--- tony banks <[EMAIL PROTECTED]> wrote:
> Hello
>
> I found related question on vmail.cgi in the mailing
> list but that didn't answer my question. I did copy
> the vmail.cgi to /var/www/cgi-bin/ but still gets
> the following error message when I access
> http://XXX.XX.XX.XXX/cgi-bin/vmail.cg
sip.conf you can set autocreatepeer=yes in sip.conf
and anyone can
place calls
through the system.
-John
--- "Olle E. Johansson" <[EMAIL PROTECTED]> wrote:
> jerk face wrote:
> > In sip.conf I have the following:
> >
> > context=OUTGOING
>
AIL PROTECTED]> wrote:
> how did u setup your asterisk for this:
>
> "I can also start a call through Asterisk to a VoIP
> provider, but there is a problem after the first
> ring:"
>
>
> - Original Message -
> From: "jerk face" <[EMAIL
First off, here is what I want to do:
SIP Clients -> SER -> Asterisk -> VoIP provider
Where SER will handle communications between SIP
clients (since I would prefer that my SIP clients not
use all of my bandwidth)
Asterisk will handle calls to a VoIP provider
I have read that people have similar
Hey,
--- Philipp von Klitzing
<[EMAIL PROTECTED]> wrote:
> Hi!
>
> > I am getting the following error message:
> > Got SIP response 403 "That is ugly -- use From=id
> next
> > time (OB)" back from 195.37.77.101
> >
> > I'm not quite sure what that means. Does anybody
> know
> > what I might ha
I am trying to make an outgoing call using an iptel
account using Asterisk. I have followed a how-to for
asterisk and iptel found at
http://www.voip-info.org/tiki-index.php?page=Asterisk%20sip%20client%20SER
I am getting the following error message:
Got SIP response 403 "That is ugly -- use From=
The following is from zapata.conf.sample:
; Ring groups (a.k.a. call groups) and pickup groups.
If a phone is ringing
; and it is a member of a group which is one of your
pickup groups, then
; you can answer it by picking up and dialing *8#.
For simple offices, just
; make these both the same
;
--- "Evan P. Hall" <[EMAIL PROTECTED]> wrote:
> -Original Message-
> From: jerk face [mailto:[EMAIL PROTECTED]
> Sent: Thursday, December 04, 2003 9:02 AM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] vmail.cgi with Redhat 9.0
>
>
> >
ED]
> [mailto:[EMAIL PROTECTED] On
> Behalf Of Olle E.
> Johansson
> Sent: Thursday, December 04, 2003 1:36 PM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] vmail.cgi with Redhat
> 9.0
>
> jerk face wrote:
>
> > I recently switched from Mandrake to R
--- Lists <[EMAIL PROTECTED]> wrote:
> On Thu, 4 Dec 2003, Olle E. Johansson wrote:
>
> > jerk face wrote:
> >
> > > I recently switched from Mandrake to Redhat and
> I
> > > noticed that vmail.cgi does not work with the
> default
> > >
--- "Olle E. Johansson" <[EMAIL PROTECTED]> wrote:
> jerk face wrote:
>
> > I recently switched from Mandrake to Redhat and I
> > noticed that vmail.cgi does not work with the
> default
> > apache installation that comes with Redhat.
> > Here is
I recently switched from Mandrake to Redhat and I
noticed that vmail.cgi does not work with the default
apache installation that comes with Redhat.
Here is what I get in my error logs:
[Thu Dec 04 11:59:57 2003] [notice] suEXEC mechanism
enabled (wrapper: /usr/sbin/suexec)
[Thu Dec 04 11:59:58 200
ED]> wrote:
> - Original Message -
> From: "jerk face" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Monday, November 17, 2003 3:53 PM
> Subject: [Asterisk-Users] SIP calls no longer work
>
>
> > Hello,
> > I'm having
Hello,
I'm having a problem with SIP. More specifically, I
can't make any calls using SIP.
I have had an iConnectHere account and Free World
Dialup account working for quite some time, and now
all of a sudden I can't make any SIP outgoing calls.
PBX*CLI> sip show registry
Host Us
In the source file for ADSI I found this:
static struct adsi_key_cmd kcmds[] = {
{ "SENDDTMF", 0, send_dtmf },
/* Encoded DTMF would go here */
{ "ONHOOK", 0x81 },
{ "OFFHOOK", 0x82 },
{ "FLASH", 0x83 },
{ "WAITDIALTONE", 0x84 },
/* Send line
I am trying to set up an IAX2 trunk between two
servers.
