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6975 Union Park Center, Suite 450
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1-800-678-3748 ext. 124
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I think it's only being tested with GSM right now, but I'm not aware of
any reason why you couldn't use another codec. Maybe Mark can enlighten
us with some details?!? (Thanks Mark for implementing this! I know I'll
use it a LOT!)
Jared
On Mon, 2003-03-17 at 03:04, Roy Sigurd Karlsbakk wrote:
easily push data across the
network at night when there aren't any calls.)
Hope that helps...
Jared Smith
On Thu, 2003-03-20 at 10:57, David Luyens wrote:
Hi,
I would like to use * as a compression box.
Between 2 sites I have an E1 leased line.
So would it be possible to use 1 port
the
G.729 code on a trial basis?)
Jared Smith
On Thu, 2003-03-27 at 14:01, Lenny Post wrote:
I'm personally more interested in the performance of the codec (ie what kind of raw
power will I need to run it, how many can I run at once on a decently powered box
etc...)
Lenny
My dual-proc Xeon boxes didn't share IRQs across CPUs until I installed
the kernel-utils RPM and made sure the irqbalance service was
running... Just a word to the wise!
Jared Smith
On Tue, 2003-06-10 at 09:52, [EMAIL PROTECTED] wrote:
H, I to appear to have an odd mix of interrupts
To execute the s extension automatically when you pick up the phone,
you need to put that channel in immediate mode. (I'd tell you how to do
it, but I can't remember the syntax off the top of my head.)
Jared Smith
On Tue, 2003-06-10 at 09:57, Thomas Haeger wrote:
Hi all,
i have read
The Cisco 7960 (or 7940 for that matter) is about the nicest hardphone
I've seen so far... I don't think you'll be disappointed on features.
The price, however, might disappoint you :-)
Jared
On Wed, 2003-06-25 at 13:44, Chris wrote:
I've got Asterisk up and running nicely using a couple of
I've had problems with RedHat 9 and Asterisk... You might want to try
downgrading to RedHat 8 or using another distribution. (I think the
problems might be related to the NPTL threads stuff.)
Jared
On Wed, 2003-06-25 at 13:52, The Traveller wrote:
Heya Mark (and others),
Here's an update on
I've never tried it with SIP, but I have faxed between to asterisk boxes
on the same network via IAX and IAX2. The secret was to set the codec
to ulaw or alaw. (Certain codecs, such as GSM, compress the data too
much for the fax machines to be able to communicate effectively.)
Jared
On Thu,
If I understand correctly, each codec has a cost associated with
converting it to another codec. I would think that Asterisk would
choose the lowest-cost codec first, but I'm not positive that's what
happens.
Jared Smith
On Thu, 2003-07-03 at 11:35, Eric Wieling wrote:
Which codec
There's got to be a way... I think zttool shows a red alarm on an X100P
when there's no phone line plugged into it (and I would guess when
there's no voltage on the line.) My guess is that it gets the info from
/proc/zap-something-or-other, but I'm just guessing.
Jared
On Tue, 2003-07-08 at
You may also want to check out http://www.cepstral.com/. Their voices
(especially the domain voices) sound as good as I've ever
heard.
(Now it's just a matter of getting Asterisk to talk to it...)
Jared
On Tue, 2003-07-15 at 13:50, Florian Overkamp wrote:
Hmm that sucks, and it pretty much
Would you mind giving us a few examples on how we can make festival
sound better? (Some sample festival configs would be nice!)
Jared
On Tue, 2003-07-15 at 14:41, Chris Albertson wrote:
--- Jeff Noxon [EMAIL PROTECTED] wrote:
Many of you are familiar with how lousy Festival sounds.
ATT
Unfortunately, I've found several problems with Asterisk running on
RedHat 9. (Most of my problems only happened under high call volume.)
For that reason, we've rolled back to RedHat 8 on all of our servers.
It's worked great for us.
