This is the right suggestion:
Run something like the following:
[h...@mouth tmp]# echo this is a test | newhire.php
If the script runs, check your maillog (/var/log/maillog) to see if
there's any evidence of what may have happened.
Geraint Lee wrote:
Check you can run the script from th
I would like to do the following:
Dial an extension in Asterisk
The extension runs an application which dials a number (like a hybrid of
DIAL and READ). The dialed number is a box which does nothing but play
DTMF tones then hangs up
The first box captures the DTMF tones to a variable
Dial
Steve Edwards wrote:
On Fri, 27 Feb 2009, James Sneeringer wrote:
If you can get the outgoing directory (or a reaonable parent) on its own
mountable partition or volume, you could accomplish this with disk
quotas. It won't control how many Asterisk processes at once (does it
even handle
There are sites which will show you the ratecenter and the company.
Try the following to see all NPANXX's in that ratecenter:
http://www.telcodata.us/telcodata/ratecenter?ratecenter=SCRM%20MAINstate=CA
To find the ratecenter:
http://www.telcodata.us/telcodata/telco?npa=916exchange=854
Try 01
Broadvox
Nhadie wrote:
Hi,
Anyone knows a DID provider that can do both outbound and inbound?
Regards
Nhadie
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update
I use them both; my legacy dialplan is all .conf and new stuff is .ael.
I find AEL to be the better option when jumping around, but that's
just my opinion.
Mik
Alan Lord (News) wrote:
Hi all,
I built my first asterisk using the traditional (?) .conf files and
constructs.
I recall
This is a good point. Knowing the outcome is important, and hopefully
it will get more people to post once the problem has been resolved.
David fire wrote:
one very important thing that will help to improve the list is when
someone solve the problem put an email [solved]original subject
It seems to me that everything one may want to know would be contained
on voip-info.org
People don't ask stupid questions because of a lack of a FAQ to read,
they ask stupid questions because they're too lazy do to the footwork.
Robert Broyles wrote:
I think we'd be better off posting a
If you find something on a WIKI that is outdated, guess what you have an
opportunity to do . . .
Noah Miller wrote:
It seems to me that everything one may want to know would be contained
on voip-info.org
Hmm. Dangerous statement. There are many things on the WIKI that are
quite
Don't over think this, guys. Again, the point of having a WIKI is to
allow for customization. A landing page for Asterisk documentation
within voip-info.org is all you need, not a whole new source of
documentation.
Jai Rangi wrote:
**
I understand. As someone else already
What you should do, assuming that each DNS request is invalid and
returns nothing, is add a fake domain on your box that all of these
requests will point to. That is, if mydomain.com is the DNS name it's
looking up, add mydomain.com to a named server on the same box. Make
sure you include
I had done something similar in the past, and I have one suggestion that
may be helpful; the call files are set to be checked every second, which
can bottleneck the system a bit. You can modify the code (pbx_spool.c)
to check in fractions of a second, which should keep the calls more
fluid.
Aby,
Assuming you're building Asterisk from source, you can change the
following in the scan_thread function:
change-
sleep(1);
to-
/*sleep(1);*/
usleep(10);
This will change the delay from 1 second to 10 microseconds (0.1
second).
There's actually a document included with the source code which will
take you through setting up an agent callback system. You can find it
in 'doc/queues-with-callback-members.txt'.
The 'AgentCallBackLogin' application has some issues, and since you can
do the same thing with your dialplan,
Require that the user is logged in, and that the form has random
text-image verification. Just my 2¢.
Chris Earle wrote:
Hi all,
I'm writing some code to do a web 'click to dial' sort of thing. Where the
surfer puts in their number and some php/asterisk API code Originates a call
out to
I'm finding that my Asterisk server is bombarding my DNS servers with
lookups like the following:
Queries
5060-b7bfce38: type A, class IN
Name: 5060-b7bfce38
Type: A (Host address)
Class: IN (0x0001)
One call alone has a handful of requests
couldn't figure it out..any ideas ??
