Nik Martin [EMAIL PROTECTED] wrote:
Of course, big brother wants his say in the matter.
http://www.wired.com/news/politics/0,1283,63884,00.html?tw=wn_2polihead
Apart from section 4(c), I can't find any text that would justify the
following quote from the article:
Under Sen. John
Hi Patrick,
I can't tell much from this brief description. Send me a console log.
Regards,
Steve
Patrick J. Conroy wrote:
Hello All,
I have downloaded and installed spandsp and downloaded rxfax, etc and
rebuilt asterisk with app_rxfax. I have added the following to my
extensions.conf:
Hi Mike,
To get something like:
Coarse carrier frequency 1832.96 (4)
Training error 927.702492
Training failed (convergence failed)
something is horribly wrong. The carrier should be 1700Hz, not 1832.96Hz :-)
Do you have a codec mismatch, or are you using a codec other than u-law
or A-law?
Hello All,
I'm not certain that I have my firewall setup to perform traffic
shaping correctly for *. I'm using a m0n0wall running on a Soekris 4501
platform. I have traffic shaping set to provide IAX port 4569 with
priority outbound access from LAN to WAN. All other outbound access is
lower
Andrew Kohlsmith wrote:
On Thursday 17 June 2004 11:38, John Fraizer wrote:
If you have PRI service into your * server, it is possible - though not
always easy - to set the ALI database information specific for each ANI
(DID number) that you use. I do this with our PRI's. Depending on
which
I don't think I was missing the point. Hardware and software
development are very much the same -- In my original proposal, I
suggested a type of communal development -- engineers would receive a
share of the company proportional to the time they donated. I have done
this type of development on
Hi!
I see in the archives a brief thread between Barton and w last
November 2003 about streaming to the Internet. Id like to use an
Asterisk to mediate multiple VOIP calls originated from the Internet
to the studio to be mixed then passed out to an encoding PC thence
back to Internet
CLI:
Hi!
Any idea or code on How to allow users to change their voice mail
password over the Phone.
The only way io know is tochange in voicemail.conf file and restart
asterisk.
- dial VoiceMailMain
- enter 0
- enter 4
- now change your password
- confirm the new password
READY!!
Philipp
I'm running Asterisk CVS-HEAD-05/24/04-17:37:48 on kernel
2.4.25-gentoo-r3. I have a Digium TDM-400P card with 4 FXO ports. Here
are the pertinent files:
zaptel.conf:
fxsks=1-4
loadzone = us
defaultzone=us
zapata.conf:
[channels]
context=north_in_pots_vip
group=1
signalling=fxs_ks
Does anyone know of any Cellular providers that will allow VOIP
connections to thier cellular phones? I'd love to be able to call a
cell phone without using the PSTN. I would think it would make sense
for the provider, they still get to charge normal minutes, but don't
have to burn a connection
Andrew Kohlsmith wrote:
Do you have information on how to do this? This is *precisely* what I want to
do. I assumed you set this up with your telco and then set the caller ID to
the # matching the address you wanted, leaving the telco to do the address
match.
In discussions with my telco,
Any body used patch,
http://bugs.digium.com/bug_view_page.php?bug_id=0001719
to get the callerid for BT Line.
I applied the patch successfully but could not get it to work.
Any help.
Here are the logs:
-- Starting simple switch on 'Zap/1-1'
Jun 17 18:22:31 NOTICE[426000]: chan_zap.c:4811
Hi All,
I just setup an asterisk system equiped to with a 4 FXO port board to
interface it with my phone system (Inter-Tel). I realize that I'll be
running our of ports once I start showing what can be done with it.
