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 1&2 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
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 => n,Goto
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
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 1&2 simultaneously, then my cell phone being off
would stop the phones in step #
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 dialpla
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
>
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
se
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
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
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:
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 ceased
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: asterisk-users-boun...@l
On Tue, Apr 27, 2010 at 8:48 PM, Aditya Kumar wrote:
> 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 .
> so How can I
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 Do
Try: exten => bob,1,Dial(SIP/ext-sip/${EXTEN},20) ?
--
_
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http://www.asteri
Do you mean you want
exten => bob,1,Dial(SIP/ext-sip/${EXTEN},20)
You want to call out via sip user ext-sip to that system's extension bob?
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Apr 28, 2010, at 10:50 AM, Aditya Kumar wrote:
> Thanks Steve, I corrected spel
Thanks Steve, I corrected spelling that but still having issue :-)
Issue:
when some one calls bob, I want asterisk to add @DOMAIN and make the call.
but it is not working .
--
Config files:
sip.conf
[ext-sip]
type=friend
context=phones
qualify=yes
host=external.proxy.com
extensions.conf
It seems to me that you're doing this the "hard way". How about this:
[context]
exten => _.,1,Set(GROUP()=1)
exten => _.,n,Goto(${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-
Hi,
Does enabling a jitter buffer in sip.conf make sense if the call is pure SIP?
--
I think it should help on the Asterisk receiving side in case of unreliable
bandwidth.
Vieri
--
_
-- Bandwidth and Colocation
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
--
_
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)
[cont
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: +1-760-468-386
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 curre
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 Blad
> 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
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
du
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
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, 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 c
- "Luis Morales" 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
--
___
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
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 P
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) {
(gd
Redfone it's good!
On Wed, Apr 28, 2010 at 10:07 AM, Olivier CALVANO 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 hav
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
> to make util.c not choke when doing commands/setting variables. I wonder
> how this "rewrite" would work?
> sub set_variable
>
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 i
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
>> a
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 quit
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
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, Apr 28, 2010 at 7:58 AM, Tim Nelson wrote:
> - "Olivier CALVANO" 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 s
- "Olivier CALVANO" 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
> same site
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";
Just a hunch - add ; 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 - Non-Commercial Disc
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
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
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: [asterisk-u
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 yo
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.
--
___
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 ca
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 out
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---
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 lit
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
wrote:
>
> Hi guys:
> i
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 ca
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, "wri
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() doe
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("" )
[Synopsis]
Regular Expression
[Descr
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 set
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 call
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
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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