> Not at the moment but, just in case, I will try to use tear gas, first ;-)
>
I have found that with the right diet, teargas is not necessary.
PaulH
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - Septemb
2008/9/22 Steve Totaro <[EMAIL PROTECTED]>
> As said before, paging would work well. A walk through of the
> building would be helpful too.
In this case, walking all the way through the building is not possible (way
too long).
>
>
> Although rash, pull the fire alarm, making sure to remove the
2008/9/21 Gordon Henderson
<[EMAIL PROTECTED]<[EMAIL PROTECTED]>
>
> On Sun, 21 Sep 2008, Olivier wrote:
>
> > Hi,
> >
> > I've got this case :
> > When the last staff member is about to leave and lock offices, he would
> like
> > to notify everybody "Offices are about to be closed" so that (s)he
2008/9/21 <[EMAIL PROTECTED]>
> Hi Olivier,
>
> What type of handsets are you using in-house?
Hi,
I'm using this one http://www.voip-info.org/wiki/view/Thomson+ST2030
I'm not familiar with its paging functions but I think it's time to study
them ...
(from memory, it should be possible to specify
Another asterisk box set up to be the network side of that link?
mark morreny wrote:
> Hi,
>
> I have Asterisk setup to run on SS7, and I would like to test it out
> before getting the line from my telco.
>
> Is there any testing or simulation tool that I can buy to simulate a
> E1/SS7 link
Hi,
I have Asterisk setup to run on SS7, and I would like to test it out before
getting the line from my telco.
Is there any testing or simulation tool that I can buy to simulate a E1/SS7
link?
Could anyone give some suggestions?
Thanks alot for your help in advance.
Regards,
Mark
___
Hi all,
We are compiling a list of exhibits & tutorials to cover at Voiceroute. We
will be twittering & doing impromptu videos. We are looking for votes &
suggestions on what people would like us to cover. Pls send suggestions to
[EMAIL PROTECTED]
The team at Voiceroute have our blackberries & iph
Hi,
On my call back system, I have the script as follows:
[calback]
exten => s,1,NoOp(* STARTING CALLCHEAP\'S CALLBACK SYSTEM *)
exten => s,n,Set(CALL=${CALLERID(number)})
exten => s,n,Set(DESTINATION=myCallback.2000.1)
exten => s,n,Set(SLEEP=5)
exten => s,n,System(/var/lib/asterisk/bin/callback
hi all , please need help for an
asterisk version 1.4.21.2
i created a write func odbc list records files in sql table:
[R]
dsn=connector
write=INSERT INTO ast_records (filename,caller,callee,dtime) VALUES
('${ARG1}','${ARG2}','${ARG3}','${ARG4}')
prefix=M
and set it in dialplan :
exten => _0X.,n
Le lundi 22 septembre 2008 à 16:10 -0400, Jason Martin a écrit :
> Thanks for the reply.
>
> I tried the option below but it did not yield any different results.
Have you read this ?
http://www.the-asterisk-book.com/unstable/manager-interface-ajam.html
Have you reading the log on your CLI when
I am preparing to connect an asterisk box with a redfone fonebridge to a
T1 service provider. I am doing this by testing first with another
asterisk and a Sangoma card playing the role of telco.
I formerly had this test configuration operating flawlessly as a PRI
connection. But I discovered
Thanks for the reply.
I tried the option below but it did not yield any different results.
---
Hello,
Try with :
[testuser]
> secret = testpass
> read = all
> write = all
> deny=0.0.0.0/0.0.0.0
> permit=127.0.0.1/255.255.255.255
read = system,call,log,verbose,command,agent,user
write = system,c
(Replying to my own reply...)
On Mon, 22 Sep 2008, Steve Edwards wrote:
> On Thu, 18 Sep 2008, Steve Anness wrote:
>
>> So here is the deal. I have an Asterisk server here at work that I
>> have recently taken over and the boss is wanting the server to do a
>> lot of things that it didn't do befo
On Thu, 18 Sep 2008, Steve Anness wrote:
> So here is the deal. I have an Asterisk server here at work that I
> have recently taken over and the boss is wanting the server to do a
> lot of things that it didn't do before. I have already configured much
> of what he wanted including a voice messagi
Good day to all--
First off let me say that I have been very pleased with the mailing
list. I have learned a ton of stuff just reading other peoples
questions and comments. I really enjoyed the VOIP Conference call on
Friday morning. Still working on figuring out the best approach to
cu
Good day to all--
First off let me say that I have been very pleased with the mailing
list. I have learned a ton of stuff just reading other peoples
questions and comments. I really enjoyed the VOIP Conference call on
Friday morning. Still working on figuring out the best approach to
cu
Philipp Kempgen schrieb:
> Philipp Kempgen schrieb:
>> Ira schrieb:
>>> At 09:29 AM 9/22/2008, you wrote:
... except in some countries, the phone numbers vary in length in the
same city. Say in Hamburg, Germany, your number can be as short as 5
digits or as long as 10. You really have n
There are two bug reports with patches that might (?) be able to help:
http://bugs.digium.com/view.php?id=7403
http://bugs.digium.com/view.php?id=12215
Philipp
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - S
Philipp Kempgen schrieb:
> Ira schrieb:
>> At 09:29 AM 9/22/2008, you wrote:
>>>... except in some countries, the phone numbers vary in length in the
>>>same city. Say in Hamburg, Germany, your number can be as short as 5
>>>digits or as long as 10. You really have no way of knowing.
