Here is a starting point:
http://www.voip-info.org/wiki/view/Asterisk+dimensioning
Not really what you need, but still. When you figure out something -
add here :-)
Has anyone put together a public list/wiki/info sheet on what the
various maximums/rules of thumb are? Seems a better idea than
On 28 Apr 2010, at 06:53, Aditya Kumar wrote:
exten = bob,1,Dial(SIP/${exte...@ext-sip,20)
Where did you define EXTERN?
S
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New to Asterisk? Join us
On Wed, Apr 28, 2010 at 07:12:57AM +0200, Claire Sinn wrote:
Hi,
a few month ago, I tried to install zaptel for my Beronet BN8S0 pci card... I
gave up and took hfcmulti/lcr. Now dahdi (2.2.1.1) seems to support the card
and I'm very interested to get it to work.
But how to get rid of
I have upgraded Asterisk from 1.4.22 to 1.4.30 and I have noticed I am
getting a lot of errors like this on the console :-
ERROR[23912]: utils.c:968 ast_carefulwrite: write() returned error:
Broken pipe
I have tracked it down to a perl AGI script which performs our own CDR
recording. It is
Hi guys:
i need to set an extension in my dialplan in which it divert calls if the
extension contain specific series ,For example :
I need to divert calls which contain to specific extension (contain ,not
start or end with), as i know i should set Gotoif command but i dont know what
to
Are talking about something like
exten = _..,1,Noop(Have in this extension)
There is also this function that can be used to look for sub strings inside a
string.
core show function REGEX
-= Info about function 'REGEX' =-
[Syntax]
REGEX(regular expression data)
[Synopsis]
Hello list,
using asterisk 1.4.25.1 and realtime queues.
I would like to use the parameter 'membermacro' so I've added a field in
my mysql-table queues, but this is not working.
Anyone knows how I can execute a macro when the queue is answered by a
queuemember ?? Also the command queue()
Two suggestions - 1. Make sure your AGI has the proper syntax/handling -
just because it works doesn't mean that it will be happy in the more
restrictive environment of a dialplan call.
2. If you are 100% certain that #1 has been addressed, change utils.c line
968 from
ast_log(LOG_ERROR, write()
Am Mittwoch, 28. April 2010 09:58:14 schrieb Tzafrir Cohen:
On Wed, Apr 28, 2010 at 07:12:57AM +0200, Claire Sinn wrote:
Hi,
a few month ago, I tried to install zaptel for my Beronet BN8S0 pci
card... I gave up and took hfcmulti/lcr. Now dahdi (2.2.1.1) seems to
support the card and I'm
GotoIf($[${CALLERID}:.*333.*]?your_extension) (untested)
Something like that (fix variable name to suitable). Check Asterisk regular
expressions.
http://www.voip-info.org/wiki/view/Asterisk+Expressions#Regularexpressions
On Wed, Apr 28, 2010 at 3:49 PM, wassim darwich wassimdarwi...@yahoo.com
Hi!
Why is asterisk so slow in sending the call info via STDIn in these cases?
Is there any way this can be fixed?
Your AGI script is faulty: In at least one place you have missed to READ
the output right after you have issued a command. So go check your script
(agi debug might help a little
Check out this snippet from Tilghman Lesher (one of the true Asterisk
Guru's)
http://www.mail-archive.com/asterisk-users@lists.digium.com/msg220482.html
It's a NAG (my term) introduced in the jump from 1.4.22 to 1.4.23 and
carried out through the rest of the 1.4 tree.
-Original Message-
Hello listers,
Still plodding along in the 1.4 tree, though I've started
to dabble in 1.6 land. Today's adventure involves a 2600 line dialplan. My
friend Google only points me to an antique java script and a bunch of GUI
dialplan creators. What is out there that will point
Am Mittwoch, 28. April 2010 09:58:14 schrieb Tzafrir Cohen:
On Wed, Apr 28, 2010 at 07:12:57AM +0200, Claire Sinn wrote:
Hi,
a few month ago, I tried to install zaptel for my Beronet BN8S0 pci
card... I gave up and took hfcmulti/lcr. Now dahdi (2.2.1.1) seems to
support the card and I'm
Danny Nicholas wrote:
Check out this snippet from Tilghman Lesher (one of the true Asterisk
Guru's)
http://www.mail-archive.com/asterisk-users@lists.digium.com/msg220482.html
Thanks but that appears related to AMI not AGI.
