Hi,
(B
(BI'm using Asterisk-1.0.0 on Fedora Core 1
(B
(BDate: Mon, 21 Feb 2005 14:22:13 +0900 [zone:Tokyo],
(B[EMAIL PROTECTED] mentioned in msg: Re: [Asterisk-Users] Asterisk
(BH323 support that ...
(B
(B For channel asterisk-oh323-v0.6.5
(B need
(B
Then how can i use web interface to configure?
On Mon, 21 Feb 2005 15:30:55 -0600, Kristian Kielhofner [EMAIL PROTECTED]
wrote:
Kiran Vahaja wrote:
Hi Folks,
I installed [EMAIL PROTECTED] on my PC. It went through the installation
and all. But now i get a command line login window.
Kiran,
From another pc on your network log into the web page by
http://ipaddress-of-the-asterisk-server/maint
Cheers,
Dean
p.s. RTFM
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kiran
Vahaja
Sent: Monday, February 21, 2005 4:05 PM
To:
could you help me out with this? I have a posting on this list, bu
nobody has replied yet. Titled "why can't I make IAX calls between 2
asrterisk servers"? I'd appreciate.
-chuks.
Original Message Subject: Re:
[Asterisk-Users] bridging iaxtel calls to PSTNFrom: "Michael
Graves"
On Mon, Feb 21, 2005 at 08:42:33AM +, Julian J. M. wrote:
Check your soundcard controls... maybe it's recording what you hear
or PCM, thus sending it again to the other party.
Are you saying that when using a sound card with your softphone the PCM should
be set to 0?
I never knew that...
Hello all,
This might be one for Digium, but I would like to see some type of Wiki
that people would have to wade through before they would get the
information on how to subscribe to the list.
This wiki should cover most of the basic stuff that gets asked over and
over again just to help
I think someone did reply to this. Don't bother trying to use iaxtel for
your connections. Its down far more then its up.
If you want help setting up an iax connection directly between your two
systems, then post what you've got that pertains to this from iax.conf
and extensions.conf. No one is
The same has been proposed several times over the last nine months,
both on -users and -dev, and its simply been ignored.
From: Kristian Kielhofner [EMAIL PROTECTED]
Subject: [Asterisk-Users] Suggestion for noise reduction on Asterisk-Users
Date: Mon, 21 Feb 2005
What it is necessary to create a termination:
Asterisk server ?
card ?
type of network ?
another software ?
Thank you.
Découvrez le nouveau Yahoo! Mail : 250 Mo d'espace de stockage pour vos mails !
Créez votre Yahoo! Mail sur http://fr.mail.yahoo.com/
Are their any good chooses for IAX Adapters?
-Thanks
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
Hello,
actually I did, but nobody responded to that. So, here it is
one more time:
___
Hello,
can someone tell me what's wrong with this? I can't make toll
free calls via iaxtel. Here's the definition in my extensions.conf
[iaxtel-trunks]
;
;outbound 1-700 and toll free calls go
I'll second PaulH's recommendation of the Telular SX-5e units. They
plug into an FXO on the Asterisk machine (or wherever). Put anyone's
capped SIM into the thing, and you're communicating with the GSM
network. As a little added bonus, there's a serial port for sending
and receiving text messages.
Hello,
Can anyone help with this please?
thx,
chuks
Original Message Subject:
[Asterisk-Users] Why can't I make inter IAX calls between2 Asterisk
serversFrom: [EMAIL PROTECTED]Date: Mon, February 21, 2005
11:04 amTo: asterisk-users@lists.digium.com
Hello,
two questions:
1: How
Rich Adamson wrote:
The same has been proposed several times over the last nine months,
both on -users and -dev, and its simply been ignored.
Rich,
I think that I said something like that somewhere in the message. But
it turned into much more than I thought it would, so I am not sure...
Do
Guys.