Jared Smith
On Sat, 2003-08-02 at 12:55, Scott Stingel
problems
debugging the threads because of the NPTL threads stuff. He thought it
might be some kind of a race condition.
We rolled back to RedHat 8.0 and have seen a lot fewer problems. Feel
free to catch me on the IRC channel (my nick is jsmith) if you want more
details.
Jared Smith
On Mon, 2003-08-18 at 10:48, John Brown wrote:
Can I use a x100p from digium to receive
and originate calls to the PTSN ??
Yes.
Jared Smith
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On Mon, 2003-08-18 at 14:59, Brian West wrote:
Maybe we can pester kram to make that an option. monitor.conf anyone?
bkw
Well, while we're in the let's pester Mark mood... why not have him
fix res_monitor so it writes to just one file! That would sure make me
a lot happier...
Jared Smith
packet loss on either leg) with silence.
This is certainly the right thing to do, whether or not we combine the
two channels inside of res_monitor.
Maybe we should make it a configuration option? Or maybe I should just
shut my trap and go back to manually mixing the files?!?
Jared Smith
In order to get the version to update, you can run make update in the
asterisk directory, instead of doing a cvs update.
Jared Smith
On Fri, 2003-08-22 at 20:01, Andres wrote:
When I checkout the latest asterisk version, then do a make clean and make
install, shouldn't the show version
Checkout the dialplan.xml file...
Jared Smith
On Mon, 2003-08-25 at 16:45, Ray Burkholder wrote:
No, this is not the case currently with any of the Cisco SIP software
loads that I am aware of. If you find this to be incorrect, please
let the list know. Cisco has not deployed much
I upgraded to the latest and greatest from CVS today, and now SIP pickup
groups appear to be broken. Can anyone else tell me whether or not
they're seeing the same problem. If anyone out there can verify the
problem, I'll submit it as a bug.
Jared Smith
Yes.
Jared Smith
On Wed, 2003-08-27 at 13:38, Bartosz Jozwiak wrote:
hello,
Is this configuratoin possible:
--FXO
--FXO
On Wed, 2003-09-03 at 10:20, Paul Lambert wrote:
When multiple calls are in session between two IAX servers do the voice
frames from the various calls get put into a single packet to conserve
on total packet rate?
If you are using IAX2 trunking, then the answer is yes.
Jared Smith
Or for those of you who are visually inclined:
http://www.jaredsmith.net/misc/cables/
Jared Smith
On Mon, 2003-09-08 at 07:49, Linus Surguy wrote:
And if we are going to get carried away: RJ45 CAT5 CAT5e
- Original Message -
From: Thilo Salmon [EMAIL PROTECTED]
To: [EMAIL
I've seen it a bunch... but Cisco claims it's not happening. :-(
Jared
On Mon, 2003-09-08 at 11:04, Travis Johnson wrote:
Hi,
We are having a problem with Cisco 7940 and 7960 phones when the PC is
plugged into the 2nd ethernet port on the phone. It will drop the PC's
connection for about
Have you tried opening a case with Cisco? Just a thought...
(If you do find the answer, please post it to the mailing list. I'd
really appreciate it.)
Jared Smith
On Mon, 2003-09-08 at 11:28, Travis Johnson wrote:
Hi,
We tried that already. No difference.
Travis
Daryl G. Jurbala
On Mon, 2003-09-08 at 11:28, Travis Johnson wrote:
Hi,
What was your fix? A seperate cable to each phone and PC? :(
Yup... thank goodness we ran 4 Cat5 cables to each station.
Jared
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I have the same problem with Cisco 7960s. I have found however that if
you do the *8 from a Zap channel, then the called phone stops ringing.
But if I do *8 from a SIP device, the called phone continues to ring and
ring for at least a minute.
I think Mark is working on fixing this (or at least
with a Service Pack?). Outlook
2000 (which most Outhouse.. er, I mean Outlook users are using) does not
block the executables. In fact, they can be run just by previewing the
message.