Server A has the following in iax.conf:
[general]
...
[ServerB]
type=friend
trunk=yes
host=dynamic
secret=myPassword
context=myContext
Server B has the following in extensions.conf:
[outgoing]
exten=>_40X,1,Dial,IAX2/ServerB:[EMAIL PROTECTED
notworthy that even if he finds your
> "451" it may still check
> the other server to see if there is anything else
> beginning with 451 since
> that could allow a "matchmore".
>
> Mark
>
> On Fri, 17 Oct 2003, jerk face wrote:
>
> > I h
I have tried all of that.
I have found that the order of includes don't matter
at all. Regardless of where they are placed, I have
to wait for Asterisk to check the other server for the
extension before dialing the local one.
--- Florian Overkamp <[EMAIL PROTECTED]> wrote:
> Hi,
>
> At 06:50 17-
I have two Asterisk servers, one of which uses a
switch statement (Server 2).
On Server 2, the dialplan is as follows:
[provider]
switch...
[default]
include=>provider
exten=>451,1,Dial,Zap/1
...
(No extensions defined for Server 2 are "can_match"
(eg. exten=>_9XX...))
The problem is that when
You need a T1 crossover cable.
You can find a pin diagram here:
http://www.gcom.com/home/support/t1crossover.html
--- Jose Quinteiro <[EMAIL PROTECTED]> wrote:
> Hello,
>
> So all the pieces are finally here, and I'm ready to
> play. I remember
> reading on this list that the connection Chann
I have two Asterisk servers:
Box 1: Static IP address
Box 2: Dynamic IP address
I have Box 2 registered to Box 1 along with a switch
statement.
I can make calls from Box 2 to Box 1, but I am
wondering how can I make calls from Box 1 to Box 2?
Do I need to register Box 1 to Box 2, or is there a
si
no interface cable and I am unable
> to find the pinout.
> Even TAC couldn't help me :(
>
>
> Jeremy McNamara
>
>
>
> jerk face wrote:
>
> >I am looking into the possibility of buying a Cisco
> >7960 with a 7914 expansion module. I know a lot
I am looking into the possibility of buying a Cisco
7960 with a 7914 expansion module. I know a lot of
people are using the 7960, but I haven't read much
about the 7914 and I was wondering if anybody has used
this module with Asterisk?
-- Thank you for your time
_
Has anybody else out there had a problem with
transfers not being detected?
Occasionally I will want to transfer somebody, so I'll
hit the # key and instead of the Transfer application
starting, the # tone is played.
My hardware is T100P connected to an Adtran TA 750.
I have relaxdtmf=yes in zapa
Quicktime will play .gsm files.
--- Dante Alzamora <[EMAIL PROTECTED]> wrote:
> Does anyone know if there is a GSM player for
> windows?
>
> Dante
__
Do you Yahoo!?
The New Yahoo! Shopping - with improved product search
http://shopping.yahoo.com
___
I was searching through the app_adsi.c file and found
some events and functions that are not used in the
sample ADSI scripts.
One of these functions is DELAY. I can't get this to
work. Has anybody got this to work?
I'm trying to create a HangUp soft key using the
following code:
KEY "Hangup" IS
Critchfield <[EMAIL PROTECTED]> wrote:
> On Wed, 2003-09-24 at 08:41, jerk face wrote:
> > Ok, here is the real gdb output.
> >
> > This GDB was configured as
> > "i586-mandrake-linux-gnu"...
> > Core was generated by `a
I am running Mandrake 9.1 if that makes a difference.
--- Patrick <[EMAIL PROTECTED]> wrote:
> On Wed, 2003-09-24 at 15:41, jerk face wrote:
> > Ok, here is the real gdb output.
> >
> > This GDB was configured as
> > "i586-mandrake-linux-gnu"...
> &
erisk core.6044, sorry
>
> On Tue, 23 Sep 2003, jerk face wrote:
>
> > I keep getting segmentation faults when I do a
> reload.
> >
> > Here are the core file outputs from gdb:
> > (I have three of them and they produce the same
> > output)
> &
I keep getting segmentation faults when I do a reload.
Here are the core file outputs from gdb:
(I have three of them and they produce the same
output)
(gdb) core core.6044
Core was generated by `asterisk'.
