How to enable cdr_mysql.conf in Asterisk 1.6?
I have installed asterisk-addons which compiled mysql support,
module show is showing cdr_addon_mysql.so
but cdr_mysql.conf was not created in /asterisk directory
Is there any configuration file to enable mysql support?
Comping cdr_mysql.conf from
http://hostseries.com/asterisk-cdr-logging-in-mysql/
http://www.asterisk.net.au/tutorial/10/
http://www.voip-info.org/wiki/view/Asterisk+cdr+mysql
http://www.spiration.co.uk/post/1327/asterisk-addons%20setting%20up%20mysql%20cdr%20for%20Asterisk
On Fri, Oct 30, 2009 at 11:35 AM, Joseph
Moises Silva skrev:
You mean you cannot see AsyncAGI events? did you enable agi in the
read= parameter in manager.conf for your Java application user?
Yeay!! Thank you! No, I have not. And I suspected that I had to put something
there, I've googled mad for it
but have not found one document
Even though certain things should be discussed in private and that certain
things should require a second thinking before stating them, I don't think you
should impose a limit.
Are you a Digium guy? An asterisk developer? Who are you?
Apart the informational value of these lists I should hope
Hi Bart,
Am Donnerstag, den 29.10.2009, 16:36 -0700 schrieb Bart Fisher:
I don't understand this message:
[2009-10-29 16:31:51] WARNING[28510]: rtp.c:1997 ast_rtp_settos:
Unable to set TOS to 184
You did not tell us, which version of asterisk You are running.
The kernel restricts setting
On Fri, 2009-10-30 at 09:53 +0100, Karsten Wemheuer wrote:
Hi Bart,
Am Donnerstag, den 29.10.2009, 16:36 -0700 schrieb Bart Fisher:
I don't understand this message:
[2009-10-29 16:31:51] WARNING[28510]: rtp.c:1997 ast_rtp_settos:
Unable to set TOS to 184
You did not tell us, which
Thanks CF.
The requirements are
Panasonic TDA will have all the PSTN lines from Telco company. Asterisk Box
will get phone lines from TDA.
Now it works fine when take an Extension from TDA and Put it in Asterisk BOX
(TDM400P).
Asterisk Box recieves call exactly ok but when asterisk box need to
Hi all,
I have tried to configure inbound routes in my elastix box 1.5.2 but i don't
get how to associate it to a Trunk i have created.
Also configuring outbound routes is proving to be a challenge. I have read the
Elastix without tears.pdf guide but seems not much information is provided in
On 30 Oct 2009, at 11:08, PATRICK KANGETHE wrote:
I have tried to configure inbound routes in my elastix box 1.5.2 but
i don't get how to associate it to a Trunk i have created.
Also configuring outbound routes is proving to be a challenge. I
have read the Elastix without tears.pdf guide
Hi,
I have a PRI euroisdn link between an Alcatel PBX and Asterisk.
I'm having some trouble with overlap dialing.
Suppose I dial '874053' from an Alcatel extension ('7034') where '87' is an
Alcatel prefix of type ARS Prof.Trg Grp Seiz.with overlap.
I'm expecting Asterisk to receive '1004053'
Hello
Since SIP/RTP is a pain to use with road warriors who need to connect
from any location over the Internet, I'd like to get them some IAX
phones instead.
For those of you using this protocol instead of SIP, what would you
recommend as IAX hardphones and Windows (and ideally Mac) softphones?
Vincent wrote:
Since SIP/RTP is a pain to use with road warriors who need to connect
from any location over the Internet, I'd like to get them some IAX
phones instead.
What gives you that idea?
--
Alex Balashov - Principal
Evariste Systems
Web : http://www.evaristesys.com/
Tel :
On Fri, 16 Oct 2009 10:08:14 +0300, Stelios Koroneos
skoron...@digital-opsis.com wrote:
I did it with PS3 and Asterisk 1.2 about a year ago
With Yellow Dog linux running on PS3
Was not using any of co-processors though, just the main cpu.
Thanks for the feedback.
I assume you're kidding?!
RTP is mangled/blocked by most hotspots and mid-size company firewalls...
