You must use GSM with iaxtel and Voicepulse for now... I talked to the guy
from voicepulse and they said ilbc might be turned sometime in the future.
But not sure.
bkw
On Tue, 14 Oct 2003, Stig Hess wrote:
I'm having trouble configuring these services the way I want. Basically I
prefer using
Are you using the recommended pwlib and openh323 tarballs?
bkw
On Mon, 13 Oct 2003, CW_ASN wrote:
Hi all:
I've got some core dumps when I use chan_h323. I dial an extension using
h323, routed thru an E100P (like a H323-ISDN_PRI gateway). Sometimes *
hangs, sometimes not. The client used
show channels?
On Thu, 9 Oct 2003, duncan wrote:
So whats the best way to find the maximum number of concurrent calls in
this setup:
IAX2 Trunk using GSM over a 512k internet line.
thanks
duncan
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Record(/tmp/testing:gsm)
Thats what I use.. and it works.
bkw
On Thu, 9 Oct 2003, Lists wrote:
If I do something like
exten = 1,1,Record(/someplace/somefile|gsm)
It does not record I end up getting
-- Executing Record(SIP/mlh-04d0, |gsm) in new stack
exten = 1,1,Record(filename|gsm)
If I dial # while in a call nothing happens. I was transfering using
the 7960 transfer function which gives me a dial tone and then I dial
700 which gives me a busy tone I also tried to dial #700 but as soon as
you push # on a 7960 it dials since # its used to signal the end of the
dial
I've implemented a bit of a workaround.
I've setup the dial plan 2 in my system as the call park prefix.
When you want to park a call, you blind transfer to 2 where is your
extension (eg: 27011). The call is parked, and you will immediately receive
a call announcing the park
Im having a similar problem with my 7960 when I receive two incoming
calls I cannot join them.
ya you can't join them. That sucks.. but you can park one call, go back
to call number 1. Press conf. Dial the parking orbit.. then press join!
bkw
exten = h,1, will not work if you park a call then pick it back up. You
are flipping the call direction from what Mark told me. Whats wrong with
CDR data? is that not good enough to tell call lenght?
bkw
On Tue, 7 Oct 2003, mattf wrote:
The way I worked around this is to log the uniqueid in
I dont see it as a bug.. I see why it don't work.. and why people think it
should. Enable # transfers.. and setup call parking to get around this.
Also if you conf very much look at app_meetme.
bkw
On Tue, 7 Oct 2003, Juan J. Sierralta P. wrote:
On Tue, 2003-10-07 at 12:09, Brian West wrote
Not yet.. but I sure wish we could... :)
On Tue, 7 Oct 2003, Juan J. Sierralta P. wrote:
Hi,
It is posible to put a call in the parking lot with a SIP phone as a
Cisco 7960 ?
Anyway, how can I put a call park on a FXS line ? Is there any magic
digits ?
--
Juanjo sin .sig
Yes but you can't do native sip tranfers to parking. Thats what I want.
And thats what I was talking about. You can't say use a Cisco 7960 and
hit transfer then dial 700 then transfer. WONT WORK.
bkw
On Tue, 7 Oct 2003, Andrew Joakimsen wrote:
You need to enable transfer:
Dial
Dialing
Works fine on my 7960 with 5.3 firmware.
bkw
On Mon, 6 Oct 2003, Babak Pasdar wrote:
Hello,
I am trying to conference two or more calls on a Cisco 7940 phone. When I have one
inbound call and one outbound (I initiate the second call by pressing conference) I
get the join button at the
use
mailbox=500
instead of [EMAIL PROTECTED]
[EMAIL PROTECTED]
since he doesn't have his stuff in the default context
bkw
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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
Sep 2003, WipeOut wrote:
Brian West wrote:
You can't anymore MySQL was ripped from Asterisk because the client libs
are GPL.
bkw
On Tue, 30 Sep 2003, Manoj K Gupta wrote:
I still don't get this.. Asterisk is GPL (with an option of a commercial
licence), MySQL is GPL(with an option
Have a way to specify it in the src? I would like to try the 8k between a
few servers and see how it sounds.
bkw
On Tue, 30 Sep 2003, James Golovich wrote:
On Tue, 30 Sep 2003, WipeOut wrote:
Whats the default SPEEX bitrate set to in Asterisk?
