On Thu, 22 Apr 2004, Steven Critchfield wrote:
...
VoIP phones have the benefit of linear growth cost. A phone costs $X,
and for the most part will cost $X no matter how many lines you roll
out. So a new extension is just $X increase, and your system is just $X
x N extensions to deploy. Also
http://bugs.digium.com/bug_view_page.php?bug_id=0001429
Saying numbers is not always easy, especially if you want one software to be
able to do it in many different languages with different syntaxes for
how to construct numbers like one-hundred-twenty-four or
fem-hundra-tjugo-åtta or
hey people,
i've installed an asterisk version on my Linux Distribution.
Then i wanted to compile the packages for H323 support.
The openh323 and pwlib libraries compiled without any error. But at least i
got one in /~/asterisk-oh323-0.5-10/asterisk-driver/chan_oh323.c in Line
1128. Too many
hi
I want to setup my intel 537 modem to place and
receive pstn call. So far Asterisk registers my
modem.
What do i do next. I couldnt find any help setting up
my modem.
p.s: can any one tell me the full extent of modem
functionality in asterisk ( AND please dont mention
digium cards... i am
Thanks Jeremy,
The problem is ended now.
But... when i use de h.323 show codecs nothing happens... my h323.conf
have the lines:
disallow=all
allow=all ; turns on all installed codecs
;disallow=g723.1 ; Hm... Proprietary, don't use it...
;disallow=all ; Disallow all codecs
;allow=ulaw
Brian D'Arcy wrote:
Hello all,
Im having a nightmare of a time trying to get stable results with SIP
clients on Asterisk. I cant seem to find a configuration that works!
In our office, we run a Sonicwall Pro 200, which is a sip aware,
stateful firewall.
We've discovered that certain
http://bugs.digium.com/bug_view_page.php?bug_id=0001476
If you're calling voicemail from IAX clients and want voicemail or other IVR
prompts to be in some other language than english, this is a patch that you
need to test.
This patch allows you to set the default language for a user/peer so that
Ok thank you!
I am going to recompile new kernel and I'll see
what's going to happend.
Steven's a bit touchy, but he's right. You've got a kernel mismatch
here. Probably you need to recompile the modules under your new kernel.
John
Bartosz Jozwiak wrote:
On Fri, 2004-04-23 at 13:04,
--On Friday, April 23, 2004 5:13 PM -0600 Rich Adamson
[EMAIL PROTECTED] wrote:
My /etc/asterisk/festival.conf looks like this:
My conf looks the same.
My extension looks like this:
exten = 603,1,Answer()
exten = 603,2,Festival('this is a test testing 1 2 3')
Try
I have a Cisco 7970 that I am tying to get working. The
phone powers up and registers but nothing else. I am using the development
build. Any thoughts?
The message I get when the phone powers on and registers.
*CLI skinny debug
Skinny Debugging Enabled
-- Starting Skinny
Oops, I forgot my footnote. Sorry everyone.
On Sat, 24 Apr 2004, Vic Cross wrote:
There are two[1] SCCP channels available for Asterisk:
snip
[1] chan_sccp comes in two flavours, really -- the original by Theo, and a
revised edition by Lambda Solutions that added better support for the
7920.
On Sat, 2004-04-24 at 10:09, Vic Cross wrote:
Oops, I forgot my footnote. Sorry everyone.
On Sat, 24 Apr 2004, Vic Cross wrote:
There are two[1] SCCP channels available for Asterisk:
snip
[1] chan_sccp comes in two flavours, really -- the original by Theo, and a
revised edition by
Note to self: Don't be a dumbass and open up an unsolicited attachment
and execute the content, especially when it was sent off-topic to a
mailing list.
Any of you who did - please report to building 666 for termination.
You've now proven yourself to be simply sucking up oxygen from others.
Set up some .procmailrc filters or whatever filters you like and move
anything with [asterisk- in the subject into a different spool. Then,
you don't get bothered and can read whenever you want just by looking at
the spool.
You're old-school Randy. I'm surprised you haven't done this
This is not a duplex card. It won't work with *
Regards,
Steve
Alejandro Acosta wrote:
Hello,
I just wanted to know if any of you has successfully (or know about)
installed the Dialogic 4xFXs ISA CARD D/41 ESC? Does it work with *?,
if so, how is the driver called?
Thanks a lot for your
Hi Steve-
Out of curiosity, what do you mean by a duplex card in this context?
Cheers
Scott
Scott M. Stingel
President,
Emerging Voice Technology, Inc.
Palo Alto California London England
www.evtmedia.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
I am using the snom 200 with
Phone type
snom200-SIP
Version
snom200-SIP 2.04g
Bootloader URL
http://www.snom.com/download/snom200-boot1.9.bin
Firmware URL
http://www.snom.com/download/share/snom200-2.04o-SIP.bin
I am using asterisk stable tree. I had
Most Canon models seem to be working OK. Serge's Canon drops the line
every time during negotiation, as though it has decided spandsp doesn't
support compatible capabilities. It is not clear why this is happening,
though. The modes which spandsp is asking to use are the most widely
used ones.
Hi Russ,
Thanks for your feedback! I hadn't received any other responses from
anyone, so I was starting to worry that I was one of the few having
these erratic issues.
I might ping Sonicwall, being a good customer and all, maybe I can get
some information out of them. I've always liked using
Check the Redirection on the web interface of your Snom 200. If it says
When Busy that's your problem. It should say Never.
Also make sure on Sip-Lines your line appearance says All or else you
will have the same problem.
Hope this helps :)
Greg
Gregory P. Scasny
Golden Technologies Inc.
