I want to record audio in ADPCM format. According to the show codecs
output of Asterisk, it looks like it supports adpcm. But I do not know what
to tell the RECORD FILE directive in my AGI script.
The RECORD FILE command usually has this form:
RECORD FILE filename format timeout [BEEP]
It
Hi,
I'd like to recommend http://www.telappliant.com
They responded far faster than a few others I contacted.
I placed my order (2 digium E1 cards , 2 ip phones) friday at 15:00.
They emailed the invoice 15 minutes later.
I paid at my bank at 16:00, and faxed them the proof.
The goods arrived
The TE410P should run fine under both Redhat 9 and Fedora Core 1. My first
question is: Are you running a new CVS version? Maybe there have been bugs
introduced with all of the recent changes. I'm running under December
versions - works ok, except for problems experienced under very heavy
Stephen I think hit it on the mark. I could not figure
out how sidetone could be heard as an echo and how it
could be loud. On my GS phone the sidetome is very low
but with zero delay.
The best way to test the side tome is to talk to the
voicemail or record application running on a local
This article below came up on the newwire. The RTCP XR RFC was published.
Will Asterisk be supporting this function in a future release? Does anyone
know if any phone vendors are going to be supporting it?
Thanks
Lee Goodman
Our Technology Update this week is about one of those
mechanisms.
John Baker wrote:
[...]
What you (and alot of businesses) don't understand about complaints is that
they generally don't come from people who don't want to do business with
you, but rather they come from people who do. They point out problems so
that they CAN do business with you. That's why you
There are somes products available claiming to connect a BT headset,
a cell phone and a phone land line all together.
I've found some :
http://www.geekzone.co.nz/content.asp?contentid=2079
http://www.clipcomm.co.kr/
The clipcomm BS-A101 sample price is : $570. It's VoIP land phone
that can
Steve.
Flosys makes fixed cellular interfaces.
Although our main products come with FXS ports we designed the
interface as interchangeable modules. One of our interface modules
is a T1/E1 interface (based on an Infineon Falc56).
So yes we do support digital interfaces.
We also have a TDM
Chris, thanks very much for this tip:
On Mon, 26 Jan 2004, Christopher Lee wrote:
My testing involves calling from a SIP handset to a dummy extension setup to
answer and playback the tones I want to check.
; Test Australian ringing tones - indications
exten = 906,1,Answer
exten =
The only thing I can think of in respect to analog DID lines is answer
supervision.
DID lines provide one way - outbound audio - before answer and cut through
bidirectional audio only after answer. But this could happen also outside
the local switch so local calls will still have bidirectional
I'll answer my own post since I plunged ahead while my original post was
stuck in my outbox over the weekend...
I notice the zaptel Makefile option
the mark2 option KFLAGS+=-DAGGRESSIVE_SUPPRESSOR
is now gone.
Ok, I found zconfig.h for KFLAGS+=-DAGGRESSIVE_SUPPRESSOR
How are
I have an * box at my home office. I have my GS 101 at work that registers
fine with it. However, when I try to dial voicemail from the phone @ work,
it sometimes doubles the digits I enter. Example: I designated 8500 as
my voicemail, so when I dial it...and it prompts for mailbox/password
Hi Jeroen1
I think that´s maybe a bug
I really don´t found the problem in my logs, i´m starting it by hand :-(
I update you if i can figure it out.
regards
Miklos
- Original Message -
From: Jeroen [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, January 26, 2004 11:23 AM
Hi!
anyone out there running an ISDN pbx behind an Asterisk server with an
AVM active card? At this moment Asterisk is connected to the internal
ISDN bus of the PBX, but I want to control the calls coming in from the
outside, before sending them of to the PBX.
Can't be of help here - I have
john wrote:
I notice the zaptel Makefile option
the mark2 option KFLAGS+=-DAGGRESSIVE_SUPPRESSOR
is now gone. Does simply adding these options still compile in a certain
echo can - or is there an other method of activating a particular can. I
have not had to update my machine that is connected to
Ok now that we have a Asterisk server running quite
well, we want to put it onto a more appropriate device, i.e. not a beige box
computer, but perhaps some kind of embedded linux appliance.