The default bitrate for speex (at this
Any nat involved? and what codec's are you trying?
On Tue, 30 Sep 2003, Kevin wrote:
When I dial with my Grandstream 101 telephone to another sip phone or
Zap FXS, the call rings, but no audio is passed. Eventually the call
gets disconnected. The same thing happens if I dial the
I would also like some more info on this whole mysql being taken out of
the core asterisk install. I understand its because of the dual lic. that
digium has.. gpl and comercial... why can't mysql be non-existant in the
comercial version. Then mysql would be compatible with asterisk?!? Or am
I
I recall someone saying that hdparm is embeded in the codec and
Registration binaries.. and that is a violation of the GPL. But thats
voiceage's doing. Anyone care to shed some light on this?
bkw
On Mon, 29 Sep 2003, Eric Wieling wrote:
On Mon, 2003-09-29 at 09:40, Mark Spencer wrote:
I ment BSD lic. mybad. :P
On Mon, 29 Sep 2003, Brian West wrote:
I recall someone saying that hdparm is embeded in the codec and
Registration binaries.. and that is a violation of the GPL. But thats
voiceage's doing. Anyone care to shed some light on this?
bkw
On Mon, 29 Sep 2003
Personally, I *love* MySQL, and I'm a bit surprised by their sudden change
from public domain (and maybe LGPL) to GPL for their client libraries...
Who can we bug at mysql to see if we can get that changed?
bkw
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ie a perl script. :)
On Mon, 29 Sep 2003, Mark Evans wrote:
As far as I am aware the CDR logging can only log
to MySQL or CSV text file..
It doesn't matter, we could write an automatic import routine which can
import that CVS into anything :)
Regards
Mark
http://www.loligo.com/asterisk/current/
I'm sure he has a few in his sip.conf examples
On Mon, 29 Sep 2003, Leif Madsen wrote:
Hi all,
I would like people to email me at 'leif at hacklocalhost dot com' some
example configuration files for VoIP providers which * can register
with. I am
You can't anymore MySQL was ripped from Asterisk because the client libs
are GPL.
bkw
On Tue, 30 Sep 2003, Manoj K Gupta wrote:
Hi list,
I am trying a scenerio where the * will take the email and mailbox number from the
Mysql database for sendming mail to a voicemail user. I have seen
Some ISO firmwares have bugs.. we ran into this and had to downgrade to
get it to work correctly.
bkw
On Fri, 26 Sep 2003, Bartosz Jozwiak wrote:
I have fixed i already.
And still it does not want to work :(
- Original Message -
From: Sean Figgins [EMAIL PROTECTED]
To: [EMAIL
Just a heads up.. you can't loop switch statements
ie
BOX A switch = BOX B
BOX B switch = BOX A
show dialplan will show the switch but not the dialplan of the remote
switch.
bkw
On Fri, 26 Sep 2003, Lee Goodman wrote:
Hello
I want to setup a IAX trunk between 2 asterisk servers. I also
What must be done for everyone software to be able to play on the same
playground without the parents watching?
bkw
On Fri, 26 Sep 2003, Mark Spencer wrote:
Just FYI, MySQL stuff has been pulled from Asterisk since apparently now
the client libraries are under GPL and not LGPL (and thus are
and to route the call over the IAX
trunk to BOXB
Lee
- Original Message -
From: Brian West [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, September 26, 2003 10:56 PM
Subject: Re: [Asterisk-Users] Configs for IAX IAX trunk
Just a heads up.. you can't loop switch statements
ie
I used it with my 7960 also. all you do is set the Preferred Codec to
g729 and put the below in your [general] section.. and its hould just
work.
disallow=all
#allow=g723.1
allow=g729
allow=ilbc
allow=gsm
allow=ulaw
On Fri, 26 Sep 2003 [EMAIL PROTECTED] wrote:
I've used it with the Cisco
This is simple to do..
voice-port 1/0/0
connection plar
!
voice-port 1/0/1
connection plar
!
dial-peer voice 1000 voip
max-conn 4
destination-pattern
req-qos guaranteed-delay
codec g711ulaw
ip precedence 5
no vad
session target ipv4:x.x.x.x
!
in h323.conf set the
to di with a 2600 that has a T1 adapter on a high-density high-density
voice port adapter?