On Sat, 2004-04-24 at 12:38, Steve Underwood wrote:
Most Canon models seem to be working OK. Serge's Canon drops the line
every time during negotiation, as though it has decided spandsp doesn't
support compatible capabilities. It is not clear why this is happening,
though. The modes which
Hi Scott,
It cannot play and record at the same time. It can play audio and listen
for DTMF. It can record audio while listening for DTMF. That is about
its limit. Most Dialogic cards are like this. The newest JCT cards
permit full duplex operation, but still have issues with excessive
On Saturday 24 April 2004 18:39, Brian D'Arcy wrote:
Hi Russ,
On a side note, I tried IAX2 last night for the first time using
IAXPHONE. HOLY CRAP I'M IMPRESSED!!! Everything just *works*, period.
I might just use softphones until IAX hardphones are released and say
screw SIP.
I'll second
test message
___
Asterisk-Users mailing list
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http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
http://bugs.digium.com/bug_view_page.php?bug_id=0001019
This patch allows to bind RTP flows to a specific interface, additionally the SDP session descriptor get's coherent with the same address
that is used for RTP traffic, this includes sip-sip and sip-voicemail and others(not tested, but
Greg Scasny wrote:
Check the Redirection on the web interface of your Snom 200. If it says
When Busy that's your problem. It should say Never.
this was already set like this
Also make sure on Sip-Lines your line appearance says All or else you
will have the same problem.
Account 1 was on 1
Hi Eric and Steve :)
My fax numbers (both Canon faxes)
3717201651
3726062501
- Original Message -
From: Eric Wieling [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, April 24, 2004 8:46 PM
Subject: Re: [Asterisk-Users] TxFax/SpanDSP problems
On Sat, 2004-04-24 at
Ian White wrote:
On Apr 22, 2004, at 23:48, Olle E. Johansson wrote:
Geert Nijpels wrote:
Ian White wrote:
On recent releases of the snom200 firmware, the MWI indicator will
turn on, but won't turn off when the message has been checked. It
works on firmware 2.03o, but not in 2.04g or newer.
in sip.conf
[general]
port = 5060 ; The TCP/IP port for SIP communiations
bindaddr = 0.0.0.0 ; Address to bind to. 0.0.0.0 all addresses
on server.
context=other ; Default for incoming calls
disallow=all
allow=ulaw
allow=gsm
in extensions.conf
[general]
I'm sure you know that if there is no matching [sipentry] for an
incoming call it will be allowed thru and will take it's settings from
[general] in sip.conf. Try setting something like context=INVALID in
[general] and then set a context= line for each [sipentry]. That way if
a connection
Hi Frederic,
What does not work though is when the phone is ringing, nobody else can
call the phone anymore.
By this, I presume you mean when the phone hasn't yet been answered?
We are seeing problems where if a snom user is on the phone and another call
comes in, then the person he/she was
Asterisk will accept unauthenticated calls, defaulting to the context
specified in the general section.
Therefore only the call to extension 88 should work.
If both, 77 and 88, are working for you then, yes, something is broken.
- Original Message -
From: Paul Mahler [EMAIL
This looks like something is not working as it should be.
Please open a bug report, add a full SIP debug copied from an asterisk started with
'-dr' including the text between the packets.
Also add your extensions.conf and sip.conf
State your platform (O/S) and the version of Asterisk
Hi,
We had a
problem this evening where asterisk was running but unable to receive any iax or
sip traffic. We restarted * and then it was fine. In hindsight we should have
assessed whether this was a deadlock situation.Has anyone else seen this
problem?We're using the -head cvs of April
On Wednesday 21 April 2004 21:48, Nicolas Bougues wrote:
This last hop may be the source of your problem. Since I believe it's
not a trans-continent link, it's either :
- a very congestioned link
- a router with serious problems at hop 13 (or maybe 12).
You should contact whoever manages
Hi everyone,
After running a few succesful tests with SIP and
video codecs(windows messenger clients) I was wondering
if anyone had managed to use video with other
channels, or even cross channel video calls.
I successfully installed asterisk-oh323, as it
seems h323 and sip both use h261
Any ideas on why when I call an extension from an outside
line, the ringing is very choppy?
[moved to asterisk-users, as this is not a development question]
At 1:40 PM -0400 on 4/24/04, Sudhir Kumar wrote:
I would like to hear from any of you who has done any kind of
benchmarking on a robust hardware that can handle large call volume,
preferably with G.729 codec involved.
We are in the
Does anyone have a T400P running on an Athlon XP with four T1s? Would
96 channels require dual processors?
Thanks,
John Todd wrote:
[moved to asterisk-users, as this is not a development question]
At 1:40 PM -0400 on 4/24/04, Sudhir Kumar wrote:
I would like to hear from any of you who has
Has anybody successfully configured asterisk to use Galaxy Voice for inbound
/ outbound calls and what is required to make this work?
Dave
---
Dave Tipton
[EMAIL PROTECTED]
817-858-9841 Home
469-223-8506 Cell
___
I'm new to Asterisk -- don't have it set-up yet, just comparing options.
Can I do this:
Wildcard T100P going into an Adtran Total Access 750 or 850, purchased
from ebay -- most of those come with a few FXS cards; swap one quad-FXS
with a quad-FXO card for four CO lines and up to 20 analog
Jay,
I have had a lot of trouble with the FXO ports on Adtran TA750. Unless
the incoming POTS lines have a balance impedance, they will buzz very
bad.
Gregory P. Scasny
Golden Technologies Inc.
http://www.golden-tech.com
219-462-7200
-Original Message-
From: [EMAIL PROTECTED]
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