Program terminated with signal 11, Segmentation fault.
#0 0x401519fc in ?? ()
I have no
I was just wondering if there was a way that you could
have two calls on one line and switch between the two
without initiating a threeway conversation?
I would imagine that Flash is the way to do this, but
when I Flash twice, a 3-way call is initiated. If I
turn threeway off, then I can't transfe
I have that line in my iax.conf
--- Rich Adamson <[EMAIL PROTECTED]> wrote:
>
> > Has anybody had a problem registering their IAXtel
> > account?
>
> My account is working fine using the following in
> iax.conf:
> register => username:[EMAIL PROTECTED]
> towards the bottom of the [general] s
Has anybody had a problem registering their IAXtel
account?
I just signed up for an account and followed the
documentation on iaxtel.org and my registration is
always rejected.
When I type "iax show registry", I get the following
output:
Host UsernamePerceived
Ref
I would like to do the following:
"A" calls "B"
"C" calls "A"
"A" hears call waiting beep and flashes the line to
talk to "C"
::Here's where I run into a problem::
"A" hangs up on "C" and immediately returns to a
conversation with "B"
The only way I have got this to work is if "C" hangs
up. Then
he FXS
> setup through the TA750
> console.
>
> -wade
>
> > -Original Message-
> > From: jerk face [mailto:[EMAIL PROTECTED]
> > Sent: Monday, September 08, 2003 4:30 PM
> > To: Asterisk
> > Subject: [Asterisk-Users] Adtran TA750 MWI problem
> &
I recently set up Asterisk with an Adtran TA750. All
is well except the phones do not show the MWI.
I have configured zapata.conf properly, as all phones
will receive a stutter dial tone if there is a message
waiting in it's assigned mailbox.
Does anybody know how I might fix this problem?
Thank
That is possible.
As for what you should look for:
There's a special type of card you need to buy. I'm
not sure exactly what the type is, but you should be
able to find it in the archives or better yet,
somebody else will reply to this and know what I'm
talking about.
--- Peter Pauly <[EMAIL PRO
It's my asterisk.adsi file that I changed to suit my
needs.
I was just looking at the file name and not thinking
while I was typing the email.
--- "Wade J. Weppler" <[EMAIL PROTECTED]> wrote:
> Where is the telantek.adsi file?
>
> > -Original Message
I guess that would be asterisk.adsi for the rest of
you
--- jerk face <[EMAIL PROTECTED]> wrote:
> I am working with the telantek.adsi file, and I was
> wondering how I would create a softkey for Transfer.
>
> I tried making a key definition and using SENDDTMF
> "#&q
I am working with the telantek.adsi file, and I was
wondering how I would create a softkey for Transfer.
I tried making a key definition and using SENDDTMF
"#", but that didn't work. Is there another way I
could do this?
Also, does anybody have any ADSI scripts for use with
Asterisk that they wo
I don't have any instructions for setting up my
soundcard (that was done automatically when I
installed my operating system).
As for oss.conf:
autoanswer=yes
context=whatever context you want to put paging in
As for extensions.conf
...
[context specified in oss.conf]
exten=>1234,1,Dial,console/
I am testing out vmail.cgi
I can listen to my messages, but I can't forward them
to another user.
I get the following error message:
Software error:
Invalid new mailbox
That doesn't tell me much, so I'm hoping that somebody
will be able to help me out.
Thank you for your time.
__
e question:
> >
> >What you mean with "unlocked" ?
> >
> >-----Ursprungliche Nachricht-
> >Von: [EMAIL PROTECTED]
> >[mailto:[EMAIL PROTECTED]
> Auftrag von jerk face
> >Gesendet: Mittwoch, 27. August 2003 18:31
> >An: [EMAIL PROTECTED]
> &
I just received an unlocked ADSI phone and I am
playing with the ADSI script.
I was wondering how I can include Voicemail functions
(Check new messages, Delete message) into the soft
buttons.
I checked in app_voicemail.c and it looks like these
functions have already been programmed.
Is there a v
Oops ... I found out my problem
span=
--- jerk face <[EMAIL PROTECTED]> wrote:
> I am using a crossover cable.
> My channel definitions are:
> fxoks=1-22
> fxsks=23-24
> in zaptel.conf
>
>
> --- Alex Lopez <[EMAIL PROTECTED]> wrote:
> >
> &g
>
> That would stop it from not working,
>
> also make sure that you have a span definition on
> the zaptel.con file.