BTW... Because I am lazy, what does plar do?
-Sean
On Wed, 24 Sep 2003, Brian West wrote:
This is simple to do..
voice-port 1/0/0
connection plar
!
voice-port 1/0/1
on a high-density high-density
voice port adapter?
BTW... Because I am lazy, what does plar do?
-Sean
On Wed, 24 Sep 2003, Brian West wrote:
This is simple to do..
voice-port 1/0/0
connection plar
!
voice-port 1/0/1
connection plar
!
dial-peer voice 1000 voip
http://voipstore.atacomm.com/shops/Browse.aspx/27934028032-27934130944.htm
bkw
On Wed, 24 Sep 2003, Mike Hjorleifsson wrote:
Does anyone sell a preinstalled asterisk server ?
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http://www.chagres.net/products/voip/phones.html
bkw
On Thu, 25 Sep 2003, Aaron Martin wrote:
Does anyone know of any reliable supplier for Grandstream phones?
I tried dealing with David Li from Grandstream, but after emailing him an order in
August, and asking how he wanted payment, I
Dream on!... as of now they dont.
On Thu, 25 Sep 2003, Gary wrote:
I would probably be interested except when will their products actually
support GSM codecs ??
On Wed, 24 Sep 2003 21:34:36 -0500 (CDT), Dave Weis wrote:
On Thu, 25 Sep 2003, Aaron Martin wrote:
Does anyone know of any
I have tried for over a month off and on to get iaxtel for inbound to
work... and tonight after alot of troubleshooting we noticed this:
iaxtel inbound will use the last entry in your iax.conf to auth against.
So if [iaxtel] is at the top and say [voicepulse] at the bottom. An
inbound call will
their really isn't much fixed between 4.4 and the 5.x stuff but at the
time thats all I had. So I put that on the phone. So far everything
works like a champ. Not one problem.
4.4
http://www.cisco.com/en/US/products/sw/voicesw/ps2156/prod_release_note09186a008016096f.html#63943
Resolved
Its fairly simple.. meetme isn't that big you can find where the hooks are
its commented in the code.
bkw
On Mon, 22 Sep 2003, Chee Foong wrote:
Hello,
Is there a asterisk developer guide/source code doc or something like that?
I want to see if I can implement the admin menu function for
[EMAIL PROTECTED] needs to fix their spam filter. Please stop using it or
learn to configure it.
+ 1 Sep 22 AntiSpam UOL (6828) RE:Re: [Asterisk-Users] MS Outlook
Thanks,
Brian
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Here's one thats way out in left field... don't use call pickup! :P
Problem solved sorta!
bkw
On Mon, 22 Sep 2003, Jared Smith wrote:
On Mon, 2003-09-22 at 15:42, Manuel Marn Garca wrote:
Please help! When I try to place a call pickup from a cisco phone 7960
using *8 the call is picked
-A INPUT -s x.x.x.x -p udp -m udp --dport 1:2
-j ACCEPT
-A INPUT -s x.x.x.x -p tcp -m tcp --dport 22 --syn -j
ACCEPT
...
Sunny
--- Brian West [EMAIL PROTECTED] wrote:
I'm trying to get some iptables rules that work with
asterisk but for some
reason I keep blocking everything
I bet your jack is wired backwards.. :) Try checking that out.
bkw
On Sun, 21 Sep 2003, Asterisk PBX wrote:
The echo canceller algorithms aren't doing anything. We get extreme
echo during the conversation, it appears even before the call connects,
the echo is there...
This only happens
Just on a side note can you please put a realname in your name field on
your email client. Everytime I see Asterisk PBX I think gee more
voicemail.
bwk
On Sun, 21 Sep 2003, Asterisk PBX wrote:
My partner found it!!
