I've just down loaded Xten Lite and it is now build 1079.
It now finds the NAT firewall type and has loads more to configure.
But it doesn't work on my poor W95 tablet PC.
--
Dave Cotton
Directeur
Linux Autrement
193 rue Marcel Cerdan
84270 Vedene
04 90 23 30 81
Internet Sheriff Technology
- Original Message -
From: WipeOut [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, October 01, 2003 12:09 PM
Subject: Re: Web Admin - was:Re: [Asterisk-Users] CDR Web Search Frontend
I think there is room for everyones ideas, the more the better.. The
biggest problem I see
- Original Message -
From: Jamie Carl [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, October 02, 2003 1:58 AM
Subject: Re: [Asterisk-Users] CDR Web Search Frontend
As for the rest of this discussion, I have already started
work on this Asterisk Web Interface. (visit
Hi All
I have that error message:
WARNING[49159]: File chan_sip.c, Line 435 (retrans_pkt): Maximum retries
exceeded on call [EMAIL PROTECTED] for seqno 102
(Request)
What can be the problem?
Thanks!
miklos
___
Asterisk-Users mailing list
[EMAIL
Title: Message
The device is seen
in linux pnp:
isapnp: Scanning for
PnP cards...isapnp: Card 'Quicknet Internet PhoneJACK'isapnp: 1 Plug
Play card detected total
and I've installed
the drivers from the openh323 dev... but I can't get * to see
it.
Does anyone have
experience with
The location of the guest / iaxtel section having to be at the end is,
as it turns out, a configuration error on iaxtel. I hope to have it
straightened out shortly.
Mark
On Thu, 2 Oct 2003, Bartosz Jozwiak wrote:
Sometime yes sometimes no :) But thats the life :)
Ok but I fixed it. Just put
Tjardick van der Kraan wrote:
Is this a Jazz-inc copyright project or are you willing to just open it all
up and make it an astweb team effort ?
If you look at the sourceforge page its GPL..
http://sourceforge.net/projects/astweb/
Like i and others have said in earlier posts would be good to
Title: Message
its a fair
question: does anyone know any?
Bryan
Nolen
Lead Developer
http://Arc.Net.AU
http://cdonline.com.au
its a fair question: does anyone know any?
I'm afraid this doesnt answer your question and is a bit of a shameless
plug, but we have just started offering IAX (and SIP) termination in the UK,
so if this helps anyone out, please feel free to contact me.
Linus
Again, we need to seriously consider moving this to a separate mailing
list and getting a 'Features' thread started, as well as a 'Mission'
thread. These should get everyone's feet on the same path.
I agree that the web administration application needs to be be something
different than simply
Ok, see, now you're confusing what I said. Nowhere did I say I had the
102D. I said he never mentioned that it was the 102, irregardless of
the D. I *DO* have the 101, which is what he was talking about. No, it
doesn't mention it's the 101.
This argument has now proved silly, especially
No one else would step up to the plate so I
did.
Here is your list.
HAVE SOME FUN!
need any more just ask.
Troy Settle wrote: With all the discussion about licensing
issues and the sort, I think it's time for a full blown 3rd party
application to work with Asterisk while at the same time not
Well, that's odd.. Can you, then, with IAX, determine in which section
(first, second, last, etc...) you read your configuration in iax.conf,
rather than matching up with passwords?
--
Josh Roberson
Indigent Networks
1.877.677.9647 x1
[EMAIL PROTECTED]
-Original
the link is at www.pawbell.com
- Original Message -
From: sip
To: [EMAIL PROTECTED]
Sent: Thursday, October 02, 2003 8:57 AM
Subject: Call it Asterisk-Addons and let us go have some
fun?
No one else would step up to the plate so I
did.
Here is your list.
HAVE SOME FUN!
need any
Troy Settle wrote:
Again, we need to seriously consider moving this to a separate mailing
list and getting a 'Features' thread started, as well as a 'Mission'
thread. These should get everyone's feet on the same path.
I agree that the web administration application needs to be be something
Does
anyone know if there is a GSM player for
windows?
