No snap shot is needed! You are able to check out the 1.0 branch from
cvs. Only bug fixes will go in this branch so you can automate the
checkout/update and rpm build process and produce daily 1.0 rpms.
To check out code from our STABLE 1.0 Branch CVS repository for Asterisk
ONLY:
# cd
I don't think that we've reached 1.0 stable, though, have we?
branching is an essential precursor in order to allow stablisation of
the current featureset to happen in a different space to the addition of
new features.
Personally I welcome this - both branch HEAD should benefit :)
However,
Post your config and we can see whats up..
bkw
On Fri, 6 Feb 2004, Mark Farver wrote:
We're still have problems with the outgoing voice message interfering
with the touch tone detection. Often the first touch tone pressed will
be detected twice. If I configure asterisk to not play the
Isn't the demo codec 1 channel only? Then one side is g729 and the other
is what?
do a sip show channels
bkw
On Fri, 6 Feb 2004, Wes Marderness wrote:
Hi,
Running Version 0.7.2, I receive the following error when attempting to
connect two SIP Devices.
WARNING[16399]:rtp.c : 1204
Nope.
bkw
On Fri, 6 Feb 2004, Chris Clifton wrote:
On the 7960's with *, when an internal sip line is dialed, is it possible
for the 7960 to display a status on the lcd that 'this ext is busy', etc. if
the line is in use ? Does this happen by default ?
Thanks,
Chris Clifton
their
'buddy watch' presence feature. Anyone else used this on recent polycom
soundpoint ip 500 or 600 phones with * ?
Chris Clifton
- Original Message -
From: Brian West [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, February 06, 2004 7:45 PM
Subject: Re: [Asterisk-Users] busy
wan
environment.
- Chris Clifton
- Original Message -
From: Brian West [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, February 04, 2004 1:58 PM
Subject: Re: [Asterisk-Users] Cisco 7960 bug in 6.1 evident in Asterisk
Does the first line, backup and emergency proxy go
cvs checkout asterisk-sounds
bkw
On Thu, 5 Feb 2004, Lance Arbuckle wrote:
I can't find Allison saying Do Not Disturb Anybody got this If
not, is there a place to submit generic requests for sounds ???
-Lance
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exten = s/6145551212,1,Answer
that will.
bkw
PS read the handbook
On Thu, 5 Feb 2004, Andrew Kohlsmith wrote:
Please do not start new threads by replying to a post; click on the
asterisk-users email address and start a new one; you break threading and
bury your own questions when you do
if you load wcfxo first then wcfxs its like this:
fxsks=1-2
fxols=3-4
reverse if you load wcfxs first.
bkw
On Thu, 5 Feb 2004, Steven E. Frazier wrote:
History:
1. Added X100P to my system
2. Added TDM400P (2 port)
Worked fine so far
3. Now I want to add an additional X100P
Is the
Question.. is the 7960 on the same subnet as your asterisk server? I have
a 7960 registered with 3 diffrent asterisk servers. All 6 lines. Running
6.1 and has 12 days of uptime.
bkw
On Wed, 4 Feb 2004, John Todd wrote:
So, I've managed to consistently lock up my Cisco 7960 (SIP 6.1) to
I have one word for you... LAZY!
bkw
On Wed, 4 Feb 2004, Christopher Lee wrote:
Out of interest, does anyone know if it's possible to get the 7960 to start
accepting a number while on-hook, without having to press NewCall, the
line button, or speaker button?
This is just something I was
features.
Anyway I've opened a TAC case with Cisco and will await their response,
which I'm guessing already will be no, can't hurt to ask tho :-)
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Brian West
Sent: Wednesday, 4 February
I agree. app_cepstral is a damn fine app and has been banished to the
edges of the earth because the theta engine isn't open src. I even added
a standalone build for app_cepstral... so you can download it.. make it
and install it without much trouble. :(
Andy maybe we can go thru and pickout
SayUnixTime will do that just give it the format you want.