Problem solved...
The error was a syntax error in the zapata.conf
RFC 3389 - Real-time Transport Protocol (RTP) Payload for Comfort Noise
(CN)
Turn off CN and you will be fine.
bkw
On Sat, 20 Sep 2003, Don LeBlanc wrote:
Hello All,
We are running Asterisk on a linux server as a SIP proxy with Cisco ATA 186's at the
subscriber end. For long distance we
Try gdb asterisk /etc/asterisk/core.2035
bkw
On Sat, 20 Sep 2003, Dan Fernandez wrote:
Last week I did a CVS update and since then I haven´t been able to run asterisk
normally. The strange thing is that I have even go back to previous versions (0.5.0)
and I am seening the same problems.
I'm trying to get some iptables rules that work with asterisk but for some
reason I keep blocking everything and or locking myself out of the box..
mybad does anyone have any configs they would like to share that allow
asterisk and ssh from x ip?
TIA
bkw
Doesn't matter it should still work. Here is a hint.. dont use
passwords/secrets it will then work!
bkw
On Fri, 19 Sep 2003, Xisco wrote:
That's true if always there to connect two asterisk servers, but I'm doing
some proves in order to connect one asterisk server with another SIP server.
Ya I just got my phone and upgraded it to 5.3 without a problem. It works
perfect with *
bkw
On Wed, 17 Sep 2003, Travis Johnson wrote:
Yes. 30 phones in production environment. No problems so far. :)
Travis
At 08:21 PM 9/17/2003 -0500, you wrote:
Anyone running the 5.x firmware on
Just as simple to call your telco and have those turned off then its not
an issue ever!
bkw
On Wed, 17 Sep 2003, Ariel Batista wrote:
I would like to prevent * from dialing 900 and 976 numbers. I setup the following
settings in extensions.conf. But this does not seem to work! I don't know
Thats all going to depend on the speed of your DSL...
bkw
On Wed, 17 Sep 2003, Senad Jordanovic wrote:
Hi,
What are real life bandwith stats for * supported codecs?
Is it true one can run 6-32 conversations over DSL, as stated in this list?
Senad
Ya fire your network admin :P Firewalls shouldn't be blocking cvs and
such if they do then your admin is way too anal.
bkw
On Wed, 17 Sep 2003, Dan Austin wrote:
So I've been trying to pay attention, but I hadn't seen any updates on
SourceForge.
I inferred from the thread I could get a
Nope works fine here...
NUFONE ROCKS!
bkw
On Wed, 17 Sep 2003, Peter Pauly wrote:
Is anyone else having trouble dialing 800 numbers
through Nufone? I'm getting the SIT tones no matter
what number I dial. Normal long distance works fine.
I don't think it's my dial plan (it was working
6KHz != 6kbps
bkw
On Tue, 16 Sep 2003, Alex Zarubin wrote:
Is there a way to play adpcm, 6KHz in asterisk? If yes, where can we get
this codec?
Thank you.
Alex Zarubin
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Don't forget nufone can do outbound for 2.9 cents a min also.
bkw
On Mon, 15 Sep 2003, Steve Haehnichen wrote:
-= On Tue, 16 Sep 2003 00:07:39 +0200, Michael Koehler [EMAIL PROTECTED] said:
You get a Budgetone for free at Nikotel if you charge your account
there with 100 bucks. The
I never has problems with CVS I think the issues were with zaptel and
the order in which the wcfxo and wcfxs were loaded. Not totally sure.
bkw
On Sun, 14 Sep 2003, Timothy Soos wrote:
Hello All,
There have been some reports of person(s) checking out the latest version of
Asterisk
by loading
in a specific order, which i find strange...
On Sun, Sep 14, 2003 at 11:46:41AM -0500, Brian West wrote:
I never has problems with CVS I think the issues were with zaptel and
the order in which the wcfxo and wcfxs were loaded. Not totally sure.
bkw
On Sun, 14 Sep 2003
Have you tried:
exten = _9,1,Dial(OH323/${EXTEN:[EMAIL PROTECTED])
bkw
On Sat, 13 Sep 2003, Michael Manousos wrote:
Cerrajetto wrote:
Hello:
I am testing Asterisk with oh323.
