it does...
Try power cycling the phone. The Grandstreams seem to get flakier the
longer they are up. Normally I notice it when they the phones stop
allowing incoming www connections. A power cycle always cures it.
-wolfgang
--
Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang
--
Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/
Direct SIP URL Dialing: http://www.wsrcc.com/wolfgang/phonedirectory.html
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was good for, but
then again I am pretty sure they allowed the volume to be increased
due to customer complaints of the phones being too quiet. :-)
I wonder if these same phones with a decent in-the-ear earphone and a
mini boom-microphone would have the same problems.
-wolfgang
--
Wolfgang S
-head from Nov-1 and sip
phones that all can reinvite.)
I'm surprised that folks using asterisk for gatewaying into the pstn
haven't raised a stink. Having a trunk stuck on a for-pay call can
easily cost folks lots of money.
-wolfgang
--
Wolfgang S. Rupprechthttp://www.wsrcc.com
payphone scams
by configuring their asterisk to only answer the 800 calls made from
normal residential phones.
-wolfgang
--
Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/
Direct SIP URL Dialing: http://www.wsrcc.com/wolfgang/phonedirectory.html
bills for mysterious
one-minute phone calls from Denver, Colorado. If you find such a call,
be sure to contact your phone company. For more information on this
scam, click herei. (Thanks to ConsumerWorld.org for this tip.)
WIRELESS WATCH
-wolfgang
--
Wolfgang S. Rupprecht
the echo. I always wanted to fill the Sipura handset with
modeling clay and see if that helped things any.
(The echo was only a problem on direct sip-to-sip calls. Any calls
going into the PSTN seemed to always be processed by an echo-can, so
it wasn't noticed there.)
-wolfgang
--
Wolfgang S
desire to have it on *37.
Which one is more common these days? Can I just make an extension
and assign the pickup code to it?
exten = *37,1,pickup(SOMETHING_TBD);
-wolfgang
--
Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/
Direct SIP URL Dialing: http://www.wsrcc.com
until after the spa841
reboots.)
-wolfgang
--
Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/
Direct SIP URL Dialing: http://www.wsrcc.com/wolfgang/phonedirectory.html
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day and 05:54:03
(crashed during a call)
People have been reporting a finicky ethernet connector, so maybe that
is the reason the phone does not answer to any traffic?
Yea, this phone has that problem too. ;-) Some cables just don't
work.
-wolfgang
--
Wolfgang S. Rupprecht
in the IVR with caller-id's like
CAR PROMO or VOIP CALL.
If spam/spit is ever a problem, I'm simply routing previously unseen
calls to a turing test of the same type and anyone that has previously
called (and/or been called) gets to bypass the turing test.
-wolfgang
--
Wolfgang S. Rupprecht
Wolfgang S. Rupprecht writes:
If there is enough interest, maybe the greater asterisk community
could adopt some semi-official mapping tables. I'd be willing to
periodically generate a flat mapping file and an extension.conf
dialplan snippet from sipbroker's list or whatever else is deemed
/voipeer.html
It looks interesting, but these things always seem to be scuttled or
reduced to glacial progress by the telecom interests.
VOIP peering isn't something that should require years of meeting to
make happen.
-wolfgang
--
Wolfgang S. Rupprechthttp://www.wsrcc.com
really be _s-. and
not s-..
-wolfgang
--
Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/
Direct SIP URL Dialing: http://www.wsrcc.com/wolfgang/phonedirectory.html
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secret from the asterisk crowd, or is
everyone else using asterisk for phone spamming from call centers and
the last thing they want is folks to be able to call them back and
give them an earful over disturbing their dinner?
-wolfgang
--
Wolfgang S. Rupprechthttp://www.wsrcc.com
a flat mapping file and an extension.conf
dialplan snippet from sipbroker's list or whatever else is deemed more
neutral or useful if there was any interest in such.
