,
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://www.asterisk.org/techtips
Thanks!
-S
Steve Sokol
Asterisk Marketing Director
Digium, Inc.
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look like?
I use this with good results:
sox ${INPUT} -c 1 -s -w -r 8000 ${OUTPUT}
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Besides, I feel that FreeSwitch is the most stable.
I like 1.2 so I went with Callweaver for many installations.
Thanks,
Steve Totaro
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-S
Steve Sokol
Asterisk Marketing Director
Digium, Inc.
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or that the call failed to answer, not that
somebody terminated the call.
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you describe where this would be needed and could not be
accomplished with existing tools like ssh and sudo?
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the output is unreliable. Kind of hit or miss,
sometimes you get more that you expect, sometimes less.
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for the event: http://www.asterisk.org/techtips
I hope to see you there!
Cheers,
-S
Steve Sokol
Asterisk Marketing Director
Digium, Inc.
PS. If you would like to suggest a Tech-Tips topic or would like to present a
tutorial, please let me know. We're always looking for cool new things you can
do
Hi,
Has anyone seen G.711.0 in real world use? The spec was published quite
a while ago, but as far as I can tell there is no RFC defining the SDP
and RTP details needed to deploy it, and nobody advertises that they
support it in their products.
Steve
handle.
I can then encode with all the codecs I need.
If I ever get to where I can use HD codecs, I still have the originals
from the studio.
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Steve Edwards sedwa...@sedwards.com Voice: +1
reason
2) Routing - Sometimes devices cannot route to each other directly
3) ITSP calls. Many SIP providers will not accept a redirect
and I am sure there are many more...
Cheers,
Steve
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On Fri, 4 Mar 2011, Steve Edwards wrote:
I'm starting a new project similar to a previous project where I used
OpenSER to front a bunch of Asterisk servers.
Now that OpenSER is gone, OpenSIPS and Kamailio seem the likely
candidates.
I'm leaning towards OpenSIPS because it's in EPEL so I
,
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Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
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. Attachment is my
extensions.conf
On Mon, Mar 7, 2011 at 10:16 AM, Steve Edwards
I don't think Jeremy intended for you to copy his example literally.
Do you really have your endpoints pointed at
'[outbound-or-wherever-you-dial]?'
I suggest you take a step back and read 'Asterisk: The Future
for your AGI library to see how to access the AGI
environment variables -- the cruft Asterisk writes to the STDIN of your
AGI before any of your requests.
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.
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think the name sounds more 'professional' when
discussing architecture with clients :)
Which do you use and why?
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apart?
If you post a link to a sample input file and a 'degraded' output file,
this may provide more clues.
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, an AGI,
where I have full access to the database API and real debugging tools.
I think database commands in the dialplan are just ugly.
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Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867
using a 'tin cans
and string' mesh with carrier pigeons for out of band call signaling and
having a problem with poop buildup on the endpoints -- I might propose
using Asterisk :)
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-
Steve Edwards
,
Steve T
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in [many|most|all]
countries.
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On Wed, 2 Mar 2011, sean darcy wrote:
That would be a great idea, but would stretch my limits.
Isn't that what makes it fun?
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,
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Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
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to make it work for any of the common chips used in
those devices. Its not hard to do, though. Source code exists which is
not a million miles from that required to hook a USB winmodem into DAHDI.
Steve
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this kit (http://nerdvittles.com/?p=720)
is pretty hot stuff. Sangoma makes a 2 FXO port USB thingy that looks
interesting.
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with it. Personally, I use C because it's
the sharpest tool in my toolbox.
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doesn't allow empty loops.
Bash has a thing about syntax too. Note you're not 'done' with your second
loop.
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,
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Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
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in advance,
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Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
,
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Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
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help please
No details, no help.
Crank up verbosity on the CLI and see if the messages yield a clue. If
not, please post the console messages.
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Steve Edwards sedwa...@sedwards.com Voice: +1
.
While the documentation on the protocol is clear, nobody gets it right the
first time -- which is why I always suggest using an established library
for the language of your choice.
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Steve Edwards
*Bump* No takers? Perhaps no-one else thinks this is a bug?
