cvs update your libpri
regards
Martin
On Mon, 31 Mar 2003, Alex Zarubin wrote:
Hi,
There are several channels on the PRI span with the periodic warning:
WARNING[9226]: File chan_zap.c, Line 5437 (pri_dchannel): Ring requested on
channel 21 already in use on span 1. Hanging up owner.
1.
You must be using some old code. Try to use code from CVS. Instructions
are on www.digium.com
regards
Martin
On Fri, 28 Mar 2003, [ISO-8859-7] ÓôáìÜôçò ÊåêÝò wrote:
Hello everybody.
I have a test box with asterisk and till now I have successfuly made it
work with iax.
I'm trying to load
I would like to known if these T2 links are related to the E1
stuff that everybody talk about on this forum. In other words, can I
If you're in Europe than your T2 are 99.9% E1's (30 voice channels + 1
signalling)
link the free T2 card of the Bosch to a Linux box with an E100P
interface and
Will to ports on this card be able to act as FXO as well, or just as FXS?
Maybe later.
But there was some posting about FXS to FXO converter a few weeks before
???
If the answer is yes, can we control which ports do which in any
combination?
Why not ?
Finally, can this card coexist with
The same as you go over the number of PRI channels ?
regards
Martin
On Thu, 27 Mar 2003, James O. Sizemore III wrote:
Quick question what happens if you go over
your channel licenses?
Mark Spencer wrote:
So it looks like the best codec is the GSM codec as far and badwidth
vs voice
It's there
On Mon, 24 Mar 2003, Darrell Eldridge wrote:
Is the macro functionality (described in the draft
handbook Version 2, Section 4.3.11 Using Macros)
already available? I'm having trouble making it work
and wonder if I need to keep trying or wait until it's
in the code.
Greetings Asterisk users.
When I launch Asterisk, I get the following
Asterisk CVS-03/20/03-16:56:24, Copyright (C) 1999-2001 Linux Support
Services, Inc.
Written by Mark Spencer [EMAIL PROTECTED]
=
[
Sure. You configure it as HDLC or CISCO HDLC and you have
hdlc0 interface to send data.
regards
Martin
On Thu, 20 Mar 2003, David Luyens wrote:
Hi,
I would like to use * as a compression box.
Between 2 sites I have an E1 leased line.
So would it be possible to use 1 port of an E400P card
.
regards
Martin
On Thu, 20 Mar 2003, David Luyens wrote:
Thanks Martin, could you point me into the direction on how to do this?
David
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Martin Pycko
Sent: Thursday, March 20, 2003 7:10 PM
To: [EMAIL
You have to add
immediate=yes
to zapata.conf to the declaration of this channel.
Then right after someone picks up the phone asterisk will just
right to 's' extension of the specified context.
regards
Martin
On Tue, 18 Mar 2003, Don Pobanz wrote:
We have need of a ringdown circuit in an
It's fixed now.
what's the intended behaviour of ${variable:a:b}?
it's the same as substring application
given that ${exten} = 501234
until yesterday ${exten:2} would give '1234'
and it does now
with current CVS ${exten:2} is '50' while ${exten:2:4} is '1234'
how do I just strip
Note that the message comes from chan_iax2.c that is under developement.
It uses iax.conf as well as chan_iax.c
regards
Martin
On Sat, 15 Mar 2003, John Vozza wrote:
Thanks to all who set me straight on the codec format stuff...
I have a remote asterisk system running on my laptop which
Of courese:
exten = 9998,1,Dial,SIP/9998|30|tTm
Notice when you don't use the timeout you do have to use the options
separator | like this:
exten = 9998,1,Dial,SIP/9998||tTm
but I think that T is not yet implemented
regards
Martin
On Fri, 14 Mar 2003, WipeOut . wrote:
Thanks the 'show
The formats that asterisk uses are #define'd in
asterisk/include/asterisk/frame.h
RTP formats are #define'd in asterisk/rtp.c
regards
Martin
On Fri, 14 Mar 2003, John Vozza wrote:
I've been trying to find a list of codec format numbers so I can more
clearly understand the following message;
IAX is short and I like it. Besides if that additional '2' irritates you
then anyways in the near future when IAX2 is working fine ppl will switch
eventually to IAX2 and then we'll refer to IAX2 as IAX
Martin
On Thu, 13 Mar 2003, Mark Spencer wrote:
What do you all think of renaming IAX2
or Packet Telephony (Simple) Protocol
On 13 Mar 2003, Karl Putland wrote:
What about ITP
Internet/IP
Telephony
Protocol
On Thu, 2003-03-13 at 09:40, Mark Spencer wrote:
LIghtweight
Voice over IP
Exchange
Or:
Lightweight
Internet
Voice
Exchange
Mark
It's dynamically changed to Yes when the fax gets detected on this
channel.
regards
Martin
On Wed, 12 Mar 2003, Darrell Eldridge wrote:
I still haven't been able to get fax detection going,
but I came across something: when I execute zap show
channel 47 one of the parameters shown is Fax
You may try to add
relaxdtmf=yes
just before channel = 4 in zapata.conf
regards
Martin
On Wed, 12 Mar 2003, Brian J. Schrock wrote:
I am using background, the pbx-invalid stuff should (if DTMF
recognition is working correctly) not get played.