-wolfgang
--
Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang
doesn't have any wideband codec. Even the low-end Grandstreams can do
16kz audio (which sounds really nice when you talk between two of them
with reinvite).
-wolfgang
--
Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/
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[EMAIL PROTECTED] (Bruce Komito) writes:
If you're going to promote your product, you might consider making sure
your web site is up, before giving out the URL.
And he could also lose that flash animation when promoting to an
opens-source/linux audience.
The fordvoice web site has a big blank
[EMAIL PROTECTED] (Matthew Boehm) writes:
Why even have the ability to set callerid name/number if end offices don't
honor it?
VOIP is bigger than just PSTN-gatewayed calls via some specific
company. The end goal is to connect the VOIP islands directly. That
is already happening at some
[EMAIL PROTECTED] (Chuck Ramirez) writes:
Looking at the source code I noticed that rand() is
used four times to get a callid. Is that safe enough?
RAND(3) OpenBSD Programmer's Manual RAND(3)
NAME
rand, srand - bad random number generator
...
Is there
[EMAIL PROTECTED] (GliTcH) writes:
I'm trying to investigate going to a different manufacturer, but I
don't like the Cisco ATA-186's very much and they're too pricey, so I
don't know where to go next. voipsupply has a pretty big collection,
maybe I'll order 1 of each for testing.
[EMAIL PROTECTED] (dean collins) writes:
You will be very disappointed at the call quality if you try and run
other software on an asterisk box, pc interrupts and processing glitches
just don't 'play well' with voice.
This isn't that much of a problem if you structure your phone system
to be
[EMAIL PROTECTED] (Iassen Hristov) writes:
Anyone having problems with registration to * from a SPA-841?
I have a spa-841 (firmware 3.1.1a) on my desk right next to me and it
registers just fine w. ~current asterisk (CVS-HEAD-03/16/05-08:43:40).
The one problem I do notice that the phone is
[EMAIL PROTECTED] (Steve Prior) writes:
The recorded prompts by Allison are more in line with the very
language structured text forecasts typically seen by pilots
There are home weather stations with computer interfaces that simply
tell you the current stats (temp, pressure, humidity, wind
It looks like asterisk isn't honoring EOF on stdin.
file add-phonelist:
database put cidname 200551234 name 1
database put cidname 200551235 name 2
database put cidname 200551236 name 3
database put cidname 200551237 name 4
asterisk -rn add-phonelist
What I see is an infinite stream of
a way to save a dialplan or to diff
it against factory defaults. It makes it kind of hard to post a
definitive recipe.
-wolfgang
--
Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/
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[EMAIL PROTECTED] (Brian Dingman) writes:
I doubt that was the problem. I would be interested in hearing what
else you did besides that to get it working.
If he had a bad entry in the /etc/hosts file that could have been a
problem.
The hosts files does require periodic maintenance.
is that the contact wires have a slightly higher than normal
angle and end up hitting the plastic lip of the rj-45 plug instead of
resting on the gold spade contacts.
-wolfgang
--
Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/
___
Asterisk-Users
[EMAIL PROTECTED] (John Goerzen) writes:
I've heard good things about VoipJet here, so I was going to set up an
account. Then I noticed their Terms of Service here:
https://www.voipjet.com/tos.php
Ignore them. There are plenty of players and you don't need to deal
with one that has NDA
[EMAIL PROTECTED] (Dan Weber) writes:
On Sat, 5 Mar 2005, Wolfgang S. Rupprecht wrote:
Does broadvoice participate in e164.{arpa,org,info}?
Yes
Does this change mean that non-customers can't call broadvoice
customers with a pure SIP call by routing the call to
sip.broadvoice.com?
Calls
in the
dial-plan is mostly determined by their ping times.)
Too bad not many providers do a database dip into e164.org. If they
did, I'd still be able to get some calls when my DID provider was
down.
-wolfgang
--
Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang
something, but better/cheaper/faster to cut out the telco middleman.)
