The scheduled guest is JR Richardson of Nntegrated Solutions in
Dallas. He wrote a widely-consulted white paper on the subject and I
hope we can get background as well as answers to any questions we may
have, so come on by:
http://www.AsteriskUsersConference.org
Here's a powerpoint of his
Nitesh,
I've messed with the Lumenvox starter kit. If you are serious about
this field, I think it's a must see. It was easy to set up and there
are demos available. Their support is excellent. There is a quiet
mailing list where questions are never ignored and most problems are
solved AFAIK.
On 8/8/07, Dean Collins [EMAIL PROTECTED] wrote:
Hmm beginning of the end of free trixbox by the sounds of it.
Dean, I thought you were on the conference call when Kerry discussed this in
detail. There is no plan to dump the free version as I understood it.
/r
ps to all: the conference can
On 8/8/07, Michael J. Liberatore [EMAIL PROTECTED] wrote:
First problem, the card with 4 FXO ports is fine until there is a storm in
the area, then all 4 lines are massively static filled making phone calls
barely understandable until the system is rebooted or the zaptel modules are
unloaded
I have been a VP connect customer for a few years, mow traffic,
outgoing only. I have had very good experiences and they are usually
the lowest cost for a USA route, often less than .01/min retail.
/r
On 8/8/07, John Meksavan [EMAIL PROTECTED] wrote:
Has anybody use Voicepulse Connect for
This week, the second part of connecting to the outside world using
TDM, ATA and even... IAX hardphones with compilable software.
More on topics and guests:
http://groups.google.com/group/asterisk-users-conference
Instructions:
http://www.AsteriskUsersConference.org
IRC on freenode.net:
I received the original message at 7:01 AM today
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
This Friday, part II of TDM solutions including ATA that do IAX and
SIP without opening the box and installing a card. Your experience in
this area would be appreciated.
You can find us here:
http://www.AsteriskUsersConference.org
Also, a Google group has been created for discussions and
After a recent update I now get messages from one IAX channel about
the wrong format of the sound files when using Playback. I have tried
to force ulaw on that channel, but that doesn't stop the problem. Am I
missing something, like maybe g729 files?
Steve,
On 8/3/07, Steve Totaro [EMAIL PROTECTED] wrote:
I just tried to call in after creating an account.
After the call connects, enter the show id: 22622# and your_PIN#
I dial in and am asked for the podcast id, I enter 22622# and am told
that my passcode is not correct. I also tried
of providers.
I've had time to learn a lot about the real world of all this stuff
and I'm willing to share what I know. How about you?
On 7/29/07, randulo [EMAIL PROTECTED] wrote:
Hi,
I am going to be on the road for the next few days and with the
variable delay on the mailing list, I am
Hi,
I am going to be on the road for the next few days and with the
variable delay on the mailing list, I am posting this now, 4 days
before the conference. If you haven't yet listened or participated,
please consider doing it. We have a great kernel of people at all
levels of expertise and ideas
On 7/29/07, Don Kelly [EMAIL PROTECTED] wrote:
Note that some of us newbies have posted the same question two or three
times because we didn't see our own post (let alone a reply) in a timely
manner.
True. I could swear that when I post to biz, I get a post confirmation
message immediately but
On 7/27/07, Jared Smith [EMAIL PROTECTED] wrote:
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
On 7/27/07, dave cantera [EMAIL PROTECTED] wrote:
randulo,
I could not get into the conference today... the SIP line was busy, no
matter what I do, the website thinks I'm not logged in and gives me the
login page. after I login, anything I want to do brings me back to the
login page... so I
You can listen or join the Asterisk Users Conference Fridays at 12:30 PM
EDT
Today's subject suggestions:
FAX capabilities, what's your solution?
Multiple asterisk server implimentation: ENUM, DUNDI or even two servers
connected
Your subjects?
Share your ideas, ask your questions!