SayUnixTime([unixtime][|[timezone][|format]])
unixtime: time, in seconds since Jan 1, 1970. May be negative.
defaults to now.
timezone: timezone, see /usr/share/zoneinfo for a list.
defaults to machine
- Original Message -
From: Brian West [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, February 04, 2004 02:23 PM
Subject: Re: [Asterisk-Users] talking clock
SayUnixTime will do that just give it the format you want.
SayUnixTime([unixtime][|[timezone][|format]])
unixtime
Andy's code and my code are the same code basically. I cleaned up a few
things and added the noanswer option. Other than that Andy did all of the
hard work.
bkw
On Wed, 4 Feb 2004, Brian Capouch wrote:
[EMAIL PROTECTED] wrote:
Feedback for the list. I compiled Andy's code. Installation
Voicepulse told me that there was no additional charge to enable trunking.
GASP SWOON!!! You received a response out of voicepulse?
bkw
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To
No it uses the linux theta libs and header files.
bkw
On Wed, 4 Feb 2004, Andreas Anderson wrote:
Hi Brian,
Andy's code and my code are the same code basically. I cleaned up a few
things and added the noanswer option. Other than that Andy did all of the
hard work.
is cepstral a special
http://asterisk.bkw.org/other/cepstral.tar.gz
bkw
On Wed, 4 Feb 2004, Brian Capouch wrote:
I'm prolly showing my ignorance here, but where *is* this code?
I've done a search at the bugs site and it came up dry. It's not in the
CVS contrib tree.
Don't know where else to look.
Thx.
B.
AM -0600 2/4/04, Brian West wrote:
Question.. is the 7960 on the same subnet as your asterisk server? I have
a 7960 registered with 3 diffrent asterisk servers. All 6 lines. Running
6.1 and has 12 days of uptime.
bkw
On Wed, 4 Feb 2004, John Todd wrote:
So, I've managed
http://asterisk.gnuinter.net/files/changelogs/asterisk-ng.ChangeLog
Compair from the last release date to today. :)
bkw
On Wed, 4 Feb 2004, Miguel Cavazos wrote:
would be a good idea to put it on the changelog, i see its there but it
doesnt really inform nothing.
Miguel Cavazos
On Wed,
'b' -- run AGI script specified in ${MEETME_AGI_BACKGROUND}
Default: conf-background.agi
(Note: This does not work with non-Zap channels in the same
conference)
NEXT!!! :P
bkw
On Wed, 4 Feb 2004, PBXtech wrote:
how? I dont see anything in there to listen for
($returncode);
}
On Wed, 4 Feb 2004, PBXtech wrote:
ok, makes sense.anyone got an AGI script that will pasrse DTMF? or
want to start one?
Thanks
Brian West wrote:
'b' -- run AGI script specified in ${MEETME_AGI_BACKGROUND}
Default: conf-background.agi
(Note
You might be able to.. not totally sure. Oh and Janets TT wasn't all
that.
bkw
On Wed, 4 Feb 2004, PBXtech wrote:
Could you use this setup to sense the end of a meetmet call to execute a
cleanup script also?
Brian West wrote:
#!/usr/bin/perl
use Asterisk::AGI;
$AGI = new
[macro-screen]
exten = s,1,Answer
exten = s,2,Wait(1)
exten = s,3,SetVar(SCREENFILE=/tmp/${TIMESTAMP}-${UNIQUEID})
exten = s,4,Playback(pls-rcrd-name-at-tone)
exten = s,5,Record(${SCREENFILE}:gsm|10)
exten = s,6,ParkAndAnnounce(silence/2:${SCREENFILE}:PARKED|20|${ARG1}|${ARG2})
Try this.
bkw
I do this:
print SET MUSIC ON default\n;
:P
On Wed, 4 Feb 2004, Matthew B Marlowe wrote:
Im the first to admit I am not a programmer. Im trying to learn peal to
accomplish one small script for my company.