Unfortunately, this is not the time or place for the Outlook flame
war...
Jared Smith
misunderstanding... this company resells used Cisco
equipment. I don't think they actually used these phones, they're just
reselling them.
Jared Smith
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hurry up and fix it.
Jared Smith
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You probably have a zapata.conf file specifying a channel. Just a
guess...
Jared
On Thu, 2003-09-25 at 13:38, Senad Jordanovic wrote:
Hi,
I have enabled ztdummy in order to have * compile it.
I can modprobe ztdummy with no problems.
The sole reason, i need ztdummy is to heve musiconhold
to use 4 fxo's from a TDM400 card, sound gets lousy. If I
manually destroy one of the zap channels (e.g. zap destroy channel 4),
sound gets good again.
Did you check interrupts? Try cat /proc/interrupts and see if
anything is sharing an interrupt.
Jared Smith
, because I've never actually tried four
cards in the same box at the same time.)
Jared Smith
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I think you need [EMAIL PROTECTED], where voicemailcontext is
your voice mail context. (I'm assuming you're using VoiceMail2.)
Hope that helps...
Jared Smith
On Fri, 2003-10-03 at 12:01, Babak Pasdar wrote:
I have a cisco 7940 with the following sip.conf config:
[Desk1.1]
type=friend
over time.
I've got trunk = yes, and it works for me.
Jared Smith
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) and I'll pass it along.
Jared Smith
I know it's bad form to reply to yourself (especially after the thread
has been dead for a while now), but I thought I'd let everyone know I
finally hacked together the app and posted it at
http://www.jaredsmith.net/misc/cisco7960/Directory-0.1.tgz. It's
?) Please help me before I blow something up!
Don't use an ethernet crossover cable... You need at T1 crossover
cable. Please check out http://jaredsmith.net/misc/cables/ (one end
should be straight through and the other end T1 Crossover).
Jared Smith
to get it to run with Asterisk. (In other words, the
version of festival that came with RedHat *will not* work.) You also
need to be sure you start the festival server (as explained in step 3 of
the README.festival file) before you start Asterisk.
Jared Smith
. People have shown interest in what I am doing, and here is the
evidence that it is not vapourware.
[snip]
Thanks Steve! I know a lot of us have been anxiously waiting for
something like this, and I just wanted to publicly thank you for your
time and effort.
Jared Smith
if calls are actually going down the trunk is
to type iax2 trunk debug at the Asterisk CLI when you have calls going
between server A and server B.
Jared Smith
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On Tue, 2003-10-21 at 09:44, Ryan Tucker wrote:
Does anyone have a quick and dirty script for defragmenting mailboxes?
[snip]
Note the gap between 0003 and 0009. This is caused by a somewhat common
situation, and it tends to bite us somewhat often. :-)
If not, if I get a chance, I'll
I know this is going to sound like a strange question, but here goes:
Does anyone know of a SIP softphone that has either a button or a
programmable soft-key to send the current call to VoiceMail?
I'd appreciate any ideas you might have.
Jared Smith
,
that you use PHP version 4.3.0 or later, due to the updated CLI stuff.
Feel free to contact me off-line if you'd like some examples.
Jared Smith
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things. While they might not
have as much user-contributed code as CPAN, PHP is certainly a lot more
than just perl lite.
/me runs off to put on some flame-resistant clothing
Jared Smith
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Check out the ZapBarge application if you're wanting to do this on Zap
channels.
Jared Smith
On Wed, 2003-10-29 at 13:51, John Haigh wrote:
Hi,
Is it possible to listen in on an existing call already, say between
the caller and callee? So the 3rd person would listen to the caller
It's my understand that they are db levels. (And, if I remember my
electrical engineering classes from college, a 3db increase effectively
doubles the volume.) I hope that helps...
