if i have this setup:
[sip users] -- [asterisk] --- [as5300] --- [pstn]
asterisk will talk to as5300 using sip. i will use as5300 as a trunk on the
asterisk so sip users can call out to pstn.
what i would like to is do prepaid on those trunks, not on the sip users. sip
users can call any
hi i been installing asterisk on a new pc...i donwloaded asterisk-1.6.0-beta7 ,
zaptel-1.4.10 and libpri-1.6.0-beta1 everthing install and configure perfectly
but zaptel gives me errors adn when i reboot my system it says that no devices
are configured...i have a sangoma a200 card
Thanks for the hint Patrick I appreciate it.
On Tue, Apr 22, 2008 at 3:02 PM, Rob Hillis [EMAIL PROTECTED] wrote:
Using _. is going to result in warnings. A much better practice is to
use _X.
Ali Jawad wrote:
Thx again patrick it worked, I used
[google-in]
exten =
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of gilbert
saunders
Sent: 23 April 2008 08:18 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] zaptel
hi i been installing asterisk on a new pc...i donwloaded
asterisk-1.6.0-beta7 , zaptel-1.4.10 and
Hi list,
We've been using 2b.co.hk to provide a Hong Kong SIP DID and it's been
working for a few months.
However, recently we've found that DTMF is no longer working and any
menu selections entered by a caller are not detected.
We have been using g729 and dtmfmode=rfc2833. I tried changing
In article [EMAIL PROTECTED],
The Asterisk Development Team [EMAIL PROTECTED] wrote:
The Asterisk development team has released versions 1.2.28, 1.4.19.1, and
1.6.0-beta8.
All of these releases contain a security patch for the vulnerability described
in the AST-2008-006 security advisory.
On Wed, Apr 23, 2008 at 09:19:26AM +, Tony Mountifield wrote:
In article [EMAIL PROTECTED],
The Asterisk Development Team [EMAIL PROTECTED] wrote:
The Asterisk development team has released versions 1.2.28, 1.4.19.1, and
1.6.0-beta8.
All of these releases contain a security patch
2008/4/22 Tzafrir Cohen [EMAIL PROTECTED]:
[snip]
A different approach:
[company-base](!)
; common settings
[company-A](company-base)
; specific for company A
[company-B](company-base)
; specific for company B
[company-C](company-base)
; specific for company C
Keep in
Hello everybody.
I was looking for the solution but nothing found. I have this in my
extensions.conf:
exten = 233,1,SetAccount(queue1)
exten = 233,2,Queue(queue1|rn)
exten = 233,3,NoOp(${QUEUESTATUS})
exten = 233,4,NoOp(${DIALSTATUS})
But when the call is placed in the queue and somebody
I've got the next AEL:
TEST=${X-CALLID};
NoOp(${TEST});
where X-CALLID=ctprueba-1208953210.12 (passed by a custom sip header)
Executing I've got the error:
[2008-04-23 13:24:01] WARNING[15638]: ast_expr2.y:742 op_minus: non-numeric
argument
-- Executing [EMAIL
On Wed, 2008-04-23 at 01:06 -0400, Matt Watson wrote:
I can;t imagine what headaches you'd have going from 1.4.11 to 1.4.19.1...
that is a minor version upgrade... no real change in functionality
Yeah, that's the theory anyway. :-)
thats basically 8 versions of bug fixes...
And what
On Tue, 2008-04-22 at 22:59 -0700, Nhadie Ramos wrote:
i want to create a billing system to monitor only the trunks and also
to load amounts on those trunks. is this possible? will i be able to
use app_prepaid for this?
TBH, I don't really understand your description, but I will say that I
Eric Dantie schrieb:
I've got the next AEL:
TEST=${X-CALLID};
NoOp(${TEST});
where X-CALLID=ctprueba-1208953210.12 (passed by a custom sip header)
Executing I've got the error:
[2008-04-23 13:24:01] WARNING[15638]: ast_expr2.y:742 op_minus: non-numeric
argument
Philipp Kempgen schrieb:
Eric Dantie schrieb:
I've got the next AEL:
TEST=${X-CALLID};
NoOp(${TEST});
where X-CALLID=ctprueba-1208953210.12 (passed by a custom sip header)
Executing I've got the error:
[2008-04-23 13:24:01] WARNING[15638]: ast_expr2.y:742 op_minus:
Sorry, bad expressed, what I want to know is how can I do this in AEL:
I've already got a variable X-CALLID with the content ctprueba-123456789.12
How can I copy the content X-CALLID to the new variable TEST?
something like TEST=${X-CALLID};
(The problem comes because of the operator minus).