--
Philipp von Klitzing wrote:
Hi!
Why is asterisk so slow in sending the call info via STDIn in these cases?
Is there any way this can be fixed?
Your AGI script is faulty: In at least one place you have missed to READ
the output right after you have issued a command. So go check your
Can you post the script?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gareth Blades
Sent: Wednesday, April 28, 2010 9:00 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Hi
i want change my asterisk server. Actually, Asterisk work's on a IBM
Server with a internal digium E1 card.
For High availability, i don't want now use internal E1 card.
In my new asterisk systems, i have two server and two E1 not in the same site.
I am search a hardware gateway, if possible
Danny Nicholas wrote:
Can you post the script?
Yes private stuff is in a separate file. $mode=start works fine but
answered and completed cause the problem.
I dont know if it is a problem with teh AGI script or just the newer
asterisk reporting it as an error. It doesnt effect functionality
Just a hunch - add STDIN; after line 15 and give it a whirl.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gareth Blades
Sent: Wednesday, April 28, 2010 9:47 AM
To: Asterisk Users Mailing List -
You mean as in :- ?
sub set_variable
{
my ($self, %vars) = @_;
while (my($var,$val) = each %vars)
{
if (!defined($val))
{ warn AGI-set_variable: not setting '$var' because value
was undef\n; next; }
#warn AGI-set_variable('$var','$val')\n;
- Olivier CALVANO o.calv...@gmail.com wrote:
Hi
i want change my asterisk server. Actually, Asterisk work's on a IBM
Server with a internal digium E1 card.
For High availability, i don't want now use internal E1 card.
In my new asterisk systems, i have two server and two E1 not in the
On Wed, Apr 28, 2010 at 7:58 AM, Tim Nelson tnel...@rockbochs.com wrote:
- Olivier CALVANO o.calv...@gmail.com wrote:
Hi
i want change my asterisk server. Actually, Asterisk work's on a IBM
Server with a internal digium E1 card.
For High availability, i don't want now use internal E1
Darn, that should have worked. The improvement from 1.4.22 to 1.4.23+
basically requires that every print STDOUT line be followed by a STDIN
to make util.c not choke when doing commands/setting variables. I wonder
how this rewrite would work?
sub set_variable
{
my ($self, %vars) = @_;
On Wed, 28 Apr 2010, Gareth Blades wrote:
The script does not issue any commands. The same script is called at all
3 stages but with different parameters on the command line to indicate
the call status. Works fine before the call is answered but during and
at the end of the call it quits
Steve Edwards wrote:
On Wed, 28 Apr 2010, Gareth Blades wrote:
The script does not issue any commands. The same script is called at all
3 stages but with different parameters on the command line to indicate
the call status. Works fine before the call is answered but during and
at the end
On Apr 28, 2010, at 11:30 AM, Steve Edwards wrote:
On Wed, 28 Apr 2010, Gareth Blades wrote:
The script does not issue any commands. The same script is called at all
3 stages but with different parameters on the command line to indicate
the call status. Works fine before the call is
Danny Nicholas wrote:
Darn, that should have worked. The improvement from 1.4.22 to 1.4.23+
basically requires that every print STDOUT line be followed by a STDIN
to make util.c not choke when doing commands/setting variables. I wonder
how this rewrite would work?
sub set_variable
{
Redfone it's good!
On Wed, Apr 28, 2010 at 10:07 AM, Olivier CALVANO o.calv...@gmail.com wrote:
Hi
i want change my asterisk server. Actually, Asterisk work's on a IBM
Server with a internal digium E1 card.
For High availability, i don't want now use internal E1 card.
In my new asterisk
Both of our production asterisk servers are dumping core when making writes
to our cdr tables. Here is a backtrace of the problems we are having:
#0 0x00447b1f in tdserror (tds_ctx=0x1, tds=0xb7938c90, msgno=20004,
errnum=9) at util.c:347
347 if (tds_ctx tds_ctx-err_handler) {
(gdb)
Looking at the Asterisk::AGI docs, maybe try calling ReadParse() early
in the script to read in anything from stdin?
