Displaying 20 results from an estimated 400 matches similar to: "Account code on SIP"
2003 Feb 21
0
I4l outgoing dtmf problem.
Hi.
I'm working with i4l with asterisk CVS-02/21/03-13:59:12,
plus i4l (chan modem i4l *dsp patched* and kernel 2.4.19
patched to disable dtmf).
All seems ok (apart some echo issues that seems gone
with mec2 aggressive suppressor), but outgoing dtmf
doesn't work . or at least I hear the very first part
of the dtmf, but then it seems suppressed.
here's my modem.conf
[interfaces]
2003 Mar 03
0
Asterisk log rotation
Hi.
Has anyone provided an easy way to rotate
asterisk log files into /var/log/asterisk.
I want to do that, because I prefer
to have full logging enabled in the debug file
and the messages file, but could became pretty
big. Same apply for cdr-csv files.
I wanted to setup a logrotate rule, but was
thinking if I must use a kill -HUP to asterisk.
(never tried HUP with asterisk... don't know
if
2003 Apr 02
0
Zap flash bug?
Hi.
I'm experiencing that bug with flash on zaptel.
That's the problem:
Zap/A call Zap/B
Zap/B flash transfers to Zap/C
Now Zap/A is online with Zap/C
Till now all ok...
but now if Zap/C wants to transfer again,
it can't... the debug says that it got a
WinkFlash when call not up or ringing
(as attached below, Zap/10 is Zap/C in my example)
Apr 2 09:14:01 DEBUG[32789]: File
2003 Apr 11
1
Strange Sip problem?
Hi.
I'm getting a strange sip issue, with
latest cvs. I was tring the *8 extension
for call pickup on sip, but I forget
to define the callgroup & pickupgroup
in sip.conf . Now when I dial *8 from
the crisco phone and hangup, the channel
in asterisk don't go down and I'm not able
to dial from the phone again.
If I do a softhangup on the rem. console
it does nothing and the
2004 Sep 22
1
Status of conference calls at Astricon ?
On late august, there was a thread about
setting up some meetme conferences to
be able to follow Astricon remotely.
This indeed could be nice for those
that can't attend for various reason.
And of course is a demonstration of
Asterisk capabilities... :)
(Astricon without a remote conference
for guest is like a big it expo without
internet connections...)
I have some bandwidth here, so can
2004 Jul 30
5
Non standard usage of X100P card.
I have two X100P card in my box. I want to connect regular phone (not the
phone line!) to one of thse cards. Does anybody think about the same?
I don't really want an expensive solution buying additional card with FXS
port, I prefer to make something by myself. It'll be great if somebody can
point me to technical materials or show electric scheme of such converter. I
believe it should
2004 Jul 26
6
Can't dial SIP<->EuroISDN (HFC-S based PCI ISDN card): Unable to create channel of type 'Zap'
Hi,
I'm trying to set up an Asterisk pbx based on a Fedora Core 1 Linux box
(customized kernel version 2.4.24). I want calls from my SIP soft-phones
to simply be dumped onto the PSTN line via a BRI (EuroISDN). I have a cheap
HFC-S based PCI ISDN card connected to the NT1+ interface, so I need zaphfc.
I've read everything I've found at www.voip-info.org, then I've downloaded
the
2005 May 16
2
a problem sourcing a file using chdir=TRUE
Dear R-users,
I used to give commands such as:
> source(file="~/path/to/file.R", chdir=TRUE)
but with the latest v. 2.1.0 it does not seem to work anymore.
I tried to figure out what it was going on and it seems that the string
for which
> class(file)
[1] "character"
is changed to
> class(file)
[1] "file" "connection"
when the connection is
2003 Nov 19
1
Installing RXlisp
Dear R users,
I was trying to install the package RXLisp by Duncan Temple Lang on a MDK
9.1 Linux machine running R 1.8.0 installed from a RPM.
Unfortunately I had a problem loading the shared library into R. Since
I'm a Linux newbie I was not able to solve the problem. Maybe some of
you can help me.
First of all I downloaded the source archive for Xlisp-Stat and
for the RXlisp package.
2003 Feb 10
2
problems using lqs()
Dear List-members,
I found a strange behaviour in the lqs function.