-wolfgang
--
Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/
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their single
DID listed, it strikes me that the real utility would be to have a
blocks of 100 or 1000 numbers assigned to folks, so they could have
each of their voip phones directly dialable from anyone that queries
your db.
-wolfgang
--
Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang
)
* reject any mail that has re: or the same subject line as
other msgs, but no references. (This one needs to be done
very carefully.)
-wolfgang
--
Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/
Microsoft xbox power cords: Finding innovative new
to tell
the user app that a socket is associate with an ip/esp encapsulation?
(And yes, I know that one of the common ipsec implementations strips
the ipsec headers and then sends the unwrapped packet back through the
IP machinery a second time.)
-wolfgang
--
Wolfgang S. Rupprechthttp
--
Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/
Hate software patents? Sign here: http://thankpoland.info/
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haven't updated your firmware to
the current version, you might want to do that first. It might save
you quite some time.
http://gs-firmware.gratissip.dk/firmwares/latest/
-wolfgang
--
Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/
Hate software patents? Sign here: http
is dialing a
7-digit or 10-digit outside number that starts with 6 and they
dither a bit after dialing 4 digits, it will end up calling the
corresponding inside number. That hasn't been a problem in practice.
-wolfgang
--
Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/
Hate
server again and changed the
GS's config, it loaded and burned just fine.
-wolfgang
--
Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/
Hate software patents? Sign here: http://thankpoland.info/
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calls are interactive and sorting the robo-caller from
the real people shouldn't be hard. The only thing bugging me is, is
there a law that would prevent a telemarketer from lying and pressing
the key for I am not a telemarketer.
-wolfgang
--
Wolfgang S. Rupprechthttp://www.wsrcc.com
be an asterisk
driver that took 8 relay closures via the parallel port and could
initiate a different pre-canned phone message for each of them.
Eg. Water has been detected in the computer room, There is smoke in
the computer room etc.
-wolfgang
--
Wolfgang S. Rupprechthttp
(propolice)
GNU ld version 2.14 20030612
In case you are curious, these are my openbsd/amd64 patches:
http://www.wsrcc.com/wolfgang/ftp/asterisk-openbsd35.patch
-wolfgang
--
Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang
things without tipping off the same two endpoints? What
about VPN tunnels? Are they illegal now by the same logic?
-wolfgang
--
Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/
openbsd amd64 http://www.wsrcc.com/wolfgang/ftp/asterisk-openbsd35.patch
a cost-saving measure. The hard jobs will still
have to be done by sending people out into the field.
-wolfgang
--
Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/
openbsd amd64 http://www.wsrcc.com/wolfgang/ftp/asterisk-openbsd35.patch
number like 5% or 10%).
-wolfgang
--
Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/
openbsd amd64 http://www.wsrcc.com/wolfgang/ftp/asterisk-openbsd35.patch
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and number
on a per-call basis?
(Eg. Can family members have the sip name aka caller id, indicate
the real calling party's name?)
-wolfgang
--
Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/
openbsd amd64 http://www.wsrcc.com/wolfgang/ftp/asterisk-openbsd35.patch
. (If
anyone knows how to handle that, sure would appreciate a suggestion.)
Given the choice between hearing dead air and hearing the tones, I
think I'd rather hear the tones. At least I know something is
happening.
-wolfgang
--
Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang
.
(I'll try to hack the registration issue here and post some GPL-ed
patches if I get it working.)
-wolfgang
--
Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/
openbsd amd64 http://www.wsrcc.com/wolfgang/ftp/asterisk-openbsd35.patch
[EMAIL PROTECTED] (James Jones) writes:
you can not ping that address because ICMP is turned off.
Do you mean *all* ICMP is turned off or just icmp-echo-request /
icmp-echo-reply?
-wolfgang
--
Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/
openbsd amd64 http
,
and then actively uses whichever one was up and had the lowest
round-trip-delay.