(From the docs)
# pull AGI variables into %input
%input = $AGI-ReadParse();
--
_
-- Bandwidth and Colocation
FWIW, I would take your STDERR references and give them another handle,
since you're not really trying to produce a CLI/Console output.
The symptoms you have described in this thread are 100% compliant with AGI
protocol violation (their term not mine) - the last suggest I would give
you is to do
- Luis Morales faston...@gmail.com wrote:
Redfone it's good!
Redfone makes a nice gateway(they also have very good support), although it is
TDMoE. The OP specifically mentioned they want a gateway which provides SIP
connectivity.
--Tim
--
On Wed, 28 Apr 2010, Gareth Blades wrote:
The script does not issue any commands. The same script is called at
all 3 stages but with different parameters on the command line to
indicate the call status. Works fine before the call is answered but
during and at the end of the call it quits
Hi:
Thanks for your answer.
i tried your suggestion (exten = _..,1,Noop) but it didnt work ,i think
(_..) is wrong formula to mean that number contains those coz asterisk
didnt matched the call with extension , is there any other formula? i will
write down wht i want to exactly to
On Wed, 28 Apr 2010, Ryan Bullock wrote:
Looking at the Asterisk::AGI docs, maybe try calling ReadParse() early
in the script to read in anything from stdin?
(From the docs)
# pull AGI variables into %input
%input = $AGI-ReadParse();
early == before (any interaction with Asterisk || exit)
Steve Edwards wrote:
On Wed, 28 Apr 2010, Gareth Blades wrote:
The script does not issue any commands. The same script is called at
all 3 stages but with different parameters on the command line to
indicate the call status. Works fine before the call is answered but
during and at the end
Steve Edwards wrote:
How do you reconcile your assumption that the Perl module is reading
STDIN and your statement that your AGI quits before asterisk has
finished sending the information about the current call via STDIN.
On Wed, 28 Apr 2010, Gareth Blades wrote:
Only that if I put a 3
Steve Edwards wrote:
Steve Edwards wrote:
How do you reconcile your assumption that the Perl module is reading
STDIN and your statement that your AGI quits before asterisk has
finished sending the information about the current call via STDIN.
On Wed, 28 Apr 2010, Gareth Blades wrote:
On Apr 28, 2010, at 1:00 PM, Gareth Blades wrote:
Steve Edwards wrote:
Steve Edwards wrote:
How do you reconcile your assumption that the Perl module is reading
STDIN and your statement that your AGI quits before asterisk has
finished sending the information about the current call via
On Wed, 28 Apr 2010, Fred Posner wrote:
Did I miss where the code was posted?
Yes. In my mail reader it is Gareth's second post.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice:
Sorry for the simple question.
I'm trying to match sipprovider.nocredit but the following doesn't execute
NoOp (it runs context but not context-custom). What am I doing wrong?
[context]
include = context-custom
exten = _.,1,Set(GROUP()=1)
exten = _.,n,Goto(destcontext,${EXTEN},1)
On Apr 28, 2010, at 1:12 PM, Steve Edwards wrote:
On Wed, 28 Apr 2010, Fred Posner wrote:
Did I miss where the code was posted?
Yes. In my mail reader it is Gareth's second post.
Thanks. Wish I hadn't looked now.
--fred
http://qxork.com
--
Hi,
Does enabling a jitter buffer in sip.conf make sense if the call is pure SIP?
SIP client---ASTERISK SIP---Internet SIP provider
I think it should help on the Asterisk receiving side in case of unreliable
bandwidth.
Vieri
--
It seems to me that you're doing this the hard way. How about this:
[context]
exten = _.,1,Set(GROUP()=1)
exten = _.,n,Goto(${EXTEN},1)
exten = sipprovider.nocredit,1,NoOp(No credit left)
If I'm wrong (happens every once in a while), Google Asterisk 302 redirect.
-Original Message-
Thanks Steve, I corrected spelling that but still having issue :-)
Issue:
when some one calls bob, I want asterisk to add @DOMAIN and make the call.
but it is not working .
--
Config files:
sip.conf
[ext-sip]
type=friend
context=phones
qualify=yes
host=external.proxy.com
extensions.conf
Do you mean you want
exten = bob,1,Dial(SIP/ext-sip/${EXTEN},20)
You want to call out via sip user ext-sip to that system's extension bob?
