Abel Monzon wrote:
and then in my softphone I call to 1 the asterisk log say this:
-- Launched AGI Script /usr/local/share/asterisk/agi-bin/a2billing.php
== a2billing.php: Failed to execute
'/usr/local/share/asterisk/agi-bin/a2billing.php': No such file or directory
--
Hi
First off, you replied a previous mail to the list, and hence your
message appears as part of a previous thread. To post a new message
start a new message.
Also,
On Sun, Oct 26, 2008 at 01:47:03AM -0400, Abel Monzon wrote:
Hello is my idea or this is a bug? The thing is that I have in my
Also check the file permissions and if you are using a RedHat like OS, check
the SELinux.
And about using a2billing,I recommend you to use version 1.4.21 or less.
On Sun, Oct 26, 2008 at 3:30 AM, Tzafrir Cohen [EMAIL PROTECTED]wrote:
Hi
First off, you replied a previous mail to the list, and
On Sat, 25 Oct 2008, Joseph L. Casale wrote:
X100P.
Yeah I saw these but they are single port and I need at least 2 ports. I
only have 1 free pci slot as well.
OpenVox.
Gordon
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2008/10/24 Wilton Helm [EMAIL PROTECTED]
I've been following this thread and trying to sort out what is wanted,
what is available, and why. Comments to the following would be appreciated
and might be useful to others.
1. Why would anyone originate a FAX via VoIP? If it has to go through
Joseph wrote:
I'm using Linksys SPA3102 adapter and have a strange ring tone:
Long-Short-Short or Long-Long-Short-Short
Does anybody know which setting adjust this ring tone on SPA3102
Sipura rings normally. I'm not sure if setting are on Regional Tab or User Tab
Interestingly, I get
On Sat, 2008-10-25 at 11:54 -0600, Joseph L. Casale wrote:
I need to increase reliability at an office as SIP/Internet provider outages
are causing some issues.
What would be the least expensive analogue card that people are using
reliably?
If its for reliability, i wouldn't recommend
Olivier wrote:
2008/10/24 Wilton Helm [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
I've been following this thread and trying to sort out what is
wanted, what is available, and why. Comments to the following
would be appreciated and might be useful to others.
1.
Hi!
I just tried to call a friend using jingle, but I got refused. Errorcode was
502, he tried to call me, heard it ringing once and then it stopped.
I used:
originate jingle/gtalk_account/[EMAIL PROTECTED] [application]
I'm registered to googletalk, but this should mean no harm, or
I had the same problem and fixed it with this change:
Yes, under the Regional Tab, then under Ring and Call Waiting Tone Spec,
then under Ring Waveform:
I change it from: Trapezoid to Sinusiod.
Now the inbound calls to the FXS ring with a more US ring cadence.
Hope this helps.
eric84
On Sun,
Hi Julien,
Gtalk channels work with GoogleTalk clients. Empathy (based on the
Telepathy framework) has a Gtalk implementation that is reported to
work with Asterisk, too.
Jingle channels should work with other Jingle implementations, but
there are only a few of them around. One reason is that
Hello Philippe!
Do I need a googletalk client? Or can I just use asterisk's originate CLI
command? I was under the illusion I could. Otherwise it's a bit problematic. I
canonly use text-based applications and they better support JACK audio
Connection Kit, for my soundcard is not simple
The originate command should work. Make sure that the user you're
placing the Gtalk/Jingle call is in the buddy list and has Jingle
capabilities. The 'jabber show buddies' command will give you that
info.
Cheers!
Philippe
On Sun, Oct 26, 2008 at 3:57 PM, Julien Claassen [EMAIL PROTECTED] wrote:
Hi!
There's something strange. I have entered a couple of buddies. On has Jingle
capability and two have resources (Home and Telepathy), but my own account
does have no resource, I put myself in the buddies list. Is tat supposed to
be?
And again about those ports: Accept the 5222 port, do
OpenVox.
Gordon
Appreciate that pointer, those are fairly cheap!
Thanks,
jlc
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Steve Underwood [EMAIL PROTECTED] writes:
That list rather poorly supports your argument. The PAP2 and the PAP2T
do *not* support T.38, despite numerous arguments you'll find to the
contrary. Personally I believe Linksys, the manual, and the menus.