My Inter-tel system supports MGCP connectivity to a gateway. Is there any
James Sutton wrote:
I see in the archives a brief thread between Barton and w last
November 2003 about streaming to the Internet. I'd like to use an
Asterisk to mediate multiple VOIP calls originated from the Internet
to the studio to be mixed then passed out to an encoding PC thence
back
Kevin P. Fleming wrote:
Andrew Kohlsmith wrote:
Do you have information on how to do this? This is *precisely* what I
want to do. I assumed you set this up with your telco and then set
the caller ID to the # matching the address you wanted, leaving the
telco to do the address match.
In
That's not gonna happen for a while. The incoming connection costs them
hardly anything (that's why some providers have unlimited incoming
minutes), and the handful of people who would actually use such a
gateway would justify the cost of setting it up. Additionally, you'd
need to use some type
Another solution to get uk caller id is to get a meteor unit, andy powell
kindly wrote a few agi scripts and the perl script for the meteor to
integerate it fully with *.
-Original Message-
From: Kannaiyan Natesan [mailto:[EMAIL PROTECTED]
Sent: 17 June 2004 6:20 PM
To: [EMAIL PROTECTED]
Kannaiyan Natesan [EMAIL PROTECTED] wrote:
Any body used patch,
http://bugs.digium.com/bug_view_page.php?bug_id=0001719
to get the callerid for BT Line.
I applied the patch successfully but could not get it to work.
Any help.
[snip]
My Zapata.conf:
usecallerid=yes
John Fraizer wrote:
Your telco really can't *prevent* you from doing PS/ALI. They don't
have to make it easy though. Even with PS/ALI, the same database is
updated. It's just you doing the update vs the telco doing it. Doing
things the Right Way TM isn't always easy but, in the end, it is
[snip]
My Zapata.conf:
usecallerid=yes
ukcallerid=yes
Change those two lines to simply usecallerid=uk.
I changed as you said and restarted asterisk. Still doesn't work.
-- Starting simple switch on 'Zap/1-1'
Jun 17 19:24:48 ERROR[262160]: chan_zap.c:4759 ss_thread:
Are there any online retailers that carry the Uniden UIP series phones? I
did a quick Froogle search to no avail.
See:
http://www.voip-info.org/wiki-Uniden
Jim
James H. Thompson
[EMAIL PROTECTED]
___
Asterisk-Users mailing list
[EMAIL
I was just noticing that my asterisk box is trying to register with my
providers on both IAX1 and IAX2. They do register on IAX2, but it seems like
it keeps trying to register with IAX1 also.
Is there a way to disable this behavior, or am I just misunderstanding it. I
just want to make
It works fine for me.
make sure you only have
usecallerid=uk
in the config.
If you also have
usecallerid=yes
set it will default to the US style.
Make sure you have the uk settings in zaptel.conf. Can you see the callerid
with a std phone on the line?
Chris
- Original Message -
I have the following settings chris. Also i confirmed with BT that caller is
enabled on my line.
Let me know if I need to modify anything. Thanks.
zaptel.conf:
fxsks=1
loadzone=uk
defaultzone=uk
zapata.conf:
[channels]
busydetect=1
busycount=7
relaxdtmf=yes
callwaiting=yes
Just a Question. I would like to know if TDMoE follows specifiaciones of
TDMoIP RAD protocol that says that there is a compression of 16/1 when
you do TDMoIP.
Manuel Marin Garcia
TRANSTELCO S.A. DE C.V.
Campos Eliseos 9050 B4 – Cd. Juárez, Chih. 32452 - México
Oficina: +52 656 692 11 09 – Fax:
I was just noticing that my asterisk box is trying to register with my
providers on both IAX1 and IAX2. They do register on IAX2, but it seems like
it keeps trying to register with IAX1 also.
Is there a way to disable this behavior, or am I just misunderstanding it. I
just want to
I have encountered this same issue. I do notice that resolution is
coming up as zero. Here is a dump from the console as a fax comes in:
Jun 17 13:07:09 VERBOSE[1217669936]: -- Executing Answer(Zap/1-1,
) in new stack
Jun 17 13:07:09 VERBOSE[1175706416]: -- Accepting call from '' to
Matt,
I tried 1.2 out, but could not get multiple lines to register to the
same sip channel. I am by no means an expert, so it is possible that
somone else could figure out how to do it. Oh well, it would be nice if
it worked.