See
On 7/20/07, Andrew Joakimsen [EMAIL PROTECTED] wrote:
Do you have any patches against your Asterisk, Zaptel or Kernel?
Actually are you using anything but the factory Kernel?
I'm using an older Slackware. The problem came in March or so with
1.2.14 I think.
Besides that I just wouldn't advise
Hi John,
On 7/18/07, John covici [EMAIL PROTECTED] wrote:
I wonder what version of Zaptel you are using -- sounds like you have
not installed a new version or you are using an older one.
Actually I just made and installed zaptel 1.2.19
then made asterisk 1.2.22 and I am getting this same error
On 7/19/07, randulo [EMAIL PROTECTED] wrote:
Actually I just made and installed zaptel 1.2.19
then made asterisk 1.2.22 and I am getting this same error
Fixed!
I just noticed that I asked about this same problem in March and got
a workaround (edit makefile) from Tzafrir. Could someone explain
Here's a quick report about this:
During the entire hour, there were always around 25 people watching
the video stream. I've recorded it locally and will put the pertinent
stuff (questions and Mark's answers) up on http://asterisktv.com bit
by bit as soon as I have them edited. The audio
In conjunction with Mark Spencer's visit to our Paris office, we'll be
kicking off Asterisk TV (http://asterisktv.com) live during the weekls
Asterisk Users Conference. I believe someone from Lumenvox will be
back with us on the conference, now that I've had a chance to play
with their speech
Reminder: A few weeks ago, Stephan and Stephen were with us on the
Asterisk Users Conference which you can download the MP3 here:
http://recordings.talkshoe.com/TC-22622/TS-23423.mp3
Or you can listen to it here:
http://www.talkshoe.com/talkshoe/web/tscmd/tc/22622
Good luck with the book,
Hi all,
I'm looking for an asterisk user (can be a n00b who knows enough about
asterisk to ask intelligent questions or a brilliant specialist) to
talk about what they do with asterisk. I would like to have a co-host
next week, someone who uses video via the web (it's a Flash
application that can
Just add the softphone to the dial command. If it's not connected
nothing will bad happen
and the regular phone will ring. Whenever the softphione is registered
it will ring as well. If the other phone is a SIP phone, you could use
IAX as the softphone with the same username and password.
Hi,
Quick reminder that the conference is happening today at 12:30 PM EDT.
I'd like to talk more about updating to 1.4. I now have a test box
running asterisk 1.4.5, CentOS 5 and Lumenvox speech rec software.
Seems to be fine except for some double NAT issues that could be
router specific.
On 6/22/07, randulo [EMAIL PROTECTED] wrote:
Quick reminder that the conference is happening today at 12:30 PM EDT.
Listen to the conference here:
http://www.talkshoe.com/talkshoe/web/tscmd/tc/22622
The little round orange Listen button will open a player. You can
also just download an mp3
What phone are you trying to dial? (ZAP/1?)
What is the CLI output when you dial?
What number are you dialing? 601?
On 6/21/07, Matt Scott [EMAIL PROTECTED] wrote:
Dear All
I have a problem with a TDM400 card with 4 x FXS modules.
The card carries extensions only and there are no incoming
On 6/14/07, Matt [EMAIL PROTECTED] wrote:
Ahh I didn't see that in the first post. Yes Mr. SpamSucks is correct.
You should use 8khz @ 16bits. Using 8khz @ 8bits will sound like a drowning
goat under water.
I am a specialist in low bit width audio: My TRS-80 was able to input
audio for
I totally puzzled by this situation. I have asterisk 1.4.4 behind NAT.
All SIP peers are working properly to place or receive calls.
Any SIP peer or friend whether NATted or not will become UNREACHABLE
if qualify=yes.
I have identical peers on the other asterisk 1.2.16 production server.
In
I am seeing this too on both Polycom and Linksys phones, as well as
external SIP peerns not behind NAT, such as FWD. I've posted a
couple of times about it, but I don't see the posts.
