you consider
that this device is really plug and play, it remotes configure
everything.
Hopefully a bridge mode will appear in a later firmware upgrade
(which, for Sipuras, are frequent and readily available on their
website).
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to
overcome this. In the meantime, get a Sipura 2100, supports 2 729 calls and
has both WAN/LAN ports. Personally, I dislike the lack of LEDs on the 2100.
My 2100s have 3 LEDs, plus 2 for each RJ-45 port. Instead of just 2
for the SPA-2000.
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, no dialplan support, no auto-upgrade (well, they
recently added some kind of support). Voice quality is OK.
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the SPA-2100 (still) lacks is
a bridge mode, where the LAN and WAN ports would act just like a
switch, so that you can easily chain devices without routing/NAT. Just
like most IP phones do.
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will
only generate a dial tone if registrered. BTW, you can easily check on
the Sipura web interface that the dial tones are parametered there.
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be more precise.
G.722 might be interesting : 64 kbps, 7 kHz. It's not free.
Otherwise, MP3 or OGG might be ok ?
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of device talking Vorbis to Asterisk. Does it
exist ?
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on the
other side (with proper hardware).
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that kind of devices, so there is no echo generated. So, basically, no
echo cancellation required. Unfortunatly, it's impossible to know from
the caller point of view whether the call will need echo cancellation
or not.
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, and your PBX users might not like it.
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will
notice it if you look at your D-channel dumps.
- use high quality NICs and switches.
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--- ztd-eth.c.old 2004-02-01 06:53:58.0 +0100
+++ ztd-eth.c 2004-07-11 00:51:45.0 +0200
@@ -251,7 +251,7 @@
{
struct ztdeth *z
. Unfortunatly, as far as I understand,
Asterisk is not really designed to handle more than one caller id
number.
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is not an issue? Does Grandstream have
improved echo cancellation scheduled for a future firmware upgrade?
The Sipuras are definetly better than the HandyTones. I've heard that
the forthcoming GS firmwares will enhance echo cancellation
performance, though.
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discussion!
Err, all (1.0.4.x at least) GS firmwares support TFTP autoconfiguration !
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It was solved by using a PCI 2.2 compliant motherboard (i865
based). It's quite an odd behaviour, and it's still not clear to me
why it happens. I initialiy thought it could be solved by fixing the
FPGA VHDL, but I'm not an expert in that field.
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(particularly if the IRQ line is shared). However, it seems to be
detected by the driver (and it should print a warning).
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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 westloc.com
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itself tens of SIP phones connected to it. And it
would alert the SIP phones when it receives MWI over the IAX channel
from the central server.
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and chassis are considerably more expensive than their
standard counterparts, if they exist at all.
You can't just hot (un)plug a standard PCI board : the bus is not
meant for this, you have most chances of destroying your board and/or
your bus.
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[217.146.224.41:4569]/3 in the events messages. So I have no way
to know it's really mine.
Event the final Response message doesn't state the UniqueId of the
call.
Maybe I missed something obvious.
Any idea ?
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.
ActionID is specified in the Response message. It would be useful with
interleaved Requests/Responses, which is not the case here.
However, the ActionID is not specified in the Event messages I get
meanwhile, so I can't track call progress nor channel allocation.
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caller id is
supposed to be right, at least for its numerical part.
I'll try to use the name part of the caller id for that purpose.
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always cristal clear.
So your problem is probably related to network congestion
somewhere on your side, on the remote side or in transit. traceroute
is your friend.
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).
By the way, anyone can tell me the commercial license price of these GPL
software: GNU Bayonne, GNUCOMM and Asterisk.
I believe no commercial licence exist for GNU whatever. For
Asterisk, you should contact Digium directly (but then again, you
should use zaptel hardware).
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if it comes
back a few ms later). Thus Asterisk uses echo cancellation, which
may alter data transmission. Hopefully tone detection deactivates
it though.
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a fax extension, but a wildcard one (_.). I would like
these detections to be simply ignored. Is there any way to do it ?
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unkillable.
Restarting Asterisk (after kill -9) solves the problem.
It seems to me that the Q921 layer in libpri has an unrecoverable
error (such as the fd being wrong/closed).
Anybody know where it could come from, and/or what should be done to
avoid it ?
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On Thu, Mar 11, 2004 at 05:12:24PM +0100, Klaus-Peter Junghanns wrote:
Nicolas,
does your TE405P share the irq?
No, it's alone on IRQ 17 (with IO-APIC).
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this setting is used. Do I really have to set one ?
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=47
bchan=63-77,79-93
dchan=78
bchan=94-108,110-124
dchan=109
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handle that load reliably ? What's the
application ?
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to be 2 Mbps (2 inbound, 2 outbound, really), and
Fast Ethernet to be 100 Mbps (again, 100+100 in full duplex), I shall
consider U-law to be 83 kbits/s, not 166.
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.
No problem here (5 fax machines). Solved the ringing detection problem
I had with the GS 286.
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+
range, isn't it ?).
The adapter works with evey kind of phone I tried, but did not work
with two different fax machines. Am I simply out of luck with these
fax ? Does my ATA look defective (tried two of them, however) ?
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(via an ISDN PRI), if you hangup the SIP phone the
message doesn't reach Asterisk, which keeps on ringing the ISDN leg.
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:) can this
unit connect directly to a cable modem?
I think so. However, what is not clear at all is whether you can
connect anything else to the DSL line.
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work well behind
most NATs (even cascaded NATs).