-Tor
Tor Roberts wrote:
Matt,
Thank you very much, I will try the
Thanks for your time, Ty Purcell,
Yes, it's just as you said, when I create a new profile, it opens a
window with the fields(Proxy Domain,Account,Password,CallerID). but it
doesn't let me to input value to these fields. I don't what's the matter.
does anybody know how to deal with it?
thanks
Rad's TDMoIP uses DSP chips on each end of the link to compress the data.
Gary
Just a Question. I would like to know if TDMoE follows specifiaciones of
TDMoIP RAD protocol that says that there is a compression of 16/1 when
you do TDMoIP.
Manuel Marin Garcia
TRANSTELCO S.A. DE C.V.
I have it working with the X100P no problems, on both BT and Telewest lines.
Anybody got it working on the TDM400P yet?
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Kannaiyan
Natesan
Sent: 17 June 2004 19:59
To: [EMAIL PROTECTED]
Subject: Re:
Interesting, the size for the tiff files are:
1728x76 file size 1839 bytes
1728x159 file size 4605 bytes
1728x61 file size 1417 bytes
1728x77 file size 2071 bytes
on 4 different files originating from the same fax. The image area is
all white with a few dots at the top and when I change the
Thanks for yor answers!
-Now I've got it running just
fine:-)
NRB
I am having problems with a call queue setup
where voicemail seems to be getting in the way.
Here's what I've found:
When I have 2 users logged in as Agents (using
AgentCallbackLogin) they will both ring when a
new call comes in.
If, for instance, one of the users makes an
outgoing call and a new
I get this error when i receive the call on the line. Also I have Call
Sign on my telephone line, does that affects this anyway?
Call Sign -- two telephone numbers for a single telephone line with two
distinct ringtone.
Jun 17 21:04:50 ERROR[360464]: chan_zap.c:4759 ss_thread: zt_get_history
hi;
i am using snom 200 and i have a sip registrar running on one machine
called xyz.com with a provisioned user called [EMAIL PROTECTED] and i have an
asterisk box on another machine. I dont want to use the asterisk box as
my sip registrar and would like to use the other machine. How can
It using DSP chips makes no difference, it can be done in software
aswell... (in theory, if its open, and the algorithmic complexity is low)
Secondly, I think just the Name explains the difference between TDMoE
and TDMoIP... it's 2 different things... actually TDMoE offers very much
overhead..
Claudio,
Claudio.loletti wrote:
Hi Jorge!
Our application rom version is 4fxosip.102
boot version is boot.104
I think we need to upgrade the app rom to version 103.
I get into welltech ftp server and found a file called
4fxosipN2004_05_17.BIN. Do you know if that is the last version for the
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
ng kar fei wrote:
| How did you do that? need to cross-compile it?
| --- Simon Brown [EMAIL PROTECTED] wrote:
|
|I have successfully used SJPhone on my iPAQ 5450
|with asterisk.
|
|Simon
|
|
|
|From: [EMAIL PROTECTED]
Rui,
Yes, it's just as you said, when I create a new profile, it opens a
window with the fields(Proxy Domain,Account,Password,CallerID). but it
doesn't let me to input value to these fields. I don't what's the matter.
Edit the profile, and on the Initialization tab and make sure the Inquired
On Thu, 2004-06-17 at 21:02, Kannaiyan Natesan wrote:
I get this error when i receive the call on the line. Also I have Call
Sign on my telephone line, does that affects this anyway?
Call Sign -- two telephone numbers for a single telephone line with two
distinct ringtone.
Jun 17 21:04:50
Also I believe the patch is in the driver to detect distinctive ring
tone for home/business calls. Look in the archive to set it up for
a BT line.