The manuals and the menus for PAP2T talk
Daniel Hazelbaker [EMAIL PROTECTED] writes:
I can answer both of those with a single point. We just switched
(entirely) to Asterisk a few weeks ago. We looked, very briefly, at
various ways to get rid of the physical, analog, fax machines. They all
ended with the answer People can't figure
Hi Julien,
On Sun, Oct 26, 2008 at 4:51 PM, Julien Claassen [EMAIL PROTECTED] wrote:
Hi!
There's something strange. I have entered a couple of buddies. On has Jingle
capability and two have resources (Home and Telepathy), but my own account
does have no resource, I put myself in the buddies
Steve Underwood [EMAIL PROTECTED] writes:
Even the big floor standing office MFPs typically only offer T.37 or
T.38 only through an expensive option card.
Medium MFP's almost all support T.37. They call it scan to email,
but they do it (as far as I can tell) in a way that is compliant with
In this application what are the pros and cons of using a multiport ata vs a
tdm400/800/2400?
Eric
On Sat, Oct 25, 2008 at 10:54 AM, Joseph L. Casale
[EMAIL PROTECTED] wrote:
I need to increase reliability at an office as SIP/Internet provider
outages are causing some issues.
What would be
Well, so asterisk seems to think, that I'm not connected, for I don't see a
resource Asterisk or Talk with my name.
That shouldn't really be. :-(
Any ideas on fixing this?
Kindest regards
Julien
Music was my first love and it will be my last (John Miles)
What influence the ring tone and patterns is the setting under Regional Tab
Ring1 Cadence: 60(2/4) (this is default setting)
and selections under User 1 Tab
Default Ring: 1
I was thinking about it yesterday Sinusoid vs. Trapezoid setting as all the
units I have several old Sipura unit and few
Strange, are you both connected to Google's XMPP server? Sometimes it
takes a little time before retrieving your roster on Gtalk. Does
Asterisk appear as connected on your friend's buddy list?
Also, what does the 'jabber show connected' say?
Cheers,
Philippe
On Sun, Oct 26, 2008 at 5:53 PM,
in multiport sipura/Linksys you cannot access individual ports you have to
address them by the group
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric Fort
Sent: Sunday, October 26, 2008 12:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
A previous issue has popped up and once again I'm out of ideas. During
the evenings it seems that the TDM channels will spike (dahdi_monitor)
and will refuse to listen for audio of any type, this includes DTMF.
The only resolution I know of is to stop Asterisk and restart the
dahdi service, but
Evening Philippe!
Here's what jabber show connected says:
Jabber Users and their status:
User: [EMAIL PROTECTED]/Talk - Connected
Number of users: 1
I'll have to ask my friends, what their clients say. Although I suppose as
my friend already send me a text message he
Sunday, October 26, 2008, 12:31:16 PM, Hans wrote:
On Sat, 2008-10-25 at 11:54 -0600, Joseph L. Casale wrote:
I need to increase reliability at an office as SIP/Internet provider outages
are causing some issues.
What would be the least expensive analogue card that people are using
Hello, I want to know if some body use Asterisk on Freebsd 7.0 release? My
problem is that, when I call to any extension and the asterisk need to
reproduced a file GSM o MP3, whatever, that have a lot of noise... Only not
have noise when the extensions is not avalaible. That happen only in
Thanks Matt,
I will check them.
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Date: Sat, 25 Oct 2008 14:28:27 -0400
Subject: Re: [asterisk-users] Fresh installed box
Hi Torintino,
1.
Login to FreePBX, Go to extensions, Select the extension you
want
On Sun, 26 Oct 2008, Eric Fort wrote:
In this application what are the pros and cons of using a multiport ata vs a
tdm400/800/2400?
For me it would be much easier to access the on-board TDM card and less
wiring/mains units. (power units)
Gordon
Eric
On Sat, Oct 25, 2008 at 10:54 AM,
On Sun, Oct 26, 2008 at 4:51 AM, Gordon Henderson
[EMAIL PROTECTED] wrote:
On Sat, 25 Oct 2008, Joseph L. Casale wrote:
X100P.
Yeah I saw these but they are single port and I need at least 2 ports. I
only have 1 free pci slot as well.
OpenVox.
Those look great, and on top of the price
On Fri, Oct 24, 2008 at 10:19 PM, Chris Walton [EMAIL PROTECTED] wrote:
The 3.1.0 firmware allows you to create up to 10 custom softkeys.
This is all documented in Polycom's SIP 3.1 Admin Guide.
Should I post some examples?