I've got most of the script accomplished and I simply want to play music
on hold
missing something here? Wont a print SET MUSIC ON
default\n; just print that text?
Don't I need to execute it somehow with $AGI-?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian West
Sent: Wednesday, February 04, 2004 11:04 PM
To: [EMAIL
.
Is it anywhere? I'd like to read up on AGI as much as I could.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian West
Sent: Wednesday, February 04, 2004 11:04 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Music on hold inside
well each call is like an http request... you can't keep state from one to
the next like that.
bkw
On Wed, 4 Feb 2004, John Todd wrote:
ParkAndAnnounce is kind of a misnomer; you don't necessarily need to
announce.
I wholeheartedly agree that a variable (${PARKINGSLOT} perhaps?)
should be
How do you start asterisk? using safe_asterisk? or what cli options do
you give it?
bkw
On Tue, 3 Feb 2004, John Todd wrote:
At 8:31 PM -0700 2/3/04, Rohde wrote:
I've looked online through both google and bugs.digium.com and cannot seem
to find this problem anywhere, so i'll ask its
You can't use that with agentcallbacklogin.
bkw
On Mon, 2 Feb 2004, Eric Wieling wrote:
Can anyone confirm that the feature listed below works? I'm using
AgentCallbackLogin and it never seems to log the agent off if they don't
answer.
/etc/asterisk/agents.conf
; Define autologoff times
Correct me if I'm wrong here but when a message bounces and the mailer/mta
generates a bounce message shouldn't the from field have in it instead
of an email addres (ie. [EMAIL PROTECTED]).
The list was nailed with over 13,000 bounce messages(and they keep coming)
from ONE list subscriber and I
generally ARE things like
MAILER-DAEMON. The envelope sender used in the SMTP conversation and
Return-Path: should be .
On Sat, Jan 31, 2004 at 01:20:02AM -0600, Brian West said:
Correct me if I'm wrong here but when a message bounces and the mailer/mta
generates a bounce message shouldn't the from
Nope I do make install all the time with asterisk running without ONE
problem.
bkw
On Sat, 31 Jan 2004, William Waites wrote:
While your problem is most likely bad RAM as other
replies have suggested, there is another thing to
keep in mind.
Some implementations of dynamic module loading
That sending notify to endpoints that aren't registered has been fixed
recently but we have one more bug that causes 0.0.0.0
bkw
On Sat, 31 Jan 2004, Clif Jones wrote:
I noticed this too and it is a pain to look at. I saw it because some
of my SIP phones were turned off and
the NOTIFY's for
2004-01-26 14:12 markster
* channels/chan_sip.c (1.284): Don't send VMWI when we're not
registered
Yes that was fixed on the 26th.
On Sat, 31 Jan 2004, Clif Jones wrote:
I noticed this too and it is a pain to look at. I saw it because some
of my SIP phones were turned off
test
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OMG I want to say If I find the person that has the mail server Never
mind its fixed now. But please check your mail filters to make sure they
bounce mail properly.
bkw
On Fri, 30 Jan 2004, Brian West wrote:
test
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Asterisk-Users mailing
Rich,
Thanks... I just double checked everything and she's still moving
along. You shouldn't see those nasty virus stuffs on the list anymore
either. :)
L8tr,
bkw
On Thu, 29 Jan 2004, Rich Adamson wrote:
Brian,
Great job fixing up the list server... postings are happening very
It wasn't just me Malcolm and I both fixed it!
bkw
On Thu, 29 Jan 2004, Frankie Gravato wrote:
Hello Rich,
Great Job Brain aka Bkw
Posts are getting up really quickly
thanks for fixing it
-Crontibs
--
Best regards,
Frankie ([EMAIL PROTECTED])
mailto:[EMAIL PROTECTED]
had to email a message to [EMAIL PROTECTED]
from your member email account and it would be accepted. My mail server says
digium is getting the message, so I'm confused as to why it isn't showing up
on the list.