On 6/3/07, Ian Clough [EMAIL PROTECTED] wrote:
Hi
I have FC6 system in the office running SVN-trunk-r63567
On 6/13/07, Akpome Akpoguma [EMAIL PROTECTED] wrote:
I have been using wav files with sample rate of 8khz and 8 bits and I find
the sound quality really poor.
8khz is correct, if you are using 8 bits, you need to use 16 bits if
I'm not mistaken.
___
On 6/14/07, Jaswinder Singh [EMAIL PROTECTED] wrote:
What does sip show peers output ? Also set a timeout in millisec like
qualify=200 instead of qualify=yes
Doesn't matter. I've used qualify=2000
There is another thread about this now, OPTIONS response from the
phone is ignored.
I now have both 1.2 and 1.4.4 boxes.
Each asterisk is behind NAT on a fixed ip with all the externip,
nat=yes, and forwarded ports etc set up.
I have two multiline SIP phones, Linksys 941 and a Polycom ip500.
THese both work normally on the 1.2 box. I took the exact configs from
sip.conf and
Oops, I had some problems and was offline unable to remind you about
the conference yesterday.
LISTEN to recent recordings: http://x2z.eu/astusers.htm (Flash
player, will autostart)
THIS WEEK: Stephan Winterberg and Stephen Boche tell us more about the
new book, whick looks like a great
In short, the 's' extension is not a catch-all.
The use of 's' can be confusing. The best example I have of the use of
's' is when a ZAP call comes in on an analog line. IIRC, the book says
something to the effect that 's' is for when, upon arrival in a
context, the call has no other place to
Almost every example on voip-info.org and O'Reily assume you are using an
FXO or FXS card.
I am 100% internet based.
This 4 year old article will go a long way in explaing the basics,
with examples. John Todd also has had his own heavily commented
extensions and other config files online all
Asterisk Users Conference Fri 12:30 PM EDT
** see http://x2z.eu for instructions on how to join or listen
RussellB from Digium and others will be around to tell us about the
recent devcon and hopefully answer any questions about the near future
of asterisk branches, what's new, etc.
Hope you
I was not clear. I removed the PDF conversion only because the faxes
were ot being recived correctly in most cases. The PDF conversion
worked fine, but who needs a bunch of blank PDF? May as well have
blank TIFF :)
On 5/29/07, Doug Lytle [EMAIL PROTECTED] wrote:
randulo wrote:
All spam faxes
You could also use the cid syntax in the extension
exten = s/ObnoxiousCallerId,1,Goto(getlost)
On 5/30/07, Steve Finkelstein [EMAIL PROTECTED] wrote:
Hi all,
I'm looking for some rudimentary insight on GotoIf() which seems to be
failing on me in my dial plan. All I basically wish to do is
Here is what I am working with now!
[204]
exten = 204,1,Wait()
exten = 204,2,Answer
exten = 204,3,Playback(demo-congrats)
exten = 204,4,Hangup
exten = s,1,Dial,(teliax) --
exten = s,2,Hangup
___
--Bandwidth and
So what is the bottom line? Does it work or not. I've heard stories it
works, it doesn't work, it kinda sorta works when it's not raining out side.
Everything under the rainbow.
I have comiled stats on my asterisk pbx over the last three years.
All spam faxes arrive perfectly readable. For
As the error message shows, you have wrongly numbered priorities
[brad]
exten = 204,1,Wait()
exten = 204,2,Answer
exten = 204,3,Playback(demo-congrats)
exten = 204,4,Hangup
exten = 204,2,Dial(Zap/g2,20) --
;exten = 204,Voivemail(u100)
Well, I've run out of ideas :)
On 5/22/07, Vincent [EMAIL PROTECTED] wrote:
Must be one of those problems that are solved in 2 seconds with the
right click or line in a configuration file... when you know what
you're doing :-)
___
--Bandwidth and
Quick reminder that this exists and is today.