My question is: can Asterisk route some calls thru a second h323 gateway (a
h323 - PSTN gw)?
${CHANNEL} doesn't work because it contains their uniqueid on the end such
as SIP/111-asdf
bkw
On Thu, 11 Sep 2003, Adam Goryachev wrote:
I use a single queue for all incoming calls, and different people login at
different times to handle the calls, however, quite often, people forget to
issue. If they are using Asterisk is it not possible to record calls
automatically. I have not reviews the CALEA requirements, must access be
Yes it is very possible to record calls with *. I record all in and
outbound calls.
bkw
___
Asterisk-Users
Accually you can MAKE * stand in the call path.
bkw
On Thu, 11 Sep 2003, Timothy Soos wrote:
On Thursday 11 September 2003 02:26 am, John Todd wrote:
Hello All,
Is it possible to monitor and record a SIP to SIP call? If so, how?
I gathered from some previous posts this would not
On the grandstreams if I recall the docs are incorrect on how the transfer
feature works. Transfer + EXT + Transfer
bkw
On Tue, 9 Sep 2003, Hielke Christian Braun wrote:
Hello,
hope somebody can help. I have setup a queue which maps to some
Budgetone SIP phones. When a call is answered,
Dude,
NuFone so totally ROCKS... I have yet to have any issues. The
last issue I had wasn't even related to NuFone.. but this stupid Nachi
worm nailing our routers and causing packets to be dropped. Other than
that the call quality is excellent. Customers can't tell the diffrence.
bkw
Also it wasn't a proven exploit. They said it could allow an attacker to
obtain remote and unauthenticated access. And if pigs could fly I
would be a rich man!
bkw
Read the security vulnerability. It referenced CVS as of a certain
date. If you aren't keeping up with CVS changes, why are
one and tested. Second its trivial to make one, if you see what is wrong
in the code.
Original advisory should have been posted here at the date of release,
or announced by someone, but it wasn't... I guess some people are too
busy, can't blame them.
Brian West wrote:
Also it wasn't
I agree they should stay at the dialplan level.
bkw
On Mon, 8 Sep 2003, John Todd wrote:
I know *8 kind of works with SIP
but what about the rest should they work
do they work with a zap device?
*0# sends flash
*8# remote call pickup (pickup phone in your group)
*67# disable caller id
HAHA I haven't used zap channels so I wasn't aware they existed there! :P
bkw
On Mon, 8 Sep 2003, Tilghman Lesher wrote:
On Monday 08 September 2003 01:56 pm, Brian West wrote:
I agree they should stay at the dialplan level.
It's not a matter of staying; it's a matter of moving. Those
I think the hold time needs to be announced only once when the caller is
injected into the call queue. Otherwise callers will hear hold times
shoot up if you have a few long calls.
bkw
On Sun, 7 Sep 2003, David C. Troy wrote:
Troy,
These is all good feedback; I did my patch primarily based
You didn't just say MS-ASF
MP3's good..
bkw
On Sat, 6 Sep 2003, John Brown wrote:
is there a clean way to have MOH (Music on Hold) source
its audio from say a MS-ASF streem ???
got a radio station that wants to have their MOH come
from their ASF based netbroadcast
Just tell em its ASF.. like the would know the diffrence.
bkw
On Sat, 6 Sep 2003, John Brown wrote:
On Sat, Sep 06, 2003 at 04:56:22PM -0500, Brian West wrote:
You didn't just say MS-ASF
Yup I did, ducking under the table... Customer requirement
MP3's good..
They rock
Just in case you guys haven't been paying attention Grandstream sliped in
some diffrent colors on the IP phones and looks like they released the
ATA-286 (Cisco is gonna have kittens I suspect)
bkw
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A friend and I have recently added the ability to announce the callers
position in the call queue every x seconds.. or even just inject an
anouncement every x seconds. All setup in queues.conf and can be setup
per queue.
