in this format:
You may want to read the man pages for curl and wget -- both can submit
POST requests.
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broken
loose in the server room, but it¹d be nice to have something a bit more
accurate and reliable.
How about a loop with Please press pound to continue?
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Steve Edwards sedwa...@sedwards.com
Happy Friday everyone,
Is there a way to log the negotiated codec that was used for each call
in CDR or in a separate log file?
This is for SIP-based calls, if that matters.
Perhaps there is some variable that can be queried as part of the
dialing script;
Or is it possible to grab the codec
productive citizens, led by the mysterious John Galt, progressively
disappear,* not devotees of Ayn Rand and her philosophy.
*) http://en.wikipedia.org/wiki/Atlas_Shrugged
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From: asterisk-users-boun...@lists.digium.com on behalf of Steve Edwards
Continuing to veer off-topic...
Maybe you should re-read Atlas Shrugged.
On Fri, 9 Jul 2010, Mike Ely wrote:
And no thanks: I've already read that execrable book, and found it to be
nothing more than overwrought
:
First thought is that you can put a timeout on your calls, but that is
just a band-aid.
Also not fixing the source of the problem, but rtpholdtimeout and
rtptimeout may help.
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users and they have read access to the file.
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at all times. I want a relationship where they want to do
business with me, not where they have to do business with me because I
have them over a barrel.
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file. Here you again have full power
of programming language in you hand.
Won't show dialplan, sip show [peers|users], etc. and a bit of
scripting undo most of this security.
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specific scripts)
/usr/local/etc/openser/dispatcher.list
/var/spool/cron/*
~/.my.cnf
across multiple projects and many hosts.
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work to the degree allowed
and lets the next guy have a starting point when the bus hits him.
The BOBW is to provide real value to continuing the relationship, not
holding your customer hostage.
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,
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On 6 Jul 2010, at 10:34, Jonas Kellens wrote:
what is the use of realtime SIP peers when you always need to reload the sip
configuration as if you were just putting your SIP peers in sip.conf ??
Did you enable caching by any chance?
S
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variable.
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-${EXT}})
exten = s,n,dial(${EXT-${EXT}})
exten = s,n,hangup()
Off the top of my head and untested, but I think this may work for you.
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On 2 Jul 2010, at 12:29, John Novack wrote:
regardless, people will post either way, and wasting archive space
complaining about either one is pointless.
I was mainly pissed off about him directly replying to people (i.e. me) rather
than the list. troll It was you lot that started the
prompts you found on the net but one of
them had a funky header and it exposed a hither-to unknown bug?
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On Fri, 2 Jul 2010, Gordon Henderson wrote:
The file is at http://unicorn.drogon.net/firewall2
Lots of cool stuff in here. It's going to take a bit to understand it
all :)
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On 1 Jul 2010, at 15:52, unsero...@aol.com wrote:
[Jul 1 16:30:50] ERROR[3954]: pbx.c:2902 ast_func_write: Function
CONNECTEDLINE not registered
Same happens trying function CALLEDID.
I am using Asterisk 1.6.1.20.
What do i have to do to use this function or alternatively the
On 1 Jul 2010, at 16:25, unsero...@aol.com wrote:
Sorry, what does this mean? Only in trunk?
If you look in the post you quoted
This feature is in Asterisk trunk and will be present in the upcoming 1.8
release.
First sentence.
S
--
On 1 Jul 2010, at 16:56, unsero...@aol.com wrote:
Sorry, i wanted to know what is in trunk means.
So it seems to mean is in the pipeline for the next version.
DON'T reply to people off list. And stop bloody top posting.
Steve
On 30 Jun 2010, at 13:48, Gareth Blades wrote:
By ITSP do you mean a SIP provider?
ITSP: Internet Telephony Service Provider
S
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the heck out of trying to
debug an AGI by peppering it with VERBOSE or syslog() statements.
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on Local/*...@custom-callfwd/n for application
Playback(hello-world) (Retry 1)
What does the call file look like before you mv it to the spool directory?
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set which results in that.
Steve
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On 28 Jun 2010, at 13:08, Jerry Geis wrote:
It works fine with I call the SIP phone directly - however -
when I first call the Local channel - then Dial the SIP phone
the SIPADDHEADER doesnt seem to do anything.
Are you adding the header before or after you dial the local channel?
S
--
On 28 Jun 2010, at 15:36, Jonas Kellens wrote:
Does this mean I have a patched asterisk ? (I ask this because some
applications require a non-patched asterisk version)
Yes.
What is then the unpatched version of Asterisk 1.4.30 ??
The one you have before you apply the patch?..
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whatever it is that you are doing
that causes you to double-post EVERYTHING?