Chris
On Thu, 2004-06-17 at 21:02, Kannaiyan Natesan wrote:
I get this error when i receive the call on the line. Also I have Call
Sign on my
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Rich Adamson wrote:
|I was just noticing that my asterisk box is trying to register with my
|providers on both IAX1 and IAX2. They do register on IAX2, but it
seems like
|it keeps trying to register with IAX1 also.
|
|Is there a way to disable this
Good question Steve. My setup is basically:
Fax Machine - PSTN - X100P - Asterisk - RxFax
I'm not even sure, does Asterisk do encoding if its not sending the call
to a SIP device, or over IAX?
In the mean time I configured Asterisk to send faxes to a SIP extension
(SPA-2000) and it receives
TDMoIP is nothing else like IAX2 with trunking, i would say. And a
compression of 16/1 (payload bandwidth!) sounds like g723.1 to me.
Just a Question. I would like to know if TDMoE follows specifiaciones of
TDMoIP RAD protocol that says that there is a compression of 16/1 when
you do
Hi there, im having some problems with my asterisk box, it seems codec is the principal cause of it. Reading in some forums i found that i can get the new codec_g729 from ftp://ftp.digium.com/pub/telephony/asterisk/g729/new_codec_binary/codec_g729b.so i checked it but the directory
I have some patches that may help you. I'll isolate them and put them in
the asterisk bug database and post the location.
Cheers,
Rich
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of AK
Sent: Thursday, June 17, 2004 8:28 AM
To:
Kannaiyan Natesan [EMAIL PROTECTED] wrote:
I get this error when i receive the call on the line. Also I have Call
Sign on my telephone line, does that affects this anyway?
Call Sign -- two telephone numbers for a single telephone line with two
distinct ringtone.
Jun 17 21:04:50
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Jason A. Pattie wrote:
| You might look into using iaxComm or ZiaxPhone (available from
| http://iaxclient.sourceforge.net/). I have successfully gotten iaxComm
| to compile and work on an iPAQ H3670 running GPE/Familiar Linux in the
| past.
Fantastic. That solved my problem.
I did reloaded the drivers now by entering,
modprobe -r wcfxo
modprobe -r zaptel
modprobe zaptel
modprobe wcfxo
output:
-- Starting simple switch on 'Zap/1-1'
-- Executing MySQLput(Zap/1-1, cid/cid=07751625432) in new stack
-- mysqlput: family=cid,
Hi,
I can't compile Asterisk on a Debian machine.
gcc -g -o asterisk -Wl,-E io.o sched.o logger.o frame.o loader.o config.o channel.o
translate.o file.o say.o pbx.o cli.o md5.o term.o ulaw.o alaw.o callerid.o fskmodem.o
image.o app.o cdr.o tdd.o acl.o rtp.o manager.o asterisk.o ast_expr.o
g729 must be licensed from voiceage, through digium. You can purchase
licenses from digium's online store @ digium.com.
-g
On Thu, 2004-06-17 at 16:37, Carlos Medina wrote:
Hi there, im having some problems with my asterisk box, it seems codec
is the principal cause of it. Reading in some
Hello all,
I am very new to asterisk I have been using it for sometime
but now I want to maintain it myself I have built my own server and am trying
to get my cisco ata 186 to register I am having a problem I get this un 17
17:34:54 NOTICE[1116941120]: chan_sip.c:6715 handle_request:
Given the myriad of telehpone cards available I like to ask this forum for
the following combination:
Asterix on Linux redhat (9.0 or Fedora)
10 analog extension using conventional phones (lets say Panasonic kx-ts3
analog)
4 analog lines coming from our telco
So i will need 3 TDM40B (total 12
Hi Tor, Thanks for trying. I opened up a case with Polycom today to look
into this. I spoke with a tier 2 engineer. He didn't know the answer but was
going to research it and get in contact with the polycom engineering team if
necessary.
