inband
canreinvite=no -- You may want to add this line for
Hi guys,
Being a true Linux geek, I've never been too much into sounds or sound
files other than a few .mp3 songs I got. My question is pretty
straightforward and simple. I see that the music format of choice for
asterisk is .gsm. What can I use to listen to files in .gsm format and
what is
At 10:34 2003-06-15 -0400, you wrote:
Hi guys,
Being a true Linux geek, I've never been too much into sounds or sound
files other than a few .mp3 songs I got. My question is pretty
straightforward and simple. I see that the music format of choice for
asterisk is .gsm. What can I use to listen
I changed it to use the extension number for the username and secret and
now it's working fine. Very strange, but I'm glad it works.
Thanks
dave
On Sun, 15 Jun 2003, Thomas A. Roberts wrote:
Dave,
In my (limited) experiance with Asterisk, your header in sip.conf must be
identical to the
But you're using a packaged version of asterisk?
Have you tried with downloading, compiling and installing
from cvs?
Matteo.
Il dom, 2003-06-15 alle 14:46, Moshe Yudkowsky ha scritto:
Oops. Here is the listing I promised.
(Ignore the attempt in the log to load codex_speex.so -- I was
Hi,
There is any available GSM file player for Windows, compatible with the
Asterisk GSM format?
I receive the voicemail messages by mail as attachment and the sound is in
GSM format.
Thanks,
Dan
- Original Message -
From: Moshe Yudkowsky [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent:
At 18:31 2003-06-15 +0200, you wrote:
But you're using a packaged version of asterisk?
Have you tried with downloading, compiling and installing
from cvs?
The pre-packaged .deb always gave me this problem -- stop now never
worked, but at least it loaded all the dialplans when I'd start it.
The
I have used Tiger jet usb phones and works with asterisk but with Open h323
softphone in h.323 or eyepmedia sip softphone , it`s only a USB speaker and
mic , what wee need to ahve is the dialpad working into this device and the
best of all this Hardware is $20.00 bucks
regards
Humberto
Humberto, what hardware was that?
Gene
-Original Message-
From: Humberto Atristain V. [mailto:[EMAIL PROTECTED]
Sent: Sunday, June 15, 2003 12:39 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] InternetPhoneWizard
I have used Tiger jet usb phones and works with asterisk but
I also have the tigerjet usb phone. I was wondering if it would work too.
does the tiger work fine with 723.1 ? do you know if the pci board they sell
is compatible with asterisk ?
thanks,
- Original Message -
From: Humberto Atristain V. [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent:
Ok,
this has really freaked me out, but in a good way - sort of.. I've made no changes at
all to my system, save messing with ADSI. However this has nothing to do with ADSI.
The thing is all of a sudden my DECT phones have started reporting caller id, and not
just the number, the name too!
And that I do get some of the channels loading, e.g., the modem channel:
open(/usr/local/lib/asterisk/modules/chan_modem.so, O_RDONLY) = 8
And if I load the apps via load app_playback.so,
open(/usr/local/lib/asterisk/modules/app_playback.so, O_RDONLY) = 25
It would seem that the most
Moshe == Moshe Yudkowsky [EMAIL PROTECTED] writes:
Moshe The sox program will convert wav into gsm:
And can also be used for playback. The sox package usually comes
with a script called play; you can use that for easy playback of most
audio formats:
play foobar.gsm
-JimC
Is this correct?
I see the 100 Trying on REGISTER frequently, but if it's not valid, we can
take it out. It serves no really effective purpose.
2. 10.3 Processing REGISTER requests. The 5th paragraph states that the
registrar has to know the set of domain(s) for which it maintains
bindings.
switch = IAX/[EMAIL PROTECTED]/incoming
is really the same as this:
exten = .,1,Dial([EMAIL PROTECTED]/${EXTEN})
(that's = dotcomma1)
Almost, except that if, for example, you have an analog phone, as you type
each digit in the switch case, Asterisk will ask the other host is this a
valid
Hi,
Something similar on Windows 2K/XPor Pocket PC too?
Thanks,
Dan
- Original Message -
From: James H. Cloos Jr. [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, June 15, 2003 10:56 PM
Subject: Re: [Asterisk-Users] .gsm files
Moshe == Moshe Yudkowsky [EMAIL PROTECTED] writes:
Hi,
I'm having trouble using Ringing with a SIP client. I'm trying to
give the caller the impression that the line hasn't been answered,
whilst listening for various extensions to be dialled.
Here's is the extension:
exten = s,1,Wait(1)
exten = s,2,Answer
exten = s,3,DigitTimeout,3
exten =
1. 8.2.6.1 Sending a Provisional Response says that UASs SHOULD NOT issue
a provisional response to non-INVITE requests.
From my message yesterday * appears to be sending a SIP/2.0 100 Trying to
X-Lite's REGISTER request before sending the SIP/2.0 200 OK message.
Is this correct?
[EMAIL PROTECTED] (Mark Spencer) writes:
Is this correct?
I see the 100 Trying on REGISTER frequently, but if it's not valid, we can
take it out. It serves no really effective purpose.
I think that it's only on REGISTER messaegs that it shouldn't be
used. Perhaps previous RFCs didn't
Is there a way to configure voicemail to do reminder paging? I would like
to configure some voicemail boxes to send an e-mail message to a pager
every 10 minutes until the message is retrieved.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
3. I have another SIP account (sip:[EMAIL PROTECTED]) which I would like
to use within asterisk both for dialing out and for receiving calls.
I see that sip.conf has a line
register = [EMAIL PROTECTED]/1234
where 1234 is the local asterisk extension. From chan_sip.c, line 1390
Are there any self described or otherwise gnophone experts out there?
Maybe I'm a stickler for pain but I'm having a lot of gnophone problems
and conventional wisdom tells me that they should be able to be figured
out. However, after reading all the documentation, reviewing the files, I am left
Mark has fixed the REGISTER issues to be more RFC compliant. I've
created a new thread so that those of you who got bored with the old
thread might read this new one. The feature that has just been added
was added a while ago, but now it actually seems to _work_. :-)
If you have a SIP
Trying to
configure voicemail with H.323 all I get is the following errorswhen I call
123, 666, 665, 664 or 031. I'm a newbie at this so, I think itmight be a
simple fix.[chan_oh323.so] = (OpenH323 Channel
Driver) == Parsing '/etc/asterisk/oh323.conf': Found
0:00.004
OpenH323 Wrapper
You have to modify the sourcer code yourself.
- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, June 15, 2003 6:20 PM
Subject: [Asterisk-Users] VoicemailMain
Hello guys
Is there anyway for me to change the sounds that are presented in
VoicemailMain?
On Sunday 15 June 2003 20:20, [EMAIL PROTECTED] wrote:
Hello guys
Is there anyway for me to change the sounds that are presented in
VoicemailMain? For instance, instead of it saying mailbox, I would
like it to say something like please enter your mailbox number now.
Is there a way for me to
Yes, have a look in /etc/asterisk/manager.conf
Regards,
Shaun
- Original Message -
From: Alvaro Parres [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, June 16, 2003 1:41 PM
Subject: [Asterisk-Users] GASTMAN AUTH QUESTION
Hi,
Any of you know where to define the user and
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