Jared Smith
On Thu, 2003-10-30 at 11:28, Dan wrote:
Hi,
For me, in order to get the same sound level
with the motherboard. It's works
great. (And yes, I'm using it in a 2U rack-mount case.)
Jared Smith
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(and myself) have been busy working on the second edition of
Asterisk: The Future of Telephony. Work is progressing on the Cookbook,
it's just not coming along as quickly as they'd like.
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looking for.
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?
You can define a channel variable in your call file, and that channel
variable will be exposed to the dialplan. Simply put a line in your call
file that looks like:
Set: variablename=somevalue
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, although it's not
perfect either. Hopefully the Asterisk development community will
eventually get around to rewriting much of the AMI actions to make their
output easier for programs to parse.
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the asterisk console straight from startup:
I'm probably asking the obvious here, but were you able to successfully
register your codec with the Digium registration server? Hase your
ethernet MAC address changed since you registered the codec?
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to
match the new Asterisk 1.4 settings.
Another common problem is that a couple of new items have been added to
asterisk.conf, so I typically renamed asterisk.conf before installing
1.4, so that I get the new version of asterisk.conf as well.
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buy them from Digium's website at
http://www.digium.com/en/products/voice/g729codec.php
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. This is a PCI card, and will work in either a 3.3
volt or 5 volt PCI slot.
[1] http://www.digium.com/en/products/hardware/te120p.php
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really make this clear.
I agree. Luckily, I know Digium's marketing department is working to
improve the information on the website so that it's clearer which
hardware is appropriate for different situations.
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, but I
haven't received any kind of estimate on how long this might take. (As
I understand it, the bandwidth for the service was being provided by a
third party, so the box may not be readily accessible.)
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.
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On Thu, 2007-07-19 at 12:15 -0400, Nitesh Divecha wrote:
Does the same procedure works for updating Zaptel, Libpri, and
Asterisks-Addons?
Yes, you should be able to install the new version over top of the old
version without any problems.
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for vpmocto64
didn't turn up much.
The TE220 is a very new card, so I'm not sure there's a lot of info
about it on Google yet.
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the
; prompt has finished
exten = 1,1,SayDigits(1) ; say one
exten = 1,2,Goto(s,1) ; go back to the menu
exten = 2,1,SayDigits(2) ; say two
exten = 2,2,Goto(s,1) ; go back to the menu
Hopefully that will get you started in the right direction.
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using Asterisk over 5
years ago).
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religion. In short, we hope to see you at
AstriCon, whether or not you choose to check out Digium/Asterisk World!
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. That should give you
some more advanced DTMF information.
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On Tue, 2007-07-24 at 10:48 -0600, Joseph wrote:
I have one Digium adapter S101I on a local network and I'm losing the
connection periodically.I'm using Asterisk-1.2.21.1
I know there have been some recent fixes to the IAX implementation in
Asterisk to handle a few scenarios when IAX
the softswitch what your Asterisk box's IP address is, but doesn't have
anything to do with you sending calls to the softswitch.)
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() dialplan application. Check
out more information at http://www.voip-info.org/wiki/view/Asterisk+cmd
+DISA.
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would have to have a pattern
match set up in your [international] to dial the international call
through your VoIP provider.)
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not sure that's much help, but maybe it'll spark
someone else's memory.)
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extension '0' in context 'from-zaptel' on Zap/10-1
You seem to be having DTMF recognition problems here... If I'm reading
this right, Asterisk only saw the 0 when you dialed 305.
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On Wed, 2007-07-25 at 17:25 -0400, Alejandro Acosta wrote:
May I do call report using the pinset instead of the extension?. I
mean, to know how many call where made using the pin code X.
You can map the pin code to the accountcode field in the CDR records.
The easiest way to do this is by
.
Hopefully that helps clarify things!
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On Wed, 2007-07-25 at 15:06 -0700, Peter Hessler wrote:
;; dialtone in the background isn't there any more
;; dialed '305'
;; everything from here is exactly as expected.
