On Thu, 18 Jun 2015, Matt Riddell wrote:
Did you buy the number from your carrier?
I prefer using 'rent' instead of 'buy' :)
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so you could
only make calls between extensions.
3) For the IP phones, check out ebay.com. Last year, I picked up 3 Polycom
SP 501's for $20.00 each. A little dated, but a great phone.
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On Mon, 15 Jun 2015, Steve Edwards wrote:
Although, if you lose power, you've probably lost your Internet
connection as well so you could only make calls between extensions.
And you would lose the Italian equivalent of 911. In the US, everybody
over the age of 6 has a cell phone stapled
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manageable.
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!!) installed on my Server...
Maybe fiddling with the SIP and RTP ports would help.
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...
-- Call accepted by X.Y.Z.K (format gsm)
-- Format for call is gsm
I thought GSM regurgitated by cell had issues. Can you try alaw/ulaw?
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Steve Edwards sedwa...@sedwards.com Voice: +1
of expertise, so if you think I've
misconfigured something on my side, please let me know.
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= MYSECRET
type= peer
5) I'd try and move more of the common settings to general, but these were the
ones listed on voip-info.org.
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/documentation/howtos/how-to-enable-distinctive-ringing-alert-info-for-calls-from-particular-
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) [pbx_config]
Note the format of my verbose() arguments. It makes it easy to
'cut-n-paste' in a 'dialplan show' command.
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Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867
On Fri, 29 May 2015, Steve Edwards wrote:
; admin functions
exten = _[456],1, verbose(1,[${EXTEN}@${CONTEXT}!${ANI}])
exten = _[456],n, gotoif($[TRUE =
${ADMIN}]?meetme-star-admin-menu,${EXTEN},1)
exten = _[456],n, goto(enter-room
On Fri, 29 May 2015, sysad...@reed-media.com wrote:
You may get better replies if you start a new thread rather than replying
to an unrelated thread.
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On Sun, 17 May 2015, martin f krafft wrote:
You know the Henry Ford quote about faster horses, right? ;)
I do now :)
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Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
On Sun, 17 May 2015, martin f krafft wrote:
also sprach Steve Edwards asterisk@sedwards.com [2015-05-16 23:22
+0200]:
I use a preprocessor
(http://software.hixie.ch/utilities/unix/preprocessor/) to tailor
dialplans and configuration files to each host based on the client (or
project
,execif($[${ACCEPT-COUNTER} 0],hangup)
#endif
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On Mon, 27 Apr 2015, Bryant Zimmerman wrote:
exten = _9XXX,n,Set(dialnumber=${l_HomeAreaCode}${EXTEN-1})
Missing a colon?
${EXTEN:-1}
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Steve Edwards sedwa...@sedwards.com
On Mon, 27 Apr 2015, Chad Wallace wrote:
On Mon, 27 Apr 2015 14:30:07 -0700 (PDT)
Steve Edwards asterisk@sedwards.com wrote:
On Mon, 27 Apr 2015, Bryant Zimmerman wrote:
exten = _9XXX,n,Set(dialnumber=${l_HomeAreaCode}${EXTEN-1})
Missing a colon?
${EXTEN:-1}
Does
monitor() call path and your mixmonitor()
call path?
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server?'
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as '=' when used in the dialplan. Personally, I always
use '=' as the dialplan doesn't seem to be the place for some object
oriented mumbo-jumbo -- at least to my 'C programmer till I die' eyes.
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, but I don't
remember if it was device or server discovery.
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idle.
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=${PATH}\
$ASTERISK $START_OPTIONS
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' and
'switch' in the same sentence seems confusing.
Do VOIP devices take more table entries than other Ethernet devices? I.e.
more than 1?
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Steve Edwards sedwa...@sedwards.com Voice: +1-760
= n,Dial(SIP/102,40,t)
same = n,Hangup()
Remember to set 'callcounter = yes' in sip.conf.
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be a better tool.
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at the exact same time every day. The dialplan coding
is easier and no pesky codes to remember.
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On Tue, 18 Nov 2014, Eric Wieling wrote:
Other than a few minor patches, we use stock phpagi.
Can you spare me a flat spot on my forehead and share the wealth?
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On Tue, 18 Nov 2014, Eric Wieling wrote:
diff at http://pastebin.com/wfDR6u0a
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.
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Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
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the off-list
exchanges. My apologies in advance if I've misinterpreted the situation.
I've copied the parties to the off-list exchanges in case they would care
to clarify my assumptions.
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same = n, execif($[${DNIS:0:1} =
+]?set(DNIS=${DNIS:1}))
same = n, execif($[${DNIS:0:1} =
1]?set(DNIS=${DNIS:1}))
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a URL or from a memory buffer (either C or PHP)?
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On 09/23/2014 02:17 PM, Steve Edwards wrote:
For some applications, storing recorded audio (prompts and caller
recordings) as a BLOB in MySQL has advantages.
On Tue, 23 Sep 2014, Don Kelly wrote:
I'm curious about what the advantages are of storing audio in a blob.
Wouldn't it be more
4 digit extensions, all the better.