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mik Cheez
Sent: Tuesday, June 03, 2008 8:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 911 via MAX TNT
PROTECTED]:1] Set(SIP/NPANXX7604-08c46518,
CDR(userfield)=) in new stack
Extension Changed NXX5557604 new state Idle for Notify User NXX5557555
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mik Cheez
Sent: Wednesday, June 04, 2008 11:39 AM
Something is certainly wrong here. Can you check the netmask? I assume
you're just using a full class C, which would make the netmask
255.255.255.0. Also, should I assume the trace is being done on the
Asterisk machine?
Try running a trace on all interfaces for port 5060:
]# tethereal -i
Tim Guy wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric
Wieling
Sent: 07 May 2008 20:14
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Out-Going Calleriid
The leading 0 is not part of
Have you tried kernel-smp-devel?
Andreas van dem Helge wrote:
Is Zaptel 1.4.10 compatible with RHEL 3 (2.4.21-53.ELsmp)? Because I
can compile 1.2.20.1 just fine but 1.4 says:
echo You do not appear to have the sources for the 2.4.21-53.ELsmp
kernel installed.
You do not appear to have the
-2.4.21-53.EL.i686
kernel-smp-unsupported-2.4.21-53.EL.athlon
kernel-smp-unsupported-2.4.21-53.EL.i686
kernel-source-2.4.21-53.EL.i386
kernel-unsupported-2.4.21-53.EL.athlon
kernel-unsupported-2.4.21-53.EL.i686
kernel-utils-2.4-8.37.15.i386
On Wed, Apr 30, 2008 at 2:47 PM, Mik Cheez [EMAIL
Correct me if I'm wrong, but if you run asterisk as a service this
happens. There is/was some dispute as to the fallacy of using
'safe_asterisk' anyway.
Start it at the command line to see the pretty colors.
Mike wrote:
Hi,
I`ve just made a leap from * 1.2.7 to 1.4.19. It took a while
Openser has SIP-to-XMPP Gateway and JABBER IM and PRESENCE
interconnection modules. I used the XMPP module with Google Talk and
Asterisk a while back. It resulted in a segmentation fault, but again
that was a long time ago.
Eric Chamberlain wrote:
Hello,
I’m looking for a SIP to
In 'top', you can always look at what percentage of your CPU is idle.
Subtract that from 100 and you've got your load average.
Cpu(s): 1.1% us, 0.6% sy, 0.0% ni, *98.1% id*, 0.1% wa, 0.1% hi,
0.0% si
Erik Anderson wrote:
On 10/12/07, Philipp Kempgen [EMAIL PROTECTED] wrote:
I don't
Sorry...I should have been more specific in my original reply.
In 'top', you can always look at what percentage of your CPU is
idle. Subtract that from 100 and you've got your load average.
I should have said you get your average load percentage, rather than
just average load.
Mik Cheez
%si,0.0%st
Cpu1:0.0%us,0.0%sy,100.0%ni,0.0%id,0.0%wa,0.0%hi,0.0%si,0.0%st
Cpu2:0.0%us,0.3%sy,98.0%ni,0.0%id,0.0%wa,1.7%hi,0.0%si,0.0%st
Cpu3: 0.0%us,0.7%sy,99.3%ni,0.0%id,0.0%wa,0.0%hi,0.0%si,0.0%st
According to what you are saying my load average should be 100.
Thanks,
Steve
Mik Cheez
Is your mysql.sock actually in /var/lib/mysql/ ?
RENZZO SOTOMAYOR wrote:
Hi! I am proving Asterisk 1.2.24 in realtime with MySQL 5.0.27 using
Idefisk softphones. I followed the steps of how to of voip-org but
always have this error:
Sep 25 20:29:07 WARNING[12000]: res_config_mysql.c:360
You can use Asterisk's AGI and PHP/Perl or whatever else. You'll need
to install connecting software, such as FreeTDS, to connect to SQL.