MATT---
-Original Message-
From: Brian West [mailto:[EMAIL PROTECTED
Doesn't matter the host= line is a host not [EMAIL PROTECTED]
but if you are doing a register the hostline should be host=dynamic
bkw
On Thu, 29 Jan 2004, MLS Drop for SysAdmin wrote:
As I said in my post, I have substituted @domain.net for the actual domains
in use to simplify the posting.
Works for ilbc and speex here! :P
On Thu, 29 Jan 2004, Kostur, Andre wrote:
On the flip side, I have tried the registry fix with X-Lite build 1101
against Asterisk 0.7.1 (Debian/Unstable package), and it works (at least for
speex)
Asterisk 0.7.1 from Debian/Unstable
Win2K
X-Lite build
Its coming... we are trying to work in all the bug fixes from the
bugtracker before we branch.
bkw
On Thu, 29 Jan 2004, Philipp von Klitzing wrote:
Quote:
Asterisk version 0.7.0 will be released by Monday Jan 12, 2004. Later
that week, we will create a stable branch from which eventually
both /usr/bin and /usr/local/bin will work.
bkw
On Thu, 29 Jan 2004, kemal asad wrote:
as i am trying to use asterisk and install my newly purchased ( got it
yesterday) digium cards.
i am following the very detail steps of
http://www.automated.it/guidetoasterisk.htm.
but one thing did not
I use gotoiftime to send them to voicemail during the hours I don't wanna
deal with them.
bwk
On Thu, 29 Jan 2004, Chris Albertson wrote:
--- Rich Adamson [EMAIL PROTECTED] wrote:
snip
Actually, haven't had the problem here at all.
I was about to write that I though I was getting these
was it that you did.
Some of us who run mailmain list servers might want to know.
--- Brian West [EMAIL PROTECTED] wrote:
Rich,
Thanks... I just double checked everything and she's still moving
along. You shouldn't see those nasty virus stuffs on the list
anymore
either. :)
L8tr,
bkw
that works best here.
On Thu, 29 Jan 2004, Jonathan Moore wrote:
I may be misunderstanding the question but what is wrong with using an extension
line like the following?
exten = 3,1,Dial(SIP/snom200SIP/snom100SIP/gs1,15,Ttr)
I use this to have all three desk phones in my office ring on
BTP
NEXT!!! :P
On Thu, 29 Jan 2004, Geert Nijpels wrote:
Jonathan Moore wrote:
I may be misunderstanding the question but what is wrong with using an extension
line like the following?
exten = 3,1,Dial(SIP/snom200SIP/snom100SIP/gs1,15,Ttr)
I use this to have all three desk phones in
turn echotraning on or update to CVS also try leaving the tx and rxgain at
zero
bkw
On Thu, 29 Jan 2004, Stephen R. Besch wrote:
Just updated from CVS 12-23-03 to tarbal 0.7.1. Identical settings in
zconfig.h for echo cancellation (MARK2, aggressive OFF). The echo got
worse, much worse.
I found by accident that there is a limit of 99 messages in your INBOX in
Asterisk.
The 100th attempt to record a voice mail causes the system to play your
greeting and then never record the 100th message and silently disconnect
the caller.
So...is it safe to simply use the UNIX find
I'm another outsider just reading this entire thread and Jeremy's
replies. NuFone could have the best service on the planet, but I would
NEVER do business with a company that has this kind of attitude. People
have a choice in which companies they do business with. Why would they
do business
Testing once again.
bkw
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Steve,
I have been working most of the day to get this problem solved.