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the response
hanging in STDIN where it is subsequently read by the GET VARIABLE
request.
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On 24 Jun 2010, at 12:49, Jonas Kellens wrote:
It seems as if some SIPaccounts could register and others could not. I don't
think a firewall distinguishes between phone brands or SIP accounts.
Alas 'stabbing in the dark' is all we can do until you actually provide some
information for us.
://www.sedwards.com/class-a-block-list
If you don't need to receive packets from far away places, it's a great
start.
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On 23 Jun 2010, at 18:39, Steve Edwards wrote:
Ouch. 82.0.0.0/8 is on my block list, available at:
http://www.sedwards.com/class-a-block-list
Would advise people in the UK do not use that list... 82.0.0.0/8 would block a
reasonable chunk of my users for starters..
Steve
On 23 Jun 2010, at 19:26, Steve Howes wrote:
On 23 Jun 2010, at 18:39, Steve Edwards wrote:
Ouch. 82.0.0.0/8 is on my block list, available at:
http://www.sedwards.com/class-a-block-list
Would advise people in the UK do not use that list... 82.0.0.0/8 would block
a reasonable
On 23 Jun 2010, at 18:39, Steve Edwards wrote:
Ouch. 82.0.0.0/8 is on my block list, available at:
http://www.sedwards.com/class-a-block-list
On Wed, 23 Jun 2010, Steve Howes wrote:
Would advise people in the UK do not use that list... 82.0.0.0/8 would
block a reasonable chunk of my
195.0.0.0/8
Please try again.
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is the Public Switched Telephone Network.
So, wouldn't it be more accurate to say no PRIs or POTS since both are
used to connect to the PSTN?
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Steve Edwards sedwa...@sedwards.com Voice: +1
3) Click on Astribank Drivers
Seemed pretty obvious to me. Am I missing something?
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?
I'm a big fan of compiled languages like C. You can execute XXX AGIs
written in C in the time it takes to load an interpreter and parse a
script.
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,
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(I
personally use JacORB). Check
http://www.lumenvox.com/partners/digium/applicationzone/projects/javaPizza.aspx
for some sample code on the Java part - this happens to drive Lumenvox to
handle
a call, but you can easily hack it to stick CORBA calls in.
Steve
,
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and inefficient way.
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On Fri, Jun 18, 2010 at 06:43:09PM -0700, Steve Casto wrote:
CENTOS 5.5
dahdi 1.4.3.0.1
dahdi-linux 2.3.0.1 ?
uname -r
2.6.18-194.3.1.el5PAE
[root at localhost dahdi-linux-2.3.0.1]# service dahdi start
Loading DAHDI hardware modules:
FATAL: Module dahdi not found.
wct4xxp
(which will come back to bite you later) or you have
identified a bug in Asterisk (unlikely, but we would all benefit from it's
resolution).
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is also a good resource.
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addresses in /etc/hosts?
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at *-domain when you have an issue.
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to the kernel
source I get this doing make:
You do not appear to have the sources for the 2.6.18-194.3.1.el5PAE
kernel installed.
thanks for the help
Steve Casto
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On 17 Jun 2010, at 15:58, Mike wrote:
I have a Cisco SPA525G latest firmware, and very often when I attempt a
transfer I get a network error message when I press Dial on the transfer. I
never get that erroron a simple call out Asterisk is configured for that
phone exactly the same as
-- an unexpected echo or print that
violates the AGI protocol, a syntax error, etc.
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debugging and watch the console output for clues.
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, but I don't know if anyone has publicly
released the code they use to integrate them.
Steve
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@lists.digium.com
*Subject:* [asterisk-users] TDD/TTY Support
On voip-info I found a few dated references to TDD support being in
the alpha stage and buggy.
Can anyone direct me to any newer information on this option?
Thanks
--
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Steve
in advance,
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into STDIN, but it cannot interact with the running instance of Asterisk.
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New
for failures or inconsistencies that may explain why you getting
different behaviors.
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asterisk first
installation.
After reading a couple of articles, if you still need help, please post
your questions with more meaningful subjects -- better bait yields better
fish.
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Steve Edwards
The last time Digium gets a dime from me.
-- Forwarded message --
From: Steve Totaro stot...@totarotechnologies.com
Date: Fri, Jun 11, 2010 at 3:44 PM
Subject: Re: [INSTALL #RKZ-745226]: Digium Support Survey, Partial Faxes
To: supp...@digium.com
Lets start with the stats first
advise.
Hunting is a telco feature.
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,
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Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
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giving the extension as a parameter to the AGI script if
you cannot get that from the included request variable.
Better to fix the really simple stuff before he gets to the complex stuff.
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equivalence is.)
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experience
with some Qwest PRIs a couple of years ago.