Im using IAX and FWD and I think everything is setup fine.. someobdy just
tried calling me but my phone jus ran once and sent them straight to the
voicemail.. the logs show this:
-- Accepting AUTHENTICATED call from 65.39.205.121:
requested format = ulaw,
requested
Hello,
We have a call redirection system setup inhouse to send calls from an
incoming line on a T1 to an external dialed out number:
Zap(call comes in) - Asterisk - Zap(call dials out)
The call comes in on a Robbed-bit T1 (d4/ami EM Wink) and goes out on a PRI.
We are using Asterisk release
I would like to get in contact with Brian Elton. He posted information to
this list regarding problems with an Avaya 4602, late last year. I am now
experiencing a similar issue, and would like to know if/how it was resolved.
Thank you.
Sincerely,
Trevor
I have an Avaya 4602 IP phone that was previously working with Asterisk. It
was being used elsewhere for several months, and I recently set it up again
to work with Asterisk. Everything works fine for several minutes -- I am
able to receive and make calls as expected. However, after a few
Kristian Kielhofner wrote:
Hello all,
This might be one for Digium, but I would like to see some type of
Wiki that people would have to wade through before they would get the
information on how to subscribe to the list.
How many more times do we have to read posts of I just downloaded
This wiki should cover most of the basic stuff that gets asked over
and
over again just to help reduce the amount of repetition that most of you
have probably noticed takes place here.
Problem is, Wikis in general suck and voip-info.org in particular is quite
useless except as a random
iaxtel is not working and hasn't been for some time. All of us that
have 700 numbers get the same response that you are.
Hello,
actually I did, but nobody responded to that. So, here it is one more time:
___
Hello,
can someone tell me what's wrong
BTW, I did need to suid the zttool-cli command to root, as the normal
BB
user doesn't have the needed permissions. I haven't looked into this,
but if anyone has a suggestion on a better way to do this, feel free
to
let me know.
It's called sudo
jens
John Novack wrote:
Dare I suggest that a MUCH better job of documenting would go a long way
towards eliminating the problems you mention?
Now I realize that programmers are much more interested in writing code
than documentation, as well as moving on to the next hot feature than
making sure
On Feb 21, 2005, at 23:36, [EMAIL PROTECTED] wrote:
Hello,
actually I did, but nobody responded to that.
Maybe people would look at it if you stopped sending HTML mail.
jens
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Colin Anderson wrote:
Problem is, Wikis in general suck and voip-info.org in particular is quite
useless except as a random clicky-clicky exercise. You ever use the search
on voip-info.org? It's almost like someone goofed setting it up and the
search results are ordered least relevant first.
use another pc.
--dalon
On Mon, 21 Feb 2005 15:56:26 -0600, Kiran Vahaja [EMAIL PROTECTED] wrote:
Then how can i use web interface to configure?
On Mon, 21 Feb 2005 15:30:55 -0600, Kristian Kielhofner [EMAIL PROTECTED]
wrote:
Kiran Vahaja wrote:
Hi Folks,
I installed [EMAIL
And yet another "helpful"
comment to clog up the list.
Some people use HTML
Some people top post
Some people don't read too well
Some people aren't as skilled as others in searching.
GET OVER IT
JMO
John Novack
Jens Vagelpohl wrote:
On Feb 21, 2005, at 23:36, [EMAIL PROTECTED] wrote:
Lets point them to google site:voip-info.org
or site:lists.digium.com then. We do a lot of that once they get on the
list. Why not before?
OK, Ollie J if you are listening maybe you might consider appending those
links to your monthly or weekly list etiquitte reminders. Post them daily,
even.
Colin Anderson wrote:
Lets point them to google site:voip-info.org
or site:lists.digium.com then. We do a lot of that once they get on the
list. Why not before?
OK, Ollie J if you are listening maybe you might consider appending those
links to your monthly or weekly list etiquitte reminders.
Hi,
Does it's possible to get more information about your design ?