Dante
I found this information on how to make XP have a dialpad in Windows Messenger
which was awesome news
HKEY_CURRENT_USER\Software\Microsoft\MessengerService\CorpPC2PHone
(change it from 0 to 1 and a magic new choice to make phone calls appears)
only to be crushed hours later when I realized It
Anthony Minessale wrote:
I found this information on how to make XP have a dialpad in Windows
Messenger
which was awesome news
HKEY_CURRENT_USER\Software\Microsoft\MessengerService\CorpPC2PHone
(change it from 0 to 1 and a magic new choice to make phone calls appears)
only to be crushed
I use mine all the time. Things to check or set:
Under System SettingsNetwork
1- Set the IP of you * box in Outbound SIP Proxy
Under System SettingsSip Proxy
1- Enable yes
2- Username (the name or number in your SIP.CONF [brackets]
3- Leave Authorized User blank (and remark out in SIP.CONF
On Thu, Oct 02, 2003 at 09:58:37AM +1000, Jamie Carl wrote:
...
As for the rest of this discussion, I have already started
work on this Asterisk Web Interface. (visit
http://astweb.sourceforge.net). The current release is
still only the CDR section, but things are starting to
evolve and
So where do or can you get older version?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of WipeOut
Sent: Thursday, October 02, 2003 10:50 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] WINXP Messenger SIP Client (Good News, Bad
News)
Anthony
Hi Josh, Costas, Adam, et all
We do sell the phones.
http://www.chagres.net/products/voip/phones.html
and digium cards
http://www.chagres.net/products/voip/cards.html
plus new things real soon :)
and if anyone ever has a problem, go yell at me
and I'll try like crazy to fix it.
john
On 2/10/03 3:51 pm, Joseph Finley [EMAIL PROTECTED] wrote:
I use mine all the time. Things to check or set:
Under System SettingsNetwork
1- Set the IP of you * box in Outbound SIP Proxy
Under System SettingsSip Proxy
1- Enable yes
2- Username (the name or number in your SIP.CONF
Yo all,
I have a problem with a Dutch (KPN) PSTN-line, connected to an X100P.
The call wil sound OK at first, but after 10-20 minutes, the audio will
start to crackle. Soon after that, this crackle turns into a continuous
noise and the parties won't be able to hear eachother anymore. It also
Grandstream 102D won't be available until December,
at the earliest.
I have 101's in stock now and can ship same day as
the order is funded.
102 (not the D model) are on backorder and I expect
inventory by the end of next week.
Transfering and other functions are really going to be
a matter
Check /proc/interrupts to make sure the cards are not shareing IRQs with
anything.
On Thu, 2003-10-02 at 11:43, The Traveller wrote:
Yo all,
I have a problem with a Dutch (KPN) PSTN-line, connected to an X100P.
The call wil sound OK at first, but after 10-20 minutes, the audio will
start to
Martin Pycko wrote:
We send SIP messages to that device up to 6-7 times and then we stop and
this message shows on the console.
WARNING[49159]: File chan_sip.c, Line 435 (retrans_pkt): Maximum retries
exceeded on call [EMAIL PROTECTED] for seqno 102
(Request)
So it isn't really an error then, but
Yo Eric,
On Thu, Oct 02, 2003 at 11:56:44 -0500, Eric Wieling wrote:
Check /proc/interrupts to make sure the cards are not shareing IRQs with
anything.
Sorry, forgot to mention it. All Zaptel-cards in that machine
already have their own unique interrupts. I will try moving the
cards to
Can someone post their experiences with these phones together with
asterisk, and give an impartial listing of what features they find
indispensible, and others that are a pain to have missing.
What codec configurations do people use these phones in currently with
asterisk ?
Can someone
Two IDENTICAL MACHINES (same motherboard, same RH 7.2, same *)
communicate through IAX2. Everything works ok on machine 1. On machine
2, if I try 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
WipeOut wrote:
Olle E. Johansson wrote:
I still can't get Windows messenger to register with a secret to
Asterisk.
Anthony - do you connect without registering or does Windows messenger
register properly with your * ?
/O
Have you tried forcing Asterisk to use plain text authentication for
Hi Martin
Please explain, why did you send the messages?
miklos
- Original Message -
From: Brian Capouch [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, October 02, 2003 2:04 PM
Subject: Re: [Asterisk-Users] error message 49159
Martin Pycko wrote:
We send SIP messages to
auth=plain
On Thu, 2003-10-02 at 12:36, Olle E. Johansson wrote:
WipeOut wrote:
Olle E. Johansson wrote:
I still can't get Windows messenger to register with a secret to
Asterisk.
