Hmm... I missed what Kevin and Jeremy caught...
But... you did mention TE410P and T1 circuit from the provider...
So, what exactly is between the TE410P and the T1 circuit... As you have it
configured, it should not work at all
If the far end switch is a DMS100 you should have
Is it possible to install Asterix (and libpri) to prefix like /usr/local/ so
it won't place non-managed-by-my-distro files on /etc, /usr, /var, etc ?
Thanks.
--
José Pablo Ezequiel Fernández
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Look at the thread Optimum online-upload throttling
confirmed.
It seems like throttling is done by all Cable
companies and that might affect the VoIP performance,
specially for uploading. Try when the activity in the
cable line is low (i.e. late) and see if it gets
better, or try sending all the
Well, mine started to work for a while and down again..I give up ;(
But the good news is.. I just got a DID from teliax... ;)
The part that really bother me was the recording when the number was
called two days ago the number you have call has been disconnected..
The fast busy, not
On Fri, Aug 19, 2005 at 06:35:08PM -0300, José Pablo Ezequiel Fernández wrote:
Is it possible to install Asterix (and libpri) to prefix like /usr/local/ so
it won't place non-managed-by-my-distro files on /etc, /usr, /var, etc ?
Thanks.
PREFIX will probably apply to the installed files but
On Friday 19 August 2005 18:52, Tzafrir Cohen wrote:
On Fri, Aug 19, 2005 at 06:35:08PM -0300, José Pablo Ezequiel Fernández
wrote:
Is it possible to install Asterix (and libpri) to prefix like /usr/local/
so it won't place non-managed-by-my-distro files on /etc, /usr, /var, etc
? Thanks.
Derek/Jeremy/Kevin,
Thank you all for your comments. I suspected the issue would be the
fact that we don't have a PRI but a T1. However, I decided to post
the question to the list simply because I would assume that because
the carrier is looking at our circuit configuration when answering
See comments inline!
Damon Estep wrote:
I have officially engaged in a pissing contest with the local Telco over
PRI calling name delivery.
Welcome to my world, I deal with theses guys daily! Errgiant arn't
they. We have a saying around work 'The telco is always wrong!'.
The telco
but the non head version is not working with realtime configuration?
hm, i think its a problem with app_expr.c but i will try now to copy the
app_expr.c from cvs-version
i will let you know
Nico
- Original Message -
From: Trey Scarborough [EMAIL PROTECTED]
To: Asterisk Users Mailing
I think the title more or less says it all.
Is there any such animal?
TIA
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you need a sip-provider?
- Original Message -
From: Bruce Ferrell [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Saturday, August 20, 2005 12:38 AM
Subject: [Asterisk-Users] [OT] Looking for Web based SIP endpoint
I
Hi:
I hope that someone can help with this problem that came up suddenly. I get
the following error with a system that was working well with a TDM400 as a
TDM31B (3FXS [ch1-3]; 1FXO [ch4]), and now it has this error. Note the
following command sequences which follow - modprobes then ztcfg
PRI is one of many signaling formats available on a T1 circuit. T1 and
PRI are not the same thing. Have your carrier change the signaling
format of your T1 to PRI. PRI is 23 B Channels and 1 D Channel
(Signaling) and is an end office protocol. It has many of the features
of a full blown SS7
Nico Giefing wrote:
you need a sip-provider?
- Original Message -
From: Bruce Ferrell [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Saturday, August 20, 2005 12:38 AM
Subject: [Asterisk-Users] [OT] Looking for Web
No, I need an endpoint I can put on a webpage
if you are looking for a web based sip user agent there is sip
communicator which can be loaded using java webstart.
look at https://sip-communicator.dev.java.net/
the jnlp is here:
Bruce Ferrell wrote:
Nico Giefing wrote:
you need a sip-provider?
- Original Message - From: Bruce Ferrell
[EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Saturday, August 20, 2005 12:38 AM
Subject:
I'm a newbie to Asterisk, but I'm moderately knowledgeable about phone
systems. Right now, I'm most certainly confused.