Thus far everything should be returning to normal and problems like this
should never happen again. Right now it has about 3 hours of posts to get
out and some were lost by accident during this mess. So if you see a
testing
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That is no longer a problem.
bkw
On Wed, 28 Jan 2004, Tazman wrote:
Hi, a Virus was sent from this account to the Asterisk-Users mailing list...
scan your computer for virus!!!
Ronen
- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, January 26,
test
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check out the latests cvs it has two reg files that will fix xlite or xpro
to work
bkw
On Mon, 26 Jan 2004, Wim Venneman wrote:
If this may be of any use:
I'm not an expert but I did the test with the FWD soft phone from X-ten and
iLBC SPX don't work.
Asterisk wasn't between the
testing yet again.
bkw
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What you (and a lot of consumers) don't understand is that some
customers need to get fired. Some people aren't worth doing business
with. Those are the facts.
I have done this. I have felt better after I fired a cusotmer in
situations like that.
bkw
All of the mailing lists are now filtered for virues and spam before they
reach digiums network. This will ensure that the mess that caused the
list server to break down wont happen again.
Its still playing catchup. But the mail is now flowing faster than usual
now.
Thanks,
Brian (bkw_)
Exten h isn't needed at all to record CDR info. Also exten h won't run if
you park the call.
bkw
On Sat, 24 Jan 2004, Girish Gopinath wrote:
Hi friends,
I have the entry exten = h,Hangup in my extensions.conf, and I am trying to
record the call details for billing. From the wiki i found
Have you tried to call them? Your emails could have been caught up in a
spam filer or such I use nufone daily for our 888 service. I talk to
Jermey daily. So I dont know what your beef is but your rant has no place
on this mailing list if you are having problems and have spent any time
Also on a side note... I have noticed emails from Jeremy back to me
getting caught by spamassassin because of his ip addresses on his dynamic
dsl connection.
bkw
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I have seen bluetooth do nothing but grow. More and more devices are
getting bluetooth the headsets can be had for 64 or so at buy.com
bkw
On Sun, 25 Jan 2004, Steve Underwood wrote:
Hi Don,
A large number of GSM phones and PDAs now have bluetooth. It looks
likely that through 2004 the
I think this has been fixed since 0.5.0 their was a problem with timeout's
and native bridges. Might wanna update.
bkw
On Fri, 23 Jan 2004, Wes Marderness wrote:
Hi All,
I've been having a hard time getting the AbsoluteTimeout function to work.
Is this Function working in for SIP? I've
NO it will log from a spool file if and only if you ref an extension and
not an application.
bkw
On Fri, 23 Jan 2004, Kannaiyan Natesan wrote:
There is no CDR for the call from spool outgoing,
You need to write a patch to solve the same.
Kannaiyan
- Original Message -
From: Iain
Its silence supression. Turn that off and it will stop doing that.
bkw
On Fri, 23 Jan 2004, Zot O'Connor wrote:
I am getting (with older image):
RFC3389 support incomplete. Turn off on client if possible
How do I turn that off for the DG104s? Or if I can't how do I tweak
asterisk?
I
I would love to use native Mysql also but the whole reason I wrote
cdr_odbc is to sidestep the GPL issues and still have mysql without having
to update this and that all the time.
bkw
On Thu, 22 Jan 2004, WipeOut wrote:
Tilghman Lesher wrote:
On Thursday 22 January 2004 08:01, Andrew
1. they aren't digiums.
2. they don't come with support
NEXT!!!
bkw
On Wed, 21 Jan 2004, SamW wrote:
Digium X100P / new cards are is available on ebay for $43.
http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemitem=3073050567category=3309
http://news.com.com/2100-7352_3-5144864.html?tag=nefd_lede
Little blurb but hey its on NEWS.COM.COM.COM.COM.COM.COM har har har
bkw
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disallow=al
allow=gsm
bkw
On Wed, 21 Jan 2004, Michael Welter wrote:
I've downloaded diax-0.9.6b and configured for IAX2. Calls from Diax to
* are perfect. However, when calling from * to Diax, I get the following:
channel.c:1097 ast_read: Dropping incompatible voice frame on
It was my impression that these phones had 10MB ehternet connections and not
100MB. Not that most of us would notice the difference in browsing the net,
it does defeat the purpose of having 100MB switches. (I believe I also saw
people on this list talking about strange things happening when
Can you clarify this? Does it or doesn't it work?
