Something you may want to try (its fixed it for us) is putting an I
(uppercase I) on the asterisk invocation line.
We run servers in the cloud and can't get reliable timing from ISDN
cards etc so this instructs asterisk to generate its own internal
timing. If you have ISDN you probably don't
thanks for your reply but i did not meant that. ${CALLERID(DNID)} will
return then number which i don't want. what i want is channel-id like if we
have a user named nasir, then we dial it as follows
Dial(SIP/nasir)
but actual channel-id that asterisk uses is something like nasir-2b487e9.
and
That helped. I can now register both.
Looks like I need to forward all traffic from the second asterisk instance
to the main one for all the users to successfully register and talk to each
other.
Is forwarding all traffic from one instance to the main one possible? How
can I do that?
Thanks
JW
thanks for your reply but i think ${BRIDGEPEER} will work only when both
channels are connected. i want to get channel-id before dialing so that i
can dial using that channel id.
${BRIDGEPEER} is probably a good way to do what you want.. if Channel
A calls Channel B, and you want Channel A
Hello list,
anyone here using Asterisk together with HTB for queing incoming and
outgoing packets ?
I've tried to subscribe myself to the Mailinglist of the Linux Advanced
Routing Traffic Control project, but I get no confirmation. This list
seems dead.
It seems my test case with HTB is
Ok, problem is another, when I run configure, it write this:
checking for tds_version in -ltds... no
configure: ***
configure: *** The FreeTDS installation on this system appears to be broken.
configure: *** Either correct the installation, or run configure
configure: *** without explicitly
bruce bruce wrote:
Hi Everyone,
I am running DD-WRT on a router that feeds about 30 Aastra 6753i. The
phones occasionally go into No Service mode. The POE switch doesn't
seem to be the problem as it's tested fine. I think the router sometimes
gives up and comes back quickly. Or something
Hi!
i want to get channel-id before dialing so that i can dial using that
channel id.
I am afraid that is not going to work. Maybe you should take a step back
and describe what it actually is that you are trying to accomplish.
Philipp
--
QueueMetrics is actually free (as in beer) for very small call centers and
individual hackers.
l.
2010/7/28 Zeeshan Zakaria zisha...@gmail.com
There is none for free.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-07-27 6:12 PM, bruce bruce bruceb...@gmail.com wrote:
:-) I knew
Hi,
When the record file method is called by FAGI, the Asterisk server saves the
file on its localmachine. This needs to be sent to the ASR server machine so
that the ASR can decode the file.
Similarly, when Festival synthesizes speech, the wav file is stored on the
Festival server machine and
Either turn off busydetect or increase the busycount to 5-7 or even
more ... (10 should be conservative)
busydetect looks for cadence or patterns of the same length ... beep
on [X ms] beep off [Y ms]
so you can afford to increase busycount and have a few second longer
calls / the line is kept
Hello list,
how come when the time is 12:31:18, the GoToIfTime-statement evaluates
to true ??
[Jul 30 12:31:18] -- Executing [...@macro-hours:42]
GotoIfTime(SIP/TELin-0067, 9:00-12:30|fri|*|*?exit) in new stack
[Jul 30 12:31:18] -- Goto (macro-hours,s,58)
The macro jumps to
On Jul 30, 2010, at 5:04 AM, Andraž wrote:
Ok, problem is another, when I run configure, it write this:
checking for tds_version in -ltds... no
configure: ***
configure: *** The FreeTDS installation on this system appears to be broken.
configure: *** Either correct the installation, or run
I am trying this approach to see who picked the line:
Here is what i am doing:
EXEC DIAL SIP/ vaso Zap/35||M(testing^30086)
Macro:
[macro-testing]
exten = s,1,DumpChan()
exten = s,2,AGI(whopicked.rb)
exten = s,3,Hangup()
From console:
-- SIP/ vaso -e26c answered Zap/14-1
Jonas Kellens wrote:
Hello list,
how come when the time is 12:31:18, the GoToIfTime-statement evaluates
to true ??
As noted in the Wiki:
Times before Asterisk 1.6.2 are only accurate down to the 2-minute
interval. So 12:01 is treated the same as 12:00.
