This returns a list of apps
show application fooreturns foo's information
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] ;Dial
0 on the phone for external line
exten = _0,1,Dial(Zap/4/$EXTEN) ;=== statically
allocated to Zap/4 needs to be dynamic
exten = _0,2,Goto(102)
exten = _0,102,Congestion
exten = _0,103,Hangup
I'll apreciate any help in this regard.
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considered what IAX trunking can do for you? It will reduce
frame rate as you add channels since each packet will then hold the
frames for each of the consecutive calls.
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. This was a very simple (1) shutdown, (2) remove power supply
(3) install riser card and TE410P, and (4) reconnect power cord.
Not sure, but I didn't think any of the Digium cards where PCIX
compatible. The TE410P was compatible with a 64bit slot but nothing
more.
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is to pull the board out and maybe hook up to a
different PSU with no chance of it shorting out to verify it is ok.
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deliver on those reliability expectations, and do
it soon, people are going to switch to companies that can. And you know
what? I don't blame them.
Funny how I have 2 production machines that handle fairly large call
loads without any stability issues.
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off to remove the digest option from the mailing list and use
a proper mail filter to split the list mail to a folder other than your
inbox.
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that needs to be done.
If you find that you need more functions, then you may need to move up
to a SIP phone.
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better density, but again, it wasn't
a direct fit with the zapata libraries and didn't look easy to do.
Does the Adtran way differ significantly enough to make this become
easy?
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in the outgoing
directory due to race conditions on your creation time and when you are
done writing the file.
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the canned audio then connects to the appropriate Zap
channel.
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file name and a phone number to call in
the data section.
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) exited non-zero on 'SIP
Please tell me I am doing something stupid ! as I would love to know,
before I pull all my hair out :-)
It can't be installed because you have uninstalled mpg123. Why did you
uninstall mpg123? Reinstall mpg123 and stop messing with your hair.
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on
hold from the phone??
Let asterisk do the hold not the phone.
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order.
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though.
Please learn to trim the messages. No need to bounce 3 copies of the
list added footer.
The phone channel you speak of is meant to use hayes compatible commands
and results in half duplex sound. If you can make it work, it would be
like a cb radio.
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. Where FXS is the phone/customer
side of the Communications. .
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.
If you go the SIP route, you might get a little further down the road,
but you also take a chance of integrating echo into the system.
The answers are there for someone with the requisite background
knowledge of the hardware available.
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other to know
who is where and the users spread over the many asterisk machines. With
IAX attempting to handoff calls if possible, it is possible to create
true P2P calls as well as routed calls to handle NAT or anything else.
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.
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looking for a /etc/modules.conf or /etc/modules or
even /etc/conf.modules
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name of the symbol.
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. Many of the commits will have bug number
references.
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google didn't help much, so
I'm sorta stuck not knowing where to go next.
You did most of the right work. Follow the link below to get the proper
google search string. It returns 3 links for me and the first one I
checked had the proper answer for you.
http://tinyurl.com/6ec25
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all if we
answer your homework questions?
How did you get to this point in your education when you haven't learned
that YOUR homework is YOUR responsibility not ours?
Why haven't you learned to not cross post to every asterisk list?
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. If you don't need a rack mount server,
you can find even cheaper machines around to dedicate to it.
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.
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. Stupid users who haven't learned safe computing littering the
net with all their trash.
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the PC's
PSU and cause it to either fail or pulse causing system stability
problems.
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will not happen on the
asterisk machine.
Just to recap, I am not flaming you no matter how it seemed in the
message.
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trying to be
regulated through your standard PC PSU of 300w. Won't you just love
trying to diagnose random reboots right after a phone call comes in and
over draws your PSU capacity and it goes into short protection where it
begins pulsing power.
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have the CAC set to provide timing to the line? If not, you need
to set your timing to 0 here so the TE410P card will provide timing.
Also, as a precaution, It is helpful to power cycle the machine when you
change timing.
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: 33 XT-PIC ide1
NMI: 0
ERR: 0
Any problems here?
BTW, please, at the least trim the footers off of the emails.
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...
instead it will just ring and ring and ring... any ideas?
Any configs?
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as well.
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upon hangup. Asterisk closes the pipes
that show up as STDIN and STDOUT for your AGI app. You need to deal with
it gracefully.
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in proper
dependency information to stop you from installing it till all the
required packages are installed.
You really should download the source, compile, and install. This will
mean that asterisk will be linked to your libraries.
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the extension/context of the voicemail box that
you sent the call to. From there it is just a matter of looking in the
appropriate directory and dealing with the way voicemail write the
information out.