Thanks,
--
Joel Vandal
- Original Message -
From: Race Vanderdecken [EMAIL PROTECTED]
To: [EMAIL PROTECTED]; 'Asterisk Users Mailing List -
Non-Commercial Discussion' asterisk-users@lists.digium.com
Sent: Thursday,
ATAs with the PA168 - a very popular chip with quite a few of Chinese
manufacturers.
-Michael
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tim Lewis
Sent: Monday, February 21, 2005 4:36 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users]
We have a call redirection system setup inhouse to send calls from an
incoming line on a T1 to an external dialed out number:
Zap(call comes in) - Asterisk - Zap(call dials out)
The call comes in on a Robbed-bit T1 (d4/ami EM Wink) and goes out on a PRI.
We are using Asterisk release
This might be one for Digium, but I would like to see some type of
Wiki that people would have to wade through before they would get the
information on how to subscribe to the list.
How many more times do we have to read posts of I just downloaded
the Asterisk and now how
Rich Adamson wrote:
This same topic comes up about every month or two, and the exact same
words are used over and over again. The last run at this was on the -dev
list about one/two months ago and shouldn't be hard to find.
If memory serves anywhere near correct (which is a stretch), lots of
folks
- Original Message -
From: James Bean [EMAIL PROTECTED]
I am going to now sit in a corner and go quietly insane while playing
the banyo with no strings.
I'd probably go insane, too, if I was trying to figure out how the heck to
play a banyo
;)
Hi,
Is there anybody out there that can e-mail me the following
files?
Get oh323 fromhttp://www.inaccessnetworks.com/projects/asterisk-oh323/Libraries/openh323-Janus_patch4-src-tar.gzGet
pwlib
Hi, all
I am doing prrof of concept system. I will have two IP phones connected to
Asterisk box. Box itself will have 1 PSTN conenction and one analog phone
conenction. A basic minimal configuration.
At the moment I am planning to use an old PII-350 with 128M of RAM I have lying
around. I can
Rudolf Ladyzhenskii wrote:
Hi, all
I am doing prrof of concept system. I will have two IP phones connected to
Asterisk box. Box itself will have 1 PSTN conenction and one analog phone conenction. A
basic minimal configuration.
At the moment I am planning to use an old PII-350 with 128M of RAM I
Hi there,
since I found a couple of reports with complaints concerning zaprtc I
thought that one or the other user might be glad to know that it works
indeed. All that was necessary was to copy all *.h files from /zaptel
into /zaptelrtc and then do make followed by make load.
Of course you'll
On Tue, 2005-02-22 at 11:41, John Novack ( Mozilla - portable ) wrote:
Rudolf Ladyzhenskii wrote:
Hi, all
I am doing prrof of concept system. I will have two IP phones connected to
Asterisk box. Box itself will have 1 PSTN conenction and one analog phone
conenction. A basic minimal
I have a server setup that runs sip no problem. I want to try a cisco phone.
how do I
a) Tell if I have skinny support loaded
b) Load it onto a debian system
Many thanks
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Thanks.
I guess, I will have to try it and see. Mine is one of those small form factor
COMPAQ boxes. I will try to get full specs from COMPAQ/HP.
What about load Asterisk puts on processor if you do, for example, IP-IP call
and IP-PSTN call? Since I will use Polycom phones, I will use SIP.
John Novack writes
Dare I suggest that a MUCH better job of documenting would go a long way
towards eliminating the problems you mention?
Now I realize that programmers are much more interested in writing code
than documentation, as well as moving on to the next hot feature than
making sure
I have two comments:
a. It maybe doesn't work because of the PCI specifications the box support.
