On Mon, 2006-06-19 at 18:45 -0400, Gary Reuter wrote:
On 6/19/06, Daniel Salama [EMAIL PROTECTED] wrote:
Is there anyone that could explain to me the phenomenon of Echo or
at
least point me where I can learn more?
This paper by Cisco is a great start: Echo Analysis for Voice over
IP
Anybody have any recommendations? IAX service preferred.
-Seth
On Tue, 2006-04-18 at 15:48 -0400, Wes Baehr wrote:
Well this is disappointing. Time to find somebody else...
--
Wes
-Original Message-
From: NuFone Operations [mailto:[EMAIL PROTECTED]
Sent: Tuesday, April 18,
Hello All,
I saw an add in my latest Linux Journal advertising Sangoma's new AA
series of FXO/FXS analog cards with on-board echo cancellation, but I
can't find any information at all on them. Even the link given in the
advertisement is a dead end as far as I can tell. Anybody else
seen/heard
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On Fri, 2005-07-15 at 20:44 -0700, Chris Coulthurst wrote:
Being one the many people trying to track down echo 'ghosts' I ran across
this page:
http://lists.digium.com/pipermail/asterisk-users/2004-November/071301.html
suggesting ways to adjust the gain. I have a TDM400P 2x2 config with
On Wed, 2005-07-13 at 23:34 -0700, Jason Walker wrote:
Has anyone had any luck in changing the voices for Festival and Asterisk?
I have Festival installed and working, but can not get the voice different
from the default.
Thanks,
Jason
Well, it's been a while since I had to do this
CVS HEAD instead of stable you can set the transfer
key(s) in features.conf. There is also a double ## patch floating around
somewhere for STABLE I think.
-Seth
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me know.
Regards,
Bharat M. Sarvan
app_skel.c has all the boilerplate code you need.
-Seth
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has been working fairly well for me.
http://www.xten.com/index.php?menu=productssmenu=download
-Seth
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On Thu, 2005-06-23 at 23:19 -0400, Michael Di Martino wrote:
I am trying to setup voicemail for my iaxy device, however,
i cannot get it to work voicemail never picks up. Below is my config.
Am i doing anything wrong here
From my Extensions.conf file
exten =
representation patch to
ztmonitor which isn't in stable.
-Seth
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http
heard others
complain about beeping with a Sipura ATA. Had something to do with it
incorrectly detecting DTMF when there was none. I believe it was fixed
in the newer firmware. You might try updating the firmware and see if
that fixes it.
-Seth
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started the process from within gdb. I've never tried to
attach to an already running Asterisk process.
-Seth
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stuff.
-Seth
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On Mon, 2005-03-28 at 17:58 -0400, Chris Mason wrote:
How should I get asterisk to start automatically on system restart?
http://www.voip-info.org/wiki-Asterisk+Starting+and+Stopping
Look at the section Starting Asterisk at Boot Time.
-Seth
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://www.asternic.org/ if your users can stand running a separate app
on a PC. You can use it in conjunction with call parking to sort of
simulate what you are looking for.
-Seth
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On Wed, 2005-03-23 at 14:30 -0500, Dan Morin wrote:
Does anyone know if the X100P clone cards provide the timer needed to
run MOH and the Conferencing service?
Yes, it will work fine.
-Seth
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On Fri, 2005-03-18 at 16:59 +0100, Reuben Grech wrote:
Dear All,
I am listening to blips during conversations when I have an incoming
call from an X100P card. This does not happen on all conversations.
Any clues? :)
Turn off call waiting in zapata.conf
callwaiting=no
-Seth
Seth
to be running firefox as root to install it. Make sure
you close all open firefox windows before to try to run as root. If you
have a firefox window open that was started from userland it will attach
to that process which would not have root privileges.
-Seth
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to remember anything.
-Seth
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I am getting compile errors when trying to compile speechtools...
