No, I do want call back. I want the caller to call a number, then hang
up without it being answered. They then get a call-back and a dialtone,
so they are now an extension on the PBX and can make calls.
Danny Nicholas wrote:
As I read this, it's not truly a callback; it's more of a notify;
2 sep 2009 kl. 22.40 skrev Fred Posner:
Here's the story...
Nortel system set to use g711 @ 30ms payload ... Asterisk box would
need to communicate to that box @ 30 ms and another end point at 20
ms.
I've seen discussions of setting this to a different size, but seems
to be limited to
Meetme() is the way to go. Running it on a virtual machine might not
be such a good idea bacause dahdi_dummy, needed for Meetme() might not
run. Google on Meetme() cmd asterisk and check the parameters
available. There is one for listen only mode.
Don't forget to add a conference room to
3 sep 2009 kl. 00.27 skrev John A. Sullivan III:
On Wed, 2009-09-02 at 21:31 +, Ott Rose wrote:
i have posted this before but was unable to resolve it. i have some
new info so i figured i would try again. the trace from bandwidth.com
are below. they are telling me that the ip that is
On 3/09/09 6:24 PM, Chris Mason (Lists) wrote:
No, I do want call back. I want the caller to call a number, then hang
up without it being answered. They then get a call-back and a dialtone,
so they are now an extension on the PBX and can make calls.
His second example will do that for you -
Hellos,
I know this might be an easy one but either way I am stuck...I need to
execute asterisk cli commands using php agi and get the output via the same
script.
How to I execute let's say show hints and get the output back to the
script? I have tried
$agi-exec(show hints);
but I am getting
On 3 Sep 2009, at 08:01, James Mutuku wrote:
I know this might be an easy one but either way I am stuck...I need
to execute asterisk cli commands using php agi and get the output
via the same script.
How to I execute let's say show hints and get the output back to
the script? I have
Hello,
I try to move our asterisk installation (3 Asterisk servers in different
offices connected using IAX and a lot of SIP phones, as well as ISDN
connections using CAPI) to use G.722 instead of G.711.
Asterisk 1.4.25.1 is used with the G.722 patch (the fixed one, which solves
the gain
Thank you for your reply . Do you mean my Asterisk extensions.conf must
contain a line like the followings ?
include = parkedcalls
If so , can you please let me know where I have to put this line in my
extensions.conf ?
Thank you in advance
Regards
H.Motamedi
On Thu, Sep 3, 2009 at 5:26 AM,
Thanks. Is it possible to do the same after Queue command? After
Queue command, hangup will hangup the call and won't go to the next
priority.
On Mon, Aug 17, 2009 at 7:22 PM, Trevor Hammondstre...@concipient.net wrote:
On Monday, August 17, 2009, Rilawich Ango wrote:
Thanks. DIALSTATUS
Hi,
Whenever one of my trunks becomes unreachable or reachable again..
On logs i got the msg as follows:
Jul 31 15:15:51] NOTICE[15112] chan_sip.c: Peer 'voiptrunk' is now
Reachable. (12ms / 2000ms)
[Jul 31 15:15:51] WARNING[15112] res_config_mysql.c: MySQL RealTime:
Failed to query database.
Francesco Peeters wrote:
Francesco Peeters wrote:
Does anybody else see the same behavior for VoipBuster connections?
When I trace one of the other SIP peers, I see it sends this message:
--
--- SIP read from
I have included that but my scripts goes silent at
AGI Rx EXEC Flite Hello 1215, you have dialed 1220.
AGI Tx 200 result=0
Below is my script
#!/usr/bin/php -q
?php
set_time_limit(30);
require('phpagi.php');
error_reporting(E_ALL);
$agi = new AGI();
$asm
Hi David,
Is T.38 Fax supported on both?
I can tell you that I've been having problems with various version of
Cisco IOS and T.38 on asterisk. I had a stable configuration fax-wise,
but I had to upgrade the IOS because of a Cisco bug, and my T.38 has
never been the same since. It's hard to
Hello
Previous context :- After Looking up sip and IAX2 that require
configuration at router level which may cause some problems like connection
break etc... so i left them . and start wondering if there is some
thing that dont require configuration at router layer. The task to
accomplish
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi all!
I'm investigating the possibility of using Asterisk as much for internal
communication in an office as between offices and I would like to know
what considerations you could comment to me being based on the
experience that you have had.
A
Hello,
I have conferences in my database.
I need at some moments, to access the database without asking pin access, or
with using cdr(accountcode).
Is it possible?
