CS == Curt Shaffer [EMAIL PROTECTED] writes:
CS And to respond to Alex, The box is only doing Asterisk. 2.8Ghz
CS proc with 1GB of RAM. The iptables is on the server itself.
Please don't top-post.
What about your OUTGOING chain, how does that look?
/Benny
Hi
I have a small question on CDR Database:
It's used by billing software no ?
he have only one structure of data or they have multi structure with
more information
logged ? sample: cdr simple and cdr_extended
thanks bye
___
--Bandwidth and
All,
Our users have a softphone client that supports the G723 Codec which we
want to use for bandwidth reasons, however we do not wish (or have the
funds) to license the codec on our Asterisk servers. We have G723
pass-through working between the clients just fine, however calls fail
when
You need curl-devel just try
yum install curl-devel
Damon Estep wrote:
On version
1.2.12.1 running on FC4 with
curl.i386 installed the asterisk CURL function is not registered,
perhaps in
need something else (curl-devel.i386 ?)
All,
Has anybody come across a problem with the MS-GSM codec on Asterisk? We
have a client for Windows XP that uses the MS-GSM codec. While the
codec seems to work when talking directly to Asterisk, when 2 of these
clients talk to each other through Asterisk the sound is all mangled on
I am not sure if i understood what you mean but yes asterisk cdr's can be
used for billing with some modifications of your own. Asterisk can make cdr
in csv,mysql,postgresql with complete call
info which can be used for billing system's .
On 19/11/06, Noc Phibee [EMAIL PROTECTED] wrote:
Hi
I
The CDR could be used by billing software not all billing soultions do
there account that way.
he have only one structure of data or they have multi structure with
more information
logged ? sample: cdr simple and cdr_extended
I am not sure what you are asking. You can log just about anything
Hi all,
My client has an E1r2 connection (10 channels), what Digium card do I need?
Thanks.
Lito
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Sorry, i mean 30 channels.
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On Fri, Nov 17, 2006 at 09:13:31AM +1300, Tim Uckun wrote:
Pre 1.2.8, chan_sip would try to get the lock indefinitely, and would
basically hang the thread until the lock was fetched. Since 1.2.8, it
gives up after 100 tries and logs that message.
Of course that does not explain why the lock
Thank you, you are correct. You need curl, curl-devel, and you have to
make asterisk again after installing the required packages.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Marnus van
Niekerk
Sent: Sunday, November 19, 2006 4:15 AM
To:
What happens if in your sip.conf you set
disallow=all
allow=g723,gsm
And then allow both codec in the phone?
On 11/19/06, Ray Jackson [EMAIL PROTECTED] wrote:
All,
Our users have a softphone client that supports the G723 Codec which we
want to use for bandwidth reasons, however we do not
I am trying to setup an interactive menu where a caller hits the main
menu and can then dial an extension. As far as I can tell the
Waitexten app is failing to read 3 digits and just reading the first
and then announcing that it is invalid since all extensions are 3 digits.
How do I make
Keep reading. The person that actually does the calling needs to be
registered. You can't provide the list to others either.
Kevin
Don Fanning wrote:
Oddly enough, there's really nothing stopping one from doing so in the
material I just scan through at:
Thanks for looking into this further Kevin.
I guess this knocks a 'formal' asterisk asp sharing agreement on the head.
I can understand why they have done this but also sucks for people installing
asterisk using this.
At least the formal data sets are documented so a module for lookup prior to
Hi all!
Can anyone shed some light on a problem with a php to .call file. I am
trying to input the number I would like to call, as well as the current
phone number where im located. I am able to generate a call to each number,
but I am unable to hear or say anything on either phone.any ideas
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Simon Tennant wrote:
I am trying to setup an interactive menu where a caller hits the main
menu and can then dial an extension. As far as I can tell the
Waitexten app is failing to read 3 digits and just reading the first
and then announcing that
Simon Tennant wrote:
[internal-extensions]
exten = 100,1,Goto(mainmenu,s,10)
You can't start at 10 on your menu, you have to start with 1.
Doug
-- Ben Franklin quote: Those who would give up Essential Liberty to
purchase a little Temporary Safety, deserve neither Liberty nor Safety.
Doug Lytle wrote:
Simon Tennant wrote:
[internal-extensions]
exten = 100,1,Goto(mainmenu,s,10)
You can't start at 10 on your menu, you have to start with 1.
Let me clarify a little bit, you can start at 10, if 1-9 exist.
