Apparently you didn't read the README.. Please read that over again.. it
tells you exactly what to do.
bkw
On Sun, 10 Aug 2003, Serge Mankovski wrote:
> Hi
> I am using inAccess channel driver.
> Compiled, installed. This is what I get when I am trying to start *
> --
incominglimit is already implemented for SIP. Just specify under the
endpoint how many incoming connections are allowed.
For example,
[cisco]
type=friend
username=cisco
secret=blah
nat=yes; This phone may be natted
host=dynamic
canreinvite=no ; Cisco poo
We've found FUDForum to be great.
http://fud.prohost.org/
Its free and supports forums with gateways to mailing lists and newsgroups.
>From their feature list:
a.. NNTP & Mailing List integration, allowing FUDforum to be used to archive
newsgroups & mailing
lists, as well as allow forum members to
Rattana BIV wrote:
I run with asterisk-oh323 0.5.4 from inaccessnetwork.
What message do you get in your mailbox?
asterisk-oh323 does handle correctly and passes the
called ID number.
Thanks
Rattana
Michael.
___
Asterisk-Users mailing list
[EMAIL PROTEC
I am testing a with 533Mhz Celeron/ 256MB. I guess this is certainly low
end.
Thanks,
Ricardo
- Original Message -
From: "Michael Manousos" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, August 06, 2003 3:34 AM
Subject: Re: [Asterisk-Users] Wierd Message
> Ricardo Villa
Hi,
when using multiple * boxes, there appear to be 2 choices as to how
to go about sharing cards and dialplans
1) using switch
2) using dial fail fall-through ie
exten => 1,1, dial(xyz)
exten => 1,2, dial(otherpbx/xyz)
As i see it switch could end up being recurrsive resulting in a wild
ooc
Hi Uriel,
Forgive me if you've already done this, but have you checked disk space on
the mailserver? Its caught me before and might save you hours debugging
something that isn't broke.
W
- Original Message -
From: "Uriel Carrasquilla" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Sa
Just cvs'ed about 40 minutes ago (10:15 CST 8-8-03). Segfaults when I
use a queue app in many different scenarios. When calling phone is only
member of queue I get a segfault. When 1st called extension is outside
line I get a segfault. Many other scenarios as well. Unsure how to go
about troubl
Hi again,
struggling with localization issues (so the script is not "German only")
took me a week longer than expected. (Did anybody ever get PHP's gettext
extension working??)
But finally, I've wrapped something up:
On Thu, 24 Jul 2003, Dave Packham wrote:
> I would like to see your code...
>
>
> >
>
> Why don't you just Dial() and then press the DTMFs, when the
> channel has been answered?
>
> Michael.
Main reasons..
1. Its a PITA..
2. You probably don't want to tell everyone what your pin number or access number is..
3. If the number changes cos you get a better deal somwhere else
Hi,
We are testing with G729 from remote offices to a
central asterisk machine. With a SNOM 200 the g729 is terrible. We notice the
following:
1) when dialling voicemail, the first part of the
announcement is missed.
2) the sound is very quiet, and sound quality is
terrible (tinny soun
Asterisk does not included a gatekeeper but you can
download a one from http://www.openh323.org
As for
whether or not you NEED gatekeeper, probably not. If you have a gatekeeper
you can register your h323 endpoints (openphone) with the gatekeeper, then have
asterisk register with the gate
On Wed, 2003-08-06 at 20:00, Richard Alexander wrote:
> And further to Dan's message I will add that I was able to help because
> a colleague and I are working on identifying all callerid variants with
> a view to patching * to work with as many as possible.
>
> If anyone has specific examples of
From: "Wade Weppler" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Subject: RE: [Asterisk-Users] Sip Trunk config / Least Cost Routing
Reply-To: [EMAIL PROTECTED]
Date: Thu, 7 Aug 2003 17:48:01 -0400
Why use an AGI? This seems to be easily done with the dialplan,
unless I'm missing some additional
Hi Dan,
when you get voicemail the dial tone changes... not only that but on my dect phones
i get a little specaking face icon that flashes...
Andy
>One more question about TDM20B. How can you know when you have a new
>voicemail in your mailbox?
>On ATA, the dialtone is modulated for a couple of
Hiya,
First off, thanks to everyone involved in app_queue. Its a great addition to
an already great system.
For my two penneth (or cents!), I think the following would be good:
- the fallback method should be optional if at all possible, so it can be
set up for, say, fewest calls with ring all a
it is option 2 since asterisk does not support option 1 :)
On Saturday 09 August 2003 10:19 am, Scott Stingel wrote:
> Hi all-
>
> This question is for those familiar with EuroISDN setup.
>
> I have a customer in Europe where I'm going to install an asterisk based
> system with 4 E1's. The custom
Hi,
Some questions:
1) Are you behind NAT?
2) If the answer to (1) is Yes, then be aware that
if you have the latest firmware (1.16w) then you should choose the the
appropriate setting under "NAT detection". The "Automatic" setting doesn't seem
to work some of our customers behind nat.
http://www.perldap.org/
http://perl-xml.sourceforge.net/
That's a start for you.
JT
Sorry about taking this OT
How do you go about "you could write a quick LDAP->XML dump with perl"
Please just point me to a howto.
TIA
John Todd wrote:
Nice...so mixing and matching IP and POTS is ok and
Hi,
thanks for that.
after implementing yours and "wipeout's" suggestions (thank you both),
x-lite changed its default codecs to G711a. which is great... and a way
forward.
but it still does not play sound when the "1000" is dialed.
my * is behind nat. and my test pc is as well.
Here are my sett
Thanks
WEll, not sure how to do that.Anyway.
I will move * on one of our internet servers. That should "take away"
all NAT/PAT issues.