IAX is often the only way our staff can connect while on the road.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On
My experience does not support your conclusions. In my personal
observations of situations in which I have been involved, most
allegations of serious SIP problems related to source NAT (IP
masquerading) are exaggerations stemming from lack of subject matter
comprehension. This is bearing in
overlapdial=yes in zapata.conf/chan_dahdi.conf
google it out
Martin
On Fri, Oct 30, 2009 at 6:54 AM, Vieri rentor...@yahoo.com wrote:
Hi,
I have a PRI euroisdn link between an Alcatel PBX and Asterisk.
I'm having some trouble with overlap dialing.
Suppose I dial '874053' from an Alcatel
Because RTP ports are assigned dynamically (and not necessarily
symmetrically) during call setup using SIP, you need a SIP aware firewall.
Without one, you may get SIP registration, but usually one-way/no audio
(RTP).
Most hotels and hotspots do NOT support SIP - either because they run cheap
On the contrary, SIP-aware ALGs mostly do more harm than good. While
their purpose is noble, their implementation is frequently lacking/
incomplete and conflicts with existing far-end NAT traversal
approaches taken by most service providers these days. These are also
present in commercial
hello all,
I have asterisk 1.4.26 working on CentOS 5.3 and I have the following problem:
- when I restart asterisk all the members of the queue are Invalid.
- when I make a call to one of the members, of the queue, and then
check the state, it turns to Not in use for the called phone,
I forgot to mention that I already have overlapdial=yes in zapata.conf.
Besides, overlapdial=yes is only for dialing out from Asterisk. Anyway, that
option is set.
Any other ideas?
--- On Fri, 10/30/09, Martin asteriskl...@callthem.info wrote:
overlapdial=yes in
On Oct 30, 2009, at 7:55 AM, Vincent wrote:
Hello
Since SIP/RTP is a pain to use with road warriors who need to connect
from any location over the Internet, I'd like to get them some IAX
phones instead.
For those of you using this protocol instead of SIP, what would you
recommend as IAX
On Thursday 29 October 2009 12:32:48 Carlos Chavez wrote:
On Thu, 2009-10-29 at 12:23 -0500, Tilghman Lesher wrote:
On Thursday 29 October 2009 11:49:30 Carlos Chavez wrote:
On Wed, 2009-10-28 at 23:36 -0500, Tilghman Lesher wrote:
On Wednesday 28 October 2009 17:57:49 Carlos Chavez
On Fri, 30 Oct 2009 09:33:02 -0400, John Todd wrote:
On Oct 30, 2009, at 7:55 AM, Vincent wrote:
Hello
Since SIP/RTP is a pain to use with road warriors who need to connect
from any location over the Internet, I'd like to get them some IAX
phones instead.
For those of you using this
To make sure I understand the question:
Problem: when I restart asterisk all the members of the queue are Invalid.
Things I've tried:
A. When I make a call to one of the members, of the queue, and then check
the state, it turns to Not in use for the called phone, and
the queue works fine for
On Thursday 29 October 2009 16:49:49 Neeraj Chand wrote:
Are all the astricon presentations up?
I'm especially after the one that tilghman did. I caught the tail end of
the prez when I decided to skip the session I was attending and go for
that one.
I'm flattered. While I haven't been able
On Friday 30 October 2009 01:05:26 Joseph wrote:
How to enable cdr_mysql.conf in Asterisk 1.6?
I have installed asterisk-addons which compiled mysql support,
module show is showing cdr_addon_mysql.so
but cdr_mysql.conf was not created in /asterisk directory
Is there any configuration file
it's not only for dialing in ...
setup an extension that is shorter than the number ... and also without the .
eg
exten = 1000,1,Dial()
also when you call out using the overlapdial circuit you do
dial(zap/g1/) or dial(zap/g1/10)
and the rest of the digits should come over overlapdial ...
at
On Fri, 30 Oct 2009, Vincent wrote:
Hello
Since SIP/RTP is a pain to use with road warriors who need to connect
from any location over the Internet, I'd like to get them some IAX
phones instead.
For those of you using this protocol instead of SIP, what would you
recommend as IAX
Danny Nicholas escribió:
Agent 1 could park the call and have agent 2 pick it up from the lot.