On Wednesday, March 12, 2003, at 01:30 PM,
you put some definitions in
[globals] section in extensions.conf
later you just use SetGlobalVar variable to change the
values of global variables and then you just take
the value of a variable like this: ${variable} or
like this ${${variable}} or like this ${extension_${EXTEN}} etc.
a trivial
put transfer=yes in the begining of zapata.conf
after [channels]
regards
Martin
On Mon, 10 Mar 2003, Mike Reiling wrote:
Did that... Doesn't seem to help
On Monday, March 10, 2003, at 09:49 AM, James Sharp wrote:
parkext = #700; What ext. to dial to
Do you have a proper zaptel.conf and zapata.conf ?
When you modprobe do you have anny errors ?
What does ztcfg -vv says ?
regards
Martin
On Mon, 10 Mar 2003, Brian J. Schrock wrote:
Howdy,
I just added a second USB converter from Digium and I am having a
problem. When I modprobe the driver
I think UHCI
Martin
On Mon, 10 Mar 2003, Brian J. Schrock wrote:
UHCI or OHCI?
On Monday, March 10, 2003, at 04:23 PM, Martin Pycko wrote:
We have some feedback from our customers that sometimes
they are able to run two S100U's on a signle machine.
regards
Martin
On Mon, 10
Do you have tos=lowdelay in iax.conf ?
You may also try to turn off the jitterbuffer (jitterbuffer=no).
Also make sure that asterisk is really using gsm codec.
WHen you do iax show channels in the format column it should
show number '2' = GSM.
Also when you look at Makefiles make sure that
out of asterisk is far away from the limits I would let to
go out to customers:-( )
M
On 8 Mar 2003, William X Walsh wrote:
He's using H.323 not iax
On Sat, 2003-03-08 at 09:58, Martin Pycko wrote:
Do you have tos=lowdelay in iax.conf ?
You may also try to turn off the jitterbuffer
pbx/pbx_spool.c
On Wed, 5 Mar 2003, Rattana BIV wrote:
Hi,
I wanted to know in which code source the file sample.call is proceeded when
we put it in /var/spool/asterisk/outgoing/
I try to make an application to asterisk who check when an user in H323
(netmeeting) is connect or not.
qualify=1000 in sip.conf in the phone config entry
regards
Martin
On Wed, 5 Mar 2003, Mark Spencer wrote:
But if I close my sip phone and a call goes through it will still wait
the 25 seconds before it goes to voice mail even though my Sip phone is
not even on. If I restart Asterisk and
But you can connect several asterisk boxes as one system.
regards
Martin
On Tue, 4 Mar 2003, Sphyrna wrote:
NO, THE ASTERISK HAS A PRATICAL LIMIT OF 8 E1S CURRENTLY. THE I/O ERRORS
STOP EVERYTHING
- Original Message -
From: Florian Overkamp [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
I need a system that can:
- Accept phone calls and give a greeting
That's basically playing back a voice file
- Allow users to select to either leave a message for callback or browse
FAQs
IVR menu
- Present a multi-level voice menu a user can use to drill down to find
their answers in
, but if this is the case,
then perhaps digital is the way to go.
Thanks,
Tyrone
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Martin Pycko
Sent: February 28, 2003 9:39 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Asterisk/IVR Newbie
The only problem
Yes, you can:
asterisk -vvvgcn|tee /tmp/log
regards
Martin
On Thu, 27 Feb 2003, Roy Sigurd Karlsbakk wrote:
hi
can I log all console output while having console access as with
asterisk -vvvgc
?
--
Roy Sigurd Karlsbakk, Datavaktmester
ProntoTV AS - http://www.pronto.tv/
Tel: +47 9801
You can also do that using Background application:
[transfer]
exten = s,1,Background,some-file ;it can be silence
exten = _XXX,1,Flash ;collecting the digits
exten = _XXX,2,SendDTMF,${EXTEN}
exten = _XXX,3,Hangup
[called_context]
exten =
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