Any help would be greatly appreciated,
Isaac
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On Wed, 2004-12-22 at 08:41, John Hill wrote:
Question:
Do you need to answer to detect a fax?
Yes. You need to answer the line so the calling fax will start sending
the fax tones and * can detect them.
-Seth
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On Mon, 2004-12-20 at 13:45, Me wrote:
What does t mean in a CDR entry?
The 't' probably means that the call ended up in the timeout extension.
-Seth
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or
so.
Sounds like it's working correctly. Broadvoice set's their registration
timeout pretty low AFAIK. My * boxes re-register every 20-30 seconds as
well.
-Seth
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with modifications
=
Olle also has an updated patch for CVS stable (1.03) at
http://edvina.net/broadvoice/patch.shtml
-Seth
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On Fri, 2004-12-10 at 20:02, Dealer Backup Admin wrote:
Received errors as follows.
snip
Are you using version 1.0 or CVS HEAD? The patch will probably only
apply cleanly on the 1.0 branch.
-Seth
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to work
with the host=proxy.XXX.broadvoice.com setting as they describe. I've
always had to set it to host=sip.broadvoice.com. I'd be interested in
how it works for you.
-Seth
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/wiki-Asterisk+s+extension
-Seth
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... it should be set to
dtmfmode=inband for Broadvoice.
-Seth
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On Thu, 2004-12-02 at 20:38, Gary wrote:
Now the question is can the IAXy be used to successfully drive and ADSI
phone ?
ADSI is currently only supported in Zap channels so the answer is,
unfortunately, no.
-Seth
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On Fri, 2004-12-03 at 10:09, Andrew Kohlsmith wrote:
On December 3, 2004 09:56 am, Seth Remington wrote:
On Thu, 2004-12-02 at 20:38, Gary wrote:
Now the question is can the IAXy be used to successfully drive and ADSI
phone ?
ADSI is currently only supported in Zap channels so
On Mon, 2004-11-22 at 17:03, Peter Svensson wrote:
Shouldn't the echo canceler cut out when it detects the 2.1kHz guard tone?
The comments in ecdis.h and zaptel.c seens to indicate that.
Yes it does. I see CLI output telling me so every time I receive a fax.
-Seth
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FAILED at 69.
2 out of 2 hunks FAILED -- saving rejects to file Makefile.rej
Make sure you are trying to patch the Makefile in the apps directory,
not the top-level Makefile.
-Seth
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configuration. I'm just waiting for
it to break.
-Seth
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but you can try it and see if it
works...
http://sremington.zapto.org/downloads/asterisk/spandsp/Makefile.patch
-Seth
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Do a ./configure
Asterisk doesn't have a configure script.
-Seth
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http
IP).
The setup should be identical to any other analog device. Just plug it
into a properly configured FXS port.
http://www.voip-info.org/wiki-Asterisk+and+faxes
http://www.voip-info.org/wiki-Asterisk+fax
-Seth
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Phone
that executed it.
-Seth
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or 3sec will not contain any
message and I don't want system to record them and sending an email to
me.
You were looking in the right config file. The parameter is called
maxmessage.
-Seth
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automatically.
Sorry about the confusion.
-Seth
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the patch on two separate * boxes (work and home) and both
incoming and outgoing have been working fine.
I'm using proxy.dca.broadvoice.com if that makes any difference to you.
Does sip show registry show asterisk as registered with Broadvoice?
-Seth
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.
-Seth
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the patch. Do you see the debugging registration
messages that were accidentally left in the patch on the CLI? (Assuming
you didn't take them out).
-Seth
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date: 2004/03/18 22:49:06; author: malcolmd; state: Exp; lines: +2 -0
Bug # 1247: Add patch to specify minimum vmail length and discard that
which does not meet the minimum
/quote CVS log
Grep the source you have for minmessage. If you find references to it
you should be fine.
