(less_than_or_equal_to_678_short_hair)
exten = s,103,Playback(more_than_678_short_hair)
Hope this helps!
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601, so the next line (checkgroup)
would jump to x+101.
Make sense?
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.
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Alternatively you could get someone to custom develop a solution for
you. Believe me the prices are not as bad as you may initially think!
:)
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http
Sean Kennedy wrote:
As the title suggests, I was wondering if there was another source of
documentation for Asterisk.
Related: If one wanted to contribute to documentation, who would one
contact?
The Asterisk Documentation Project comes to mind:
http://www.asteriskdocs.org
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any help or have any further questions.
Hope I haven't muddied the waters too much!
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, but it crashed
and I haven't spoken with him since. Maybe if you can track him down
he'd give you the code he's already done and you could continue from there.
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http
See my previous response to your mail. Post again if you are still
having problems.
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guess you could parse the text to look for cloudy/sunny/raining etc
and then use that to form the playback statement.
Oh well, looks like your post may be useful to him after all.
I'll just crawl back to the Daily Asterisk News now!
:)
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You could try _ALERT_INFO instead of ALERT_INFO...
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Star User wrote:
It's been down the last 5 hours at least. Anyone know what the problem is,
or when it will be back up?
Nope, no idea, but I can confirm it's down from here too!
:)
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for collocation to them!
Ooh lots of exclamation marks!
:)
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)
Description: If you call ParkCall (from the dialplan or the manager
interface), you can pass in the channel to announce the parking
destination to as well as the channel that you would like to have
parked. Bla bla bla
:)
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Kerry Garrison wrote:
Firefly supports Speex too but trying it just now I am getting no audio.
-Kerry
Type show translation in your Asterisk console to check if you have
speex installed.
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Excellent!!!
I have numerous customers who were wanting to try this out, but had ZAP
extensions! (hehe myself included)
This really makes this product I think!
Keep up the excellent work!!!
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://www.asterisk.org
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:)
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, and
then call the number.
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.
:) No problems.
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.
Therefore -r1-0 should download -r1-0-7
:)
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?
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(${ACCOUNTCODE})
And that should work.
Then other option would be to use the astdb and put a pin in for each
account.
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are at the end of the driver source, how long does it take to
put into a more accessible form?
Probably not long: just head over to http://asteriskdocs.org and see how
you can help out, or pop on to the wiki ( http://www.voip-info.org ) and
add an entry.
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'.
No, because his next line is _.,1 so it will actually use the extension.
His problem is just one of permissions. Maybe he should use a suid prog
to set the date.
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http
? There's no way that I would have time to check every post for new
news...
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be impossible to organize.
The idea would be to get some of the hardware manufacturers to support
it. If you manage to do this, I'm sure most of the community would get
behind and support/publicize it.
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Matteo Brancaleoni wrote:
I hate to say that, but the problem is that Digium doesn't do this.
Ahh I beg to differ.
I resell both Digium and Sangoma gear and provide full installation
support for both.
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Matt Schulte wrote:
Is there an SRV bounty out there yet? $500 to the first person who
implements it (correctly :-) )..
Email for details.
Add this to the bounty page on the wiki (www.voip-info.org)
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Damon Estep wrote:
http://groups-beta.google.com/group/Asterisk-test
Stuff shows up fast! Anyone have insight on this, did I miss something?
Looks like a mirror of the mailing list...
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PROTECTED])
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Hi,
we've just posted a completed interview with Kevin Fleming on the Daily
Asterisk News.
You can access the interview from here:
http://www.sineapps.com/news.php?rssid=667
:)
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stops programs registering
server/client ports?
Did you have any problems with such a program (i.e. zonealarm) and then
tried to uninstall it?
Do you have the latest version of the firefly-thirdparty.exe file?
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has
a stream on the web...
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on this.
What exactly did you want realtime audio for (bearing in mind that the
sound subsystem is not normally used by Asterisk)?
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http://www.sineapps.com
(quality of service). There is
a good wiki page (www.voip-info.org) on it. I personally use the
wondershaper script.
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Garrett Smith wrote:
I have about (500) more of those NEW IN BOX CP-7960s that many of you
purchased last month. If anyone needs more please let me know.
You have posted to the wrong list.
This type of mail should be directed to asterisk-biz
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mailing
list and so decided that because there are so many on the asterisk-users
list you would simply use that instead.