Which would be great, if Polycom weren't the Firmware-Nazis that they
On Fri, Oct 24, 2008 at 10:09 AM, Drew Gibson [EMAIL PROTECTED] wrote:
Can anyone clarify how SMS to non-mobile numbers are generally handled
in North America?
Is it possible to have SMS delivered direct to your landline DIDs? Then
have Asterisk relay it to the actual mobile DID.
When I send
If you buy your phone from a reputable place they will be able to provide the
firmware.
--Original Message--
From: Andrew Joakimsen
Sender:
To: Asterisk Users Mailing List - Non-Commercial Discussion
ReplyTo: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Benny Amorsen wrote:
Steve Underwood [EMAIL PROTECTED] writes:
That list rather poorly supports your argument. The PAP2 and the PAP2T
do *not* support T.38, despite numerous arguments you'll find to the
contrary. Personally I believe Linksys, the manual, and the menus.
The
Benny Amorsen wrote:
Steve Underwood [EMAIL PROTECTED] writes:
Even the big floor standing office MFPs typically only offer T.37 or
T.38 only through an expensive option card.
Medium MFP's almost all support T.37. They call it scan to email,
but they do it (as far as I can tell)
You could, under programming section 1.3.4 in the http interface to
configure the GW card enable DTMF Detection, that will enable Out of
Band DTMF. In the TDE they renamed this to DTMF signalling.
On Fri, Oct 24, 2008 at 2:42 PM, Richard Scobie [EMAIL PROTECTED] wrote:
Jonn R Taylor wrote:
Hello Asterisk-Users,
For some reason my CDR records for disposition and billsec are not working
correctly.
I always receive a 0 for billsec and the disposition is always at NO
ANSWER', even when I grab the calls.
I experience this with Asterisk 1.6.0.1 and Asterisk 1.4.22.
Here is
I have the same problem for Disposition when I use call files. The
duration is correct but the Disposition is always NO ANSWER. I also am
using 1.6.0.1. I did not have the problem when I was using 1.4.21
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Pedram M
Sent: Monday,
Hi all,
Yes, this might not be the proper list for this, but i have a question about
Asterisk and voice recognition.
If I want to create a menu system where the user can say different things in
the Swedish language what should I look at?
For example, i want the user to be able to say something
Other vendors, including Cisco, will provide the firmware directly. I
no longer deploy Polycom (unless someone really wants them) due to
this. Yes I can get it from the supplier but it takes a few days. I
would rather just go to Polycom.com and get the firmware when I want
to.
There is no excuse
Not sure about the Swedish, but Lumenvox has a great speech
recognition app for Asterisk.
- D
On 26 Oct 2008, at 19:53, Christian wrote:
Hi all,
Yes, this might not be the proper list for this, but i have a
question about Asterisk and voice recognition.
If I want to create a menu
Hi,
Many thanks for that info.
Is there any free solution available as well?
Many thanks,
Christian
On 2008-10-26 at 20:32 Darren Sessions wrote:
Not sure about the Swedish, but Lumenvox has a great speech
recognition app for Asterisk.
- D
On 26 Oct 2008, at 19:53, Christian wrote:
Hi
Hi All,
For some reason since moving to Asterisk 1.6. my CDR records are no
longer displaying the Clid field. The CDR records contain the Source
field be for some reason not the CID details. I am logging CDR to
mysql.
Is anyone able to help?
Regards
David.
Sphinx
http://cmusphinx.sourceforge.net/html/cmusphinx.php
Not sure how the implementation works with Asterisk but I know it's
been done (I'd google it).
- D
On 26 Oct 2008, at 20:55, Christian wrote:
Hi,
Many thanks for that info.
Is there any free solution available as well?
Many
On Sunday 26 October 2008 21:28:34 Andrew Joakimsen wrote:
Other vendors, including Cisco, will provide the firmware directly. I
no longer deploy Polycom (unless someone really wants them) due to
this. Yes I can get it from the supplier but it takes a few days. I
would rather just go to
C F wrote:
You could, under programming section 1.3.4 in the http interface to
configure the GW card enable DTMF Detection, that will enable Out of
Band DTMF. In the TDE they renamed this to DTMF signalling.
Believe me, I spent a great deal of time on this including Ethereal
captures and
Also posting this question to people working on manager interface and
dialers.
I have a simple auto dialing script (using Originate) that forwards all
incoming calls to a queue full of waiting agents instead of a meetme
conference room. I use queues rather than meetme so I can leave the
automatic
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