*From:* asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Miguel
On Fri, Oct 30, 2009 at 4:25 AM, Robert Bielik robert.bie...@xponaut.sewrote:
Yeay!! Thank you! No, I have not. And I suspected that I had to put
something there, I've googled mad for it
but have not found one document saying that I should. Now I know :)
Perhaps this is something to be added
Hi all,
When somebody leaves a message in the voicemailbox, is there a way to know the
file name of it?
I need to return the voicemail file name in the deadagi command.
Thanks,
Anahi
_
With overlapdial=yes set, when an Alcatel extension calls Asterisk, the Alcatel
user doesn't even have time to dial the second digit because Asterisk connects
it immediately instead of waiting for the rest of the digits.
In Asterisk I have an incoming context [from-alcatel] with patterns such
Assuming you arent writing your VMs to a database, the voicemail will be
in /var/spool/asterisk/voicemail/default/extension/INBOX/msg.WAV
(wav, gsm, txt depends on voicemail.conf).
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com]
Yes, but is there a way to know the filename of that message?
For example: msg0029.wav?
I know where it is saved, but if I want to return it, I need to find the last
one... and it is not recommended in my opinion...
Thanks,
Anahi Ludueña
From: da...@debsinc.com
To:
On 30 Oct 2009, at 15:18, Anahi Ludueña wrote:
Yes, but is there a way to know the filename of that message?
For example: msg0029.wav?
I know where it is saved, but if I want to return it, I need to find
the last one... and it is not recommended in my opinion...
Find the one with the most
Unless you have a long-winded message-leaver, the high sequence should be
the last one...
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes
Sent: Friday, October 30, 2009 10:27 AM
To: Asterisk Users
Thanks people,
I've already found the way...
The variable ${VM_MESSAGEFILE} contains what I need...
Bye,
Anahi Ludueña
From: a_ludu...@hotmail.com
To: asterisk-users@lists.digium.com
Date: Fri, 30 Oct 2009 15:18:25 +
Subject: Re: [asterisk-users] Voicemail file
Yes, but is
This is just a shot in the dark, but perhaps ${VM_MESSAGEFILE}?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes
Sent: Friday, October 30, 2009 10:27 AM
To: Asterisk Users Mailing List -
Just as I found that in the source
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña
Sent: Friday, October 30, 2009 10:32 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Voicemail file
You could use the AMI to find out how many messages are in the voicemail
mailbox and then work out the file name from that...
Anahi Ludueña wrote:
Yes, but is there a way to know the filename of that message?
For example: msg0029.wav?
I know where it is saved, but if I want to return it, I
Thanks Prince (good links) and Tilghman.
I'm using Gentoo installation of Asterisk-1.6.1.8-r1 that just showed up on
portage.
I've emerged(installed) asterisk-addons and this file usually creates necessary
drivers and copy cdr_mysql.conf file into /etc/asterisk (it worked in past
verions 1.2
This thread taught me my Asterisk Lesson of the day - to find the
Variables passed to the dialplan from a program, go to the source directory
and do 'grep setvar_helper app'. Here's the output for app_voicemail.c in
1.4.26.2
grep setvar_helper app_voicemail.c
Hi all,
I would like to know if there is any application replacement for the
AgentCallBackLogin() from asterisk 1.4 on asterisk 1.6. I know, from what
I've read that the call back login agent can be done using a smart dialplan
as showed on the docs. But I cannot thinks a flexible dialplan for a
Easier than poking around the source code is to add a DumpChan() step
to any area of your dialplan to see all defined variables.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Oct 30, 2009, at 8:51 AM, Danny Nicholas wrote:
This thread taught me my Asterisk Lesson
On Fri, 2009-10-30 at 08:37 -0500, Tilghman Lesher wrote:
On Thursday 29 October 2009 12:32:48 Carlos Chavez wrote:
On Thu, 2009-10-29 at 12:23 -0500, Tilghman Lesher wrote:
On Thursday 29 October 2009 11:49:30 Carlos Chavez wrote:
On Wed, 2009-10-28 at 23:36 -0500, Tilghman Lesher
Hi,
Are any of you using them yet ? How did you get on ? Have you tried the
streaming capabilities in the XML browser ? Any issues with Asterisk ?
