Un-top-posting...
On Sun, 23 Jan 2011, Michelle Dupuis wrote:
Is it possible to have a call file enter the dialplan, and then
initiate 2 outbound calls and then bridge them?
On Sun, 23 Jan 2011, Steve Edwards wrote:
A call file can specify a channel and a context/exten/priority
moments until I am able to provide you with exceptional service
today or tomorrow depending on the length of our call queue, you know?
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Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867
the extensions.conf that your Asterisk is
configured to read?
3) Do you start Asterisk with the ? command line option?
4) What is the value of 'astetcdir' in asterisk.conf?
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On Wed, 19 Jan 2011, Steve Edwards wrote:
3) Do you start Asterisk with the ? command line option?
? = '-C'
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Newline
]
or
1. mumble-mumble [pbx_ael]
or both?
(pbx_config means extensions.conf, pbx_ael means extensions.ael)
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Newline
to extensions.ael.
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' the next time
Asterisk is restarted.
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+Extension+Matching
Should get you started.
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. Also,
a 'show dialplan|dialplan show' for the executed context may yield some
clues.
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Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline
from 'XXX' to extension
'012345' rejected because extension not found in context
'DLPN_DialPlanX'.
2-If user dials 0 waits for the signal, and then dials 12345 then it
works fine.
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Steve Edwards
show 2103@context-from-previous-command' show?
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,
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Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
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On Fri, 14 Jan 2011, Tom Rymes wrote:
While we're at it, can someone please tell me whether I should be using vi or
emacs? ;-)
What 'rymes' with flame bait?
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,
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Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
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on hand, but 1.2 1.6 use set().
Also, just a suggestion to make your dialplan more maintainable, check out
the 'n' priority instead of explicitly numbered priorities.
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Steve Edwards sedwa
the priority on the 'macro' line.
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On Wed, 12 Jan 2011, Steve Edwards wrote:
On Wed, 12 Jan 2011, Gary Kuznitz wrote:
I currently have in extensions.conf:
exten = 106,1,Set(CALLFILENAME=${TIMESTAMP}_${CALLERID(num)})
exten = 106,n,Monitor(wav,${CALLFILENAME},m)
exten = 106,hint,SIP/106
exten = 106,Macro(stdexten,106,${HINT
) The Asterisk source code. Even if you aren't a C programmer grepping
through the source code can be very productive.
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Newline
,
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,
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Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
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does.
Matching is facilitated by patterns.
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,
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Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
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was on.
Any chance sipsak or sipp can handle this task? I reboot my aging SPA3K
using an http request via wget.
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Newline
,
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Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
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,
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for
'bind' in sip.conf.
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?
How about:
sip show registry
sip show peer server
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and interaction between sip.conf and
extensions.conf you will be in a better place to evaluate the merits of
using a GUI to create your dialplan or continue growing your own.
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.'
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,
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Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
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,
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Un-top-posting...
On Sun, 5 Dec 2010, Thomas Perron wrote:
Any reason why I don't get audio on the channel after it rings and the
end user picks up.
exten = s,n,Dial(SIP/callwithus/44,120,A,(demo-thanks))
On Sun, Dec 5, 2010 at 8:34 PM, Steve Edwards asterisk@sedwards.com
:0.0.0.0
unless you know what you're doing.
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decoding MP3s over and over? If you decode the files to
[wav|ulaw|slin|xxx] the files will 'just work' and you'll have more cycles
for more fun stuff like handling calls.
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Steve Edwards sedwa
On Thu, 2 Dec 2010, Steve Edwards wrote:
What does 'sip show settings' show? The first 2 settings (1.6.2.5) should
be:
UDP SIP Port: 5060
UDP Bindaddress:0.0.0.0
On Thu, 2 Dec 2010, Gary Kuznitz wrote:
In sip.conf bindport = 5060
'Sip show settings' doesn't work
to be 'unixfied?'
What does 'hexdump -C sip.conf' look like?
Does commenting (';') out line 1 change anything?