Has anyone already done this? Any suggestions
on some tidy, small, suitable linux systems to use for
If this may be of any use:
I'm not an expert but I did the test with the FWD soft phone from X-ten and
iLBC SPX don't work.
Asterisk wasn't between the connections. Just x-lite and fwd (who is an
Asterisk Server?)
The soft phone makes the connection but I can't hear any sound.
Wim
-
Hi all,
I'm having a bit of a problem using the # sign to transfer when using a
soft IAX2 client. Has anyone else experienced this problem or know of a
possible work around / fix of this problem. The following is a snippet
from my extensions.conf file. This is how the file is setup for
Title: Call Queue wait times
Hi Everyone,
Is there any specific way to get the current wait time for a queue? If not what is the best way to implement this feature? I would really like to be able to intelligently estimate wait time.
Thanks,
Matt
Hi to everyone,
I am dealing with my primer Asterisk installation and we are trying to set up a H323 server in order to use Asterisk to place calls
between NM clients (also Gnomemeeting).
I have a basic extensions.conf file:
[general]
static=yes
writeprotect=no
[default]
exten =
I have a few (more specific) questions about 'Zapateller':
1) How would you test this?? Would I need a predictive
dialer machine like the telemarketers use. OK, I could
just wait and see if it seems to cut down the unwanted calls
but that's not really a test.
2) I don't understand how I
I hope you are prepared to be mightly flamed when you complain about
nufone not responding to your emails grin
Nick
On Mon, Jan 26, 2004 at 09:09:32PM +1000, Vic Cross wrote:
On Mon, 26 Jan 2004, Jeremy McNamara wrote:
Our network and services speak for themselves. If they don't
Hello all,
I would like to set-up some direct lines so that when a user
of mine answers the phone he/she knows to say the correct intro message, so
that we can introduce ourselves as different companies.
I have played around with caller ID and can modify that
using caller ID Name
On Mon, 26 Jan 2004, Aaron Martin wrote:
I have Asterisk running with a combination of SIP and H323 clients. I am using the
OH323 module instead of the H323 one.
When the SIP clients ring each other, they can hear a ringing noise in the ear peice
to let them know that the other parties
Try with:
http://bugs.digium.com/bug_view_page.php?bug_id=214
Regards,
Gus
- Original Message -
From: Anton Yurchenko [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, January 26, 2004 11:01 AM
Subject: [Asterisk-Users] app_queue and dialplan
Hello,
I`m trying to achive
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Friday 23 January 2004 15:08, Tais M. Hansen wrote:
How can I configure the TE410P card to act as master instead of slave?
Ah, the wiring was wrong. Straight when connected to the Telco, crossover when
connected to a local PBX.
- --
Regards,
On Thu, Jan 22, 2004 at 11:44:52AM +1100, [EMAIL PROTECTED] wrote:
Has anyone from digium looked at why there is a 30 min to 3 hour lag on
messages on this list?
Continuing this...
I sent a message (Subject: Zapateller) yesterday at 17:54 GMT and it only
came through to my box today at 08:28
On Mon, 26 Jan 2004, Steve Foy waxed:
I'm just wondering about 'Zapateller'.
How exactly does it work!? I might be interested in employing it at work
here, but wondering if anyone's using it?
I think you can just put it in your dial plan:
exten = s,1,Answer
exten = s,2,Wait(2)
exten =
I'm not sure if you're trying to accomplish something specifically by using rc.local,
but I use RH9, and I used make config on both asterisk and zaptel and that created the
correct init files for me. Starts up perfect every time!
Sean
-Original Message-
From: listas iPfone
Hello Asterisk,
This is test to the list..
--
Best regards,
Frankie ([EMAIL PROTECTED])
mailto:[EMAIL PROTECTED]
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I'm another outsider just reading this entire thread and Jeremy's
replies. NuFone could have the best service on the planet, but I would
NEVER do business with a company that has this kind of attitude. People
have a choice in which companies they do business with. Why would they
do business
Hello
I downloaded the stable build of * and was
not able to find the festival patch in that build. Also i tried from CVS
and the same.
Can anyone tell me where i can find the
festival patch?
Or the patch is no more requried. i can
just compile festival and it will work??