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Esben Stien wrote:
Matt Riddell [EMAIL PROTECTED] writes:
Any plans for asterisk to support jack for realtime audio?,
So that you can have a phone on the console of Asterisk?
Yes.
And what difference is 30ms going to make? Bearing in mind that reverb
mixes with original sound (to our ears
: [Asterisk-Users] Qs about FXO/FXS cards
On Mon, 2005-01-03 at 20:14 -0600, [EMAIL PROTECTED] wrote:
Gosh, you sure are full of yourself.
Looks like you didn't get enough caffeine into your body before
responding. Ohh well
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Corvin wrote:
Matt Riddell wrote:
Corvin wrote:
Hi!
I would like to use lcdproc and asterisk.
Any hints or links? Maybe someone
has experience in such matter. I am working
on such solution. I've heard of SAPBX.
Thanks for any help.
Hi, I was working with someone on this until my BB Forum fell
1234 is the process id).
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= iaxprovider
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so
you won't be able to use it without echo outside of the United States.
Really I'd recommend getting a TDM400P card or some older customers.
:-)
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http
on the Makefile!
:-)
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Ferguson, Michael wrote:
G'Day All,
rpm -q kernel-source returns Package kernel-source is not installed
Where can I find it and install it. Asterisk evidently needs it for a
successful install.
You can do:
yum install kernel-source (although I thought you didn't need it in 2.6)
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is wrong?
chan_iax2.c:7331: internal error: Segmentation fault
Ahhha segfault in gcc is generally a bad thing (and not usually
associated with Asterisk).
How much RAM does the box have?
Can you do a memtest?
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that it can't find your soundcard.
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will be free software, though. :-)
Steve
Just thought I post that you are truly a star Steve!
:-)
Keep up the good work!
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: generally not a good idea to berate Critchy!
:-)
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Scheda wrote:
If anyone knows of a linux applicable IAX softphone,
I'd be more than willing to give it a shot, but I haven't found one so
far.
Have you tried iaxcomm?
http://iaxclient.sourceforge.net/iaxcomm/
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Mike Cathey wrote:
Except use this patch instead of the one they link to:
Wasn't the patch added to CVS?
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/join #asterisk after clicking on irc://irc.freenode.net
You sometimes get lag on IRC, but it's unlikely to be over 30 seconds...
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or
http://www.geocities.com/tdanro
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That's funny. I post to a topic called long delays in posts and my post
arrives in less than 5 minutes...
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you are happy that your current system is
secure) subscribe to bugtraq, and check on
http://packetstormsecurity.org/ for new releases.
:-)
Or you could just employ a security consultant...
But, if you have the time, it's worth learning!
:-)
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of the simplest patches in the world to apply. I can talk
you through it, or you could have a look (hint +xxx means add xxx, don't
forget that the spaces are actually tabs in the Makefile).
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to manually merge
the changes, although backporting it would probably require at least a
little C knowledge.
The other option would be to run CVS-HEAD. This would however not be
recommended for a production system.
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C F wrote:
Well thanks for the reply, until you don't test it with the Digium
TDM04B I don't think you *know* anything.
WTF!
Bad form...
It works. If *you* had *tried* the suggestions people gave you, you
would find that out.
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g723 to
slinear (i.e. uncompressed)...
Still not got it...
Hint: You don't have the g723 codec. Unless you paid like $100,000 for
the licence and then got someone to code it for you.
:-)
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...
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. Just add more boxes.
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Denis Galvão - iSolve wrote:
Same problem with 0.9.9g...
You could try contacting Dan offlist?
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Michael George wrote:
On Tue, Jan 18, 2005 at 04:16:08AM -0600, Justin Carlson wrote:
no i was using line 1 for testing /w fxs module and i never changed it
back
Also, could you show us the contents of your [routing] context in
extensions.conf?
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Eric Bishop wrote:
I logged a support issue with HP and their response was that it's not
their server that is the problem and if other cards show interrupts
(which they do) there's nothing more they can do
And you told them that this is the only server the cards don't work in?
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of config files?
What IP device?
Do you have QOS set up on your network?
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to
improve the sound quality
As above, let us know what devices you are using. I would say that the
PSTN GW is the likely culprit. Does it have any settings for
receive/transmit gain adjustment?
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Brian West wrote:
I have never had an issue with 2.6.9 with asterisk.
I second that.
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Joseph wrote:
Who else makes Analog Telephone Adapters with IAX protocol besides
Digium?