Most cheap ethernet/ethernet modems (sold as Cable/DSL routers) can do
the trick.
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On Mon, Feb 09, 2004 at 05:37:48PM -, David J Carter wrote:
Have a look at http://www.plugndial.com/aps_sample.html
I've been told by sipphone that this format is new. It's not
supported by anything on the market right now.
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works.
I suppose the phone can sit itself right behind an Ethernet DSL modem
with this setup, but I'm not sure whether/how it enables a standard PC
to work on the other ethernet port.
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for algorithms even if you do your own original implementation.
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On Tue, Jan 20, 2004 at 08:57:32AM -0600, Eric Wieling wrote:
On Tue, 2004-01-20 at 01:12, Nicolas Bougues wrote:
Yes, but with a Pentium you don't have to pay a license to use MMX in
your software, since the MMX instructions are part of the product you
are allowed to use them
) ?
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ISA 10 Mbit Ethernet controller
- a phone interface
- and some glue around that
These are quite cheap components (the most expensive part is the $6
DSP).
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cfg.txt file ?
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is
to buy a BRI card and then a PRI card.
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On Mon, Jan 19, 2004 at 10:51:02AM +0100, Nicolas Bougues wrote:
Attached is the config file I send to my Grandstream.
Change IP address Phone ID to suite.
That's great. Is it documented somewhere ?
And how do you manage tens or hundreds of phones ? Are they all in the
same
On Mon, Jan 19, 2004 at 08:44:36AM -0600, Eric Wieling wrote:
On Mon, 2004-01-19 at 02:34, Nicolas Bougues wrote:
These are quite cheap components (the most expensive part is the $6
DSP).
What *I* want to know is why someone has not made a CHEAP PCI card with
4, 8, or 16 of these DSPs
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be
expected).
There might then be possible to use some IPV6 specific stuff, but it's
quite far away.
Note that some time in the future I could be interested in it. Our
network is almost fully IPV6 enabled, and we have full, native IPV6
peering.
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through some gateway. And I agree,
this would be quite a mess from the voip protocol stand point.
But hopefully, either :
- v6 hosts are fully v4 enabled as well
- v6 only hosts will only talk to v6 peers
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minutes per day in order to have a fairly decent quality.
About 200 GB a month (total of both directions) would qualify, IMHO, as
optimised bandwidth usage in that kind of context.
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link automagically ?
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, and have
no more time to take care of it.
Now that the wbxml library supports SyncML, it should be quite easy to
make a really nice package out of it. I believe that Chuck Hagenbuch
(Horde) is working on it.
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that make five 9s reasonable at
all ?
Do you really need five 9s ? There is no such thing I'm aware of in
enterprise grade telephony. You have to go to carrier grade
equipment, which asterisk, and PCs in general, are definetly not aimed
at.
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comply to the
GPL terms. Nobody, even Digium, can revoke the GPL licence for GPL-ed
software.
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amongst servers).
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of
extensions.conf on each server.
I'm not quite familiar with IAX(2) and registration questions. Does
anybody sees any tricky problem that could arise with some kind of
auto-cascading registrations through IAX (that is, any registration,
SIP, IAX... would be forwarded on an IAX channel) ?
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to trunking ?
In fact, what would be very nice to have would be some kind of
trunking with registry/dialplans automatic exchange, so that one can
easily setup a larger virtual PBX, that would server both capacity and
redundancy requirements.
I keep on thinking :)
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. But when the user
dials again, it should work. It should be fairly OK with the client
performing a new DNS lookup before it registers, if the DNS is aware
of which box is up.
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= my_local_e164_extension
which works fine, except that now I'm at the 2 level in the context,
and I had to modify my_local_e164 extension context accordingly.
Does somebody know of a better way to do it ?
Thanks.
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? If so, I haven't found out.
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=no in the user definition
Or of course, if Asterisk thinks that it needs to process the stream :
for instance, if you want Asterisk to be able to transfer your call
(t/T options for Dial).
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user
Same thing for IAX peers.
Of course, setting up various IAX links between each server is no
problem (with registration cascading, for instance).
Is such a setup common, if at all possible ?
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On Fri, Dec 05, 2003 at 04:40:42PM +, Michael T Farnworth wrote:
On Fri, 5 Dec 2003, Nicolas Bougues wrote:
On a slightly different topic : does somebody know of a NAT-friendly
(as Grandstream means it) tftpd server ? It seems theirs replies from
port 69, which is the only thing
not sure whether this is RFC compliant.
So yes, I have the 1.0.4.17 FW files, but I have no way to have the
phone download them !
I'll try to hack a NAT friendly tftp server on monday.
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, the telco would rather run the DS3 to your building, then
split it up in a bunch of E1 on their own CPE.
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? It seems theirs replies from
port 69, which is the only thing their phones will accept.
[ If anybody wants it, I can send the 1.0.4.17 firmware by email ].
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transmission (although
there is support in libpri).
It should be possible, however, to modify chan_zap and add a SendText
application (and/or modify the Dial app) to handle this.
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Dear all,
I'd like to know if the core (demo, voicemail...) asterisk prompts
have ever been recorded in french (and are freely available).
If not, I'm willing to have them studio-recorded by a professional
speaker, and contributed back to the community. Does a message list
exist ?
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for a meeting in early january. If needed I can
arrange just about anything, a drink, a dinner, and/or hosting a
meeting at my company.
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. It involves quite complex signal processing. Look at the various
tries in the zaptel source.
An no, POTS phones don't have echo cancellers.
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