* see http://x2z.eu for instructions
Maybe JerJer (aka Put down the crack pipe) will be there to comment
on the about Nufone and their plans in Canada and elsewhere?
___
--Bandwidth and Colocation provided
So... I guess it's something in the 3102 that must be changed so that
it will finally TX/RX voice packets to remote phones (works fine when
picking up an IP phone in the same LAN as the 3102 and Asterisk).
Here's something from an old post:
Upon replacement of the Linksys, everything worked
... but when I pick up the handset, I get no voice either way, even
when I set the Linksys gateway to use a static external IP address
(STUN doesn't seem to work).
asterisk doesn't do STUN AFAIK, but I've never needed it and I use
double NAT and have since 1.0.?.
What happens when you do the
developers in the conference room until
they tell us what happened at their conference. Not sure if they'll
have escaped by then. What about it, Kevin? We'd love to hear from
you.
randulo
asterisk user and enthusiast
___
--Bandwidth and Colocation provided
Friday May 11, 2007 at 12:30PM EDT
A short reminder that you can connect with others in the asterisk
community by phone or SIP (or both obviously) during these
conferences. Anyone interested in asterisk is welcome to join the
conference.
Details are found here: http://x2z.eu
Past recordings
On 5/9/07, Zeeshan Zakaria [EMAIL PROTECTED] wrote:
they are complaining that audio goes blank for a few seconds during the
conversation, and then comes back on. It goes blank for both parties.
I had this same problem happening every 8 minutes. It ended up being a
DSL issue at the DSLAM. You
Can Absolutetimeout(n) be used in multiple places in a context?
Because I need to use it to hang up the X100P, I set it to 2 minutes. I
didn't realize that incoming calls would be cut at that time too. I want
to put Absolutetimeout just before both voicemail calls (u and b), is
this ok in the
Kevin Walsh wrote:
You must have missed the section of the documentation that says Each
time you call AbsoluteTimeout(), all previous absolute timeouts are
cancelled.
http://www.voip-info.org/tiki-index.php?page=Asterisk+AbsoluteTimeout
Thanks, I did miss that. The volume of all these documents
Hello,
I have an X100P connected to a phone line that is used for ADSL. The DSL
modem is connected directly to the wall jack. A regular phone was
connected on this line through an ADSL line filter.
The audio part works with * just like it does on the phone, but it
appears callerid is not
randulo wrote:
I have an X100P connected to a phone line that is used for ADSL. The DSL
modem is connected directly to the wall jack. A regular phone was
connected on this line through an ADSL line filter.
DEFINITIVE ANSWER:
One kind soul responded off list. At any rate, it was the phone company
For info,
I receive the mailing list on a brand new account that is not used for
anything else.
Just received, a virus (*apparently*) From: [EMAIL PROTECTED]
I suppose there may be 8,000 people getting it but just in case.
___
Asterisk-Users mailing
Brian Capouch wrote:
Does anyone know a way of knowing where a given recording of a song stands?
This is one of several sites:
http://www.pdinfo.com
Also try the site of the agencies of your region, BMI/ASCAP in the US,
SACEM in France etc.
___
Angel Gabriel wrote:
I want to be able to make calls on my laptop, and then have an * box,
route the call via a proper phone line. Is the above possible?
Yes, assuming the two machines are connected on a network, nothing would
be simpler. You'll just need some kind of card in the * box to
Hi,
I'm confused about a config we have going where there is
NAT router -- 192.168.1.101 linux+asterisk
PC -- 192.168.1.104 WinXP with X-Lite
At another location:
NAT Router -- PC X-Lite xxx.xxx.xxx.xxx
The remote works fine with *, can use the FXO line, can call FWD members
thru *,
Dave Cotton wrote:
And the 255 million people in Europe? Please not the usual, 75US$ for
the unit 80US$ for FedEx or UPS to deliver it from the US.