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follows here
[outbound-toll]
--- outbound calling info follows here
Without the details of these outbound sections, we can't tell if you
have a pattern matching problem that is causing your troubles.
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in the middle of your
call. at random times.
not sure about this one.
6) Where can one post thier configs to check / validate that they are good,
and no overloading the server ?
It is unlikely you will overload a server because of your configs.
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the software as it is a commercial solution.
I'm pretty close to writing a rule to bounce a copy back to him and the
company who sold him the software. Anyone want to provide a spam list to
add him to for being annoying?
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%, Oracle - 55%, MySQL - 33% and PostgreSQL - 8%.
I see they created this with Mysql,
78 + 55 + 44 + 8 = 185%
I'm sure if you add in the others we would get to something around 300%
deployment.
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be an authoritative source. The DB
should be authoritative as to what is correct.
You shouldn't have to babysit the DB to make sure it is doing the
correct thing. If you have to babysit it, it isn't worth it.
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will
read your message before assuming we all need to see blue.
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that are cheap to replace
and use a group of channel banks to support the phones. Maybe a bit more
expensive than the IP phones, but it is tried and proven technology.
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to learn in-line quoting as well and configure your mail app
for proper quoting as well.
- Original Message -
From: Steven Critchfield [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, March 14, 2005 3:56 PM
Subject
.
Keeping the idea of IP phones open lets you possibly add IP phones to
the network as needed. Think of the possibility to add a phone for a
student teacher/observer with private extension while leaving the main
classroom phone available as usual.
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questions that
are not on topic for this list. It is high enough traffic as it is
without talking about something completely unrelated.
And you still haven't learned to trim your messages.
- Original Message -
From: Steven Critchfield [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non
by one parent and running crying to the other
parent for some consoling. I am quick to spank users who misbehave.
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compile the source spandsp-0.0.2pre10; when i try to do the
make sentence the following errors appear:
bert.c:43:20: tiffio.h: No such file or directory
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want
them in.
Use meetme to conference them all in.
Ohh my, looks like I just put enough work for you to need to send you an
invoice for you or your customer to pay me Do your own damn homework
more often.
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On Sat, 2005-03-12 at 21:35 -0600, James Taylor wrote:
Anyone done a chat line app?
Any reason why the meetme app doesn't fullfuill your needs or did you
not bother to look?
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about how little time and effort it took to follow the above
pattern to quickly answer questions on your own.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Steven
Critchfield
Sent: Thursday, March 10, 2005 6:42 PM
To: Asterisk Users Mailing List - Non
, MySQL is just so much more easy to use, install and maintain.
phpMyAdmin makes it even easier.
If that is a deciding reason, you should check out phppgadmin sometime.
Very similar interface but for postgres.
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majority doesn't have access or
at the minimal don't use the software you are using to do config and as
it is an agi script outside of asterisk, you should go to the vendor of
PBXWare and see what they say.
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only.
I'll admit I haven't had to use this feature yet, but I see where some
people could really need it.
-Original Message-
From: Steven Critchfield [mailto:[EMAIL PROTECTED]
Sent: Thursday, March 10, 2005 1:00 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
app multiple time simultaneously as it will start
showing the ability to handle concurrent users.
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a day later than the original? Could you not look at the
suggestions provided already and work from there or provide feedback as
to why you didn't like the suggestions?
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Asterisk
for it implements the same API as
the current ISDN cards in use support.
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On Thu, 2005-03-10 at 23:10 +0100, Stefano Arata wrote:
On Thu, 2005-03-10 at 11:07 -0600, Steven Critchfield wrote:
Is there some decent reason you felt the need to post this exact
question again a day later than the original? Could you not look at the
suggestions provided already and work
the number and dial it out. The benefit is that asterisk then
logs the outgoing number and the times in CDR.
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and
the ,ps file as well.
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through the hoops to make it scale. Of course my
experience was about as much writing(inserts, updates) as reads.
For stability, I don't think there is any problems with mysql or
postgres. I only had problems with mysql under heavy load.
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and config database. As for
stability/scalability, the .org registry is on postgres.
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must use the Hayes AT command set. The command set is
not designed to allow for bidirectional comms traffic. So you would get
into a cb radio situation where one side could talk but not the other.
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On Wed, 2005-03-09 at 15:18 -0600, Matthew Boehm wrote:
Again, may be off topic but are there any cards out there supported by
asterisk that have on-board DSPs to do better 729-711 or 729-PRI
conversion?
Not yet, and I don't know if anyone is working on the drivers for such a
card.
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being
not to difficult to figure out and split and symlink back together if
need be.
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in the context or in contexts that are
included?