My next project is to add the ability to announce the callers estimated
under 20 minutes
If I ever heard a time over 20 minutes I'd hang up and call back later,
or stop doing business with the company. This limits down your number of
prompts and lowers the expectation of wait time accuracy.
Sprint PCS comes to mind on that longer than 20 min hold times! :P
bkw
: [Asterisk-Users] app_queue input needed...
On Fri, 2003-09-05 at 14:05, Brian West wrote:
A friend and I have recently added the ability to announce the callers
position in the call queue every x seconds.. or even just inject an
anouncement every x seconds. All setup in queues.conf and can
The second method, where a sliding window average of wait times in
the last X minutes is used as the sample base is a bit more
difficult, but after some thought I am think it will provide a more
accurate number. Note that an unanticipated result of this method
may be that some callers hear
Why on earth don't you just compile it?
bkw
On Fri, 5 Sep 2003, Ben Bloomberg wrote:
Would anyone mind emailing me, or maybe posting somewhere their music
on hold .so file?
thx
-ben
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I opened a request on bugs.digium.com for this feature. The 6k and 8k
codecs are very impressive also.
bkw
On Tue, 2 Sep 2003, Eduardo Goncalves wrote:
Hello,
I've read about speex's bit rate. Speex can work from 2.15kbps to 44.2kbps.
With asterisk, what's the bit rate used by
Well put.
On Tue, 2 Sep 2003, John Todd wrote:
At 11:42 AM -0500 9/2/03, Brian West wrote:
http://bugs.digium.com/bug_view_page.php?bug_id=149
bkw
On Tue, 2 Sep 2003, Eduardo Goncalves wrote:
On Tue, 2 Sep 2003 10:27:53 -0500 (CDT)
Brian West [EMAIL PROTECTED] wrote
Try System(/usr/local/bin/hetest 01 on)
bkw
On Tue, 2 Sep 2003, Josh Edwards wrote:
Question below, here is the file in question
exten = 9,1,system,/usr/local/bin/hetest 01 on
exten = 9,2,system,/usr/local/bin/hetest 02 on
exten = 9,3,system,/usr/local/bin/hetest 03 on
exten =
Sounds like an IRQ issue.
bkw
On Tue, 2 Sep 2003, Gavin Hollinger wrote:
TE410P - intermittent one way audio forcing reboot multiple times per day.
but have thought of restarting asterisk at cron time. If someone's on
the phone at that time they're only talking to themselves anyway.
It was submited to bugs and added to the base isntall of * today.
bkw
On Mon, 1 Sep 2003, Paul Cheng wrote:
Yes, keep up the good work!
On Sunday, August 31, 2003, at 09:24 AM, Brian West wrote:
I have added support for enum looks for iax,iax2 and h323. So far in
my
testing it has
Nope wasn't me selling them... I see them on ebay and thought I would pass
the info along search ebay for sip phone they are still listed.
bkw
On Sat, 30 Aug 2003, Brian Capouch wrote:
Brian West wrote:
101's for 68.00
http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemitem
I get no echo on mine.. but you can check to make sure your line isn't
reversed. A reverse wired jack can do that.
bkw
On Thu, 28 Aug 2003, Brian J. Schrock wrote:
I can minimize doing those tricks, but I cannot seem to get it to go
away.
On Thu, 2003-08-28 at 11:33, Dan wrote:
-
Well depends.. what kind of problem are you having?
http://www.cisco.com/en/US/tech/tk652/tk701/technologies_tech_note09186a0080093f62.shtml
http://www.cisco.com/en/US/tech/tk652/tk701/technologies_problem_troubleshooting09186a00800c5e33.shtml
Check those... I suspect one of those has nailed
101's for 68.00
http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemitem=3042066051category=11175
102's for 79.95
http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemitem=3042066389category=11175
Just passing what I find along
bkw
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http://bugs.digium.com/bug_view_page.php?bug_id=156
patients grass hopper!
bkw
On Mon, 25 Aug 2003, Don Pobanz wrote:
Is it possible to configure * so that if a caller reaches voicemail for
someone in Engineering, but doesn't want to leave a message they can
press zero (0) and reach the
ment can't do iax via ENUM.. my bad! :P
bkw
On Sat, 23 Aug 2003, Brian West wrote:
Ok after a few mess ups on my end I have ENUM clue! :P Too bad we can do
IAX via enum..
bkw
On Sat, 23 Aug 2003, Brian West wrote:
Does anyone have any private ENUM examples
,
all of this subject to whatever restrictions are imposed on the use of these
features.