Seems very clever technology :)
Best Regards,
--
This message has been scanned for viruses and
dangerous content and is believed to be clean.
Have you tried forward-porting it? I don't do queues or 1.6 so it's just
an academic question to me.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mariano
Lecuona
Sent: Friday, October 30, 2009 10:58 AM
To: Asterisk
Perhaps, but it only gave me what the channel was using. There were other
variables that might have been available that the grep method found.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jim Dickenson
Hi all,
I can only get a line signal when I set the phones to not register
with domain .
All phones are in the same NAT and I cannot make calls.
I am getting "Call failed : Proxy Authentication Required" in Xlite
and a busy signal when using an ATA.
Here is my extensions.conf
[internal]
I just jumped to asterisk-1.6.1.8 and I calls will not go through to my
asterisk.
Same setup with asterisk-1.4 and calls get accepted.
sip show registry (asterisk-1.6):
Host dnsmgr Username Refresh State
sip.actio.pl:5060 N 4589835105 Registered
sip show
On 10/30/09 12:05, Joseph wrote:
I just jumped to asterisk-1.6.1.8 and I calls will not go through to my
asterisk.
Same setup with asterisk-1.4 and calls get accepted.
sip show registry (asterisk-1.6):
Host dnsmgr Username Refresh State
sip.actio.pl:5060 N 4589835
I'm going to have to tackle this issue in the near future for one of my
clients, and the approach I'm thinking of taking goes something like this:
Store queue codes in the AstDB that are associated with actual queues (i.e
queuecode 1001 = sales_queue, 1002 = cust_serv_queue, etc). Have the users
How are you setting up xlite and the ata? Which version of Asterisk are you
using? What does the general section of your sip.conf look like?
On Fri, Oct 30, 2009 at 1:01 PM, Cliconnect cliconn...@cliconnect.comwrote:
Hi all,
I can only get a line signal when I set the phones to not
Thank you,
How are you setting up xlite and the ata?
Xlite
User name : 1000
Domain: IP of the server running Asterisk
Register with domain and receive incoming calls: clear
Port used in local computer : manually specify range : 5061-5062
ATA
SIP server address: IP of the server running
In asterisk 1.6 the voicemail prefix u b don't work, I have:
exten = 1,3,Voicemail(u11) and it keeps telling me: No entry in Voicemail
config file for u11
exten = 1,3,Voicemail(11) works,
Isn't prefix u suppose to play: The person at extension ... 11 ... is
unavailable, ?
--
Joseph
That would be u,11 or b,11 for unavailable 11 or busy 11 (also s,11 for
silent 11)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joseph
Sent: Friday, October 30, 2009 2:23 PM
To:
On 10/30/09 13:23, Joseph wrote:
In asterisk 1.6 the voicemail prefix u b don't work, I have:
exten = 1,3,Voicemail(u11) and it keeps telling me: No entry in Voicemail
config file for u11
exten = 1,3,Voicemail(11) works,
Isn't prefix u suppose to play: The person at extension ... 11 ... is
On Fri, 2009-10-30 at 13:23 -0600, Joseph wrote:
In asterisk 1.6 the voicemail prefix u b don't work, I have:
exten = 1,3,Voicemail(u11) and it keeps telling me: No entry in Voicemail
config file for u11
exten = 1,3,Voicemail(11) works,
Isn't prefix u suppose to play: The person at
No, Voicemail(u,11) DOES NOT WORK
Voicemail(11,u) WORKS
--
Joseph
On 10/30/09 14:27, Danny Nicholas wrote:
That would be u,11 or b,11 for unavailable 11 or busy 11 (also s,11 for
silent 11)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
On 10/30/09 15:34, Jared Smith wrote:
On Fri, 2009-10-30 at 13:23 -0600, Joseph wrote:
In asterisk 1.6 the voicemail prefix u b don't work, I have:
exten = 1,3,Voicemail(u11) and it keeps telling me: No entry in Voicemail
config file for u11
exten = 1,3,Voicemail(11) works,
Isn't prefix u
The Wiki actually does mention the 1.6 caveat. If in doubt, Jared's
suggestion is the best route (that's why they pay him the big bucks).