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Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline
in
separate directories, but all on the same development box.
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Un-top-posting...
On Sat, 27 Nov 2010, Giuseppe D'alessio wrote:
Hi guys, I'm using Asterisk 1.4 and i need a way to hangup all channels.
On Sat, 27 Nov 2010, Steve Edwards wrote:
2) Write a script to do asterisk -r -x 'core show channels', parse the
output and do asterisk -r -x
of a channel. Are you suggesting the OP
write an AGI so he can call into his system to ask it to hang up all
channels?
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Newline
,
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${CHANNEL_NAME} for
each channel.
3) Write a script to do #2 using AMI.
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,
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Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
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From: Steve Edwards asterisk@sedwards.com
I'm still not sure what you are asking for. Are you wanting to pick up a
SIP [hard|soft] phone, dial an extension and hear yourself talk? If so,
the relevant configuration files will be:
On Sat, 27 Nov 2010, Ali Khalfan wrote:
yes
,
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New
,
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as the console log from a call coming in on each channel
will help in assisting you in resolving this issue.
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Newline
in advance,
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Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
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'phone home' either without
permission or with the permission granted somewhere in the silly EULA that
nobody reads.
Am I missing something really obvious?
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Steve Edwards sedwa...@sedwards.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve
Edwards Sent: Wednesday, November 17, 2010 2:45 PM
Would you object to a 'curl' request in the script that starts Asterisk
that sends your MAC address and Asterisk version number to Asterisk.org?
Not tracking your IP
' runs a daily script that emails a tarball of the current
configuration to me.
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Newline
,
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On Sun, 31 Oct 2010, Matt Darnell wrote:
We have downloaded some royalty free music but it sounds 'fuzzy' when
we
test it with the system.
On Sun, Oct 31, 2010 at 5:34 PM, Steve Edwards asterisk@sedwards.com
wrote:
Can you post a link to the original?
On Sun
Un-top-posting...
On Sun, 31 Oct 2010, Matt Darnell wrote:
We have downloaded some royalty free music but it sounds 'fuzzy' when we
test it with the system.
On Sun, Oct 31, 2010 at 5:34 PM, Steve Edwards asterisk@sedwards.com
wrote:
Can you post a link
,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
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something or
mistyping an important detail.)
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On Sun, 31 Oct 2010, Matt Darnell wrote:
We have downloaded some royalty free music but it sounds 'fuzzy' when we
test it with the system.
Can you post a link to the original?
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-
Steve Edwards
On Wed, 27 Oct 2010, Nile Kaledon wrote:
You should use Read() instead of Background() component.
We conf file weenies call them applications.
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Steve Edwards sedwa...@sedwards.com Voice
On Wed, 27 Oct 2010, Nile Kaledon wrote:
You should use Read() instead of Background() component.
On Wed, 27 Oct 2010, Steve Edwards wrote:
We conf file weenies call them applications.
On Wed, 27 Oct 2010, Danny Nicholas wrote:
Is that like a Perl Weenie?
Yes, and you can proudly wear
appreciate any help that you guys can
give. Thanks!
Read whereever-you-keep-your-asterisk-sources/doc/README.variables.
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,
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Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
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is why everybody keeps adding me to their
MUA kill lists...
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On Mon, 18 Oct 2010, Jigar Joshi wrote:
@Gilles here are my requirement.can you please help me .
On Mon, 18 Oct 2010, Steve Edwards wrote:
Are you putting this out to bid or are you just too lazy to read ATFOT
(http://downloads.oreilly.com/books/9780596510480.pdf)?
On Sat, 23 Oct 2010
the email.
Off the top of my head...
exten = *,n,system(echo Call to ${DNIS} failed | mail -s 'Call failed'
r...@localhost)
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day? What if you get hit by a truck on your way
in and the heater runs continuously until...?
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Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline
anyone have some clue on how to fix this.
You didn't show the console log of an outgoing call, but I suspect the
problem is with your carrier.