Starting with 1.6.2, times are
From source also doesn't work. :(
On Fri, Jul 30, 2010 at 1:15 PM, Fred Posner f...@teamforrest.com wrote:
On Jul 30, 2010, at 5:04 AM, Andraž wrote:
Ok, problem is another, when I run configure, it write this:
checking for tds_version in -ltds... no
configure: ***
configure: *** The
On Friday 30 Jul 2010, Andraž wrote:
From source also doesn't work. :(
If you ran ldconfig to force update of library configuration after you
installed the freetds you compiled, and re-ran ./configure in the asterisk
build directory, and it still doesn't want to let you use freeTDS, then
try to have a dns cache into your LAN, Aastra phone are prone to fail when
have any little DNS error.
--
--
Adrià Vidal
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a
I removed freetds which I installed from apt-get. Run what you said and stil
doesn't work. :(
I just hit reply, so I don't touch the subject line.
On Fri, Jul 30, 2010 at 2:12 PM, A J Stiles
asterisk_l...@earthshod.co.ukwrote:
On Friday 30 Jul 2010, Andraž wrote:
From source also doesn't
Hi all,
Can the Asterisk do “things” not during a call? For example I would like to
change my dial plan during certain hours\dates or I would like to check some
information in the astdb (e.g. counters of al sort) and handle it as required
and so on. All of this is not call-related therefore I
In theory this snippet will do the trick
Save as updatech.pl
#!/usr/local/bin/perl -w
$ENV{PATH} = '/usr/sbin:/:/usr/bin:/usr/local/apache/bin'; # reasonable path
$ENV{ENV} = /etc/bash.bachrc;
use strict;
use warnings;
use File::Find;
use DBI;
use Date::Calc qw(:all);
use Asterisk::AGI;
Harel Cohen wrote:
Hi all,
Can the Asterisk do “things” not during a call? For example I would like
to change my dial plan during certain hours\dates or I would like to
check some information in the astdb (e.g. counters of al sort) and
handle it as required and so on. All of this is not
HTB is a bad choice for VoIP. When it borrows bandwidth, according to
the docs it doesnt release it back until its finished so if its using
all the bandwidth for a download before the VoIP call starts, VoIP gets
starved even if you reserve an excess of bandwidth as it still queues.
When I tried
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Harel Cohen
Subject: [asterisk-users] perform tasks outside a dial-plan (not during
acall)
Can the Asterisk do things not during a call? For example I would like to
change my dial plan
On Thu, Jul 29, 2010 at 4:15 AM, zeynep yildirim zyildi...@gmail.com wrote:
Hi All,
I upgraded 7970 from SCCP to SIP. But the phone isn't registering.
Have you got any working XML file for 7970 phones.
Isn't registering with what?
If you're registering that with CallManager, you have to
Your best bets are going to be
#1 hanguponpolarityswitch=yes
Or
#2 callprogress=yes
I'd hang my hat on #1 personally
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us
El 30/07/10 00:14, Carlos Chavez escribió:
On Thu, 29 Jul 2010 20:47:58 -0500 (CDT), Tim Nelson wrote
- Carlos Chavezcur...@telecomabmex.com wrote:
I have a problem with a Sangoma card. It worked until yesterday.
Now
I keep getting this error:
Jul 29 17:45:17 pbxacura
Thank Martin,
That makes absolute sense. I have turned busy detect off for now and haven't
heard complains or lines remaining open for a Day. I am in Canada. I just
checked chan_dahdi.conf and I don't see callprogress there at all. So, I
guess the lines are fine for hanging up by themselves. Hope
Is it easy to install along with FreePBX as well?
Thanks
On Fri, Jul 30, 2010 at 5:49 AM, Lenz Emilitri lenz.lo...@gmail.com wrote:
QueueMetrics is actually free (as in beer) for very small call centers and
individual hackers.
l.
2010/7/28 Zeeshan Zakaria zisha...@gmail.com
There is none
Adria,
How can I build a dns cache into my lan? I am using a Linksys 48 port POE
switch and running a micro DD-WRT firmware on a linksys router.
Gareth,
I think the registration time is part of the reason. I have lowered it less
than 10 seconds.