Anthony - do you connect without registering or does Windows messenger
register properly with your
On 2/10/03 5:59 pm, Joseph Finley [EMAIL PROTECTED] wrote:
Very basic:
Sip.conf
;
; SIP Configuration for Asterisk
;
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind to
context = sip-phones; Default for
Bryan,
IP Telephonics is developing a VoIP gateway service in Australia.
It is not yet operational.
If you want to discuss anything please email me offlist.
Peter Brown
At 23:23 2/10/2003 +1000, you wrote:
its a fair question: does anyone know any? Bryan Nolen Lead
Developer
It's a WARNING, so if you want to know why your phone doesn't work you can
read it or ignore it.
regards
Martin
On Thu, 2 Oct 2003, Brian Capouch wrote:
Martin Pycko wrote:
We send SIP messages to that device up to 6-7 times and then we stop and
this message shows on the console.
This probably has an easy solution, but I found it yet. How can I get out
of a remote console after using ssh to get into the box, making changes,
reload etc. without stopping *?
Thanks in advance.
Sincerely,
Andy Hester
Consero
___
Asterisk-Users
I'm trying to get musiconhold to work w/ my Asterisk system, and I'm not
having much success yet.
First, I noticed that nothing happened even after I had enabled all of
the options in zapata.conf setup a sample extension in extensions.conf.
Then I read something about how Asterisk uses mpg123
On Thu, 2 Oct 2003, Michael T Farnworth wrote:
The people at chagres.net appear to sell the phones.
They do in fact sell the phones, as I bought one from them :)
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
On Thu, 2003-10-02 at 14:53, Andy Hester wrote:
This probably has an easy solution, but I found it yet. How can I get out
of a remote console after using ssh to get into the box, making changes,
reload etc. without stopping *?
Not that it is clean or neat, but control-c is what I use.
--
Simply run the /usr/src/asterisk/safe_asterisk
And then type /usr/sbin/asterisk -vvvgcr
^
r being remote console and then you can do everything as if you ran it
directly and exit as you wish or STOP NOW to kill it.
Regards,
Joe
-Original
use quit or ctrl-D
Martin
On Thu, 2 Oct 2003, Andy Hester wrote:
This probably has an easy solution, but I found it yet. How can I get out
of a remote console after using ssh to get into the box, making changes,
reload etc. without stopping *?
Thanks in advance.
Sincerely,
Andy Hester
I did a google search and did not come up with anything on this. I loaded Gastman on
a Windows 2000 pro PC and it will not work. It says the following.
gastman.exe has generated errors and will be closed by Windows. You will need to
restart the program. I have tried to set the compatability
That sound you hear is the sound of mpg321 running. Do an ls -l
/usr/bin/mpg123 if it's a symlink to mpg321 then you have found your
problem.
On Thu, 2003-10-02 at 14:54, john lawler wrote:
I'm trying to get musiconhold to work w/ my Asterisk system, and I'm not
having much success yet.
on Thu, Oct 02, 2003 at 02:53:00PM -0500, Andy Hester wrote:
This probably has an easy solution, but I found it yet. How can I get out
of a remote console after using ssh to get into the box, making changes,
reload etc. without stopping *?
Thanks in advance.
Looks like exit will release
Hmm, impartial, well I sell them but I'll try :) :)
Bad things:
Base color, white... new colors coming soon
Call waiting ring is a bit annoying
The normal ring is really annoying, they are changing this
good things:
they work
cheap
easy to setup
sound quality very good
multi codec selection
Colors other than white seem to be hard to get from GS.
I've been asking for multiple weeks for something other than
white. I don't have solid ship dates on black or any other
color.
I'll let folks know when we have something other than white.
On Thu, Oct 02, 2003 at 12:45:50PM -0400, Steve
WipeOut wrote:
Anthony Minessale wrote:
I found this information on how to make XP have a dialpad in Windows
Messenger
which was awesome news
Some more crushing news is if you upgrade MSN messenger past ver 4.x
it
no longer uses SIP.. (so I have been told)..
MSN messenger does not use
Look at www.pawbell.com they have the frontend. They
even have the NAT problem fixed!
- Original Message -
From:
23
To: [EMAIL PROTECTED]
Sent: Thursday, October 02, 2003 12:01
PM
Subject: [Asterisk-Users] Front end
Hi,
Can anyone help mewith a few links to
Wow look at the choices :)
Thanks everyone for the info. I'll try them out.