If was manufactured before Jan 2000, it is quite probably that it won't
recognize the Digium cards.
b. From the point of view of load, I see no problems, I think the specs of
the machine are enough for
On Sat, Feb 19, 2005 at 11:17:14AM +0100, Kurt Bauer wrote:
So all I have to do is change the line 'switchtype=euroisdn' in zapata.conf
to 'switchtype=Q.SIG' ??? I still haven't got the point how to tell my *
Box that it is now Q.SIG aware :-o
Well, switchtype=qsig, not switchtype=Q.SIG.
Hi,
We're deploying a small VoIP solution for a group of teleworkers.
Naturally, this exposes us to all sorts of fun, most of which we seem to
have working properly. However, some NAT issues are still bugging us and
we have noticed that often these situations didn't exist when users were
Dean,
I'd be very interested in helping with this effort.
I've worked with both SGML and XML in the past (I used
to work at SoftQuad in Toronto, one of the original
providers of SGML and HTML tools), and have written
several DTDs, both for SGML and XML.
I think it would be fun to work on an XML
Try to analyze this link: Asterisk - Dual -Server:
http://www.voip-info.org/wiki-Asterisk+-+dual+servers
#Joseph
On Mon, 2005-02-21 at 15:41 -0700, [EMAIL PROTECTED] wrote:
Hello,
Can anyone help with this please?
thx,
chuks
Original Message
First off -
change:
exten = _2000,1,Dial(IAX/master:[EMAIL PROTECTED]/[EMAIL PROTECTED])
to:
exten = _2000,1,Dial(IAX2/master:[EMAIL PROTECTED]/[EMAIL PROTECTED])
On Mon, 21 Feb 2005 13:00:39 -0500, kurt x [EMAIL PROTECTED] wrote:
I have two * boxes running two differnet versions of *.
Box
Hi,
That much exists in a well standardised form (VoiceXML at
http://www.w3c.org/Voice/) and several not so well standardised forms.
It is what comes beyond the basic XML that needs to be implemented.
Regards,
Steve
beonice wrote:
Dean,
I'd be very interested in helping with this effort.
I've
Title: Message
Does anyone know of
any sip wifi phones? Only one i can find that is redily availiable is the zyxel
prestige 2000w and from what i hear it is flaky.
Kurt Fankhauser
WaveLincwww.wavelinc.com114 S. Walnut
St.Bucyrus, OH 44820419-562-6405
--
No virus found in this outgoing
Rudolf Ladyzhenskii wrote:
Hi, all
I am doing prrof of concept system. I will have two IP phones connected to
Asterisk box. Box itself will have 1 PSTN conenction and one analog phone conenction. A
basic minimal configuration.
At the moment I am planning to use an old PII-350 with 128M of RAM I
I have set up FWD via IAX service. I have tested the IAX service with
613, echo test, and 612, saytime. It all works well.
However when ringing a FWD user, I got this error all the time:
Connected to Asterisk CVS-v1-0-02/01/05-09:34:45 currently running on
chat (pid = 8282)
chat*CLI
Verbosity
Receiving calls from LiveVoip DIDs results in dropped DTMF digits.
I'm using SIP, not IAX, and I've tried this without a dtmfmode and with
dtmfmode in all the various permutations. Note that LiveVoip does not
instruct us to put any dtmfmod statement in.
The server is set to do ulaw and I've
how would one dial multiple multiline sip phones (cisco 7960) and
making sure that all the phones ring on the next available line
appearance?
I'm currently using the local channel to accomplish this but I'm
having some trouble. Here is the configs:
each cisco 7960 phone has six registrations in
These are the currently available wireless IP Phones that we are aware of.
Pulver
Zyxel 2000
Hitachi IP5000
Clipcomm CP-100E (Traditional desktop phone that is WIFI extensible with
an optional PCMCIA wireless card)
We are also currently in the midst of testing a new wireless phone from a
Anybody know a good IAX provider for Canadian DIDs?
Mohit.
--
Mohit Muthanna [mohit (at) muthanna (uhuh) com]
There are 10 types of people. Those who understand binary, and those
who don't.