On Wednesday 23 April 2008 08:37, Eric Dantie wrote:
Sorry, bad expressed, what I want to know is how can I do this in AEL:
I've already got a variable X-CALLID with the content ctprueba-123456789.12
How can I copy the content X-CALLID to the new variable TEST?
something like
On Tuesday 22 April 2008 19:34, Brian J. Murrell wrote:
On Tue, 2008-04-22 at 17:58 -0500, Security Officer wrote:
Asterisk Project Security Advisory - AST-2008-006
So given that I'm new to asterisk's svn and bug tracking tool, is it
sufficient then to apply the two patches
Try TEST=${X-CALLID}; and see how you go.
Eric Dantie wrote:
Sorry, bad expressed, what I want to know is how can I do this in AEL:
I've already got a variable X-CALLID with the
content ctprueba-123456789.12
How can I copy the content X-CALLID to the new variable TEST?
something like
Eric Dantie schrieb:
Sorry, bad expressed, what I want to know is how can I do this in AEL:
I've already got a variable X-CALLID with the content ctprueba-123456789.12
How can I copy the content X-CALLID to the new variable TEST?
something like TEST=${X-CALLID};
(The problem comes
For information purposes:
We had problems with proper dtmf recognition.
Asterisk Version: SVN-branch-1.4-r114083
SNOM FW: 7.1.30
With the above constellation dtmf tones sometimes worked and sometimes not.
Already a few hours and much testings later, we found and used the
configuration parm
Hello list,
I would like to know if anyone could suggest a possible solution to
getting the ZapRAS command to execute properly. It seems that even with
suid-root on the pppd binary that the spawned process is still not
allowed to load the plugin. When I place to option within
/etc/ppp/options
22 apr 2008 kl. 14.41 skrev Aadilkhan Maniyar:
Hi,
We have a scenario wherein the endpoint needs to send a 600 Busy
Everywhere after receiving an INVITE. I am using SIPp as this end
point. SIPp is configured as UE2.
Now when UE1 calls UE2 (SIPp) receives the INVITE and responds with
22 apr 2008 kl. 13.56 skrev Atis Lezdins:
Hi,
I experience my log flooded with warning messages like this:
[Apr 14 01:30:24] WARNING[19514] chan_sip.c: Remote host can't match
request NOTIFY to call
'[EMAIL PROTECTED]'. Giving up
I traced this down to point when we added to sip.conf
Carles Pina i Estany wrote:
Hello,
We have an Asterisk server with a TE410P Quad-Span togglable E1/T1/J1
card, 3 SPANs configured and OK and one SPAN unconfigured.
In our tests it works fine, but when it has a big laod of calls (say,
from 40 to 60) we have quality problems: some calls has
On Wed, 23 Apr 2008, Steve Davies wrote:
Wow! That took some finding, as it is little more than a footnote
(page 115-116 of Asterisk: The Future of Telephony) but is a
fantastic feature...
and always in that order? Is this feature well used and well tested???
If you are referring to
Hello, all!
According to http://bugs.digium.com/view.php?id=4909, spool
functionality was implemented in cdr_pgsql driver to be connection
failure-resilient back in '05.
This patch seems to have been merged in 1.2 branch, but in my 1.4.19
vanilla source, the code is absent from
In article [EMAIL PROTECTED],
[EMAIL PROTECTED] wrote:
Hello everybody.