Sincerely,
Andy Hester
Consero
(817)375-1244
(817)937-7977
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
-Original Message-
From: Martin Pycko
Sent: Thursday, October 02, 2003 4:13 PM
use quit or ctrl-D
Martin
From what I can tell, * doesn't honor EOF, at least I've had no luck with
it.
--
Troy Settle
Pulaski Networks
http://www.psknet.com
540.994.4254 ~ 866.477.5638
Or you can use safe_asterisk to start * then asterisk -r to connect
bkw
On Thu, 2 Oct 2003, PJ Welsh wrote:
on Thu, Oct 02, 2003 at 02:53:00PM -0500, Andy Hester wrote:
This probably has an easy solution, but I found it yet. How can I get out
of a remote console after using ssh to get
Title: Message
I was
able to get it to register just fine, but I get no sound. It connects
fine, no sound.
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anthony
MinessaleSent: Thursday, October 02, 2003 4:31 PMTo:
[EMAIL
Here is my setup
7960(A)--Firewall/PAT--dsl-Internetdsl--Firewall/NAT---7960(B)
| |
| |
7960(C)--NAT--cable- -dsl -- Asterisk
(A) can communicate with (C) only when C is
Here is my setup
7960(A)--Firewall/PAT--dsl-Internetdsl--Firewall/NAT---7960(B)
| |
| |
7960(C)--NAT--cable- -dsl -- Asterisk
(A) can communicate with (C) only when C is
Hi guys,
Don't want to ruffle feathers, but did I see Ratnakar's email address as
being @cisco.com.
Is Cisco thinking of using Asterisk? Just a thought.
Welcome Ratnakar
Peter
From: [EMAIL PROTECTED]
At 14:50 2/10/2003 -0700, you wrote:
Here is my setup
okay, you no longer have to have [iaxtel] as the last entry. It was a
config error on x...
mark
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Hi James--
I got a dialer working without too many hiccups about two
months ago. It relies on changes to chan_agent, app_queue,
a PostgreSQL backend, a Tcl-* manager interface, a bunch of
Tcl glue, and some cron jobs. The results for each call are
logged in right through the phone key pad, and
At 08:54 AM 10/3/2003 +1000, you wrote:
Hi guys,
Don't want to ruffle feathers, but did I see Ratnakar's email address as
being @cisco.com.
Is Cisco thinking of using Asterisk? Just a thought.
Well if I was a large hardware manufacturer I would certainly be testing
compatibility of my hardware
so does that mean I can now have multiple iaxtel numbers?
Doug
On Thu, 2 Oct 2003 17:56:43 -0500 (CDT), Mark Spencer [EMAIL PROTECTED]
wrote:
okay, you no longer have to have [iaxtel] as the last entry. It was a
config error on x...
mark
___
Chris are you willing to post the code?
Peter
At 19:09 2/10/2003 -0400, you wrote:
Hi James--
I got a dialer working without too many hiccups about two
months ago. It relies on changes to chan_agent, app_queue,
a PostgreSQL backend, a Tcl-* manager interface, a bunch of
Tcl glue, and some cron
On Thu, 2003-10-02 at 17:09, C. Maj wrote:
I know there was a separate list setup for discussions about
a predictive dialer, and I would like to contribute my code
there but don't remember who made the list or if it has ever
seen any traffic.
That list was set up by me back in April.
yes i work for cisco. But playing around with asterisk is purely personal.
It is in no way related to my work at cisco.
I tried using another email id yesterday, but the post never showed up. Even
though I got a mail from the news server that it was posted.
Thanks,
=ratnakar
Jon Pounder wrote:
We have an odd problem, where inbound H323 (chan_h323) calls will sometimes
cause a Ringing Congestion that appears to keep the channels open and never
release it until we kill and restart asterisk. These Ringing Congestions
start to pile up, which eventually crashes Asterisk.
H323 Gateway -
I think there's a confusion here.
There're 2 different products:
1. MSN Messenger 4.6/4.7 (Windows 9x, ME, 2K)
2. Windows Messenger 4.7/5.0 (Windows XP)
I was told MSN Messenger 4.7 works with the registry hack. Have never
tested this myself, though I'm very certain 5.0 doesn't work.
Windows
I checked out gnophone from CVS and I'm trying to build it.
