I,
show applications This returns a list of apps
show application fooreturns foo's information
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added echo?
Have you 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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ically
> > 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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gt; anything. 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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esn't need to be restarted once a week just to do its job
> properly. If Digium can't 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 machi
stiff with out
being able to cross brace.
So my suggestion 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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put both a prerecorded file name and a phone number to call in
the data section.
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To UNSU
neric wakeup script
that plays the canned audio then connects to the appropriate Zap
channel.
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rent time.
You will find a lot of suggestions to not create files in the outgoing
directory due to race conditions on your creation time and when you are
done writing the file.
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of encoding that allowed multiple D channels to
be combined into a B channel to get 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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Voicemail access isn't hard.
Beyond that, there isn't a lot 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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nt to participate better in the mailing list, maybe you would
be better 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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_
o be loaded in a certain order.
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>
> What do I have to do to get the MOH working when a caller is placed on
> hold from the phone??
Let asterisk do the hold not the phone.
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you have uninstalled mpg123. Why did you
uninstall mpg123? Reinstall mpg123 and stop messing with your hair.
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t; documentation on this channel 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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. It will not provide a dialtone to another
device.
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k boxes peered to each 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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ally don't see more than 2 lines.
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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Steve
at is typically the Office station side
> > that has much lager power requirements. Where FXS is the phone/customer
> > side of the Communications. .
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tring. 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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re
appropriate for you to watch. Many of the commits will have bug number
references.
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T
e made the linking of the zaptel module know the
precise name of the symbol.
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uestions. It becomes a distro
specific question.
Continue looking for a /etc/modules.conf or /etc/modules or
even /etc/conf.modules
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http:
ing degree with us 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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t; >
> >BTW - Giving everyone "a hug" is an expression in Brazil. Everyone
> says it... it's like saying "have a good one" or "good to see you."
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for a machine to dedicate to asterisk. As
you have seen, asterisk needs realtime speeds and when other apps get in
it's way something gets dropped. If you don't need a rack mount server,
you can find even cheaper machines around to dedicate to it.
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box).
X and graphics drivers are big hogs on memory and CPU. VNC moves the
graphical portion over to the client machine. If you need GDM, why not
get X servers for your other machines and let GDM broadcast. This should
mean your X server run from whatever other machine should be able to be
configured
; certainly isn't that difficult to design a PC board to support
> 24 fxs ports.
You are right, but you still hit the same problem just at a lower
probability of major problems. I still contend that any design is going
to need it's own external powersupply so as to not over
dot article goes out about
asterisk. Stupid users who haven't learned safe computing littering the
net with all their trash.
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lity. Think of the troubles that would cause 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.
--
good.
> 14: 138574 XT-PIC ide0
> 15: 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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h on and off, but no help.
> > >
> > > /etc/zaptel.conf
> > > =
> > > span=1,1,0,esf,b8zs
Do you 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
dget you can get Alison to do the
recordings for you. It should match up well with other pre-recorded
prompts as well.
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t the 'unavailable' message...
> instead it will just ring and ring and ring... any ideas?
Any configs?
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you downloaded a binary copy
of asterisk and the vendor of that package didn't put 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 asteri
risk doesn't send any signal 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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Aste
Of course most of us want to follow DRY (Don't repeat yourself). In
doing so, you try and let one place be an authoritative source. The DB
should be authoritative as to what is correct.
You shouldn't have to babysit the DB to make s
es in use:
>
> SQL Server with 78%, 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.
--
to fix this problem. Also
it is unlikely he can fix 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 anno
m a video guy) on a call? Every now
> and again, you hear these strange beeps, and tweets 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
> [outbound-local]
> ---> outbound calling info follows here
>
> [outbound-longdistance]
> ---> outbound calling info follows here
>
> [outbound-tollfree]
> ---> outbound calling info follows here
>
> [outbound-toll]
> ---> outbound calling inf
to
> the AGI script?
Well, you could store 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.
March 14, 2005 6:44 pm
> > > To: Lista Asterisk
> > >
> > > Sirs,
> > >
> > > I can't compile the source spandsp-0.0.2pre10; when i try to do the
> > > "make" sentence the following errors appear:
> > >
> > > ber
over for speeding, would
you eventually let your speed creep up while driving?
You are right though that the extra responses are usually not necessary.
I understand that some feel the need to smooth the waters. I equate it
to a child getting spanked by one parent and running crying to the
asking 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]>
&
r have to do POE and the added cost of the POE injectors to
get the same functionality.
