Title: Message
Yes,
it is. But why would you want to do that when yo said what you want it to
be at 6.0.
Maybe
you didn't expling what and why you want to do it in enough detail to get a good
answer.
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
people don't know. That's excusable. Others refuse to do it
because their mail client defaults to top posting (ahem, Outlook).
Please take the time to do it correctly.
Daryl
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.
[...]
Daryl
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Sean and WipeOut...that's greatexactly what I'm after. Works great.
Daryl
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, but would like to get some of the finishing touches like
this underway.
Thanks in advance for any help you can provide.
Daryl
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it at all.
Daryl
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/linux symbolic link to the directory it unpacks
to.
Daryl
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-Original Message-
From: nathan [mailto:[EMAIL PROTECTED]
Sent: Monday, September 15, 2003 10:48 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Analog FXO Card
These cards are replicas of the X100P sold for use in an Asterisk (
www.asterisk.org) phone system. They are
for the university.
do you have a cable pin-out descriptions for that purpose? thanks!
A T1 cross-over cable is:
1-4
2-5
3-3
4-1
5-2
6-6
7-7
8-8
Daryl G. Jurbala
BMPC Network Operations
Tel: +1 215 825 8401
Fax: +1 508 526 8500
INOC-DBA: 26412*DGJ
PGP Key: http://www.introspect.net/pgp
to the channel bank, which muxes
them to an E1 or T1). That way you have have fewer cards in your * box,
fewer interrrupts to poll, fewer things to go wrong.
Channel banks are arguable much more reliable than PCs and PC hardware.
Just a thought...might not be possible depending on your setup.
Daryl
those in, and it hasn't hurt so
far, but maybe i'm missing something (which is oft the case)
Since a T1 is only two pair form the telco, no. But properly made
cables ought to be fully pinned, for strength if no other reason, and
that's the standard way to do it.
Daryl
. It is possible with a CB?
Define full functionality. I'm not aware of ANY advantages (other
than cost in low density installations) for using FXO cards. Maybe you
have a functionality requirement that I'm not aware of, and is out of
the ordinary.
Daryl G. Jurbala
BMPC Network Operations
Tel: +1
an duplex
on the phone as well as on the device attached to the phone.
Daryl G. Jurbala
Introspect.net Consulting
Tel: +1 215 825 8401
Fax: +1 508 526 8500
http://www.introspect.net
PGP Key: http://www.introspect.net/pgp
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[several very good point deleted]
Thank you. Well stated, and you saved me the typing ;) Find me SIP
termination with unlimited minutes at a reasonable flat rate to US
destinations that works natively with * and I'll dump Vonage tomorrow
(and deal with the rest).
Seriouslyplease?
Daryl
.
At the moment, I believe they have DIDs in Michigan (and 800) only. I
am told they are working on agreements for more locations.
My vonage # forwarded to the Michigan DID seems to work just fine ;).
Daryl
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hardware and you're all set.
Daryl G. Jurbala
Introspect.net Consulting
Tel: +1 215 825 8401
Fax: +1 508 526 8500
http://www.introspect.net
PGP Key: http://www.introspect.net/pgp
-Original Message-
From: Hemant Kumar [mailto:[EMAIL PROTECTED]
Sent: Wednesday, September 03, 2003 3:52 PM
You have to go to Settings- #9 Unlock Config in v4+ firmware. The
unlock password no longer works from just anywhere.
Daryl
-Original Message-
From: Rich Adamson [mailto:[EMAIL PROTECTED]
Sent: Wednesday, September 03, 2003 9:58 PM
To: Asterisk-users-list
Subject: [Asterisk-Users
Considering www.asteriskpbx.org doesn't mention zapata either, care to
enlighten us at to what it does?
I've got zapter, libpri, and asterisk compiled and running. While I
haven't had the change to play with ALL of the features, all seems to be
working fine with my setup.
Daryl
-Original
Actually, that's incorrect. The code in zapata has long
since been incorporated into other code, so zapata is no
longer necessary.