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joseph
Sent: Friday,
You're attempting to connect on ports 5061-5062 but are bound to port
5060...?
What does your CLI look like during a failed call attempt?
Thanks,
--Warren Selby
On Fri, Oct 30, 2009 at 2:18 PM, Cliconnect cliconn...@cliconnect.comwrote:
Thank you,
How are you setting up xlite and the
No change on it.
Do I have to enter a command ?
I have changed the port to 5060 in both clients. Still the same problem.
thanks
Jair Santos
Warren Selby wrote:
You're attempting to connect on ports 5061-5062 but are
bound to port 5060...?
What does your CLI look like during a failed
you can get debug info a couple of ways from the asterisk CLI. I like this
command the best. sip set debug ip xxx.xxx.xx.xxx where xxx.xxx.xxx.xxx is the
of the x-lite phone. It will give you a lot of info. I haven't figured out how
to redirect output yet.
Date: Fri, 30 Oct 2009 13:05:35
Hi,
I have played with the 820 for some weeks, mostly love it excellent
speech quality. Even got the mini browser running showing my favorite
webcam, this could be put to real use too:)
Issues so far:
Some embarrassing crashes while speaking, was able to speak but all
freezed. Still a little
Setting up a new Asterisk server with Centos 5.2 and Asterisk 1.4.23
using Dahdi and getting a kernel panic - not syncing: Fatal exception
error during boot. Anyone have thoughts on what I can do to rectify
this or is this card not compatible with this machine?
Take a look at this document. This may help you on trouble shoot your kernel
panic.
http://www.novavox.co.uk/docs/install-guides/novavox-asterisk-card-installation-issues.pdf
2009/10/30 David Shauger sollost...@gmail.com
Setting up a new Asterisk server with Centos 5.2 and Asterisk 1.4.23
so you're either testing it wrong or it's been broken since that
worked fine years ago
you may try adding the . after then extension ... I don't remember
maybe it's needed
eg:
exten = 1004000.,...
but better yet
exten = 100400XX,...
Martin
On Fri, Oct 30, 2009 at 10:08 AM, Vieri
Trying asterisk-1.6.1.8 but I can not get dtmf signal to go through.
dtmfmode = rfc2833
is there still problem with dtmf in version 1.6 ? :-(
--
Joseph
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing
Hello
I just found out this:
I had a phone into the FXO ports to see why calls are not passing through,
When I ask asterisk to dial a number of 10 digits, it dials the first 9
digits, then wait 2 seconds and dial the last digit!
Any idea how to overcome this and dial the whole number 1
On 10/30/09 17:28, Joseph wrote:
Trying asterisk-1.6.1.8 but I can not get dtmf signal to go through.
dtmfmode = rfc2833
is there still problem with dtmf in version 1.6 ? :-(
Will DTMF + sip ever be solved in asterisk or it is impossible task?
I remember having problems with DTMF ever since ver.
In my previous post which included CLI from a working call and non
working call I did a diff of the two
trying to find out why libpri 1.4.10.2 does not work and 1.4.1 does work
(I am trying to upgrade).
below is the difference that stuck out. This was EXTRA in 1.4.10.2 which
is the libpri that
On Fri, Oct 30, 2009 at 10:48 PM, Joseph syscon...@gmail.com wrote:
On 10/30/09 17:28, Joseph wrote:
Trying asterisk-1.6.1.8 but I can not get dtmf signal to go through.
dtmfmode = rfc2833
is there still problem with dtmf in version 1.6 ? :-(
Will DTMF + sip ever be solved in asterisk or it
On 10/30/09 23:15, Moises Silva wrote:
On Fri, Oct 30, 2009 at 10:48 PM, Joseph syscon...@gmail.com wrote:
On 10/30/09 17:28, Joseph wrote:
Trying asterisk-1.6.1.8 but I can not get dtmf signal to go through.
dtmfmode = rfc2833
is there still problem with dtmf in version 1.6 ? :-(
Will DTMF
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