I was looking for the solution but nothing found. I have this in my
extensions.conf:
exten = 233,1,SetAccount(queue1)
exten = 233,2,Queue(queue1|rn)
exten = 233,3,NoOp(${QUEUESTATUS})
exten =
Tilghman Lesher wrote:
On Tuesday 22 April 2008 05:22, Sergey Shumeyko wrote:
I have following problem with my Asterisk installation (version 1.6.0. beta
7.1). I want to assign start record conversation to #7 and stop record
conversation to #8, but it isn't working (on previous Asterisk
2008/4/23 Steve Edwards [EMAIL PROTECTED]:
[big snip]
Steve,
Fantastic examples. Many thanks for the feedback :)
Regards,
Steve
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or
best solution would be to return quad card buy 4 single port cards put 4
servers instead of one ... but i guess this is only possible it you had a
time machine ...
On Wed, Apr 23, 2008 at 2:44 PM, Matthew Fredrickson [EMAIL PROTECTED]
wrote:
Carles Pina i Estany wrote:
Hello,
We have
Thank you for your answer.
But the Dial command has a option 'g' which means that after succes will
proceed next priorities in the dialplan. Is there something also for
Queue() because according to manual there is no option for it. So I am
looking for some other solution.
Andy
Tony
Queue will continue if called person hangs up (and there's no option).
If caller hangs up, call goes to h extension in same context. Just the
same way as Dial with 'g'. There's a change in 1.6 that allows called
channel to continue if caller hangs up, so probably something like
this could be
Hi, sorry to confused you with my question.
the normal prepaid application like astcc, if i'm not mistaken, monitors the
amount left on the user (which i usually refer as extension), what i want to do
is monitor prepaid on the trunk (or the SIP channel use to call outbound to
pstn). Is that
Strangely, working...
TEST should be ctprueba-123456789.12 but not ctprueba-123456789.12
But got the value.
Thanks.
- Original Message -
Try TEST=${X-CALLID}; and see how you go.
Eric Dantie wrote:
Sorry, bad expressed, what I want to know is how can I do this in AEL:
I've
Why would you go to the trouble of writing a PERL AGI and take the
Performance Hit of using AGI as opposed to using the built-in MYSQL
from the dial plan ?
Mike Trest - On Travel wrote:
Hi,
I suggest you look at writing a PERL agi program to handle all of
the MYSQL / DB
access and just
Carles Pina i Estany wrote:
We have an Asterisk server with a TE410P Quad-Span togglable E1/T1/J1
card, 3 SPANs configured and OK and one SPAN unconfigured.
In our tests it works fine, but when it has a big laod of calls (say,
from 40 to 60) we have quality problems: some calls has the
Why would you want a channel to continue after the caller has hung up.
I clearly am missing something here because I can't see what good that
would be. What do people do with this Continued Channel ?
What is is used for ? How Does having it help you ? ???
Atis Lezdins wrote:
Queue will continue
i have HEARD asterisk wasn't made with the idea to run on multi-core
processors in mind .. the result was that it uses one core all the time ..so
one single P4 3.4 GHZ would perform better than a far more newser quad one.
but i might be wrong. but one thing for sure check hardware compatibility
On Wed, 2008-04-23 at 09:38 -0700, Nhadie Ramos wrote:
Hi, sorry to confused you with my question.
the normal prepaid application like astcc, if i'm not mistaken, monitors the
amount left on the user (which i usually refer as extension), what i want to
do is monitor prepaid on the trunk
I want to log in database some info ( total agents logged in, busy
agents, time ... ). I have some variables and checking them.
Let me explain it from beginning:
Somebody call the queue and everyone is busy, i need to play to caller
that everyone is busy and he should call later, and log this
Nope..
asterisk*CLI dundi lookup [EMAIL PROTECTED]
1. 0 SIP/priv:[EMAIL PROTECTED]/400 (EXISTS)
from 00:1e:0b:dd:e9:99, expires in 5 s
DUNDi lookup completed in 104 ms
-- Executing [EMAIL PROTECTED]:1] Set(SIP/156-08274b60,
CDR(accountcode)=wth) in new stack
-- Executing
Am I correct in thinking that one application of this would be
monitoring what you have left for funds with a prepaid vendor?
Darren Wiebe
[EMAIL PROTECTED]
Brian J. Murrell wrote:
On Wed, 2008-04-23 at 09:38 -0700, Nhadie Ramos wrote:
Hi, sorry to confused you with my question.
the
On Tue, Apr 22, 2008 at 7:10 AM, Carles Pina i Estany [EMAIL PROTECTED] wrote:
Hello,
We have an Asterisk server with a TE410P Quad-Span togglable E1/T1/J1
card, 3 SPANs configured and OK and one SPAN unconfigured.