Thanks in
Does anyone have any ideas what might be causing this, and how to diagnose
without taking our main phone switch down? I have access to a PRI, but it
has no phone numbers that ring into it, so I can connect to real network
equipment and make calls, but cannot receive any.
it might be something
Darren,
On Tue, 27 Jan 2004, Darren McIntosh wrote:
Do you have country=au set in indications.conf? A reload brings the
indications.conf changes in for me.
Yes, I do. The cadences of all the tones are correct, which tells me that
at least part of it is working. The frequencies of the
Testing once again.
bkw
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Hello all, I am using voicepulse DID's to receive
calls via IAX to and asterisk IVR dial plan I have put together. The problem is
after 3-5mins the system cant pickup the DTMF tones I am sending... I have tried
different telephones... It just repeats menu options over and over I have to
Yep.. I am a voicepulse subscriber and can also report they don't answer
their email either.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Frankie
Gravato
Sent: Monday, January 26, 2004 5:42 AM
To: John Baker
Subject: Re[2]: [Asterisk-Users] Has Nufone
All of a sudden my list traffic appears to have dropped to a few
messages/day the past few days. I anyone else seeing this as well?
Michael
--
Michael Graves [EMAIL PROTECTED]
Sr. Product Specialist www.pixelpower.com
Pixel Power Inc.
Hello
I am interested in testing the voice
quality with another * setup in India. I am planning to setup a * server at
India and would like to know whether the audio quality would be good enough to
make freequent calls.
anyone willing to help please let me know
and i can test in your
Has anyone ever seen these errors generated by a
cisco 7960? none of our other brand phones seem to generate these
erros:
Jan 27 21:54:07 WARNING[-1147556944]:
chan_sip.c:486 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED]
for seqno 101 (Response)Jan 27 21:54:08
HI,
Once the call enters into asterisk and then that call gets transferred,
cdr does not record CLID and SRC field data for the transferred call.
Is this a bug or I am missing something?
Ta
SJ
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Hello
I am interested in testing the voice
quality with another * setup in India. I am planning to setup a * server at
India and would like to know whether the audio quality would be good enough to
make freequent calls.
anyone willing to help please let me know
and i can test in your
Hello
I downloaded the stable build of * and was
not able to find the festival patch in that build. Also i tried from CVS
and the same.
Can anyone tell me where i can find the
festival patch?
Or the patch is no more requried. i can
just compile festival and it will work??
Thanks in
Having some trouble building cdr_addon_mysql. I've installed mysql and
mysql-dev and all related rpms on a RH9 box. The box is up and running
*.
I've checked out the asterisk-addons but at make install, I receive the
following error:
./mkdep -fPIC -I../asterisk -I/usr/include/mysql `ls
Excellent note. I took a look at Broadvox and filled out their little form.
The result was
This is an automatically generated Delivery Status Notification.
Delivery to the following recipients failed.
[EMAIL PROTECTED]
Oh well.
-Original Message-
From: [EMAIL PROTECTED]
Steve,
I have been working most of the day to get this problem solved.
Thus far everything should be returning to normal and problems like this
should never happen again. Right now it has about 3 hours of posts to get
out and some were lost by accident during this mess. So if you see a
Hi, a Virus was sent from this account to the Asterisk-Users mailing list...
scan your computer for virus!!!
Ronen
- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, January 26, 2004 7:29 PM
Subject: [Asterisk-Users] Test
The message contains Unicode
And check the firmware revision in your Adtran. I believe current is L36.
Alfred R. Nurnberger wrote:
The only thing I can think of in respect to analog DID lines is answer
supervision.
DID lines provide one way - outbound audio - before answer and cut through
bidirectional audio only after
Hello all!
We have a PSTN line with four numbers calling into it. There is
distinctive ring on these lines. They are are follows:
1. standard ring
2. short ring
3. long ring
4. short ring, long ring, short ring
Based on the information I have been able to find, I have created the
following
For both the latest CVS and the packaged distribution. I dug up the patch
that Jean-Denis Girard posted to the list fall 02 to fix the usb problems
with the 2.4.20 kernel but that won't apply (not a big suprise). Is this
code already integrated? Any other know issues with ztdummy? This is on
My Asterisk server registers two FWD numbers.