I will look into this.
Cheers,
Brad
On Mon 25 Aug 2003 14:25, Brian West wrote:
http://bugs.digium.com/bug_view_page.php?bug_id=156
patients grass hopper!
bkw
On Mon, 25 Aug 2003, Don
True.. thats why I say usually.. but it does yield more successful
muxing over just throwing the files together by trying to calc the in and
out diff.
bkw
On Mon, 25 Aug 2003, David Carr wrote:
If I understand the script, your technique first calculates how much longer
OUT is than IN, then
Does anyone have any private ENUM examples that will work with *?
bkw
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I ment can't do iax via ENUM.. my bad! :P
bkw
On Sat, 23 Aug 2003, Brian West wrote:
Ok after a few mess ups on my end I have ENUM clue! :P Too bad we can do
IAX via enum..
bkw
On Sat, 23 Aug 2003, Brian West wrote:
Does anyone have any private ENUM examples that will work
PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling
Sent: Saturday, August 23, 2003 1:33 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Intresting.. hrm
The real question is: How much?
On Fri, 2003-08-22 at 23:44, Brian West wrote:
And it runs linux.
http://www.zip4x4.com
http://www.politechbot.com/p-05040.html
Funny...
bkw
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Sent: Saturday, August 23, 2003 1:33 AM
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Subject: Re: [Asterisk-Users] Intresting.. hrm
The real question is: How much?
On Fri, 2003-08-22 at 23:44, Brian West wrote:
And it runs linux.
http://www.zip4x4.com/ZIP4x4.htm
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sox infile.wav -r 8000 -c 1 outfile.gsm
On Fri, 22 Aug 2003, Dan wrote:
Hi,
I have some voice prompts in WAV format (PCM, 44KHz, 16bit, mono, bitrate
705kbps) and I want to convert them in gsm format.
Using :
sox file.wav file.gsm
The result is a gsm sound file which when is played the
I don't put much faith in that lasting too long. They don't regulate
email or can't for that matter. They don't regulate streaming video...
Its a battle they will loose. Its data packets. If they get away with
regulations on this.. whats next regulation of FTP or HTTP for that
matter?
And as
And it runs linux.
http://www.zip4x4.com/ZIP4x4.htm
Anyone seen one?
bkw
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Or you can just tftp01.sipphone.com and get it! :P
bkw
On Thu, 21 Aug 2003, Steve Meyers wrote:
On Thu, 2003-08-21 at 00:32, Brian Capouch wrote:
I have seen two references today (don't recall whether here or on one of
the other VoIP lists I read) to people having the .78 version of the
Peter,
Did you read the website? Not only does it support h323.
Inter-Asterisk Exchange (IAX)
H.323
Session Initiation Protocol (SIP)
Media Gateway Control Protocol (MGCP)
http://www.asteriskpbx.com/index.php?menu=features
bkw
On Thu, 21 Aug 2003, Peter Eckhardt wrote:
Hello,
I am
Steve,
I pay 2.9 cents a min inbound 800 and outbound. Email
[EMAIL PROTECTED] I think he is being overloaded with requests for
information. It takes him all over 30 seconds to set someone up.
bkw
On Thu, 21 Aug 2003, Steve Lane wrote:
Nufone won't answer their phones. I am very
NUFONE R0X!
Took him 30 seconds or so to set me up when I got services with him! :P
bkw
On Thu, 21 Aug 2003, Jeremy McNamara wrote:
Our phones have been working perfectly fine all day. I've personally
supported quite a few new users over the phone today and even set a
couple up.
Jeremy
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