In that case you can cancel my last question. ;)
Daryl
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* cd /usr/src/linux (you did unarchive the sources, and ln -s
/usr/src/whatever Debian called it /usr/src/linux, right?)
* make config, hit enter through the whole thing
* make dep
Go about compiling *. modversions.h is generated by make dep.
Daryl
-Original Message-
From: Phillip
them on a
patch panel and they usually have nice plastic guides that
keep your fingers away from the terminals.
[...]
Otherwise known as a 110 block.
Daryl G. Jurbala
Introspect.net Consulting
Tel: +1 215 825 8401
Fax: +1 508 526 8500
http://www.introspect.net
PGP Key and Adobe Digital
This is a known problem. I have the same situation with RH9 as you do.
I don't know if the problem has been added to the new bug tracking
system. We should check.
My workaround is to run the AGI scripts on a RH7 box and forward calls
using IAX.
Scott Stingel wrote:
Hi-
Asterisk (CVS
Brad's recent list of enhancements look good, but I haven't looked
at the code yet. If the code looks good, I hope it will be committed
to the project CVS.
Here's a partial list of enhancements that I would like to see in
Comedian Mail. I am probably interested in helping to fund the
enhancement
I'm trying to debug a problem with robbed bit signalling on a T1
coming into an Asterisk box on a T100P card. Specifically, I need
to look at the signalling timing. Is there a way to turn on this
kind of debugging in Asterisk, similar to what 'pri debug' does?
911 trunks are usually delivered to public-safety answering points (PSAP) on
analog reverse-battery facilities. (The PSAP provides battery toward the CO).
ANI is provided using MF tones. The PSAP equipment must take the ANI and use it
to submit a database query to lookup the caller's address (ALI
I am trying to make an in/out trunk group comprised of 4 DS0's using
EM Wink signalling. The first four channels of a DS1 on a T100P
are being used for the group. Outbound calls work fine, but inbound
calls fail. The other 20 DS0 channels are used for a PRI. Does the
configuration shown below
EM are in the robbed bit.
On Sat, 2003-07-05 at 13:57, Daryl Jones wrote:
I am trying to make an in/out trunk group comprised of 4 DS0's using
EM Wink signalling. The first four channels of a DS1 on a T100P
are being used for the group. Outbound calls work fine, but inbound
calls fail
Yes, but I have been able to mitigate it by setting the following
parameters. I have the problem with ATA's that are behind firewalls
and not, but mostly with the ones that are behind firewalls.
CfgInterval:1800
SIPRegInterval:100
On Thu, 3 Jul 2003, Kim C. Callis wrote:
Is it just me or do
Should Asterisk run under it's own user id, or the web server user id,
or root, or what?
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I started using Festival for the first time today and am having a problem
with zombies left behind after every time that it speaks. I'm using
Festival 1.4.3 with today's CVS of Asterisk. Everything seems to work.
The only obvious problem is that a defunct process is left behind every
call to
I'm running a pretty successful Asterisk system and recently moved our
PRI to a T100P board. The PRI was previously connected to a Cisco 2600
that was serving as a voice gateway. We are having a frequent problem with
inbound and outbound calls being disconnected shortly after they are
answered
This doesn't work for me. Voicemail says the extension number but
does not play the user's name. (Asterisk CVS-04/30/03-22:57:49)
On Tue, 17 Jun 2003, Benjamin Miller wrote:
When in voicemail they need to go into the record name section and
record their name. Then it will play their name.
Is there a way to configure voicemail to do reminder paging? I would like
to configure some voicemail boxes to send an e-mail message to a pager
every 10 minutes until the message is retrieved.
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I experienced the exact same symptoms but didn't have the confidence
to post my experience to this list because of my lack of experience with
Asterisk. I restored the June 1 version from CVS and the problem went away.
There's definitely a problem in code since June 1.
On Sat, 7 Jun 2003, John
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