bkw
On Mon, 19 Jan 2004, Asterisk User Group wrote:
I had been running an older patched CVS to get VOIP working with NAT and
everything had been running fine. I just built * on a new box with
CVS-01/18/04-12:19:25. And now I can get remote
Use account codes. That works ALOT better. If you require passwords then
look at app_authenticate.
bkw
On Tue, 20 Jan 2004, Ing Isianto Istiadi wrote:
Dear all,
I have a questions:
1. I have 3 FXO connected to 3 analog phones. But I have 5 users using those
phone. I want to be able to
Also you should never call an application directly with an call file as
the CDR info won't get updated correctly Link your app to an extension
then call it like that.
bkw
On Mon, 19 Jan 2004, Marcin Kuzmicki wrote:
Cytowanie Charles Hatchette [EMAIL PROTECTED]:
I'm trying to devise a
You can't use chan_h323 with call manager.
bkw
On Sun, 18 Jan 2004, Dan Austin wrote:
I tried to use it to create a 'trunk' to Cisco's call manager. The
0.7.1 code worked up to a point.
The call would be established, but audio was one-way from the Call
Manager. Asterisk with
Chan_h323
You can also get the same files from ftp.digium.com
I did a mirror to help people get it faster in case cvs was hammered
again.
bkw
On Mon, 19 Jan 2004 [EMAIL PROTECTED] wrote:
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Shailesh
Check your tables. I logged everything as integer.
set verbose 10 and make a call and watch it.. then do reload and watch the
output. It will unload and reload and you can check to make sure your
accually connetcing to the database.
bkw
On Sat, 17 Jan 2004, Iain Stevenson wrote:
I've just
Their is no need to telnet with perl you can just shove a notify packet
down the cisco and let it reboot on its own. Really easy.
bkw
On Sat, 17 Jan 2004, Steven Critchfield wrote:
On Fri, 2004-01-16 at 15:26, B. J. Bomar wrote:
Yes, I was wanting to do it via a script, but telneting in will
it doesn't even realise anything is wrong, but might crash at any moment :-)
ie Windows Media Player :P I finally got it to blow up on me lastnight.
EVIL THING...
bkw
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Also it will log the userfield if the uniqueid is set to on.
bkw
On Fri, 16 Jan 2004, Michael Labuschke wrote:
Hi,
either turn loguniqueid=off or add a field userfield to your sql table ( i
made it varchar 20 since i could not find any information ;) after the
loguniqueid field in the db.
Just ignore my last post.. I'm not away or something! :P
bkw
On Fri, 16 Jan 2004, Michael Labuschke wrote:
Hi,
either turn loguniqueid=off or add a field userfield to your sql table ( i
made it varchar 20 since i could not find any information ;) after the
loguniqueid field in the db.
Works fine here.. got two of em.
bkw
On Fri, 16 Jan 2004, Bill Hamel wrote:
Hi,
Just got some CISCO 7960 phones. They seem to work great except if I make any
SIP call using the speaker phone (leaving the hand set in the cradle)the call
will disconnect in about 5 or so seconds. If I pick up
http://www.bkw.org/~brian/cisco/reboot7960.txt
or you can us this handy perl script..
NEXT!!!
bkw
On Fri, 16 Jan 2004, Rich Adamson wrote:
Does anyone have a working way of having a Cisco 7960 reload its config remotely.
I
have tried some of the scripts that I have found
on the web,
Questions... happen to use webvmail?
bkw
On Thu, 15 Jan 2004, Brian Capouch wrote:
I have a user, running CVS a/o 11/23/03, who has complained about
phantom messages showing up days or even weeks after she has deleted them.