>>
>>
>> The
Ira schrieb:
> At 09:29 AM 9/22/2008, you wrote:
>>... except in some countries, the phone numbers vary in length in the
>>same city. Say in Hamburg, Germany, your number can be as short as 5
>>digits or as long as 10. You really have no way of knowing.
>
>
> The unanswered part of that, is this?
Hi!
Is it possible to specify the timezone in the GotoIfTime application?
E.g. I want to route the call if it is "9:00-10:00 in Austria/Vienna" or
"10:00 - 11:00 in New York".
This is needed for example if the time based routing for the office in
New York is done on an Asterisk server runnin
At 09:29 AM 9/22/2008, you wrote:
>... except in some countries, the phone numbers vary in length in the
>same city. Say in Hamburg, Germany, your number can be as short as 5
>digits or as long as 10. You really have no way of knowing.
The unanswered part of that, is this? Can 5 digit number, say
> When number starts with 011, and as country code and city code is
> identified, expect as many numbers as determined by country+city code
> (once you know country and city code, you know how many local digits to
> expect)
... except in some countries, the phone numbers vary in length in the
sam
VoIP usage was legalized in India a few months back -
http://www.voip-info.org/wiki/view/India+TRAI+Press+Release+legalizing+VOIP.
Please let me know if I misunderstood it.
BTW, I don't think I need a VoIP hardphone, the FXS slot would enable
any ordinary phone to connect to Asterisk. Again, plea
Hi,
[EMAIL PROTECTED] schrieb:
> In fact, after entering in Asterisk for the first time, my call is
> redirected to an other component of my system. This other equiment
> redirect the same call to Asterisk a second time.
Hm, I suppose, your "equipment" is using reinvites for that redirection.
The
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 for
may i noe wad can i do because my asterisk is working fine but the calls cannot
proceed between 2 asterisk servers.
hope anyone can help me solve this major problem.
thanks a lot in advance
Regards
_
Get in touch with your inner
Hello,
Try with :
[testuser]
> secret = testpass
> read = all
> write = all
> deny=0.0.0.0/0.0.0.0
> permit=127.0.0.1/255.255.255.255
read = system,call,log,verbose,command,agent,user
write = system,call,log,verbose,command,agent,user
Le lundi 22 septembre 2008 à 09:46 -0400, Jason Martin a écr
Hello,
I'm trying to do some simple tests with AJAM on asterisk 1.4.17 and I'm not
having much success.
Right now the http server just listens on localhost:8088. I've used lynx and
elinks for testing. I am able to get an Authentication accepted message using
login, and I can view the stored a
Hi!
Using setvar in a peer configuration (sip.conf) I can set the channel
variables for the incoming channel. Is there a similar method which
allows me to load these variables also for outgoing channels (e.g. to
load callee preferences)?
thanks
klaus
__
In fact, after entering in Asterisk for the first time, my call is
redirected to an other component of my system. This other equiment
redirect the same call to Asterisk a second time.
It is something like this (it's an IMS architecture) :
Softphone A --> Equiment --> Asterisk --> Equiment -->
On Mon, Sep 22, 2008 at 12:30:52PM +0200, randulo wrote:
> Hi,
>
> It's almost happening. Are there going to be any online feeds on
> Twitter, ScribbleLive or any audio or video streams? There are so many
> free tools to share your experiences in writing or via audio or video.
> Call a short repor
Hi,
Voiceroute will be at Astricon and we will be twittering a lot on events at
Astricon & we plan to make small short videos on exhibits & maybe tutorials
happening during Astricon
Keep updated with Astricon through Voiceroute
Twitter
http://www.twitter.com/voiceroute
Voiceroute Youtube Channels
Hi,
It's almost happening. Are there going to be any online feeds on
Twitter, ScribbleLive or any audio or video streams? There are so many
free tools to share your experiences in writing or via audio or video.
Call a short report into Utterz.com. The #asterisk IRC channel,
whatever. While I reali
Hi,
[EMAIL PROTECTED] schrieb:
> I have done what you told me to do, but nothing changed. Always the same
> problem.
If I understand your dialplan right, your * is still calling itself via
SIP, right?
This is what is called a loop. You should review your dialplan and
replace all dial(SIP/[EMAIL
Hi Stefan,
I have done what you told me to do, but nothing changed. Always the
same problem.
Here my extensions.conf
[originating_hind]
; Anonymous call
exten => _55tel_SIP.,1,Answer()
exten => _55tel_SIP.,2,SetCallerPres(prohib)
exten => _55tel_SIP.,3,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
ext
> Most distros come with a caching daemon.
Still that's not really the point... If Asterisk has all of a sudden
developed a habit of sending high volumes of nonsense DNS requests
then it's a serious issue. Besides if the requests are different for
each call the caching server is not going to help
I don't see where it is difficult to figure out.
First of all, system keeps looking up on the table as user dial each
number.
When number starts with 1, expect USA. When number doesn't start with
either 1 nor 0, expect USA too.
When number starts with 011, and as country code and city c
38 matches
Mail list logo