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On Tue, 2005-03-08 at 18:52 +0100, Florian Effenberger wrote:
Hi Steven,
Do you have wildcarded extensions in the context or in contexts that are
included?
no, I don't.
Then next step is to include your config files so we aren't just
guessing at your problem.
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added to the
driver. The TDM400P card will probably be developed for a while to come
as it the current option.
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On Tue, 2005-03-08 at 13:46 -0500, Steve Prior wrote:
Steven Critchfield wrote:
TDM400P with FXO daughter card includes 1 hour of Digium support. It is
supposed to support other line types. If you have trouble, it is likely
you will get direct support from Digium and from the community
it is mentioned on a list as
is pertaining to code I am running. I have so little time now, I don't
go looking for needles in haystacks that I don't need.
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cable
with 50 pin connectors on either side.
Go to any reputable supplier near you and they should be able to help
you look at and find what you are comfortable with to use for
installation.
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;
tmp[x] = ~((1 GAIN) - 1);
f is the frame to be written. f-data is the audio data to be written.
The for loop just uses pointer math to traverse each sample of the
buffer and increase the volume of the sample with clipping control.
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want to store in the DB or be incomplete.
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on these machines. At 32k I could only use
2.6 million inodes pointing to minimum sized files on the 80gig drive.
If you are worried about inodes, I believe it is xfs that dynamically
creates inodes as needed and won't run out.
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of being run from
a different machine.
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On Mon, 2005-03-07 at 19:19 -0500, Karl H. Putz wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Steven
Critchfield
Sent: Monday, March 07, 2005 6:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users
or at the least be respectful of those willing to give the help.
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will not work without a license. The error message above told you
that asterisk couldn't find a valid path to convert from gsm audio to
g729 audio data. Seems that should have been very obvious from the
error. It is well documented had you even decided to search.
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three-way
calling enabled on whatever interfaces you are using?
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cables. The software here behaves
much the same way, it picks the audio data out of the packet and passes
it through to the other side of the communication.
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. The customer phone number is in the database, so clicking
and icon asterisk would dial the number and connected to my speakephone
when the connection goes through.
lookup sample.call
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on the first 8 lines
; used by internal phones.
;
context=default
;user = voice00
;user = voice01
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On Fri, 2005-03-04 at 13:58 -0700, Joseph wrote:
On Fri, 2005-03-04 at 14:45 -0600, Steven Critchfield wrote:
On Fri, 2005-03-04 at 13:41 -0700, Joseph wrote:
Is it possible to dial number from the command line and passing the
connection to one of my extension (or speakerphone
in the right direction I might be able to fix it
myself...
You probably want to dump the FC kernel like a bad habit. Get a plain
vanilla kernel and see if that fixes your problems.
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and therefore
can't talk to each other.
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homework is to
read and understand the source, then you have a lot of reading to do.
You may want to look at the progdocs generated when you do a make
progdocs
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a user back out
to the search engines to do their homework, we do so as a jealous
protecting of this forum and what value we receive from it.
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holding.
You and I are much more the same than not.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven
Critchfield
Sent: Wednesday, March 02, 2005 11:28 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users
find that whiny adults
actually generate a repulsive reaction by other adults and you are
actually more likely to receive more of the same treatment.
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for over 3 years.
switch is easy, you just include it in the appropriate context and it
will contact the remote machines for dialplan completion.
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to continue
exerting your own effort if the driver was not cooperating or wasn't
even interested in getting out to help push.
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ports as long as you
don't exceed 24 ports.
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the codecs
they have licenses for and charge money for the product. Might be
interesting.
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Steven Critchfield [EMAIL PROTECTED]
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told you that it had been seen and discussed
before.
http://www.google.com/search?q=Unknown+RTP+codec+72+received+site%3Alists.digium.com
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Steven Critchfield [EMAIL PROTECTED]
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On Wed, 2005-03-02 at 19:03 -0500, Andrew Kohlsmith wrote:
On March 2, 2005 05:58 pm, Steven Critchfield wrote:
Simple search would have told you that it had been seen and discussed
before.
http://www.google.com/search?q=Unknown+RTP+codec+72+received+site%3Alists.d
igium.com
Actually
members to
be able to connect simultaneously? Or is there something in the * that can
do this?
Go read the docs. Read them one or two more times after that. You need
meetme. Read about it.
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Steven Critchfield [EMAIL PROTECTED]
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of ram and swap.
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Steven Critchfield [EMAIL PROTECTED]
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) that would
mess up the recording. A bug was filed with a patch that was accepted
during the 1.0.5 deployed time.
If you are below 1.0.5 then you are sure to not have the patch. If your
1.0.5 is older than Feb-06, you don't have the patch.
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Steven Critchfield [EMAIL PROTECTED
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