Then I noticed -- it was me :)
I guess I had too many beers.
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the evaluation of $sip_peer.
Try something like this:
system('asterisk -rx sip show peer ' . $sip_peer . ' | grep -c X-Lite')
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Steve Edwards asterisk@sedwards.com
I'm a 1.2 Luddite, so I can't tell you more about it.
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you a clue.
EAGI implies you are doing something with the incoming audio on FD 3. Any
chance you are closing FD3?
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version of Asterisk?
1.6+ has the new confbridge feature that may be of use. I'm a 1.2
Luddite, so I can't tell you more about it.
2) Why is meetme unacceptable?
3) Why is parking unacceptable?
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Steve
brought up
http://www.voip-info.org/wiki/view/DID+Service+Providers
as the first link. You didn't say where you are in the world, but
sipgate.[at|com|de] seems to be popular.
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,
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On Fri, 11 Jun 2010, Martin wrote:
if you know IP then ban with iptables
iptables -A INPUT -s IP -j REJECT
Ever play http://en.wikipedia.org/wiki/Whac-A-Mole ?
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Steve Edwards sedwa
Orlando, FL 32819
m...@accessgate.net
Office Toll Free: (888) 227-9337
Fax: (407) 352-2717
From: Steve Howes steve-li...@geekinter.net
Sent: Thursday, June 10, 2010 4:26 AM
To: d...@accessgate.net, m...@accessgate.net, jsny...@accessgate.ne
Subject: Re: [asterisk-users] Out of Office
On Thu, 10 Jun 2010, Zeeshan Zakaria wrote:
Shouldn't a moderator block emails from this email address, maybe
temporarily?
There is no moderator.
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heavy like bzip2'ing your
database dump or compiling Asterisk from source, there are still several
CPUs available for Asterisk.
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Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867
is
called when an extension is dialled.
If you enable AGI debugging, this may give you a clue. Posting the console
output may help someone help you.
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for a live introductory webinar every Thurs:
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To UNSUBSCRIBE or update options visit:
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On 8 Jun 2010, at 16:40, Jonas Kellens wrote:
I noticed that changes to realtime sip peers are not applied until a
'reload'. A 'sip reload' does not make any changes to realtime sip peers.
sip prune ?
S
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caller ID and see what
their response is.
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in advance,
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Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
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On Mon, 7 Jun 2010, Adil Zaaraoui wrote:
Hello Steve,
Thanks again for the effort.
I tried your dialplan like this in my extention:
exten= 777,1, Goto(absolute-timeout-test,777,1)
[absolute-timeout-test]
exten = T,1, verbose(1,[${CONTEXT}:${EXTEN}])
exten
getChannel().setChannelVariable('SECONDS-REMAINING', '60'). In the C
library I wrote, it is 'agi_set_variable(STATUS, SUCCESS).'
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,
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Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
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at the dial command documentation. The second parameter is the
timeout, the third is the options.
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this morning
and can't explain why coherently.
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.
Can you lookup the peer and the remaining minutes in your AGI, set these
values in channel variables and then set the timeout and dial your peer in
your dialplan?
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On Sat, 5 Jun 2010, Adil Zaaraoui wrote:
I want to write an AGI script doing this:
1-user call a number.
2-asterisk call the agi script
3-the script dial the peer
4-if the call is answered, let the call up for 1min
5-then the script hangs up the channel.
On Sun, 6 Jun 2010, Steve Edwards
while loop may help with
CPU resource consumption.
6) More detail like what version of Asterisk and some console output with
AGI debugging enabled may help.
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Steve Edwards sedwa...@sedwards.com
something in features.conf is tripping you up. I don't use
features so this is just a guess.
Maybe you could bump up debugging and verbosity and reply with the console
output.
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Steve Edwards sedwa
On 3 Jun 2010, at 14:24, Necati Demir wrote:
I want to ask how to get call duration.
Go on then
When you do ask the question you might want to include a few details. Are you
trying to get call duration during a call? If so then the cli will help 'core
show channels'. If it's after the
provided explicitly
for displaying output on the console (verbose()) rather than using the
obtuse side effect of an application whose name does not translate well
for non-native-English speakers.
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and
is more explicit. Try it -- the next guy will thank you.
I also fight windmills in my spare time :)
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On Thu, 3 Jun 2010, Steve Edwards wrote:
-Also, what about slow queries? If a query takes a few seconds to
complete, does the call wait for the query to complete or are there
timeouts for the query that could result in dropped calls?
(I prefer to call MySQL from an AGI instead
, always able to
interwork with a codec that does support it.
G.729AB or G.729A/B are the usual ways people described a codec which
uses the Annex A version of the encoding and decoding, and which
supports CNG/VAD.
Steve
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