In our tests it works fine, but when it has a big laod of calls (say,
At 04:13 PM 4/22/2008, you wrote:
I'm presently working on a project to build a scheduling system
accessible by both web and phone. on the web side one can query what
items are available when by using the time or the item as a key then
reserve for an available time slot. reservations may
Just curious, are you recording these calls because that is around the
I/O threshold for audio issues when recording all calls.
100% right, all recording should (its a must actually) be done in ram drive
then copied to disk later. an asterisk server that do recording should have
enough ram to
On Wed, 23 Apr 2008, linuxian iandsd wrote:
i have HEARD asterisk wasn't made with the idea to run on multi-core
processors in mind .. the result was that it uses one core all the time ..so
one single P4 3.4 GHZ would perform better than a far more newser quad one.
but i might be wrong. but
On Wed, Apr 23, 2008 at 11:07:01AM -0700, Steve Edwards wrote:
At 04:13 PM 4/22/2008, you wrote:
I'm presently working on a project to build a scheduling system
accessible by both web and phone. on the web side one can query what
items are available when by using the time or the item as
On Wed, Apr 23, 2008 at 2:07 PM, linuxian iandsd [EMAIL PROTECTED] wrote:
Just curious, are you recording these calls because that is around the
I/O threshold for audio issues when recording all calls.
100% right, all recording should (its a must actually) be done in ram drive
then copied
Al Baker napsal(a):
Why would you want a channel to continue after the caller has hung up.
I clearly am missing something here because I can't see what good that
would be. What do people do with this Continued Channel ?
What is is used for ? How Does having it help you ? ???
On Wed, 23 Apr
top says asterisk 1.2.25 is using multiple cores:
Cpu0 : 2.7% us, 9.3% sy, 0.0% ni, 87.7% id, 0.0% wa, 0.3% hi, 0.0%
si
Cpu1 : 1.7% us, 4.0% sy, 0.0% ni, 94.3% id, 0.0% wa, 0.0% hi, 0.0%
si
Cpu2 : 1.3% us, 4.3% sy, 0.0% ni, 94.3% id, 0.0% wa, 0.0% hi, 0.0%
si
Cpu3 :
I advise using different servers for
different tasks (with redundancy obviously).
i would really appreciate it if you gave me some hints about making
recording run on another server.
___
-- Bandwidth and Colocation Provided by
linuxian iandsd wrote:
i have HEARD asterisk wasn't made with the idea to run on multi-core
processors in mind .. the result was that it uses one core all the time ..so
one single P4 3.4 GHZ would perform better than a far more newser quad one.
but i might be wrong. but one thing for sure
May be I'm wrong but:*
timeout - the maximum time, in seconds, the call will wait in the queue.
When this time expires, the next extension, by priority, will be executed.
By default the timeout is set to 300 seconds.
So you clearly have two ways to feed your database with your statistics:
If
On Wed, Apr 23, 2008 at 11:07:01AM -0700, Steve Edwards wrote:
AGIs do not have a substantial performance hit and I think people need
to get this misconception out of their heads.
Writing AGIs in a scripting, non-compiled language may be great for
prototyping and proving concepts where
At 02:58 PM 4/23/2008, you wrote:
Re performance hit.
I actually re-wrote one of my frequently used AGI in C and even set
the STICKEY-BIT to avoid reloading the static text portions.
I noticed slightly lower disk activity level (but the perl file was
probably in memory cache too). So keeping
Asterisk was indeed written with the intention to run on
multi-core systems, and should utilize extra cores just fine.
Matthew Fredrickson
Software/Firmware Engineer
Digium, Inc.
well, i guess i was wrong ... or maybe i had outdated information
On Wed, Apr 23, 2008 at 03:29:47PM -0400, Mike Trest - On Travel wrote:
At 02:58 PM 4/23/2008, you wrote:
Re performance hit.
I actually re-wrote one of my frequently used AGI in C and even set
the STICKEY-BIT to avoid reloading the static text portions.