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are using to convert the file?
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in Asterisk security or are you just asleep at
the wheel?
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(wav). You only have to be concerned with
supplying audio encoded appropriately for the first hop in your delivery
path.
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happy.
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in an AGI you can parse this output and do something useful
with it like mute the user or return the user number to your dial plan as
a channel variable
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,
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and
guides on configuring their gear to work with Asterisk.
In a restaurant environment, I'd be looking for some sort of faceplate
overlay to make cleanup easier.
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when loaded.
You could restart iptables when your dynamic host name changes and it will
be resolved correctly with the new IP address.
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=
debug,dtmf,error,event,notice,verbose,warning
Please remember to change them back and reload when you have identified
your problem(s).
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On Wed, 22 Sep 2010, Jose P. Espinal wrote:
If you are using a script you could get the version with something like:
tar -tf asterisk-1.4-current.tar.gz | head -n1
You need a '-z' in there.
--
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On Wed, 22 Sep 2010, Jose P. Espinal wrote:
If you are using a script you could get the version with something like:
tar -tf asterisk-1.4-current.tar.gz | head -n1
On 09/22/2010 11:20 AM, Steve Edwards wrote:
You need a '-z' in there.
On Wed, 22 Sep 2010, Kevin P. Fleming wrote:
Modern
?
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,
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would probably extract
the channel count from UNIQUEID and use something like:
mixmonitor(${ANI}-${EXTEN}-${STRFTIME(${EPOCH},,%Y-%m-%d-%H-%M-%S)}-${X}.wav)
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the way it was
before Xmas or Why did you charge me $XXX 6 weeks ago Tuesday.
*) I'll leave it as an exercise for the reader to preserve the access and
change dates :)
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, you just get to
compete for excess bandwidth with your neighbors.
You are confusing DSL with cable.
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Newline
(MySQL?) database.
The external database approach would let you cobble up a web page so
somebody other than you can maintain it :)
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,
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Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
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Supermicro boxes last year for $150 each including shipping.
My client was reluctant until I pointed out they cost less than the sales
tax on the Compaq/HP box he was thinking of.
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,
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Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
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yet. IMO, trapping HUP is a prerequisite for a well written
AGI.
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Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
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back to the music.
On 29 August 2010 01:29, Steve Edwards asterisk@sedwards.com
wrote:
Check out externalivr().
On Sun, 29 Aug 2010, Julian Lyndon-Smith wrote:
Not sure how this would allow the caller to ff / rw the file currently
being played - would that portion have to be written
to
the caller, using playback (so that they have ff / rw etc), and when
finished, go back to the music.
Check out externalivr().
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,
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on a setting, how long
will it take the next guy to figure out when the setting disappears
unexpectedly?
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,
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for more info.
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.
4) Read http://catb.org/esr/faqs/smart-questions.html again.
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. We'll need to know the number in the
programming to know where to send the files. Basic stuff?
On Wed, Aug 25, 2010 at 5:22 PM, Steve Edwards
2) Use TDM instead of SIP.
3) Google for Hylafax.
On Wed, 25 Aug 2010, Terry wrote:
1. 10,000 numbers. 2,000 DID, 8000 toll-free. Max 70 concurrent
add them to my
repertoire.
*) I'm sure there's more nuggets in there but my eyes are glazing over.
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Newline
.
On Thu, Aug 19, 2010 at 11:01 PM, Steve Edwards
asterisk@sedwards.com wrote:
exten = _x.,n, set(FOO=XXX)
exten = _x.,n, execif($[${ANI} =
551212],set,FOO=YYY})
On Thu, 19 Aug 2010, Sherwood McGowan wrote:
Oh boy...Ok, first, let's
:)
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and decrement cruft in a race condition free sort of way.
Both methods still leave a small window of opportunity in comparing the
count with the threshold.
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On Fri, 20 Aug 2010, Sherwood McGowan wrote:
Good point my man...You drinking yet?
Let me check to see if I still have a pulse -- yep!
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