The PA1688 chip now supports IAX.
There are many phones/ATA units which use this chip.
There is even a page on the wiki about them.
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)).
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Matt Riddell
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was released, they changed their mind. Much to the
detriment of New Zealand internet...
So, once again, the big boys get to keep their toys, while we can but
beg for quality service...
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. That way you would know
that they came to you because they didn't enter a number, and you could
interrogate them as to what/how/why etc...
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Computer Onsite Support wrote:
Thanks you but that didn't work. Any other solutions?
Make sure you are using the correct version of mpg123...
I think from memory the correct version is 0.59r
Also, I think Redhat has simply made a symlink to mpg321 (which is *not*
the same).
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!
THIS SOFTWARE COMES WITH ABSOLUTELY NO WARRANTY! USE AT YOUR OWN RISK!
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the Authenticate line before it.
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the m option in the dial command?
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beonice wrote:
Ouch ... error while writing audio data: : Broken
pipe
What are the messages before this?
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directly access. This is the simplest way.
I think someone could write a program that could analyse you config
files. I've kinda got too many projects on the go, but maybe you could
write a spec and submit a bounty?
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Colin Anderson wrote:
Ouch ... error while writing audio data: : Broken pipe
Did you run make samples from /usr/src/asterisk?
Yeah he has config files.
What is the response you get running mpg123?
I.E. what are the first four lines?
I.E. what version?
:)
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it will be highly appreciated.
You could try altering you dial line so that it starts with a few w's:
exten = _9X.,1,Dial(Zap/1/www${EXTEN:1})
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,
Matt Riddell
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, Matt Riddell wrote:
Frank wrote:
I've been looking through the archives and have not been able to find
anyone with a similar problem but perhaps I'm not searching in the right
places. The problem is that my outbound call sometimes go though and
sometimes don't. If someone can point me in the right
.
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taf taffey wrote:
Ok thanks.
You might want to have a go at turning off html emails!
:)
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= 30Mb
22Kb x 10,000 users = 220Mb
Poor digium servers!
:)
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scroll any windows you get clicks on any calls that are in
progress. Even on FVWM.
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and the T1s came up.
Do know if this will help you, but the T1 card seems to be delicate with
power.
Hmmm strange. Makes you wonder if this is the problem that is occuring
with the HP machines...
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line surge protectors
would only protect against *huge* voltages.
Maybe there is such a thing as a power conditioner for telephone lines?
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Nils Segerdahl wrote:
Im running bristuff-0.2.0-rc2b with Florians patch.
4 Billion hfc cards in ptp mode.
Works like a charm.
4 billion hfc cards! Wow that must be some server :)
Oh a brand name - I guess I missed the capital letter.
hehe
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). In fact there may even be an example on the wiki.
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james wrote:
Jan 22 21:42:26 WARNING[234]: format_wav.c:159 check_header: Unexpected
freqency 4000
You might want to try encoding at 8000Hz instead of 4000Hz.
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Roger Schreiter wrote:
Hi,
I put a
pickupgroup=0
line for each user in sip.conf.
As far as I'm aware, pickup groups are only for zap interfaces...
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Cheers,
Matt Riddell
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http
this issue?
The fact you don't have speex compiled...
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Cheers,
Matt Riddell
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there's no way
I'd be paying it!
Do you have a list of these numbers so that I can call them though FWD?
:)
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Matt Riddell
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...
[extensions]
exten = 100...
[outgoing]
ignorepat = 0 ...
[internal]
include = extensions
include = outgoing
Then you put context=internal for any phones that are internal (I.E.
SIP/FXS/IAX etc), context=incoming for any FXO lines/trunks...
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Cheers,
Matt Riddell
, I was wondering if someone was going to point this out. I was
about to post it myself!
Pretty early morning for you Duane?
:)
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Matt Riddell
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the other end?
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Matt Riddell
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Asterisk-Users
).
Bummer. Glad I don't run a business in Israel. Thought it was bad here
in New Zealand! I'd hate to have my business phone cut off because
someone saw an increased call volume!
:)
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Cheers,
Matt Riddell
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in Israel would be seen as selling it.
HmmmI've offered my line but haven't used bellster to make any
calls. Is that still reselling? What about if someone comes in off the
street and says they've been in an accident. Are you not allowed to let
them use your phone?
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Cheers,
Matt Riddell
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