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Apr 28, 2010, at 10:50 AM, Aditya Kumar wrote:
Thanks Steve, I corrected
Try: exten = bob,1,Dial(SIP/ext-sip/${EXTEN},20) ?
--
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-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
Thanks a lot Jim and Ryan.
It worked with changing the order as you suggested.
--
Few more questions on Dial plan:
use case:
when some one in my pbx calls 100.200, I have translations well defined, Media
also (media via asterisk) --Works.
when some one calls bob, or for any names I am adding
On Tue, Apr 27, 2010 at 8:48 PM, Aditya Kumar adityakumar...@yahoo.comwrote:
Hi All,
pl help me with this basic question.
I have a users (soft clients) with usernames having Alphabetics.
I want to use Asterisk as my server.
How should I have the dial plans as there are no numbers involved
We've been here, done this; This is a 1.6 NEW and Specific message to tell
you that Asterisk can't start it's canary-monitor thread and that under
certain conditions, you might be about to lock up. Look through the earlier
posts in April.
-Original Message-
From:
All,
I just noticed this in my logs, and am rather lost as to what module
it pertains to. I would assume pseudo-realtime priority for the process,
but I am looking for a little confirmation from the group:
[Apr 28 12:28:36] WARNING[20773] asterisk.c: The canary is no more. He
has
Danny Nicholas wrote:
We've been here, done this; This is a 1.6 NEW and Specific message to tell
you that Asterisk can't start it's canary-monitor thread and that under
certain conditions, you might be about to lock up. Look through the earlier
posts in April.
-Original Message-
From:
Am Mittwoch, 28. April 2010 16:21:44 schrieben Sie:
On Wed, Apr 28, 2010 at 03:56:04PM +0200, Claire Sinn wrote:
Am Mittwoch, 28. April 2010 09:58:14 schrieb Tzafrir Cohen:
On Wed, Apr 28, 2010 at 07:12:57AM +0200, Claire Sinn wrote:
Hi,
a few month ago, I tried to install zaptel
On 23/04/10 10:31 AM, Bryan Jacobs wrote:
Don,
No, I'm not trying to say there's a problem with generating the tones.
The issue is that my phone is still holstered, connected to the car via
Bluetooth. I have steering-wheel buttons for receiving calls and
hanging up, but I don't have a safe
On 25/04/10 7:00 AM, bruce bruce wrote:
Adobe Air and Adobe FMS are good examples of VoIP working flawlessly
over TCP. We are actually developing a flash phone which needs only TCP
to transmit both signal and audio.
Ok, let's look at that (UDP vs TCP for realtime stream). Let's call the
On 27/04/10 7:33 PM, 675842709 wrote:
when i install asterisk addon ,i got error here
chan_ooh323.c:1934: error: dereferencing pointer to incomplete type
chan_ooh323.c:1935: error: dereferencing pointer to incomplete type
chan_ooh323.c:1937: error: dereferencing pointer to incomplete type
Hi,
I have a duplicated DTMF issue with, it appears, bridged IAX channels.
I have the following setup:
PRI IAX
* PSTN ---* Dialplan
I've configured a number on the dialplan server to make and outbound
call to the pstn. This call then comes back into the dialplan
Matt,
What I think you're suggesting is:
1. followme(SIP phones, etc) - WAIT X SECONDS
2. if (!answered) { call(Cellphone) }
This is fine, except that it imposes a delay on connecting my call. If
I were to do steps 12 simultaneously, then my cell phone being off
would stop the phones in step
re-posting the question.
---
use case:
when some one in my pbx calls 100.200, I have translations well defined, Media
also (media via asterisk) --Works.
when some one calls bob, or for any names I am adding Domain and call is been
sent to the other party -- Works, no media...
For the
Here's a segment of my dialplan, I'm working on the freenum/ISN
functionality:
same = n,Set(isnresult=${ENUMLOOKUP(${EXTEN},sip,,1,freenum.org)})
same = n,GotoIf($[${isnresult} != ]?:fn-CONGESTION,1)
; set up our outgoing call state
same = n,Set(SIPFROMUSER=${CALLERID(num)})
same =
On 29/04/10 2:00 PM, Bryan Jacobs wrote:
This is fine, except that it imposes a delay on connecting my call. If
I were to do steps 12 simultaneously, then my cell phone being off
would stop the phones in step #1 from working.
If you play a message telling someone that you are being located,
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