Doug
-- Ben Franklin quote: Those who would give up Essential
This part below did say you can be a 3rd party.
If the
telemarketer is accessing the registry on
behalf of other sellers or telemarketers,
that telemarketer also must identify
each of the other sellers or telemarketers
on whose behalf it is accessing the
registry, and it must certify,
I have a request into their operations @ the FTC asking for developer access
to write a module based on their data. We'll see...
-Original Message-
From: Dean Collins [mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
]
Sent: Sunday, November 19, 2006 9:18 AM
To: Asterisk Users
On 11/18/06, Pedro Silva [EMAIL PROTECTED] wrote:
I also restarted the box and the problem is not solved :(
PS
Pedro,
I had the same problem on a test box recently. I fixed it by upgrading
the version
of FreePBX to 2.1.3.
Mike
2006/11/18, Dumpolid Exeplish [EMAIL PROTECTED]:
i also used
Doug Lytle wrote:
Doug Lytle wrote:
Simon Tennant wrote:
[internal-extensions]
exten = 100,1,Goto(mainmenu,s,10)
You can't start at 10 on your menu, you have to start with 1.
strange - I jumped into that context at 10 and numbered up from 10 - I
thought that was ok.
Also when I started
http://activa.sourceforge.net/ does this.
- Original Message -
From: Ondrej Valousek
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Tuesday, November 14, 2006 12:28 AM
Subject: [asterisk-users] Desktop integration
Hi all,
I am interested in
SugarCRM also integrates with Asterisk.
kjcsb wrote:
http://activa.sourceforge.net/ does this.
- Original Message -
*From:* Ondrej Valousek mailto:[EMAIL PROTECTED]
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
mailto:asterisk-users@lists.digium.com
On 12:46, Sun 19 Nov 06, Joe Dennick wrote:
SugarCRM also integrates with Asterisk.
I keep hearing this but here in .nl noone can give me any
demo or installation with it.
advertisement mode
Covide (http://www.covide.net) on the other hand has it up
and running and you can find everything you
Hi Lito, as good?
If need to necessary a Digium Wild Card TE110P and the libraries to protocol
mfc/r2. I like libraries developed for Steve Underwood, where it places for
download in the site www.soft-switch.org.
I hope this help
Regards
Josué
2006/11/19, Lito Lampitoc [EMAIL PROTECTED]:
Hi Andrew,
All that happens is Asterisk sets up the call using G723 as the codec
and calls then fail due to transcoding problems (since it cannot
encode/decode g723). The softphone always prefers g723 over gsm, so if
you allow both in sip.conf Asterisk always select g723 due to the
In article [EMAIL PROTECTED],
Matthew Rubenstein [EMAIL PROTECTED] wrote:
How do I set up an existing call to dial out to a new terminal which is
included in a conference with the two existing legs of the call? When
the dialplan executes the Dial(terminal) command, control does not
[EMAIL PROTECTED] has it built-in. If you worship the Google gods, you'll
find documents describing how to integrate stand-alone installations of
SugarCRM and Asterisk...
Michiel van Baak wrote:
On 12:46, Sun 19 Nov 06, Joe Dennick wrote:
SugarCRM also integrates with Asterisk.
Have a look at
http://www.monetra.com/~brad/*callerid_shell*.agi
Im sure it would not be to hard to add another 'service to the script.
On 11/17/06, Damon Estep [EMAIL PROTECTED] wrote:
What would be the simplest way to retrieve information form a CNAM
database that provides http based
Makes more sense now!
Any way of setting an Asterisk CURL timeout in case of DNS, remote
server, or WAN link failure?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Russ
Beaupre
Sent: Friday, November 17, 2006 3:15 PM
To: Asterisk Users
I am trying to restrict calls to a Do Not Call List. These numbers are in a
database of over 90 million records. I would like to know if anyone has
already worked on this. Perhaps we can call a query from a script every time
we make a call, and return a value that tells us if this number is
We have an AGI app which queries a SQL database, matching the phone number
and checking flag on whether to accept/refuse call.
Go to www.generationd.com for app smartCID.
MD
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ed Nuñez
Sent: Friday,
Yes. I will take a look at it. Thanks for the suggestion
Lito
On 11/20/06, Josué Conti [EMAIL PROTECTED] wrote:
Hi Lito, as good?
If need to necessary a Digium Wild Card TE110P and the libraries to
protocol mfc/r2. I like libraries developed for Steve Underwood, where it
places for download
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