:
Face wrote:
Well, thanks for responding. I went back to 10.10.0 and things seem to
be working fine now!
This is certainly good to know but I'd like to know why upgrading to 11
did not seem to work for you. This is the first case since it's been out
where it doesn't appear to have been
On Mon, Nov 19, 2012 at 3:51 PM, Joshua Colp jc...@digium.com wrote:
Face wrote:
Hello,
Hola,
After Upgrade to Asterisk 11.1.0-rc1 I keep getting
== Using SIP VIDEO TOS bits 136
== Using SIP VIDEO CoS mark 6
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Hello all,
I am running Asterisk 10.10.0 and I can send Message between SIP's no
problem. However, I would like to be able to send send Message to a
SIP from AMI Or CLI. I check the ListCommands On the AMI and it
don't have MessageSend. Therefore, I try the SendText.
AMI:
Action: SendText
On Wed, Nov 14, 2012 at 7:42 PM, Steve Edwards
asterisk@sedwards.com wrote:
On Wed, 14 Nov 2012, Face wrote:
Is there a way I can trigger a AGI script On SIP REGISTER event.
On Wed, 14 Nov 2012, Danny Nicholas wrote:
What you will need to do is to monitor for the SIP REGISTER in AMI
...@lists.digium.com] On Behalf Of Face
Sent: Thursday, November 15, 2012 12:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] On SIP REGISTER event trigger a AGI script
On Wed, Nov 14, 2012 at 7:42 PM, Steve Edwards asterisk@sedwards.com
wrote
Hello,
After Upgrade to Asterisk 11.1.0-rc1 I keep getting
== Using SIP VIDEO TOS bits 136
== Using SIP VIDEO CoS mark 6
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [603@DLPN_AlDimnaDialPlan:601]
Dial(SIP/601-0002, SIP/603) in new stack
[Nov 16
On Wed, Nov 14, 2012 at 7:42 PM, Steve Edwards
asterisk@sedwards.com wrote:
On Wed, 14 Nov 2012, Face wrote:
Is there a way I can trigger a AGI script On SIP REGISTER event.
On Wed, 14 Nov 2012, Danny Nicholas wrote:
What you will need to do is to monitor for the SIP REGISTER in AMI
On Wed, Nov 14, 2012 at 10:14 PM, Steve Edwards
asterisk@sedwards.com wrote:
On Wed, 14 Nov 2012, Ali Pey wrote:
You can also consider using a proxy server such as opensips or Kamailio.
They would enable you to do much more at the signalling level and many other
advantages such as better
Hello All,
Is there a way I can trigger a AGI script On SIP REGISTER event.
--
Any help would be much appreciated.
falazemi
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us
On Wed, Nov 14, 2012 at 2:38 AM, Steve Edwards
asterisk@sedwards.com wrote:
On Wed, 14 Nov 2012, Face wrote:
Is there a way I can trigger a AGI script On SIP REGISTER event.
Well, an AGI runs in the context of a channel. A REGISTER does not.
So, no.
--
Thanks in advance
--- 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.cgi
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
] 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 PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, December 22, 2003 2:42 PM
Subject
in the
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
autocreatepeer=yes
[Provider]
type=friend
username
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
;
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 have done wrong?
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
--- 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 what I get in my error logs:
[Thu Dec 04 11:59:57 2003
--- 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
apache installation that comes with Redhat.
Here is what I get in my error
: [Asterisk-Users] vmail.cgi with Redhat
9.0
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 what I get in my error logs:
[Thu Dec 04 11:59:57 2003] [notice] suEXEC
--- 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
I recently switched from Mandrake to Redhat and I
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 a problem with SIP. More specifically,
I
can't make any calls using SIP.
I have had an iConnectHere
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
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 have tried all of that.
I have found that the order of includes don't
matter
at all
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
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 I
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
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
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 Channel
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
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)
(gdb) core core.6044
Core was generated by `asterisk'.
Program terminated with signal 11
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...
Core was generated by `asterisk'.
Program terminated
this be caused by a configuration error?
--- Steven 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 `asterisk'.
Program terminated
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
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
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
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
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] section.
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 A is connected to
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
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
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?
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-
From: jerk face [mailto:[EMAIL
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]
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
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 mailboxBR
That doesn't tell me much, so I'm hoping that somebody
will be able to help me out.
Thank you for your time.
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
]
[mailto:[EMAIL PROTECTED]
Auftrag von jerk face
Gesendet: Mittwoch, 27. August 2003 18:31
An: [EMAIL PROTECTED]
Betreff: [Asterisk-Users] ADSI Programs
I just received an unlocked ADSI phone and I am
playing with the ADSI script.
I was wondering how I can include Voicemail
functions
,
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-Users] T100P/ TSU 600
installation problem
To: [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]
Each port
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:
What cable are you using, The SU600 to Digium
cards
need
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
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 problem with the T100P card.
So
far I have done
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 PROTECTED] On Behalf Of jerk face
Sent: Monday, August 25, 2003 3:01 PM
To: [EMAIL PROTECTED]
Subject
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
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
: warning: (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
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,
/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
_
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