On average I get about one call a day from someone calling
from an FWD number and leaving a pointless, under 10 second
message. It's easy to see who these people are if I look
in my CDR file I can see thier name and number. They seem to
be new FWD
Try setting canreinvite=no in sip.conf. It might be that attempts to natively
bridge the voice streams are failing.
On Saturday 24 January 2004 23:26, Chris Wilson wrote:
Hmm, The host seems to be good, I have no firewall rules in place at the
moment for the local network, and everything is
Hi,
- Original Message -
From: Adam Hart
I believe you can connect using a standard asterisk box but you'll miss
out on the extended features.
It is a standard IAX client or a softphone using a modified/non-standard
version of IAX2?
I have tried to register with my own * box without
testing
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[EMAIL PROTECTED] wrote on 01/26/2004 01:12:09 AM:
You're right, Jeremy. I made up the whole thing. I went out of my way
to
concoct a story about how I wanted to do business with you, but was
unable
to figure out how on your website, so I called and left a message and
didn't
get a
If you want to test it, once you include Zapateller in your dial plan, place
an incoming and block your caller id (from the phone your testing from) and
it will do its job.. to block incoming calls that dont produce a caller id.
J.C.
- Original Message -
From: Chris Albertson [EMAIL
I would like to correct some of the text on the GotoIf application page
on the wiki. Does somebody actively manage changes like this, or should
I fire away and make it myself?
I'm actually surprised I have permission to edit a page without prior
authorization, but it DOES state at the bottom
That is no longer a problem.
bkw
On Wed, 28 Jan 2004, Tazman wrote:
Hi, a Virus was sent from this account to the Asterisk-Users mailing list...
scan your computer for virus!!!
Ronen
- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, January 26,
Hello
When ever i make calls via a SIP provider
I keep getting this error message
Jan 29 02:09:20 NOTICE[1228887360]:
rtp.c:264 process_rfc3389: RFC3389 support incomplete. Turn off on client
if possible
any idea what is it?
Regards
Deepak
We have done this with nexcom.com appliances. We are currently using EBS
1569PS unit with a 1gig flash card. Keep in mind were only using the gateway
features of asterisk on this box. If you are using the full PBX features you
may want to use a more powerful model.
John Bittner
Simlab.net
Nexcom
test
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Hi,
I have a TE410P card anda configuration with IAX (soft phone). How * is using the chan_iax/zap ? What sequence and for what ?
Also, at zap channels there is a kind of result recognition. Meaning, the busy tone, no answer etc. So, when recognizes a status, automatically creates a record
On Tue, 27 Jan 2004, Chris Albertson wrote:
Question: Does everyone with an FWD number get these junk
calls or am I the only lucky one?
I just got an FWD number a couple days ago, but haven't had that
experience yet.
And no, I haven't tried calling you to see if you'd answer. :)
Greg
Ok I am looking at the Clipcomm solution. Bluetooth enabled phones can
place voip calls by communicating with the BS-V100/L100? Am I getting
this right? What is this feature called in phones and which cell phones
support this?
- Dustin -
Eric Bart wrote:
There are somes products available
Go ahead and edit the page. I've fixed several little errors on pages that I
didn't create. The voip-info.org Wiki is like the total-open-source Asterisk
manual. Although the total-access may be a problem in the future because all
someone has to do to delete everything is just to register and
check out the latests cvs it has two reg files that will fix xlite or xpro
to work
bkw
On Mon, 26 Jan 2004, Wim Venneman wrote:
If this may be of any use:
I'm not an expert but I did the test with the FWD soft phone from X-ten and
iLBC SPX don't work.
Asterisk wasn't between the
testing yet again.
bkw
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What you (and a lot of consumers) don't understand is that some
customers need to get fired. Some people aren't worth doing business
with. Those are the facts.
I have done this. I have felt better after I fired a cusotmer in
situations like that.
bkw
All of the mailing lists are now filtered for virues and spam before they
reach digiums network. This will ensure that the mess that caused the
list server to break down wont happen again.
Its still playing catchup. But the mail is now flowing faster than usual
now.
Thanks,
Brian (bkw_)
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