I got as far as getting a ./configure built and that to
build the makefiles and then I find compile poblems in the source.
Leads me to thing maybe 0.2.5 is still a work in progress.
true?
One more question. What software phones are
Mark,
The location of the guest / iaxtel section having to be at the end is,
as it turns out, a configuration error on iaxtel. I hope to have it
straightened out shortly.
Ok but I fixed it. Just put the guest section in iax.conf all the way on
the end.
And right now it works for me.
I've searched the site with google, but can't think of the magic words I
guess.
I got a swap out TDM30 today to replace my buzzy one.
I swapped it with the older one, swapped out the FXS modules, hooked it
up to the computer's power supply, and booted, but the wcfxs driver
won't load--it
there is no security risk, actually it reduces access. The bug meant that a
[iaxtel] section had to be the last section in the file, otherwise it would
be ignored. If you aren't having a problem with authenication in iax (people
who have access but are getting rejected), you won't need the update.
5volunteers needed to test NAT Transversal
software in realtime enviroment. Must be behind a firewall.
Reply to [EMAIL PROTECTED] if you would like to join
the test.
This message was checked by MailScan for WorkgroupMail.
www.workgroupmail.com
Is it showing up on /proc/pci? It should be a tigerjet. Does dmesg
report anything unusual? There are *some* machines which have no no 3.3V
supply. If that's the story with yours, send me your machine and I'll try
an experimental fix on it.
Mark
On Thu, 2 Oct 2003, Brian Capouch wrote:
Mark Spencer wrote:
Is it showing up on /proc/pci? It should be a tigerjet.
Yes. I put the other card back in (production machine) but over the
weekend I'll get the card in there and capture the output of lspci.
Does dmesg
report anything unusual?
Nope. Doesn't show any sign of seeing
Shaun == Shaun Ewing [EMAIL PROTECTED] writes:
Shaun - Original Message -
Shaun From: Chad R. Graham
For the first 15 seconds of a call I get echo on the ata 186 side
only. I assume after that the echo canceller kicks in but is there
any way to make it happen faster?
Shaun
So, a quick look through a full session of a call between two SIP
phones doesn't show that there is a Date: header being inserted
anywhere in the SIP headers. I _swear_ I saw that earlier, and in
fact, I recall watching Mark fix some syntax this spring on the floor
of the VON show to make the
LDM == Louis-David Mitterrand [EMAIL PROTECTED] writes:
LDM Having purchased a license for 5 g729 channels on Digium's web
LDM shop I thought registration and installation would be a snap. NOT.
LDM I followed registration instructions to the letter but it failed
LDM with that message:
LDM
Brian == Brian West [EMAIL PROTECTED] writes:
Brian Just a heads up.. you can't loop switch statements ie
Brian BOX A switch = BOX B BOX B switch = BOX A
[...]
I was actually wondering -- why?
This is something I very naturally wanted to do the first time I
configured two *'s. I wanted them
Mark == Mark Spencer [EMAIL PROTECTED] writes:
[...]
Mark No problem, it's easy to get confused :) I would, however, take
Mark issue with the GPL being evil. It's not my *ideal* license,
Mark but it certainly is good enough.
Just for the reference, while we're at it. GPL does have an issue,
On Thu, 2003-10-02 at 12:04, Jan Rychter wrote:
I'm also hearing this, with an analog phone (connected to an
S100U). Rather annoying.
Incoming calls have an entirely different problem for me, a disastrous
5-8 second crackling/clicking sound, which seems to go quiet a while
after you start
On Thu, 2003-10-02 at 07:51, Josh Roberson wrote:
Ok, see, now you're confusing what I said. Nowhere did I say I had the
102D. I said he never mentioned that it was the 102, irregardless of
the D. I *DO* have the 101, which is what he was talking about. No, it
doesn't mention it's the
Does anyone know if there's public voice detection algorithms available? I've
scoured the net for the last hour or so, and I can't come up with anything
except a few proprietary or embedded solutions.
I know dsp.c uses goertzel algorithms for DTMF detection, but how does one
detect voice?
I
Hi
I am trying to get started with asterik PBX.I need to establish a call
between an H.323 terminal (example : Netmeeting) and a SIP terminal.
I would like to know :
1)What are the configuration to be done Asterik PBX (I coild build the
source on Redhat Linux 7.3).
2)How to configure
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