I have a TDM-04B (four FXO) and four analog FXO lines running into it
from an AdTran 616. I have Asterisk working internally, although I could
use some help getting incoming
Okay, first a little background - I've been with Packet8 since a month
after they started. I found that we were outgrowing their services
and decided to move to an asterisk box in the office. I found a
service provider that offered me a reasonable rate. After a fair
ammount of testing I decided
On Friday 19 August 2005 21:27, Kevin Walsh wrote:
I'll send the modified Makefiles to anyone who needs them.
May I humbly request they be attached to a feature request on Mantis?
-A.
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Of course, after spending a full 12 hours bashing my head against the
problem, I found my low voltage contractor had reversed the tip and the
ring. *bashes head against the wiring cabinet*...
-K
Karl S. Katzke wrote:
I'm a newbie to Asterisk, but I'm moderately knowledgeable about phone
Karl S. Katzke wrote on Friday, 19 August 2005 6:30 PM:
Of course, after spending a full 12 hours bashing my head against the
problem, I found my low voltage contractor had reversed the tip and
the ring. *bashes head against the wiring cabinet*...
I feel your pain! :-)
Hi All,
I recently signed up with a VOIP provider that
supports SIP. By default the provider supports X-term.
i downloaded X-lite configured as mentioned in their
site and got it working.
These are the details for configuring x-lite
Enter the
Andrew Kohlsmith [EMAIL PROTECTED] wrote:
On Friday 19 August 2005 21:27, Kevin Walsh wrote:
I'll send the modified Makefiles to anyone who needs them.
May I humbly request they be attached to a feature request on Mantis?
I've been less than humbly requested not to do that sort of thing
Are there any compactPCI boards that work with Asterisk?
Brent
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HI
I JUST IMPLEMENTED ASTERISK AND IS SUCH A GRATE SOLUTION...I AM USING IT
POLYCOM 301 AND 501 PHONESON LAN A IAM USING G.711 AND I HAVE A 16 PORT
LINKSYS SWITCH...
THE PROBLEM IS WHEN SOMEBODY INSIDE THE NETWORK IS MAKING A CALL TO OTHER
EXTENSION (IN THE SAME NETWORK) AND FOR EXAMPLE IS
You could always post it to the wiki... No disclaimer required for posting
to the wiki...
- Original Message -
From: Kevin Walsh [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, August 19, 2005 7:57 PM
Subject:
Well hey! let me know!!! :-)
I got my max232 chip sitting out and am building a converter board right
now...
Gonna give it a shot soon as I get yer info!!! :-)
Have you done successful re-blast on one of these before?
Very familiar (well kinda) with Motorola vxworks surboards etc.
Take care!
Unfortunately, this happens all too often with LNP.
What has happened is that the numbers have been released by the old carrier
(they send an unlock request and then a delete routing request to the
routing database system - these may be refered to by different terms by
different people... i'm
Steve Gladden wrote:
Well hey! let me know!!! :-)
I got my max232 chip sitting out and am building a converter board right
now...
Gonna give it a shot soon as I get yer info!!! :-)
Have you done successful re-blast on one of these before?
Very familiar (well kinda) with Motorola vxworks
John Riek wrote:
Does anybody know what would be causing the errors
below?
I get these errors continuously until asterisk finally
quits. This happens when I make 20 simultaneous SIP
calls with the Dial Command.
chan_oss.c:291 sound_thread: Failed to write sound
chan_oss.c:200 send_sound:
[EMAIL PROTECTED] wrote:
Hi:
I hope that someone can help with this problem that came up suddenly. I
Did you upgrade Fedora Core?
Check if the udev files still contain the required entries (normally fedora
copies the old ones to 50-udev-rules.old and makes new ones).
--
Cheers,
Matt
Hi, Matt and Asterisk gurus
I encountered the same problem in my asterisk meetme.
Whenever the 3rd person joins the meeting, it creates echo in the meeting,
while 2 person meeting is fine.
I am wondering if you can give me more hint on how to configure the mixer to
have echo cancelled.
We are
On Fri, Aug 19, 2005 at 07:01:00AM -0600, Damon Estep wrote:
On the same setup, if I connect another PRI device to it that emulates
switch side signaling and includes the CNAM as a Display IE in the
setup, the SIP invite is properly formatted and * receives the calling
party name.
Does
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