So I asked her to let me know when it happened again, and she
Same here. I can't recreate the problem. I think this is a windows media
player issue.
bkw
On Thu, 15 Jan 2004, Troy Settle wrote:
I can't reproduce this either, but I do have the gsm codec installed (though
WMP won't play a .gsm file).
I play the wav49 files in Winamp with no issue.
Create a new wav49 on your system and play it.
bkw
On Thu, 15 Jan 2004, Warwick Ward-Cox wrote:
I'm having the same problem.
Warwick
- Original Message -
From: Jim Flagg [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, January 15, 2004 5:39 PM
Subject: [Asterisk-Users]
Or use this http://www.cam.org/~noelbou/1-step.html
bkw
On Thu, 15 Jan 2004, Troy Settle wrote:
I can't reproduce this either, but I do have the gsm codec installed (though
WMP won't play a .gsm file).
I play the wav49 files in Winamp with no issue.
--
Troy Settle
Pulaski Networks
UM take that =a - that a is bad
bkw
On Tue, 13 Jan 2004, Sean Garland wrote:
I have the standard parking.conf but extension 700 doesn't show up in my
dialplan Why? I can dial 701 which tells me that I don't have any
calls parked there. 700 just gives me invalid extension noise
: Tuesday, January 13, 2004 11:38 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Asterisk 0.7.0
A Ter, 2004-01-13 às 16:01, Brian West escreveu:
Why not quickly patch the source an release 0.7.1 if the bug is
critical?
Give it a few days and I bet we will. because chan_h323
You can no longer dial 700 directly nor have 700 in your dialplan. I just
tested this and park does infact work make sure you have T or t on the
dial.
bkw
On Tue, 13 Jan 2004, Sean Garland wrote:
I am having trouble with call parking... I am basically using the stock
sample files, but
It won't show the parking extension when you do a show dialplan
bkw
On Wed, 14 Jan 2004, Sean Garland wrote:
I have just set the parking extension at 701 and then the range is
702-710 and still I cannot transfer to 701. Show Dialplan doesn't show
an extension 700, although it shows all the
Yes it will work.
bkw
On Wed, 14 Jan 2004, Michael Welter wrote:
Will 0.7.0/1 work with the existing zapata-0.8.0 and libpri-0.5.0, or
have they been modified as well?
TIA
Mark Spencer wrote:
Asterisk 0.7.1 has been released fixing a few minor bugs. Thanks again to
the bug
http://bugs.digium.com/bug_view_page.php?bug_id=851
NEXT!!!
bkw
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No he renewed it... but Gododaddy did the transaction over the phone
manually and they never posted the payment so they shut it down. It
should be back by now.
Switch-1 ip is 66.225.202.72
bkw
On Tue, 13 Jan 2004, Steven Critchfield wrote:
On Tue, 2004-01-13 at 01:26, Brian Capouch wrote:
Here are a list of mirrors for the 0.7.0 tarball.
http://66.225.202.82/downloads/asterisk-0.7.0.tar.gz
http://parc.styx.org/asterisk/asterisk-0.7.0.tar.gz
http://www.bkw.org/asterisk-0.7.0.tar.gz
http://www.moctel.com/asterisk/asterisk-0.7.0.tar.gz
http://matrix.gs/asterisk-0.7.0.tar.gz
Why not quickly patch the source an release 0.7.1 if the bug is critical?
Give it a few days and I bet we will. because chan_h323 is broken also in
0.7.0 (JerJer :P but him and I stayed up till 3 am fixing it.)
bkw
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It's hard to call a realease stable until a number of people outside
the developer's lab have used it for a while.
0.7.0 was never taged nor advertised as stable. Its just the last release
before the 0.9.x branch is tagged. 0.9.x will go thru this process of
release after release till the
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