In the current system I'm running, qualify IS supported for realtime
peers/users, I'm using it right now.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Pavel Jezek
Sent: Friday, August 04, 2006 11:15 AM
To: Asterisk Users Mailing List - Non-Commercial
Use the ISNULL function. ISNULL will return a value of 1 if the string it's
fed contains data. Example:
If you set VARNAME=foo, and then grab the value of ${ISNULL(${VARNAME})}, it
will be 0 because VARNAME has data
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On B
in one
row, but when I tried to use a stored proc that returns two columns Asterisk
doesn't seem to get anything.
Any help, suggestions, links, etc would be greatly appreciated. I'll post
any progress I make here if anyone's interested.
Rushowr
__
Douglas,
Awesome! I don't know why I didn't get to the point of removing all the
spaces, probably got distracted by some shiny object ;-)
Anyway, thanks for the update!
Rushowr
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas
Gar
answer your
question, removing the space does not help. I'm actually getting a bug
report together concerning this, and I tested it with and without spaces in
multiple places in the for loop definition. I'd give examples but I don't
have access right now.
Keep up the great wor
Y'know, I was thinking about a similar idea recently, primarily because I do
a lot of work with dialplan based apps. It would be great if there was a way
to set up a _complete_ call (meaning it would include what digits to enter
when, etc) in a test and then run it against the dialplan being worked
Hello all,
Wanted to toss out a question that I've been looking into for some time now
with no real results. When a variable is given a value in the dialplan, that
obviously will take up a little memory. If you're running a rather
large/complex dialplan, you may end up with variables you don't nee
I'm not personally sure, but if I recall correctly, the astDB is cleared
whenever the Asterisk server is stopped...
Anyone else?
On Friday 14 July 2006 9:30 am, Tomislav Parčina wrote:
> In article <[EMAIL PROTECTED]>, [EMAIL PROTECTED]
> says...
>
> > It would look like this:
> >
> > exten =>
> If Asterisk database can handle large amount of data, I would prefer it
> because of stability and speed. If Asterisk database can't handle that then
> I'll have to use MYSQL (or MSSQL which I prefer because I already store CDR
> to MSSQL).
>
> With internal database I have code like this.
>
> ex
12/23/05, rushowr <[EMAIL PROTECTED]> wrote:
> Thanks,
>
> I'll probably be using this.
>
> Do you know if leaving out the setting will allow the *PBX to respond
> on multiple ports? We use 5060 and 5061 for example.
>
There's one call to bind one port.I belie
ember 23, 2005 12:26 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Emergency Information Needed: sip.conf
-bindport allow multiple ports?
On 12/22/05, rushowr <[EMAIL PROTECTED]> wrote:
> This is rather a bit of an emergency, sorry for the
This is rather a bit of an emergency, sorry for the rush but, can ANYONE
tell me if 1.2.1 allows multiple ports in the general section of sip.conf
for bindport? Example:
[general]
bindport=5060,5061
Or similar?
Thanks
SKM
___
--Bandwidth and Colocati
Is 1.2.0/1 still having problems with crashes due to having too many
connections to the manager api or has that been solved? If it is, does
anyone know roughly how many connections cause the crash or is it seemingly
random
___
--Bandwidth and Colocation
a simple m0n0wall or pfsense system running on a sub $100
pc makes a GREAT router for this and allows you to use multiple internet connections in
concurrency for speed increases
other than that, yes, 2 NICs and some creative
networking and you're done
From: [EMAIL PROTECTED]
[mailto:
I'm curious about something minor, and haven't seen anything covering it
yet:
If you want to match an 11 digit US phone number (and know that ONLY proper
numbers are being passed to this portion of your dialplan), which is more
speed & memory efficient:
exten = _1.,1,VERBOSE(1|${EXTEN})
Use the o flag to force the original callerid, not the num2
callerid.
example:
exten = s,1,Dial(SIP/200,30,ortT)
SKM
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
asterisk183Sent: Monday, December 12, 2005 2:59 AMTo:
asteriskSubject: [Asterisk-Users] CallerID
Transf
Depends on what Nortel you're using. But here's just a few
I can say from being a Nortel PBX guy a while back:
Cost (depends on the situation)
Ability to write just about ANYTHING to interact with your phone
system. Just about anything you want can be done if you apply the time
I just wanted to throw in here that I was intrigued by the question, so I
went through the "Asterisk, The Future of Telephony" book, all the notes
I've gathered, and every single link that looked even mildly promising.
>From what I see, you're asking if you can use a regular expression to match
Definitely should just copy the 18661234567 extension set and then remove
the 1. I do not believe you can use regular expressions (I take it you _do_
mean regex, and not the standard pattern matching)
That's all I got...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
(oops, wrong account, let's try again, without the work email getting it
blocked)
Am I correct in my thought that if I was to issue:
Exten = 1234,1,VERBOSE(1|${SIP_HEADER(HEADERNAME)})
In the dialplan that asterisk would return the value of the HEADERNAME
field, if there was one attached to the
I'll check it once I get some extraneous hardware issues that just cropped
up after reorganizing the server's room. Thanks for the idea to check!
Rushowr
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gordon Dewis
Sent: Saturday, January 15,
y, January 15, 2005 10:12 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Voicemail after one ring?
Rushowr wrote:
> Anyone else ever have the problem of asterisk picking up with
> voicemail after one ring on an extension? I'm using free world
Anyone else ever have the problem of asterisk picking up with voicemail
after one ring on an extension? I'm using free world dial up's IAX2 service,
and I can make calls but received calls get a voicemail pickup after one
ring. No decent answer on google, cannot find anything that seems to be
wrong
Title: Message
Is it possible to
set up Asterisk without any of the cards? I'm interested in setting it up for
the company I work for, but I would like to set it up and see how difficult it
will be before I start having the company spend a chunk on equipment.
Additionally, what
phones ca
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