-wolfgang
--
Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/
openbsd amd64 http://www.wsrcc.com/wolfgang/ftp/asterisk-openbsd35.patch
___
Asterisk-Users
(Dameon D. Welch-Abernathy) writes:
Wolfgang S. Rupprecht wrote:
Have you gotten asterisk to work for dial-out to the PSTN when using a
md5 authentication?
What I discovered via tcpdump was that the Asterisk box wasn't
responding to the authentication request for whatever reason. I
[EMAIL PROTECTED] (William Suffill) writes:
Seems quite interesting. Any suggestions of where to order one and
about how much?
Mine was $125 from www.voxilla.com. I ordered it on Sunday and had it
in my hands on Tuesday.
-wolfgang
--
Wolfgang S. Rupprechthttp://www.wsrcc.com
:5060
Jul 14 16:42:11 hsephone [1:5061]-64.142.50.224:5060
Etherdump also showed quite a few invalid syslog lines coming from the
sipura. Mostly they were missing the local0.debug. Some went to
local2.
-wolfgang
--
Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/
openbsd
[EMAIL PROTECTED] (Dameon D. Welch-Abernathy) writes:
On Tue, 2004-07-13 at 19:10, Wolfgang S. Rupprecht wrote:
I'd settle for just a way of restoring it from a file. I just got my
Sipura-3000, and it would be nice to keep the master config on disk
and under CVS.
If you generate your
in filling in one of the dozen or so
passwords.
Sorry, no detailed HOW-TO's yet. This thing can obviously be made to
do what I want of it, but it will be a while figuring it all out.
This thing really needs a wiki devoted to it. ;-)
-wolfgang
--
Wolfgang S. Rupprechthttp://www.wsrcc.com
less than between
two BT-100's. I'm going to have to bump the gain up a bit in at least
the PSTN-VOIP direction. Perhaps I need to do the VOIP-PSTN
direction too. This is going to make echo even worse.
-wolfgang
--
Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/
openbsd amd64
. I just got my
Sipura-3000, and it would be nice to keep the master config on disk
and under CVS.
-wolfgang
PS. I'm starting to feel nostalgic for the Moringstar Router and its text
config files that one could ftp to/from the unit to update its
config.
--
Wolfgang S. Rupprechthttp
to be an
asterisk scheduler problem under OpenBSD that happens even with no
chroot etc.)
-wolfgang
--
Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/
openbsd amd64 http://www.wsrcc.com/wolfgang/ftp/asterisk-openbsd35.patch
___
Asterisk-Users
asterisk
actually has (this is from -current):
;
; codecs: a_mu adpcm alaw g726 gsm ilbc lpc10 ulaw
;
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=adpcm
allow=g726
allow=ilbc
;; allow=lpc10 (robotman)
-wolfgang
--
Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/
openbsd
on the
internet and attempts to ARP for something that is halfway around the
world (eg. sends an arp-request for the sip server even if that
machine isn't local).
I like GS's sound quality and price, but their firmware clearly has
some serious corruption problems.
-wolfgang
--
Wolfgang S
authentication questions to the ATA and forwarded
the ATA's answers back to the remote SIP server? Could that be made
to work?
-wolfgang
--
Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/
openbsd amd64 http://www.wsrcc.com/wolfgang/ftp/asterisk-openbsd35.patch
? (Assuming I have my _sip._udp SRV crap set right.) Do
any commercial SIP providers lookup SRV?
-wolfgang
--
Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/
openbsd amd64 http://www.wsrcc.com/wolfgang/ftp/asterisk-openbsd35.patch
And when can we expect a patch from you for this? :P
I'd like to see this too and be willing to do this under GPL. Is that
good enough?
-wolfgang
--
Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/
openbsd asterisk http://www.wsrcc.com/wolfgang/ftp/asterisk-openbsd35.patch
)
-wolfgang
--
Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/
openbsd asterisk http://www.wsrcc.com/wolfgang/ftp/asterisk-openbsd35.patch
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