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On Sun, 7 Sep 2014, Steve Edwards wrote:
In specific, your ordering of '_xx' in the middle of 's' is odd. This would
disrupt the value of the priority in older versions of Asterisk, but it
appears that it does work in modern (I'm using 11) versions.
Disregard that. I can't even follow my own
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For future reference, a well chosen subject will yield more relevant
replies.
Better bait == better fish.
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On Mon, 25 Aug 2014, Patrick Laimbock wrote:
https://wiki.asterisk.org/wiki/displa/AST/Introduction+to+Local+Channels
s/displa/display/
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Steve Edwards sedwa...@sedwards.com Voice: +1-760
-paste into the 'dialplan show' CLI command.
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what callers are doing to reproduce the error
reliably yet.
Any clues or suggestions?
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'get_lock/release_lock.'
If you're using some other database, see what locking features you have
available.
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'slow' from time to time.
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Please don't top post.
Please keep the thread only on the list.
On Thursday, July 31, 2014 12:16 AM, Steve Edwards
asterisk@sedwards.com wrote:
I'm running Asterisk 11.11.0 on an HP ProLiant DL320e Gen8 E3-1240v2.
1,300 calls with no audio issues.
On Wed, 30 Jul 2014, babak wrote
as 729, do you have gobs of excess CPU capacity? If not, you'll
just be trading 1 resource limitation for another.
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at 460,000 blocks per second. I can remember when 10,000 was big :)
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On Tue, 22 Jul 2014, Steve Edwards wrote:
How about something like:
asterisk -rx core show channel SIP/spa841-0003\
| awk '/Call Identifer/ {gsub(/[][]/,); print $3}'
Or:
asterisk -rx core show channel SIP/spa841-0003\
| awk -F
, this is a warning, not an error.)
2) Please show a few more lines of console output (with verbose and
debug set high) to give us some context.
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Colin wrote:
Can you explain?
I'm guessing 'core show application MixMonitor' should give you a good
start.
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,${EXTEN},1)
So are the quotes now a requirement? (Quotes makes it look like you are
comparing strings instead of numbers to my old eyes.)
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. Most script-kiddies
won't bother to 'port-scan' to identify the new port number.
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On Wed, 7 May 2014, Meriem Abid wrote:
salut,
je suis entrain de developper une application...
You will have better luck if you can post in English.
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a file system that isn't
maintaining atime/ctime/mtime/etc as expected, like NFS?
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it from happening, how about 'exten =
s,2,hangup()?'
If you're asking how to hang up the channel while it is in a loop, what
have you tried? Does 'channel request hangup' help?
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Steve Edwards sedwa
, Steve Edwards asterisk@sedwards.com wrote:
If you're asking how to prevent it from happening, how about 'exten =
s,2,hangup()?'
Note that you also could have added the 'missing' priority and reloaded
your dialplan and the hangup would have been executed on the next
iteration of the loop
output (with 'core agi set debug
on,' 'core set debug 99,' and 'core set verbose 99') of a successful and a
failed call.
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in the toolbox.' If C is not
in your 'wheelhouse,' use PHP or coughJava/cough. You (and the next
guy who gets to enhance and maintain this application) will be glad you
did.
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On Wed, 23 Apr 2014, Steve Edwards wrote:
I tried database access in the dialplan using the mysql() application
years ago, just to confirm I was right and I was :) What an ugly,
messy, fragile dialplan.
On Wed, 23 Apr 2014, Doug Lytle wrote:
With FuncODBC this is no longer an issue. All
,
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() or confbridge().
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of 512 on an Intel E3-1240v3 with
'pre-packaged' Asterisk so I'm good for now.
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version. Currently we're connecting AMI
to a message bus and passing events across the bus.
What do you see as the advantages of a message bus (dbus?)?
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), but I'd lose some functionality and have to re-write parts of
my application.
Any clues of what limit I'm hitting and how to increase it?
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On Fri, 21 Mar 2014, Steve Edwards wrote:
The call file on box1 originates a call to box2 and then plays a 2 hour
WAV file.
The call file on box1 originates a SIP call to box2 and then plays a 2
hour WAV file.
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On Fri, 21 Mar 2014, Steve Edwards wrote:
Googling 'dahdi pseudo channel limit modprobe' showed the secret sauce.
Oops. Guess I should complete the thread...
You can set the DAHDI pseudo channel limit in /etc/modules.conf:
options dahdi max_pseudo_channels=x
or you can set it from
On Fri, 21 Mar 2014, Steve Edwards wrote:
I'm trying to make the leap from 1.2 to 11.8.1
On Fri, 21 Mar 2014, Steve Totaro wrote:
Is there any good documentation on that process?
I haven't looked. I know they added a few of variables to the AGI
environment Asterisk passes to your AGI
is top to bottom, left to right, answers
follow questions.
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you to do what you need to do --
even if the next dialplan priority just says 'goto h.'
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hours') you are limiting your potential audience.
Posting late on a Friday afternoon can be an exercise in futility.
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? I run tons of AGIs, but I write them in C.
Does 'vmstat 5' show swapping?
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