I'll keep you posted on my findings.
-Eric
--__--__--
Hello:
I'm trying to implement an IVR for internal use for the
enterprise I work for, but the goal I'm trying to reach is that the
main menu of this IVR present itself to the user after 5 seconds he picks
up his extension (and only if the user doesn't press any key, off course).
I
On Thu, 2004-06-17 at 04:45, Michael Sandee wrote:
Am I dreaming?
Yes.
Community based development is too unreliable.
Just to refer to ongoing projects... Look at the farfon
(www.farfon.com), It's an active project in the final stages of development.
It offers the benefits
On Thu, 17 Jun 2004, PAZ wrote:
I'm trying to implement an IVR for internal use for the
enterprise I work for, but the goal I'm trying to reach is that the
main menu of this IVR present itself to the user after 5 seconds he picks
up his extension (and only if the user doesn't press any
this must be a very faq, but wiki and google are not yielding
much except for the appended message.
i want the tone, which i am currently producing with
PlayTones(dial) to stop when the caller hits a key.
as in
exten = s,1,Answer
exten = s,2,DigitTimeout,5
exten =
Hi,
The what belongs were is my big question at the moment and I personally
don't want to design anything LDAP-ish that would become my private tree
instead of defacto implementation.
You should definitely have a look at the defacto standards for storing users
and groups (check
I would guess that you have Echo squelch enabled in capi.conf. Either
disable this or use app_capiNoES (In the latest capi build) before
forwarding a call to voicemail. Beware, app_capiNoES does not check for
channel type so only use it on a CAPI channel.
Cheers
Martin
if i go off hook and dial 666 from an internal sipura spa-x000
(at extn 141), it rings straight through to extn 666.
using the same dialplan, from a cisco 7960 with 7.1 sip code
(at extn 142), i have to
go off hook
hit NewCall
punch 142 (or any valid extn in the dialplan)
hit Dial
In the mailing list archives, I found a message that indicates that the
IAXy has the ulaw, alaw, and g726 codecs, but I cannot find anything
official on Digium's site about it. The Installation Manual has an
example iax.conf file that indicates the ulaw codec, so I know that one
is good.
But
Hello. Im trying to upgrade my asterisk
installation to most current CVS version. Currently I am running
CVS-03/24/04-07:26:16 and dialing out works fine.
When I install the latest CVS, outbound dialing fails,
but inbound and internal calls work just fine.
== Spawn
John Bittner wrote:
Hi,
I have it working great. I have debian running on it with music on
hold disabled. I setup 10 cisco 7960 phones and tested the 4801 with
calls on all 10 phones at the same time through voicepulse with no
issues. I ran top with all the phones running and I was only
Create the profile
And a new windows appears:
Profile name
File name
Profile type Calls through SIP proxy
Then in SIP proxy,
click the sip proxy option
enter the Ip address of the proxy domain port
user domain
and proxy for nat and also the port (5060)
be sure u have the sip.conf file correct
i'm new to asterisk and am having trouble placing outbound calls. i
know this topic has been discussed ad nauseum in the past [1] , but i
can't seem to find a workaround and i'm wondering if my newbie-ness is
getting the best of me.
after registering the phones correctly and receiving a 200
Clint - Yes.
But it's not chan-capi that you need to worry about, it's the cards that
CAPI supports.
An AVM Fritz does work in Australia, visit www.avm.de for the local
reseller.
Other cards I am not sure about but can't see any problems as Telstra uses
the Euro ISDN standard. So theoretically
Matthew Koch [EMAIL PROTECTED] wrote:
When I have 2 users logged in as Agents (using
AgentCallbackLogin) they will both ring when a new call comes in.
If, for instance, one of the users makes an outgoing call and
a new call comes in, the caller will get sent directly to
that agents busy
Klaus-Peter Junghanns wrote:
TDMoIP is nothing else like IAX2 with trunking, i would say. And a
compression of 16/1 (payload bandwidth!) sounds like g723.1 to me.