Thank you
Cordialement,
BERGANZ François
P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.
On 14:24, Thu 03 Sep 09, ABBAS SHAKEEL wrote:
Hello
Previous context :- After Looking up sip and IAX2 that require
configuration at router level which may cause some problems like connection
break etc... so i left them . and start wondering if there is some
thing that dont require
Sorry, there are some errors, here the right question:
Hello,
I have conferences in my database.
I need at some moments, to access the CONFEERENCE without asking pin access,
or with using cdr(accountcode).
Is it possible?
Thank you
Cordialement,
BERGANZ François
Karl Fife wrote:
Any theories as to why one routine would behave differently than the other
with Echo Cancellation enabled?
In my mind, anything that alters the audio path may cause issues with
DTMF detection. As to why, I'm not qualified to say; I'm not a programmer.
You may want to
I found !
If I need to enter in a conference (without pinacces) which is in the
database (and have a pin access),
Just add ,thepinacces at the end of meetme!
Cordialement,
P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.
De :
Hi,
It seems Asterisk needs to be notified that log rotation happened tough
applications like astmanproxy or FOP doesn't need to be restarted (nor
notified of any rotation).
Is this personal observation true ?
How could this be explained ?
Regards
___
On 3 Sep 2009, at 11:39, Olivier wrote:
It seems Asterisk needs to be notified that log rotation happened
tough applications like astmanproxy or FOP doesn't need to be
restarted (nor notified of any rotation).
Is this personal observation true ?
How could this be explained ?
They don't
On Sep 3, 2009, at 2:34 AM, Olle E. Johansson wrote:
2 sep 2009 kl. 22.40 skrev Fred Posner:
Here's the story...
Nortel system set to use g711 @ 30ms payload ... Asterisk box would
need to communicate to that box @ 30 ms and another end point at 20
ms.
I've seen discussions of setting
I want to do a callback scenario. Each time asterisk receive a call, it creates
a callfile, sends back the hangup signal and dial back the extension.
Here the default CDR logging is enabled.
If a dial attempt is failed then a CDR is generated. How I do a trick to stop
CDR logging for all
Sorry guys.
My bad!
As you can see, the command on prior message is incorret.
I've changed to:
Dial(SIP/${EXTEN}|20|RtTL(30:6:2))
and it's working now.
Thanks and best regards,
Mauro.
Mauro Sergio Ferreira Brasil escreveu:
Hello there!
I'm testing Dial call limit option on
Have your callfile work through a context instead of dialing. The context
can disable CDR.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faheem
Sent: Thursday, September 03, 2009 7:57 AM
To:
On Thursday 03 September 2009 02:47:05 Armin Schindler wrote:
Hello,
I try to move our asterisk installation (3 Asterisk servers in different
offices connected using IAX and a lot of SIP phones, as well as ISDN
connections using CAPI) to use G.722 instead of G.711.
Asterisk 1.4.25.1 is used
Hello.
I've been trying to use the AMI to originate phone calls.
I'm trying to call the SIP phone 'zoiper' with the SIP phone 'yziquel'.
So, the AMI interaction is:
Action: originate
Channel: SIP/zoiper
Exten: yziquel
Priority: 1
Timeout: 30
Context: internal
Response: Error
Message:
Hi everybody
I have a problem and want to know if anyone has already seen it before :
I try to use web-meetme.3.1.0 and follow these instructions
http://sourceforge.net/docman/display_doc.php?docid=48924group_id=164788
1) when i do make command in cbmysql folder, errors happened
Tzafrir Cohen escreveu:
On Tue, Sep 01, 2009 at 08:53:04AM -0500, Danny Nicholas wrote:
This may be dumb and/or obvious, but did you do these steps?
1. dahdi_genconf dahdi modules user to make sure all of the configuration
files are up to standard
this is an R23 connection, so I dont think
No such device is sometimes an indication that /etc/init.d/dahdi start did
not load the driver.
What does /etc/dahdi/modules look like?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joao Gomes
Pereira
Sent:
it looks like this:
tail /etc/dahdi/modules
# Autogenerated by /usr/sbin/dahdi_genconf (Dahdi::Config::Gen::Modules)
on Wed Jun 24 12:41:26 2009
# If you edit this file and execute /usr/sbin/dahdi_genconf again,
# your manual changes will be LOST.
wct4xxp
Danny Nicholas escreveu:
No such
here are my logs when I start the dahdi driver:
/etc/rc.d/init.d/dahdi start
Sep 3 18:02:39 catumbela kernel: Found TE4XXP at base address fdcff000,
remapped to f88a8000
Sep 3 18:02:39 catumbela kernel: TE4XXP version c01a0164, burst OFF
Sep 3 18:02:39 catumbela kernel: FALC version:
Here it is:
[r...@catumbela ~]# lsmod|grep wct4xxp
wct4xxp 242176 0
dahdi 197640 5 wct4xxp
[r...@catumbela ~]#
dmesg is in attach
:)
Danny Nicholas escreveu:
Okay. What is the output of these commands?
dmesg
lsmod|grep wct4xxp
-Original Message-
On Thu, 2009-09-03 at 06:30 -0300, Daniel Bareiro wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi all!