Senad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Nathan
Littlepage
Sent: 09 August 2003 11:53
To: [EMAIL PROTECTED]
Try kill -9. Also make sure there's not more incidence of it running. I
don't know if this is the case with asterisk but some programs spawn
multiple processes. If your using RedHat and you've installed the startup
scripts you can execute /etc/rc.d/init.d/asterisk stop. You can replace
stop
On Tue, 2003-08-05 at 07:46, John Schmerold wrote:
> OK. Thanks - I think :-)
>
> I'll go trolling on Ebay, see what comes up. Given that most of my projects
> take 6 months or so to get off the ground, I hate to put a bunch of money into
> this anyway. So, for <$1,000, I can put a 6 x 18 uni
kill -9
--
Todd Lieberman
800-675-3078
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jim Friedeck
Sent: Friday, August 08, 2003 5:50 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Killing runaway PBX
How do I stop asterisk when it is in a bad moo
Agreed. I was *just* thinking that it would be incredibly useful for me to
have a way to get in touch with someone who has implemented Asterisk in a
way similar to the way we're trying to do it. I debated on asking on the
list, but since I'm new, I'm not sure if that breaks any rules.
- Devon
> "Jeremy" == Jeremy McNamara <[EMAIL PROTECTED]>:
Jeremy> Jan Rychter wrote:
>> Please try to find a better solution.
>>
Jeremy> The DSP Group owns G.729. There is nothing anyone can do, they
Jeremy> have us by the family jewels. We use iLBC and found it to be
Jeremy> very acceptable
Hi!
I've been trying to use the Voicemail (and Voicemail2) applications with
an SIP Phone (X-Lite, for those who cares), when I use g.711(a/u) codec,
it works perfectly with inband (it detects the whole mailbox (in my case
10007)), but not with rfc2833 (in this case, it only detects 107 as the
mai
> And you should take a look at queues.conf for some comments detailing the
> various queue distribution algorithms, ringall, roundrobin, leastrecent so
> on so forth.
I wanna see if anyone else has seen this result?
All except of which do not work. The only method I can get working is
ringall.
Hi,
On Fri, 2003-08-08 at 22:46, Jayson Vantuyl wrote:
> On Fri, Aug 08, 2003 at 01:06:32AM +0200, Armand A. Verstappen wrote:
> > On Thu, 2003-08-07 at 22:17, Wade Weppler wrote:
> > > Any idea if these fixes will get added to CVS?
> > check http://bugs.digium.com/bug_view_advanced_page.php?bug
AT&T 957 analog sets
AJ
On Fri, 8 Aug 2003, WipeOut . wrote:
> What phones are you using?
>
> > In my recent new Asterisk installation I'm having users complain that if
> > they answer a call on call waiting while talking on an existing line they
> > are then unable to park a call without one
I'm wondering if there is any command-line interface available for
working with values stored in astdb. Of course, I can run "asterisk
-rx "database show" " or other commands like that, but I was hoping
for a local command that would allow manipulation or output in some
other form. Is astdb i
you have to make /etc/zaptel.conf
and /etc/asterisk/zapata.conf
match on the same type of signalling ..
should work then :)
cheers
Dave
- Original Message -
From: <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Friday, August 08, 2003 12:24 AM
Subject: [Asterisk-Users] Newbie Issue
>
Is it possible to know what application? The extension I'm daling is very
simple:
exten => 1001,1,Dial(SIP/1001,15)
exten => 1001,2,Voicemail2(u1001)
As soon as the Voicemail picks up the NOTICE line appears multiple times on
the console.
Thanks,
Ricardo
- Original Message -
From: "Mart
I think that's a very good idea. When I started to become active in * last
December the list was much less congested and Mark usually responded to
requests, comments and patches within a few hours. Now things are clearly
taking off - good for * and Digium but it's sort of losing the community
On 08 Aug 2003 11:39:47 -0500
Steven Critchfield <[EMAIL PROTECTED]> wrote:
> >
> > Fax sometimes goes without problem and sometimes the fax machine can't send
> > the fax.
>
> If your SIP link is over a long line, or that has much in the way of
> variable latency, then this will just have
Hi to all,
does Digium's X100P, TDM40B & E100P works in
PCI-X slots? I want to install those cards with new Intel server
boards
Thanx in advance,
Victor...
I have been running busydetect=yes, using BUSYDETECT_MARTIN and am
having hangups during calls.
The busycount=6 workaround seems to be doing the job, but I was
wondering if there was any value in using BUSYDETECT_TONEONLY or
BUSYDETECT_COMPARE_TONE_AND_SILENCE, as well as BUSYDETECT_MARTIN or a
- Original Message -
From: "The Traveller" <[EMAIL PROTECTED]>
> Hey all,
>
> As there seem to be some problems with DTMF-signalling between chan_sip
> and several clients, due to which many could not properly dial a number
> at the dial-tone of the XS4ALL-gateway at FWD-number "42442",
Hey all...I'm brand new to * and I want to convert my home into a pbx
type setup. I've figured out that I want a Wildcard X100P to bring my
single POTS CO into my Linux box. My problem is that I'm sure sure what
I need to do to get my analog phones connected up into a structured
phone system. I
could you send me the exact syntax for rxgain / txgain?
I think that might help towards my problem
becuase i'm having to turn the handset volume all the
way up ..
thanks
Dave
- Original Message -
From: "WipeOut ." <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, August 05, 2003
Hi Klaus-Peter,
On Wed, 2003-08-06 at 12:33, Klaus-Peter Junghanns wrote:
> always use latest chan_capi. the bug is fixed in 0.2.4a.
> today 0.2.4b is online which fixes some issues with sending
> dtmf and a small enhancement to capiECT.
I checked the site, but can't find the 0.2.4b version. The
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