-Seth
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]
exten = s,1,Dial,Zap/4
[outgoing]
exten = _NXX,1,Dial/Zap/1/${EXTEN:1}
This should be exten = _NXX,1,Dial(Zap/1/${EXTEN:1})
-Seth
kindly pointout my mistakes/errors and helping me out.
Thanks In Advance .
Adnan Ahmed.
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internal housekeeping.
If all of tagnames get confusing (unfortunately the tag naming standard
appears to have changed over time) just stick with v1-0.
-Seth
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but my understanding is
that it's only currently implemented in the Zap channel driver. I don't
have paperwork to back that up though :)
What are you trying to do? Run an analog ADSI phone through an IAXy or
something?
-Seth
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Botham
Client Support Consultant
Computer Software Group plc
www.computersoftware.com
Looks like you have mixed CVS versions. Here's how somebody else fixed
it...
http://lists.digium.com/pipermail/asterisk-users/2004-October/070073.html
-Seth
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__
Check the Generic heading:
http://www.asterisk.org/index.php?menu=hardware
-Seth
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. The actual
extension that gets used just depends on which context you drop the
incoming / outgoing call into.
Maybe if you give us some more specifics / config examples we can help
you out more.
-Seth
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,102,Busy
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On Thu, 2004-11-04 at 10:10, Mike Shultz wrote:
Quick Question that I hope someone can answer. Will Asterisk work
with basic PCI FaxModems instead of those expensive cards listed on
the hardware page?
No. The wcfxo driver only works with a very specific Intel/Ambient
chipset.
-Seth
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IS Flash OR Flash
FLASH
ENDKEY
...and then use it in one of your subroutines like so...
SHOWKEYS flash
Hope that helps.
-Seth
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,${CALLFILENAME},m)
exten = _8.,3,Dial(ZAP/g1/${EXTEN:1})
exten = _8.,4,Congestion
exten = _8.,104,Congestion
http://www.voip-info.org/wiki-Asterisk+cmd+monitor
-Seth
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+SetGroup
-Seth
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) as keywords for
search at http://www.voip-info.org for details.
rgds
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The command you are looking for is zap show channels. If * complains
that no such command exists then you installed zaptel after you
installed Asterisk. Recompile Asterisk and you should be fine.
-Seth
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according to ast_false().
-Seth
Also, callprogress seems to be quite buggy for lots of people, I would
try turning that off.
Jeremy
On Thu, 28 Oct 2004 15:39:58 +0800, Asterisk Mania [EMAIL PROTECTED] wrote:
my zapata.conf
busydetect=1
busycount=7
callprogress=yes
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On Wed, 2004-10-27 at 16:19, Me wrote:
Is it possible to transfer a caller to another internal extension with a
plain analog phone attached to an ATA?
Sure. Just make sure you have the 't' and/or 'T' option enabled in your
Dial command.
-Seth
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can load the core dump into
gdb and get a backtrace to see exactly where it is failing.
-Seth
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patches applies to the 1.0 branch.
BTW... if you want to convert your existing cvs HEAD over to the stable
branch you can run update -r v1-0 instead of checkout.
-Seth
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tone from within the context that you pass to it.
Just make sure you secure it enough that you feel comfortable against
unwanted access.
-Seth
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the first time a voicemail is left. Allison will
walk you through recording busy and unavailable messages once you have
logged into your mailbox. Just follow the prompts.
-Seth
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functions by modifying the
asterisk.adsi file using stuff gleaned from the app_adsiprog.c file,
but docs would be really helpful at this point.
Tia,
Dennis
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the unix command line. my
festival.conf file is set up per the wiki
Did you apply the patch in /usr/src/asterisk/contrib? The
/usr/src/asterisk/contrib/README.festival document might have some info
to help you.
-Seth
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Phone
On Sun, 2004-08-22 at 07:35, Roland Zagler wrote:
Please can someone send me the .tar.gz file of spandsp, the site is
offline and i didn't find it anywhere!
http://sremington.zapto.org/downloads/asterisk/spandsp/ until the
opencall.org DNS servers are back up.