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 m
You
might want 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"
>
> Se
rchase fancy phones for the
classrooms. You could use analog telephones 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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Steven Critchfield <[EMAIL PROTECTED]>
_
out those who will
read your message before assuming we all need to see blue.
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To UNSUBS
for your billing, just know it doesn't
have to dial out of your system.
Use an IVR menu system to direct the person to the chat room you 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
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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re.
Just a quick thought here, as the vast 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 th
leaving just 69284k
of real application space used.
Second example.
245084k + 17652k = 262736k total memory
36216k + 21952k = 58168k buffers and cache
262736k - 58168k = 204568k actual memory for applications.
--
Steven Critchfield <[EMAIL PROTECTED]>
___
ier-grade' service and we use MySQL everywhere.
>
> IMHO, 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
conclude that it is possible to use.
optional steps
8. think 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
> Critch
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
> > suggest
usable if the drivers for it implements the same API as
the current ISDN cards in use support.
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> I'm not sure if it is in the driver sources or in the asterisk sources.
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 from there or provide feedback as
to why you
e MySQL showing
faster. A test that would probably show less of a gap is running
whatever testing app multiple time simultaneously as it will start
showing the ability to handle concurrent users.
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t to disk faster. Caching helps for reads 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, 20
er mysql
supporting it this directly, I think I remember the file structure being
not to difficult to figure out and split and symlink back together if
need be.
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Asterisk-U
vers for such a
card.
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rnal modems 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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___
s a lot of
press though as being an easy to install and config database. As for
stability/scalability, the .org registry is on postgres.
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h
perience has been that postgres is better at scaling than mysql
without having to jump 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 pr
create a "print job" that creates the .call file and
the ,ps file as well.
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he dialtone and validate there is one, why don't you just go ahead and
capture 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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_
= 32767.0;
if (tmpf < -32768.0)
tmpf = -32768.0;
tmp[x] = tmpf;
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 po
in connector. Then you connect it with a 25 pair 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.
x27;&&' has no
> right operand
Looks like a linux source code on your computer problem not a asterisk
and/or distribution incompatibility.
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ffectively
doubles the volume as it is saving and removes the doubling on playback.
No other audio format is given the same treatment.
> If that's the case, then why has bug 2023 been lurking without any such
> comments for many many months?
If it is lurking fo
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
em anymore, you may not get new features 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 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
gu
ed.
Do you have wildcarded extensions in the context or in contexts that are
included?
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To UN
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-
ll. FastAGI should
also be capable of being load balanced if that process is doing too much
work to be responsive upon scaling. AGI is not capable of being run from
a different machine.
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On an 80gig drive I have there are 9.4million inodes.
Block size is (I think) 32k 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 ino
need to essentially rewrite formats/* to deal with whatever file
formats you want to store in the DB or be incomplete.
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ven't asked for help, but once you
enter a public forum and request the help you need to be following the
rules or at the least be respectful of those willing to give the help.
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;
> 201,password,name,[EMAIL PROTECTED]
>
> Voicemail delivery and all works great but when I check sip extension ext1
> (analog phone using a Granstream ATA 286), the stutter tone signaling
> message waiting does not work.
SIP dialtones come from the SIP device. Look up the config on you
ver itself not in
> asterisk. Any help would be appreciated, and I can code a bit in c so if
> someone can point me in the right direction I might be able to fix it
> myself...
You probably want to dump the FC kernel like a bad habit. Ge
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
se the default context on the first 8 lines
; used by internal phones.
;
context=default
;user => voice00
;user => voice01
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and accounting
> application. 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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__
network patch 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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ll to get the recorder in the mix? So do you have three-way
calling enabled on whatever interfaces you are using?
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ce?
> If i buy a licence could solve my problem?
G729 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 documente
to feel it is important to do the searching first
and not expect hand 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
s full of venues
where you need to meet specific criteria before you are considered
worthy of interacting.
All that to lay the ground work to say that when we send 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 f
t.
Homework questions shouldn't be community work. If your 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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_
InterTel play the same role and therefore
can't talk to each other.
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To U
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+
idea.
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.digium.com
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code for an IAX2 firmware
and get help from them. They would be able to incorporate the codecs
they have licenses for and charge money for the product. Might be
interesting.
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Aste
programmable, you get a third FXS port and
you can mix and match ports 1-8 FXO and 16-24 FXS ports as long as you
don't exceed 24 ports.
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bother. You would be even less inclined 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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ng switch now 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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should use that whiny line on your next employer and see if it
helps you get the job. Of course you might 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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Steven Critchfield <[EMAIL PROTE
1 - 100 of 1956 matches
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