___
Asterisk-Users mailing list
Would enabling Busydetect really help if Asterisk thinks it detects an
On-Hook?
MATT---
-Original Message-
From: Rich Adamson [mailto:[EMAIL PROTECTED]
Sent: Monday, February 21, 2005 7:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Zap
Any chance that is a bad number??? I do not see anything that would
cause this unless there is a problem with the number you are trying to
dial.
Maybe do am iax debug to get more info??
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of david kwok
Sent:
Please let me know the answer to this one.
I set up FWD today and I am having the same problem.
Thanks for the iax debug tip.
Race Vanderdecken
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Robert
Webb
Sent: Monday, February 21, 2005 9:32 PM
To:
Hi,
Thanks to Andrew Kochetkoff for sending Asterisk-oh323 files while
inaccessnetworks web page was down.
Now, I have a problem when compiling Asterisk-oh323 versions 0.7.0 or 0.7.1.
I get the following error:
/usr/include/_G_config.h:52: confused by earlier errors, bailing out
make[1]: ***
We did our proof on concept on a celeron 600 - which was fine to run 2 software
and 2 hardware phones off.
Original testing was with sjphone and xlite. (software phones)
We didn't fit a TDM card until we set up our first box, which was a dual
p3-1000.
Later,
PaulH
-Original Message-
Hi all,
I am using * as a PBX for a Broadvoice VoIP account. It had been working
well since about last November, although not perfectly (similar
disconnection problems, although I am pretty sure it had to do with my
PPPoE setup, but I think these issues were resolved). As of a few weeks
ago,
I've been using FC2 with Kernel 2.6.9, the hardest thing for me was getting
my capi startup script right, you should not have any capi related stuff in
modprobe.conf. I have included my startup script. If you are using a DID
or Point to Point line for the Fritz! then change protocol=2 to
no, but you have to install the wanpipe from sangoma as well
you can get it at: frp.sangoma.com/linux/current_wanpipe
On Mon, 21 Feb 2005 08:09:45 +0200, Altus Snyman [EMAIL PROTECTED] wrote:
Good day all
Is there any difference in the sangoma zaptel.conf and zapata.conf then
other cards
And I forgot - once we were finished with the 600 we gave it to Jurgen.
Caring and sharing,
PaulH
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Hales
Sent: Tuesday, 22 February 2005 2:03 PM
To: 'Asterisk Users Mailing List - Non-Commercial
(Thanks Paul) And that same box now has a TDM-400 card in it, all 4
ports used. Two ATs are registered with the server as well. Most of
the time, it doesn't even break a sweat. I would not want to use it
for anything close to production though.
On Tue, 22 Feb 2005 14:25:00 +1100, Paul Hales
Seems I have the same problem, when I call a FWD number or they call me, my
phone rings once and then the console says everybody was busy and they get
my voicemail...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of david kwok
Sent: Lunes, 21 de Febrero
Its not flaky at all. We have 2. The only bad thing is its lack of power.
I'm not that too familiar with WiFi devices but it only has about 2hrs worth
of talk time and about 10hrs of standby time. I'm not really sure on the
standby time, but it had a full battery when I left it on my desk at 5 on
Hello All,
I have integrated my asterisk server to cisco call manager, now in the
process of doing video for asterisk. I understand from the wiki that
asterisk supports Wooksung WVP-2000 SIP (hardware phone). Are there any
others in the market, I mean hardware phones.
I know sometime last year
Sounds like I'm going to have to wait and hope some new phones are
released.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matthew
Boehm
Sent: Monday, February 21, 2005 7:55 PM
To: Asterisk Users
Subject: Re: [Asterisk-Users] sip wifi phone?
Its not
Title: Message
Hi Kurt,
I agree the Zyxel WiFi 2000w is a very
difficult phone to use. Extremerely laggy in the interface etc.
I just sent back 5 to my supplier as they
were not very good. The only option I can see is a Cisco based one, but you
need a mint to buy it.