No Dave, more like $125 for the unit and $80 FedEx :)
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[EMAIL PROTECTED]
hank smith wrote:
hello
Hi hank
I am not sure where to ask this question at so please except my
apologise if this is the wrong list.
Why not post to the fwd forum?
I need to ask if any one has got firefly sip version to work with fre world
dialup?
Not that I've heard. There is a new beta out
[EMAIL PROTECTED] wrote:
It appears that the X100P (FXO) does somehow not passes the
'hangup' signaling *.
snipped your scenarios
I am having the same issue on a normal analog POTS line (but in France
so you never know what other signalling anomalies there may be.)
The h signal never happens on
Can anyone ell me if they've had experience on the continent with caller
ID on analog POTS lines?
Here in France, we currently do not have it enabled on any of out lines,
but I would consider paying for it if I thought * via X100p cards would
be able to detect it properly.
I know for example
Dave Cotton wrote:
Caller ID works perfectly here in France and I can confirm that the full
UK number is also transmitted correctly. I can't answer for US numbers.
Hi Dave, so we're neighbors? Funny the FT 3131 service never returns a
number from an international call. Do you know if the X100P
Andy Powell wrote:
http://www.ainslie.org.uk/callerid/cli_faq.htm
Lots of info there
Thanks Andy, that looks interesting. I probably should have also done a
search on the subject, but my questions are specific to use for * and X100P.
r
___
I have the dev kit installed and the X100P answers calls and * routes
them as expected. I am not able to dial out at all:
[analog-out]
exten = _9.,1,Dial(Zap/1/$EXTEN:1)
exten = _9.,2,Congestion
included up in the default section
shouldn't this take any call beginning with 9, strip the 9 and
David J Carter wrote:
hi,
Try
exten = _9.,1,Dial(Zap/1/${EXTEN:1})
Holy cut and paste! That should make a difference, thanks. (not at the
office to find out)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of randulo
Sent: 12 March 2004 14:54
To: asterisk
I've scoured the list about the TDM400 PCI 2.2 issue and I'm more
confused than ever. Some say the card is compatible but PCI 2.2
compatible mobo is not necessary. Note: too bad there's no good way to
remove false answers from the archives.
I just received the card and put it on a mobo that is
Bob bevins wrote:
Did you plug the power into it?
Yes, sorry I *knew* when I began that I needed to say I plugged it in,
but I forgot anyway. I've even tried different power plugs (since some
have had noise from sharing power connex with the hard drives.
Nathaniel Powning wrote:
Did you try using the free technical support that comes with the dev kit?
I will do that, but I thought the huge number of configs of those of you
on this list using this stuff in the real world would be a bigger
knowledge base. I also was hoping to find someone close
as an
Ensoniq 1317 or something -(I am not near that box at the moment.)
Will * work with this card or what cards will it work with? I really
want to be able to have dialup music. I have an old DAL CardD+ ISA
soundcard but I'm assuming that won't ever work.
tia,
randulo
Hi,
Andrew Thompson wrote:
If by dialup music you mean music-on-hold, a soundcard is not required
for that, go to the wiki and read.
I do mean music on hold, or in this case music on demand.
http://www.voip-info.org/wiki-Asterisk
You mean the part that says Asterisk needs no additional hardware
hank wrote:
are you using alsa drivers?
Forgive me, I just installed Slackware two days ago, I'm not up to speed
yet, but I see ALSA mixer app is there. I also saw somewhere that the
soundcard is muted at boot time and needs to be manually unbooted using
the alsamixer app. I ran that and it
Steven Critchfield wrote:
All those snd- modules sounds exactly like alsa. The error message is
probably related to the chan_oss module trying to get access, but not
having a OSS driver to talk to. This isn't a problem, but if you don't
want to see it, put a noload = chan_oss in modules.conf for
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