>
>
>
> Message: 7
> Date: Mon, 25 Aug 2003 12:52:12 -0700 (PDT)
> From: jerk face <[EMAIL PROTECTED]>
> Subject: RE: [Asterisk-User
t; not automatically done
> (like the Adtran Total Access series).
>
> Mind you, you should still have a sync light on the
> T1 card...
>
> -wade
>
> > -Original Message-
> > From: [EMAIL PROTECTED]
> [mailto:asterisk-users-
> > [EMAIL PROTECT
My zapata.conf is located in /etc/asterisk and my
zaptel.conf is located in the /etc directory.
--- "Adams, Gavin" <[EMAIL PROTECTED]> wrote:
> > -Original Message-----
> > From: jerk face [mailto:[EMAIL PROTECTED]
>
>
> > I seem to be having a prob
I have just received a T100P and an Adtran TSU 600 in
the mail.
I seem to be having a problem with the T100P card. So
far I have done the following:
vi zaptel.conf
fxoks=1-22
fxsks=23-24
...
vi zapata.conf
...
signalling=fxo_ks
...
channel => 1-22
...
signalling=fxs_ks
...
channel => 23-24
I th
de) = @_;
print STDERR "Return code for Zap/1-1 is $returncode\n\n";
}
James Golovich <[EMAIL PROTECTED]> wrote:
On Thu, 21 Aug 2003, jerk face wrote:> I'm having some trouble getting the channel status with an AGI script.> > #!/usr/bin/perl> > u
I'm having some trouble getting the channel status with an AGI script.
#!/usr/bin/perl
use Asterisk::AGI;
$AGI = new Asterisk::AGI;
my %input = $AGI->ReadParse();
$AGI->channel_status('Zap/1-1');
I am now stuck, and don't know how to get the return codes:
-1 There is no channel that mat
/usr/src/usr/include/mysql/errmsg.h
The version of MySQL that I'm running is 3.23.57-1
Could you tell me where mysql/errmsg.h is located on your
distribution? We can update the Makefile to look there for that
header.
-Tilghman
_
Prot
You should instead update the appropriate Makefile to change:
-I/usr/local/mysql/include
to
-I/usr/src/usr/include/mysql
or you could just do:
cp -a /usr/src/usr/include/mysql /usr/include/mysql
I did that, and I still get the same error:
[root cdr]# make cdr_mysql
cc -fPIC -I/usr/local/mysql/incl
I updated asterisk this morning "cvs update -dA"
When I try to run Asterisk (asterisk -vvvc), I get the following error:
[cdr_mysql.so]WARNING[16384]: File loader.c, Line 226 (ast_load_resource):
/usr/lib/asterisk/modules/cdr_mysql.so: undefined symbol: mysql_init
WARNING[16384]: File loader.c, Li
near initialization for `mysql_lock')
cdr_mysql.c:40: warning: data definition has no type or storage class
make: *** [cdr_mysql.o] Error 1
Any help is always appreciated.
On Wednesday 13 August 2003 03:24 pm, Jerk Face wrote:
/usr/src/usr/include/mysql/errmsg.h
The version of MySQL that I'm r
I'm trying to compile the cdr_mysql module, but I am receiving error
messages.
I have installed mysql-devel.
Here is the output of make cdr_mysql:
cc -fPIC -I/usr/local/mysql/include -I/usr/include/mysql -c -o
cdr_mysql.o cdr_mysql.c
cdr_mysql.c:30:26: mysql/errmsg.h: No such file or direct
I am trying to compile the cdr_mysql module but I am getting errors. I have
MySQL version 4.0.11a installed on my box (Mandrake 9.1).
As far as MySQL packages, I have installed:
MySQL-4.0
MySQL-client
MySQL-devel
MySQL-common
libmysql
I have the latest CVS source for Asterisk.
When I run make cd
Some of my end users have reported to me that occasionally they'll be in the
middle of a conversation and the call will be dropped. I have yet to catch
anything unusual when debugging the channels.
Has anybody had this problem before, if so, how did you solve it?
My hardware:
2 X100P
1 TDM40B
T
Does the T100P support message waiting on an Adtran TSU 600E with FXS cards
installed?
So basically:
Asterisk w/T100P -> Adtran TSU 600E -> Analog phone
Will I be able to receive the stutter dial tone on the analog phone?
Thank you for your time
___
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