Thanks
On Feb 21, 2005, at 3:12 PM, Jon Gabrielson wrote:
What are the advantages/disadvantages of using
a ZAP FXS port versus using one of the many
small ethernet FXS devices on the market. The
ZAP FXS talks directly to asterisk over PCI. Is this
an advantage? The ethernet devices I assume
speak
Tell me about.
I was on a project once that tried to use DELL PDA's with a soft phone
in them to be wi-fi back to asterisk. Unless it sat in a cradle,
attached to a wall outlet, you couldn't make it through a 10 minute
call.
Lucky if we got the PDA thing to stay awake for a call to come in.
One
That's my plan as well.
PaulH
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kurt Fankhauser
Sent: Tuesday, 22 February 2005 6:05 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] sip wifi phone?
Sounds
Hi,
I wish to initate calls from a web interface, by clicking on a link and then
connecting to the automatic outgoing call by picking up an analogue phone.
I've got one fxs and one fxo and I wish to automate the call using a call
file (which I can do now). How can I pick up a handset and
Guys..
Anybody ever had problems with noise on calls after certain amount of
minutes? It happening to me since yesterday.. after you place or get a call
using your SIP Phones and X100P cards, after around 3 or 4 minutes of talk,
I get noise like interference and either they ant hear me or I cant
Kurt Fankhauser wrote:
Sounds like I'm going to have to wait and hope some new phones are
released.
Kurt,
Check out my message from October:
http://lists.digium.com/pipermail/asterisk-users/2004-October/067769.html
and here is a link to the Broadcom page:
what about senao SI-7800H?
this is the link:
http://www.senao.com.tw/english/product/product_wireless01_outdoor_1.asp?pgtl=Wirelesstp1id=02tp2id=06proid=000131
On Mon, 21 Feb 2005 23:42:30 -0600, Kristian Kielhofner [EMAIL PROTECTED]
wrote:
Kurt Fankhauser wrote:
Sounds like I'm going to
What did the Telco have to do? I want to get our outbound callid working
properly...
PaulH
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rod Bacon
Sent: Friday, 18 February 2005 4:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
hi,
anybody have tried to connect from asterisk PBX to
textron voip gateway?...a 4FXO gateway..h323 capable...
=
Kolosos
Philippines
__
Do you Yahoo!?
Read only the mail you want - Yahoo! Mail SpamGuard.
On Mon, 21 Feb 2005 23:26:24 -0600
Anton Krall [EMAIL PROTECTED] wrote:
Guys..
Anybody ever had problems with noise on calls after certain amount of
minutes? It happening to me since yesterday.. after you place or get a
call using your SIP Phones and X100P cards, after around 3 or 4
Good day all
I registered at a few sip server in different countries
Now I want to route outgoing calls for that country threw that sip
server and all the others there my own pstn,ZAP card.I already
registered asterisk with them.
How would my extensions.conf look.This is what I have but no matter
Hi
If you going to run digium hardware, please make sure
that machine does PCI
2.2.. otherwise you will NOT be able to get the card
to work.
the slowest machine we found was a pIII that supported
pci 2.2
chow
Liaan
- Original Message -
From: Paul Hales [EMAIL PROTECTED]
To: 'Asterisk
Greetings *`s,
I am having what appears to be a small problem, but the frustration is
erally getting to me, what am I doing wrong here ?
I used AMP to set up a custom menu, so if caller presses 1 it goes to
ext200, if caller presses 2 it goes to ext201 etc etc...
Now I have created a third
Trevor G. Hammonds wrote on Monday, 21 February 2005 2:54 PM:
I suspect this may be related to the MWI indicator and the mailbox=
statement in my sip.conf file, as the last time I was using this
phone with *, it was not set up to use voicemail.
I have confirmed that this issue is directly
101 - 186 of 186 matches
Mail list logo