I'm investigating the possibility of using Asterisk as much for internal
communication in an office as between offices and I would like to know
what considerations you could
hi folks.
i have several remote sites with total of 200 sip phones connect
to our Asterisk server. i want to minimize bandwidth usage and
thinking about getting a Digium TC400B transcoder card. what are
your experience with it? how's the quality? also if there are
120 active channels in used.
Hello,
The company I work for recently purchased 2 Rhino CB24s and a Rhino
PCI-E R4T1. The channel banks are plugged into the R4T1, as well as 2
PRIs from our telco. The CB24s are for all internal analog phones.
Most of the phones are setup in batphone mode, which is
immediate=on in the
On 3/09/09 10:39 PM, Olivier wrote:
Hi,
It seems Asterisk needs to be notified that log rotation happened tough
applications like astmanproxy or FOP doesn't need to be restarted (nor
notified of any rotation).
In logrotate we just add a command to be run after rotation to do:
asterisk -rx
On 4/09/09 2:41 AM, Guillaume Yziquel wrote:
Hello.
I've been trying to use the AMI to originate phone calls.
I'm trying to call the SIP phone 'zoiper' with the SIP phone 'yziquel'.
So, the AMI interaction is:
Action: originate
Channel: SIP/zoiper
Exten: yziquel
Priority: 1
Timeout:
On 4/09/09 3:24 AM, harry R wrote:
Hi everybody
I have a problem and want to know if anyone has already seen it before :
I try to use web-meetme.3.1.0 and follow these instructions
First off, (even though I don't understand French) your error is that
ast_config_load in the version of
Hello.
Matt Riddell a écrit :
To start with I'd do (just rearranging but makes me feel better):
Action: originate
Channel: SIP/zoiper
Context: internal
Exten: yziquel
Priority: 1
Timeout: 30
Callerid: yziquel
Thank you for your answer.
But also, are you sure that the extension
Trying to do something like this in the sip.conf under my incoming provider
profiles:
setvar=CDR(accountcode)=${EXTEN}
It seems to show up in the CDR but it's showing up exactly like this
${EXTEN}.
Is there a way to stuff the DNIS (number dialed) into the accountcode for
CDR?
I have already
Todd Routhier wrote:
Trying to do something like this in the sip.conf under my incoming
provider profiles:
setvar=CDR(accountcode)=${EXTEN}
Set(CDR(accountcode)=${EXTEN})
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety,
On 4/09/09 10:41 AM, Doug Lytle wrote:
Todd Routhier wrote:
Trying to do something like this in the sip.conf under my incoming
provider profiles:
setvar=CDR(accountcode)=${EXTEN}
Set(CDR(accountcode)=${EXTEN})
Nah he's trying to do it in sip.conf
Basically what you should do is add the
The Asterisk Development Team has announced the release of Asterisk 1.2.35,
1.4.26.2, 1.6.0.15, and 1.6.1.6. These releases are available for immediate
download at http://downloads.asterisk.org/pub/telephony/asterisk/
These releases have been created in response to an IAX2 denial of service
Asterisk Project Security Advisory - AST-2009-006
++
| Product | Asterisk |
Hi,
I'm having trouble with chan_mobile. mobile search is give me only
headset device even if they are laptop or cellphone. I want to use my
nokia 6630 with asterisk and I cannot understand what is missing. I
already google that problem and read the forum.. but i'm not getting
out of this
Hi All,
I'm receivig DTMF from my provider in RFC2833 but my provider send in Event
Duration the value 0 and when asterisk forward this DTMF to PSTN asterisk play
the DTMF very fast.
Anybody now how fixes this problem ?
Thank You,
Bruno Rodrigues___
yes, callfile work through context. When control is in the dialplan
context/extension/priority, I can enable/disable CDR's. Problem comes when
asterisk dial a call and user is busy or did not answered the call. In this
case a CDR is generated. No CDR should be generated on busy or failed call
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