-Seth
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On Wed, 2004-08-18 at 23:20, Adam Goryachev wrote:
On Thu, 2004-08-19 at 07:01, Seth Remington wrote:
On Wed, 2004-08-18 at 14:45, David Filion wrote:
Does anyone know of an alternate source for spandsp? opencall.org is
down and all the links returned by Google just point to the dead
/spandsp/
-Seth
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,Voicemail,u1001
exten = 8441,106,Voicemail,b1001
You have the 'm' option enabled in your Dial() command. That will play
MOH to the calling party until the called channel answers.
-Seth
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FAILED errors.
Try this patch instead:
http://sremington.zapto.org/downloads/asterisk/spandsp/Makefile.patch
-Seth
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sure to run make install from the top level Asterisk directory.
My guess is that you are running it from the apps directory. The top
level Makefile defines _GNU_SOURCE and the lack of it causes the
PTHREAD_MUTEX_RECURSIVE error.
-Seth
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. Just
another quirk of the GS to add to the list.
-Seth
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?
You can get the RC1 from cvs... just use the command cvs co -r
v1_0_rc_1. Or you can use cvs co -r v1-0_stable to get the old
stable version (not recommended).
-Seth
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--as each new post arrived
the oldest post would drop off?
Thoughts?
If you use their advanced search page
(http://www.google.com/advanced_search) you can restrict your results to
the past three, six, or twelve months.
-Seth
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and the 'T' extension when
used inside a macro?
Christopher is asking about AbsoluteTimeout so 'T' is correct.
t (lowercase) is the timeout extension in a context (set with
ResponseTimeout for instance).
T (capital) is Timeout on AbsoluteTimeout.
-Seth
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: Channel and File.
Channel is the channel that you are monitoring and want to change the
filename being recorded to and File is the new filename.
-Seth
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Seth Remington
SaberLogic, LLC
661-B Weber Drive
Wadsworth, Ohio 44281
Phone: (330)335-6442
Fax: (330)336-8559
gives the very
unhelpful help that was mentioned in the original post.
I had to grep the source in order to find the proper parameters. In fact
none of the show manager command ... CLI commands give any of the
required parameters. Some patches would probably be in order.
-Seth
--
Seth Remington
,Sample User,[EMAIL PROTECTED],,attach=yes
-Seth
--
Seth Remington
SaberLogic, LLC
661-B Weber Drive
Wadsworth, Ohio 44281
Phone: (330)335-6442
Fax: (330)336-8559
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of the function where the
error occurred an the parameter that had the out-of-bounds memory
address (if applicable). If your C skills are up to snuff to can try and
debug it yourself. If not open up a bug on the bug tracker with all of
the info you have collected.
-Seth
--
Seth Remington
SaberLogic, LLC
661-B
that are concatenated with
'' and the prompt from the first one in the list will be used.
-Seth
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Seth Remington
SaberLogic, LLC
661-B Weber Drive
Wadsworth, Ohio 44281
Phone: (330)335-6442
Fax: (330)336-8559
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On Thu, 2004-08-05 at 13:57, Wayne wrote:
Seth Remington wrote:
In voicemail.conf set up your Admin mailbox:
101 = 101,Admin Mailbox,,,delete=1
This will allow you to record your Admin prompt but the delete=1 will
auto delete the message from this mailbox. Then in extensions.conf
up
for voice if you use faxdetect.
-Seth
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Seth Remington
SaberLogic, LLC
661-B Weber Drive
Wadsworth, Ohio 44281
Phone: (330)335-6442
Fax: (330)336-8559
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with '2'. It
doesn't even read the rest of the digits '000'.
You might consider adding a DigitTimeout command at the beginning of
your IVR.
http://www.voip-info.org/wiki-Asterisk+cmd+DigitTimeout
-Seth
--
Seth Remington
SaberLogic, LLC
661-B Weber Drive
Wadsworth, Ohio 44281
Phone: (330)335-6442
and that extension looks weird to me.