Suppose I have the following data:
y <- c(7.6, 7.7, 4.3, 5.9, 5.0, 6.5, 8.3, 8.2, 13.2, 12.6, 10.4, 10.8,
13.1, 12.3, 10.4, 10.5, 7.7, 9.5, 12.0, 12.6, 13.6, 14.1, 13.5, 11.5,
12.0, 13.0, 14.1, 15.1)
x1 <- c(8.2, 7.6,, 4.6, 4.3, 5.9, 5.0, 6.5, 8.3, 10.1, 13.2, 12.6, 10.4,
10.8, 13.1, 13.3, 10.4, 10.5, 7.7, 10.0, 12.0,
2003 Sep 28
0
Asterisk CVS viewer on line
Hi to all.
I've put on line a cvs viewer for asterisk source code.
Is based onto the suite horde+chora.
The website is http://asterisk.espia-net.net
The cvs modules shown are
* asterisk
* asterisk-addons
* zaptel
* zapata
* libpri
* libr2
* libiax
* libiax2
* gnophone
* phpconfig
* gastman
all revisions, branch , comments & whatever cvs is has been
preserved. this could be a sort of
2003 Oct 19
0
Flastman 0.0.1-pre-alpha
Hi.
My first 'snapshot' of flastman is out.
Flastman stands for FLash ASTerisk MANager.
written in flash, this first version is just
a proof of concept, ie doesn't nothing except
for logging in/out & displaying manager events
while logged in.
But is realtime & in any flash-enabled browser.
Not very useful yet, but I'm going to improve it.
For the hardcore testers, grab it
2003 Nov 24
0
Picking an open channel (FXO port) for outbo und calls
Thanks to everyone for your quick responses to this question. I'm very
excited about the Asterisk project, and the growing community seems to be
very active these days. Hopefully when the time comes for our county's
transition to VoIP we may be able to go for an Asterisk-based solution.
--
Tony Kava
Network Administrator
Pottawattamie County, Iowa
-----Original Message-----
From:
2004 Jan 31
1
asterisk php status viewer
since I was annoyed this morning, I
wrote this simple php script to output
channel status from asterisk manager.
<disclaimer>
that's very bad written, nor commented...
I wrote that just for fun
</disclaimer>
and if someone will use that / improve
it , just lemme know.
http://asterisk.espia-net.net
(wrote with php 4.3.3 and depends
on Event: StatusComplete, so a recent
* cvs
2003 Apr 08
1
Wiki for the * community.
Hi 2 all.
I was thinking to start a little web site with phpwiki,
to let the * community build a sort of shared
documentation 'bout * & related.
That because in a wiki "place" all grows faster,
and is also the right place to share experiences.
For example it's right to have documentation
about * installations, ie who has done what with asterisk
Till now we don't know
2003 Apr 24
0
TDM40B Heating ?
Hi.
I was testing my brand new TDM40B
and noticed that when the kernel module
is loaded (so the card is powered on),
the chip onto the ringer module (the one
below the S/N label) is pretty hot
(I can't touch it for more than 2 seconds).
Is that normal ?
Or too much hot could be dangerous ?
Matteo
Matteo Brancaleoni
Espia - Emmegi Sys Admin
http://www.espia.it
2003 Nov 14
0
Hidden bug in *8 call pickup with Sip
Hi.
Seems that there's a hidden bug in sip call pickup
with *8.
If I pickup a Sip ringing phone from another sip phone
on the very first ring (before the first ring completes),
the ringing phone doesn't stop ringing, however the call
is connected ok.
But, if I pickup the call after the first ring,
(after the first, the second, or during the second,
or other silly combinations always
2004 Jan 20
0
[A-bit-OT] Power Over Ethernet Discovery process
Hi,
Since someone asked, here's how POE standard does discovery
process for a POE device. of course is a passive detection...
but that's why you don't have POE always-on on a POE enabled
switch port....
you can find more info in article area of
http://www.poweroverethernet.com
and full specs @ http://www.ieee802.org/3/af/index.html
You will find a resistance value in the quote
2004 Apr 26
0
Re: [Asterisk-cvs] asterisk BUGS,1.7.2.1,1.7.2.2
Hi, about select vs poll differences, like in cvs message below
<snip>
>
> Modified Files:
> Tag: v1-0_stable
> BUGS
> Log Message:
> Update 1.0 BUGS file to be aware of select vs. poll stuff
<snip>
> +* The number of channels that can be simultaneously run through Asterisk
> + may be more limited than in development releases due to the use of
> +
2004 May 28
0
E1 channel bank problem
Hi all.
I have and E1 channel bank from Loop Telecom.
there's a little issue with it, I cannot ring
the phones on fxs interface, but can connect
without issue them.
What happens:
I dial the phone on port 1, asterisk says
"Zap/1 is ringing", but the phone on the
analog port doesn't ring. but if I take
off hook the ringed phone, asterisk detects
the answer at they're bridged