Doesn't Linux ignore the sticky
hi sir,
yes that would be it, but instead of having a prepaid provider, i will setup my
own as5300 and asterisk will talk to that. is that possible in astcc, astbill
or a2billing?
regards,
nhadie
Am I correct in thinking that one application of this would be
monitoring what you have left for
(As you know) I'm trying a X101P card without success; some of
the problems are:
On Fri, 18 Apr 2008 11:30:46 -0400
Steve Totaro [EMAIL PROTECTED] wrote:
Is this region specific?
I bought that card in 2006 and I didn't realize the difference:
how can I understand if my card is European or
I did not mean to stir up a hornet's nest or religious war :)
The BASIC QUESTION I was trying to ask is this...
Since the MYSQL add-on provides a way to interface with MySQL
what is it that one gains or is trying to gain by writing their OWN
AGI script to do the interface ?
The only
On Wed, Apr 23, 2008 at 10:31:27PM +0200, giuliano curti wrote:
(As you know) I'm trying a X101P card without success; some of
the problems are:
On Fri, 18 Apr 2008 11:30:46 -0400
Steve Totaro [EMAIL PROTECTED] wrote:
Is this region specific?
I bought that card in 2006 and I didn't
Ok, I'm not aware of this feature in astcc and I can't speak for astbill
or a2billing. I do know that I coded it into astpp and it's called
vendor rating in there. It works but it's not used a lot at present.
Darren Wiebe
[EMAIL PROTECTED]
Nhadie Ramos wrote:
hi sir,
yes that would be it,
Jeremy,
It is not the dip peer that is failing but the dial plan:
-- Goto (macro-dundi-lookup,400,1)
[Apr 23 12:46:44] WARNING[2269]: chan_sip.c:2898 create_addr: No such
host: 192.168.4.51/400
[Apr 23 12:46:44] WARNING[2269]: app_dial.c:1191 dial_exec_full:
Unable to create channel of type
I'm fairly sure SIP will never work unless I hard-code peers everywhere, which
isn't going to happen. The only reason I want to use it is for the call-limit
option.
Looking at sip channels there is no option to pass the extension after the IP,
it's always [EMAIL PROTECTED], or [EMAIL
Take a look at this setup, it does not use passwords on the sip peers
or the mappings in Dundi. As long as you inside your network this
maybe the way to go.
http://www.voip-info.org/wiki/view/DUNDi+Enterprise+Configuration+SIP+with+no+passwords
You could also look at the incominglimit and
On Thu, 24 Apr 2008 00:13:56 +0300
Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Wed, Apr 23, 2008 at 10:31:27PM +0200, giuliano curti wrote:
(As you know) I'm trying a X101P card without success; some of
the problems are:
[cut]
yersteday I downloaded the zaptel-1.4.0, patched with oslec-0.1
On Wed, 2008-04-23 at 08:52 -0500, Tilghman Lesher wrote:
Please understand that that's NOT the only security fix that has gone in
during that time. If this is the only thing that you fix, you're likely to be
vulnerable on several other levels. See our full list of security disclosures
at
On Wed, 23 Apr 2008, Al Baker wrote:
The BASIC QUESTION I was trying to ask is this...
Since the MYSQL add-on provides a way to interface with MySQL
what is it that one gains or is trying to gain by writing their OWN
AGI script to do the interface ?
I like doing serious work in an
On Wed, 2008-04-23 at 15:41 -0600, Darren Wiebe wrote:
Ok, I'm not aware of this feature in astcc
Keep in mind that astcc is simply a tool that keeps a database of
minutes used for some entity (typically a calling card) and calculates
those minutes used against a pre-charged amount. The number
thank you sir, i will try to check on that. i haven't really tried astcc yet so
i really dont understand how it works right now.
also, do you have any reference on using app_prepaid? can't find some sample
config, i would like to see how i can use that. do you think app_prepaid is
suited for
Thx so much for taking the time to share.
Damn Insightful Damn Helpful
THANKS!
Steve Edwards wrote:
On Wed, 23 Apr 2008, Al Baker wrote:
The BASIC QUESTION I was trying to ask is this...
Since the MYSQL add-on provides a way to interface with MySQL
what is it that one gains or is
71 matches
Mail list logo