OK, I missed this in the first email you sent... Asterisk is playing
dialtone *on top* of the background message the first
() application. In your case, you might want to add a custom
queue log entry every time the caller rejoins the moh queue, saying
something to the effect of Caller XYZ has rejoined the moh queue for
the 10th time or something like that.
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there when
searching for a channel next time (round-robin)
R - round-robin, highest to lowest
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On Fri, 2007-07-27 at 06:26 -0700, bilal ghayyad wrote:
For example: what is the best (shortest) way to search
for information related to the command playbak()?
I find that the fastest and most up-to-date information regarding the
dialplan applications is the online help in the Asterisk CLI.
starting Asterisk.
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. As calls go out to your SIP provider,
assign them to a particular group, and check the group count. If the
group count is higher than what your SIP provider allows, then play
congestion (or some other message) back to the caller.
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On Fri, 2007-07-27 at 11:45 -0400, Baji Panchumarti wrote:
Any plans for a sequel ? I'll order 10 copies in advance :-)
Yes, the second edition of the book will be out very soon now. I'm glad
to hear you enjoyed the book. Hopefully you'll like the second edition
even better. :-)
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always add a VPMOCT064 module to it,
but it doesn't come bundled with the card.)
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bring that up so that someone didn't mistakenly open up their
firewall for TCP traffic instead of UDP traffic and wonder why IAX
traffic wasn't making it through.
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On Mon, 2007-07-30 at 21:45 +0800, GNUbie wrote:
Where can I find a list of description for each sound files provided
by the asterisk-sounds-main Debian package?
The file core-sounds-en.txt should contain the text of each of the sound files.
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.
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. :-)
On the serious side, one might see it as an example of how easy it is to
add a new CLI command to Asterisk (complete with compiler flags, command
auto-completion, and the whole nine yards).
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/my_iax_peer/5551212,30)
Obviously my example isn't that robust... it's simply meant to
illustrate the idea. (It depends on the SPA3102 returning a status code
that maps to CONGESTION if it's already in use... I don't have an
SPA3102, so I can't tell you how it actually performs.)
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On Tue, 2007-07-31 at 09:00 +0200, Florian Arthofer wrote:
So, if my ISDN-number is for example 1234567, then i should, if i dial
1234567-123, see _something_ on the console and at least i should hear
it ringing on the phone i place the call with. Am i right?
OK, ordinarily this would be true,
On Mon, 2007-07-30 at 09:31 -0400, Jared Smith wrote:
The second major difference between the cards is echo cancellation. The
TE212P comes with an echo cancellation module installed, while the TE220
card comes without one. (You can always add a VPMOCT064 module to it,
but it doesn't come
.
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algorithm.
[1] You can read the code for that function at
http://www.asterisk.org/doxygen/1.4/app__queue_8c.html#94119419d819a6af1b06f79ed4133192
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be in apps/app_queue.c near line 1390.
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on the right-hand side of the page.
If you know of someone who we should invite to speak, we're open to
suggestions as well. Feel free to email me (off the list, please!) with
your recommendations.
I'm looking forward to a great conference, and hope to see you all
there!
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Community
is the CPE or PBX side.
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probably the
easiest (and a great way to get started with Asterisk, I might add).
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the network. Asterisk has an IP address for
the peer and is trying to call it, but the peer isn't responding.
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that clarifies things for you.
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if you
called the Congestion extension directly in your example), then it won't
be set.
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captured. You can either use Wireshark itself to do the
network capture, or you can capture the traffic with tcpdump and then
pull the file into Wireshark at a later time.
Inside Wireshark, go to Statistics, RTP, Show All Streams, and then
select a stream and hit the Analyze button.
--
Jared
-voipcare-for-asterisk.html
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Jared Smith
Community Relations Manager
Digium, Inc.
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a replacement card.
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Jared Smith
Community Relations Manager
Digium, Inc.
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