Regards,
Steve
On 7 February 2011 16:45, Steve Davies davies...@gmail.com wrote:
Hi,
The following IAX config (slightly edited) causes an issue for me in
version 1.6.2.16.1, where my CDR data is unreliable.
[user1]
type=friend
,
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Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
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and the channel is destroyed.
I'm guessing you would have better luck kicking off an external process
that checks the channel status via AMI.
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Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867
The action and username lines were followed by pressing ENTER.
The secret line was followed by pressing ENTERENTER.
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external and 6 internal lines??
There re USB and Ethernet devices (Xorcom, Sangoma, Sipura/Linksys/Cisco,
and others) that can interface analog phones to your Asterisk server.
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On Fri, 18 Feb 2011 09:19:01 -0800 (PST), Steve Edwards
The action and username lines were followed by pressing ENTER.
The secret line was followed by pressing ENTERENTER.
On Fri, 18 Feb 2011, Gilles wrote:
Thanks for the tip. I figured this out after a while ;-)
I can now successfully
you're not trapping?
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investigate this
elsewhere but report back about the solution.
I also tried this with a 6757i and a 6753i with no problems (blind and
attended) on Asterisk 1.6.2.16.1. Have you updated the handset
firmware to 2.6.0.2010?
Cheers,
Steve
be the only one there.
What you are describing looks to me like a third party controlled
transfer, and not a barge-in at all.
I suspect that the Asterisk Manager API action Redirect will be your friend.
Regards,
Steve
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On 15 Feb 2011, at 03:39, Jeff LaCoursiere wrote:
minipbx*CLI show uptime
System uptime: 41 years, 7 weeks, 6 days, 3 hours, 26 minutes, 46 seconds
Last reload: 8 hours, 3 minutes, 51 seconds
What's the highest current 'genuine' one on-list?..
klein*CLI core show uptime
System uptime: 2
paying it. The destination, GUID, CDRs all stored in a database. We
also recorded using Orecx and and tied that into the same database and
had full integration with the home brewed CRM.
Thanks,
Steve Totaro
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On 15 Feb 2011, at 13:17, Richard Kenner wrote:
Of course not! It would be useless if that were the case: the whole
point here would be that you need the master encryption key.
Here's a possible design:
- There's optionally a file in the config
directory called master_key. It contains
dramatically.
I note there have been changes since then (128.0.0.0 was assigned to RIPE
back in November), so if anybody wants to 'refresh' and post changes,
please do.
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On 11 Feb 2011, at 22:37, Danny Nicholas wrote:
In 500 words or less (if possible), please explain what is a legal
music-on-hold file?
Depends on the country, and what licence you posses. Googling 'countryname
hold music regulations' may help.
S--
/asterisk/sounds/' to your
path yielding:
/var/lib/asterisk/sounds/home/abejide/Desktop/a.*
is this what you want?
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Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
On Sat, 12 Feb 2011, ayodele abejide wrote:
I am having problems playing files with the playback command...
And don't hijack other people's threads :)
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,
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the first time.)
If you crank up verbosity and debug do you get any clues?
The CLI command 'agi set debug on' may also yield some clues.
--
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-
Steve Edwards sedwa...@sedwards.com Voice: +1-760
dialplan should include extensions like:
exten = s-BUSY,1, verbose(1,[${EXTEN}@${CONTEXT}])
exten = s-BUSY,n, ...
--
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-
Steve Edwards sedwa...@sedwards.com
On 8 Feb 2011, at 13:30, Shariq Khan wrote:
Can i add ${HANGUPCAUSE} in CDR after the Dial command using h extension? I
want to add the Hangup reason of call in userfield of CDR.
http://www.google.com/search?q=asterisk+hangupcause+cdr
Top result... Should do it
Steve
On 8 Feb 2011, at 14:52, mehran khajavi wrote:
i searched a lot but i couldn't find the answer
.
i have two openvox(fxo/fxs) card so I have 24 ports!
Ok!
on first card i have 12 fxs and on the second i have 12 fxo
i want to then one person calling from dahdi/13 forward it to dahdi/1
when
a plausible voice adjustment.
On Sat, Feb 5, 2011 at 9:44 PM, Steve Underwood ste...@coppice.org
mailto:ste...@coppice.org wrote:
On 02/06/2011 05:39 AM, Bruce B wrote:
Hello,
Are there any other other voice changer applications to
Asterisk other than the one from
that was used to authenticate the call ie.