Jeremy McNamara
exten = 011001,1,Answer
exten = 011001,2,Playback(tt-monkey)
-Seth
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Seth Remington
SaberLogic, LLC
661-B Weber Drive
Wadsworth, Ohio 44281
Phone: (330)335-6442
Fax: (330)336-8559
the administrator to override the default mailer command
with a defined command. Mailcmd takes a string value set to the desired
command line to execute when a user needs to be notified of a voice mail
message.
http://www.voip-info.org/wiki-Asterisk+config+voicemail.conf
-Seth
--
Seth Remington
SaberLogic
On Mon, 2004-08-02 at 12:34, Steve Hanselman wrote:
Delete it and cvs update will retrieve it.
cvs update -C app.c would also work. -C retrieves the clean copy
from the repository and saves your local changes into another file.
-Seth
--
Seth Remington
SaberLogic, LLC
661-B Weber Drive
in.)
make[1]: *** [app_rxfax.o] Error 1
make[1]: Leaving directory
`/usr/src/install/cvs/2004-07-29/asterisk/apps'
make: *** [subdirs] Error 1
You need to make sure that _GNU_SOURCE is defined or else you will get
this error.
-Seth
--
Seth Remington
SaberLogic, LLC
661-B Weber Drive
Wadsworth, Ohio
sprinkled through the code (specifically the adsi_process() function).
-Seth
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SaberLogic, LLC
661-B Weber Drive
Wadsworth, Ohio 44281
Phone: (330)335-6442
Fax: (330)336-8559
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-17ae'
__SNIP__
Did you also apply the /usr/src/asterisk/contrib/festival-1.4.3.diff
patch?
-Seth
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Seth Remington
SaberLogic, LLC
661-B Weber Drive
Wadsworth, Ohio 44281
Phone: (330)335-6442
Fax: (330)336-8559
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of FDM mode. Comedian Mail is
sort of a combination of the two. It downloads and stores some stuff
(like softkeys) the first time you connect, but it is also SDC in the
way it transmits the number of messages you have and the name and
callerid of each one.
-Seth
--
Seth Remington
SaberLogic, LLC
661
}) /* use the zap */
exten = _91NXXNXX,2,Congestion
The above example would allow the sales people to dial 9 to dial a long
distance number on the Zaptel card. The context you use for admin would
explicitly use the vpb card instead.
-Seth
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Seth Remington
SaberLogic, LLC
661-B Weber Drive
://sremington.zapto.org/weblog/2004-07-04_14.52.21.html
-Seth
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SaberLogic, LLC
661-B Weber Drive
Wadsworth, Ohio 44281
Phone: (330)335-6442
Fax: (330)336-8559
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to send DTMF via SIP INFO
instead of in-audio.
-Seth
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Seth Remington
SaberLogic, LLC
661-B Weber Drive
Wadsworth, Ohio 44281
Phone: (330)335-6442
Fax: (330)336-8559
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Seth Remington
SaberLogic, LLC
661-B Weber Drive
Wadsworth, Ohio 44281
Phone: (330)335-6442
Fax: (330)336-8559
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)
[global] should be [globals]
-Seth
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SaberLogic, LLC
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Phone: (330)335-6442
Fax: (330)336-8559
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and a different one in vpb.conf.
-Seth
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Seth Remington
SaberLogic, LLC
661-B Weber Drive
Wadsworth, Ohio 44281
Phone: (330)335-6442
Fax: (330)336-8559
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On Tue, 2004-07-27 at 09:55, Harry Schechter wrote:
Nicholas Bachmann wrote:
Is there a good place to find Asterisk consultants?
There is a list here --
http://www.voip-info.org/wiki-Asterisk+consultants
-Seth
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Wadsworth, Ohio 44281
Phone: (330)335-6442
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