Then you can either pass arguments to your script or use environment
variables to set Asterisk variables.
Here's a good place to start:
Did you confirm that the file exists?
/var/lib/asterisk/agi-bin/rabot.agi
Also, in your script (wherever it actually is), put a space between php
and -q
#!/usr/bin/php -q
Karim H wrote:
Hello,
I have succeded in compiling and configuring My TDM Card and asterisk,
all works fine.
But I
aka. the FTP site is back up
dave cantera wrote:
ed,
do you positively have to have 1.4.0?
just download 1.4.9 or 1.4.8... 1.4.0 is too old...
I can email you 1.4.8, 1.4.5, 1.4.9...
I just downloaded 1.4.9 from:
This would let you include/ignore a leading 1
1{0,1}[2-9]{2}[0-9]{8}
Brent Torrenga wrote:
Hola,
What would a valid regexp in Asterisk be to identify a NANP number, i.e.,
NXXNXX?
Sincerely,
Brent A. Torrenga
Torrenga Engineering, Inc.
907 Ridge Road
Munster, Indiana
My mistake...you're correct...should have tested it.
Mojo with Horan Company, LLC wrote:
But those are not REGEX expressions, those are asterisk dialplan
pattern-matching expressions. great for the X in:
exten = _X.,1,blah
but not for use with REGEX() function.
I think it would be
Use auto dial. You can have as many calls as you wish.
http://www.voip-info.org/wiki-Asterisk+auto-dial+out
Porier, Jeremy M. wrote:
Are there any scripts out there that would help me stress test two boxes
that are setup back to back with 4 PRI connections? We're having
problems with
Mani,
I've gotten the same result both dialing from a gtalk client to SIP, as
well as an SIP call to gtalk. You can run a jabber debug before the
call is placed to see more debug info on what's causing the crash. With
the module in Beta, I believe it's just a bug that needs to be worked
out.
I'm running sippeers and sipusers in my extconfig, and everything runs
perfectly when a client is registered (ex. registers to port 1000), but
when it re-registers the client is set to port 5060. This behavior does
not take place if I use the static files.
Both in my sip_buddies table for db,
That was an April Fools Day joke...unfortunately whoever started it
didn't think about what it would do to the Archive.
Justin Selleck wrote:
Is asterisk 2.0 real? Running in c#? I see references to it but
cannot find it anywhere.
Olle E. Johansson wrote:
Andrew Kohlsmith wrote:
On Wednesday 20 July 2005 20:15, Eric Wieling aka ManxPower wrote:
As I understand it, adding VAD/Silence would require redesigning the
entire RTP stack of Asterisk.
My understanding is that with the new jitter buffer both of
Why not use SIP?
list wrote:
Anybody using asterisk to talk to a lucent tnt gatekeeper via h323?
Any suggestions or recommendations about how I can get this working?
Any config examples?
thanks,
jon
___
Asterisk-Users mailing list
There is an open source version of the license:
http://www.readytechnology.co.uk/open/g729/INSTALL-041103.txt
You can view the licensing information at the following:
http://www.intel.com/software/products/ipp/
more details can be found on http://www.voip-info.org
Steven Critchfield wrote:
On Fri,
Thanks for the clarification. In that case the following should only be
considered for development.
Steven Critchfield wrote:
On Fri, 2005-02-25 at 09:25 -0800, Mik Cheez wrote:
There is an open source version of the license:
http://www.readytechnology.co.uk/open/g729/INSTALL-041103.txt
You
We're having a problem with Asterisk when we try to pass a call off to a
Lucent PSTN using SIP. This behavior does not exist with SER:
With Asterisk
An ISDN call is started, at the T1 level we receive call proceeding
and immediately we receive a Call in Progress just like the far end
party
42 matches
Mail list logo