IAX2/user2-; I specifically put a password onto [user1] so there
is no possibility that the call is authenticating there.
Am I missing something? Or is this a bug?
Thanks,
Steve
+channels
While there is a lot of out of date crap out there, www.voip-info.org is
still a valuable resource.
--
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as well.
It might help if you explained the kind of change you would like to
make, which the lobstertech module doesn't offer.
Steve
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);
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On 30 Jan 2011, at 09:21, Pezhman Lali wrote:
Faxter is an opensource email to fax gateway,
please check it, let me know if any bug.
Only bug i can see is the attitude of the developer...
As for the bugs, having the config variables liberally scattered throughout the
script makes it's use
,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
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the absolute timeout on a channel
so a caller won't consume all of your 'prepaid' (nothing is free) minutes
and drive you into unexpected charges.
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-
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ideas.
Best of luck!
murf
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(test,s,1)
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in advance,
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Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
On 25 Jan 2011, at 09:36, Andrew Thomas wrote:
Try changing 'hostname=127.0.0.1' to 'hostname=localhost' in the
cdr_mysql.conf. I seem to remember a problem I had when '127.0.0.1' and
'localhost' didn't marry up never did find out why.
I believe localhost means it can use a socket, where as
On Mon, Jan 24, 2011 at 3:37 PM, Steve Edwards
asterisk@sedwards.com wrote:
One of my clients is complaining that their customers that use U-verse
(and other cable providers) for telephone service cannot enter credit
card numbers reliably.
The issue not all digits are received in my
' when they switched.
If every SIP connection failed wouldn't you know it by now?
If it were SIP...
My client guestimates it may affect up to 20% of their customers.
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On Mon, 24 Jan 2011, David Backeberg wrote:
Can you record a few calls just to confirm the problem?
On Mon, 24 Jan 2011, Steve Edwards wrote:
If I record via mixmonitor(), I just get a bunch of clicks where the DTMF
should be. I'm assuming this is because the DSP has already taken the DTMF
in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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
be back up now. Use 0.0.6pre18
Steve
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On 22 Jan 2011, at 18:02, Carlos Chavez wrote:
Cannot allocate memory
Have you tried looking at memory?
S
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On 01/21/2011 08:37 PM, Tom Rymes wrote:
On Jan 20, 2011, at 8:52 PM, Steve Underwood wrote:
A comparison wouldn't be complete without mentioning Hylafax. Hylafax has a
great infrastructure - tools for integrating with Windows clients, and so on.
Neither spandsp or the Digium FAX code can
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
to the second caller?
Receivefax can handle hundreds of calls at one time, if your machine's
resources are up to it? Why would there be a restriction of one call?
Steve
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to test with the code you intend
to deploy.
Steve
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On 20 Jan 2011, at 17:13, Andrew Thomas wrote:
Sorry about this - testing this disclaimer problem :)
I can give you a POP3 account on my server if it stops you spamming the list?..
S
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On 01/21/2011 06:46 AM, Bryant Zimmerman wrote:
On 01/20/2011 11:47 AM, Steve Underwood
On 01/20/2011 11:11 PM, Kevin P. Fleming wrote:
On 01/19/2011 02:30 PM, Bryant Zimmerman wrote:
On 01/19/2011 02:05 PM, Bryant Zimmerman wrote:
I am working on some fax tools for some of my users. I am
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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]
or
1. mumble-mumble [pbx_ael]
or both?
(pbx_config means extensions.conf, pbx_ael means extensions.ael)
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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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.
Can anyone that is not affiliated with Digium post their stats and
reports from users using T.38?
Thanks,
Steve T
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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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show 2103@context-from-previous-command' show?
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On 17 Jan 2011, at 11:29, Hans Witvliet wrote:
Missing something obviously,
core dump / backtrace? ;)
Might be worth knocking a few of the modules out that were listing errors to
see if any of them are causing it. It's possible something not loading isn't
being handled gracefully.
S
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that I always recommend a
few POTS lines for 911 and faxing, most clients are cool with that
once you explain that there could be a liability issue.
Thanks,
Steve T
Thanks,
Steve T
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