16:1 means an avaerage of 4kbps per channel. It would have to be G.723.1
with optimistic silence compression to get that low. I
On the Cisco if you pick up the handset or push speaker, you can then dial
immediately without choosing New Call. In order to have it dial immediately
(without using Dial), then you need to implement the dialplan.xml file on
your tftp server. This does pattern matching to cause the phone to dial
Michael George wrote:
In the mailing list archives, I found a message that indicates that
the IAXy has the ulaw, alaw, and g726 codecs, but I cannot find
anything official on Digium's site about it. The Installation Manual
has an example iax.conf file that indicates the ulaw codec, so I know
g726 is 16,24,32 and 48k asterisk only does g726-32k. The iaxy doesn't do
g726 it does ADPCM as g726 is too complex for the iaxy to do.
So in this case g711ulaw/alaw is all you have to choose from.
bkw
- Original Message -
From: Michael George [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
On Jun 17, 2004, at 5:42 PM, Randy Bush wrote:
if i go off hook and dial 666 from an internal sipura spa-x000
(at extn 141), it rings straight through to extn 666.
using the same dialplan, from a cisco 7960 with 7.1 sip code
(at extn 142), i have to
go off hook
hit NewCall
punch 142 (or
If I understand you correctly, you are trying to figure out why you must
dial ext 142 prior to dialing exten 666 in order to get the 7960 to
connect you with 666 - please forgive me if I misunderstood...
Anyway, it appears as though the two contexts you have listed below have
the exact same name
If you use the mepis debian release, it's a piece of cake to install *. It
takes about 15 minutes to install Mepis and *.
www.mepis.org
Paul
Paul Mahler
[EMAIL PROTECTED]
Signate, LLC
665 Third Street
Suite 100
San Francisco, CA
94107-1901
Asterisk Services and Training
Michael M. Saunders wrote:
Can I just pay you to fix it for me.
I cant see anywhere where I use the debug
Why do you see a need to run a 3rd party channel driver? Asterisk has
native H.323 support.
Jeremy McNamara
___
Asterisk-Users mailing list
That was an excellent information which you gave in advance.
I did that too.
Intially I was bit confused with the dring, and I was worried about finding
the right pattern. But it gave me the value when i make a call after
enteringusedistinctiveringdetection=yes in zapata.conf. And I got the
I´d like to know if someone could help me with this issue:
Anybody there have the Mibs for .68 ver in 1204 ?
or meavy the .cfg file.
I found mediatrix box so hard to configure.
--
___
Get your free email from http://www.hackermail.com
Powered by
On Thu, 17 Jun 2004, [ISO-8859-15] Robin Calmegård Siurua wrote:
I can't compile Asterisk on a Debian machine.
What is wrong? :/
debian... :-(
I was only able to compile asterisk when I gave up on doing it by myself
and decided to use the debian package (.deb).
Do it via apt-get. Remember to
[EMAIL PROTECTED] wrote:
Matthew Koch [EMAIL PROTECTED] wrote:
When I have 2 users logged in as Agents (using
AgentCallbackLogin) they will both ring when a new call comes in.
If, for instance, one of the users makes an outgoing call and a new
call comes in, the caller will get sent
On Thu, Jun 17, 2004 at 11:30:11PM +0200, Robin Calmeg?rd Siurua wrote:
Hi,
I can't compile Asterisk on a Debian machine.
I couldn't get asterisk to compile with the default openh323 and libpt
packages in debian so I went and grabbed the original source for:
pwlib v1.5.2
openh323 v1.12.2
The issue with the fax machine not answering was actually a physical
wiring issue (crosstalk with another line, its an old building) that is
now resolved.
Though the above shouldn't have affected RxFax from receiving a fax,
(since the FXO card plugs directly in to the demark) I decided to
101 - 183 of 183 matches
Mail list logo