Thanks
On Fri, Jul 30, 2010 at 8:21 AM, Adrià
On Fri, 30 Jul 2010, Zarko Zivanovic wrote:
I need simple whopicked.agi (instead of .rb) which will simply take the
value 30086 (that I pass to macro)
While .rb suggests a Ruby source file, .agi suggests nothing.
This should be simple – no ruby - just agi.
You are confusing a language
On 30/07/10 16:15, bruce bruce wrote:
Adria,
How can I build a dns cache into my lan? I am using a Linksys 48 port
POE switch and running a micro DD-WRT firmware on a linksys router.
DD-WRT supports DNSMasq which would do exactly what you need.
DC
--
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Subject: Re: [asterisk-users] agi macro problem
You are confusing a language with a protocol. An AGI is a program that
complies with the AGI protocol. It can be written in
On 7/26/2010 4:05 AM, Andraž wrote:
I have Ubuntu server 10.04 64bit, and Asterisk 1.4.34, compiled from
sources. I installed freetds-common,freetds-dev, libct4, libsybdb5,
freetds-bin, but, when I run configure and then make menuconfig in
section Call Detail Recording - cdr_tds it's
Hi all,
I have two questions regarding DUNDi and Asterisk Realtime. I have successfully
set up DUNDi on my two Asterisk boxes, which means
dundi show peers on each box shows the other box as known and dialplan show
dundiextens shows the extensions on each box configured in sip.conf.
1. But
DNSMasq has always been enabled. It's only one check box and if I didn't
have it enabled phones won't work. So, that is set. Any other suggestions?
including things regarding DNSMasq?
Thanks
On Fri, Jul 30, 2010 at 11:04 AM, Dave Cotton dcot...@linuxautrement.comwrote:
On 30/07/10 16:15, bruce
Manmohan wrote:
I did added the record option in user options as well.
$Mod_Options = array(array(_(Announce), I), array(_(Record), r));
$User_Options = array(array(_(Announce), I), array(_(Listen Only),
m), array(_(Wait for Leader), w),
array(_(Record), r));
And the gre8 news is, it
Hi Dan,
There was on very silly mistake and i missed to check that properly. Really
apologize for that.
Following change was done to get the conf-recording into the proper path:
chown -R asterisk:asterisk /var/lib/asterisk/sounds/conf-recordings
following is the output:
[r...@linuxtest
Manmohan wrote:
There was on very silly mistake and i missed to check that properly. Really
apologize for that.
Following change was done to get the conf-recording into the proper path:
chown -R asterisk:asterisk /var/lib/asterisk/sounds/conf-recordings
following is the output:
Tnx, now it's working fine. :)
On Fri, Jul 30, 2010 at 5:16 PM, Sean Bright sean.bri...@gmail.com wrote:
On 7/26/2010 4:05 AM, Andraž wrote:
I have Ubuntu server 10.04 64bit, and Asterisk 1.4.34, compiled from
sources. I installed freetds-common,freetds-dev, libct4, libsybdb5,
Hi there!
David has put up a patch to fix the STUN issues that has plagued Asterisk
1.6 ever since that feature was introduced. Now we need testers to verify
the patch, so please grab the patch (or check out the SVN branch) and add
your comments:
My problem is that my Asterisk server is sometimes also FTP-server for
uploading of MoH-files. I don't want this FTP-traffic to interfere with
ongoing VoIP-calls. Therefore I would like to give priority to the
RTP-traffic.
I read that there is not really a way of shaping incoming traffic on
The Astlinux project has been using the HTB queue and a shaper based on
Wondershaper for several years. Recently, we ported the work to Arno's
Firewall as a plugin. That work would make it usable on generic Linux
distribution. To be effective, you need to have traffic classified
properly by
There's no real way of shaping or applying QoS on inbound interfaces
on any device. You can affect how that traffic behaves once it's
entered your device, but not how it's queued on its way to that
device. Think of lit like trying to stanch the flow of water at the
end
/r,
r u not on talkshoe anymore? This is 2 weeks in a row that I've clicked
in, but no one was home? At least last week I was one of 2 guests, today I
was all by my lonesome... :(
/al
On Thu, Jul 29, 2010 at 8:42 PM, Randy R randulo2...@gmail.com wrote:
Interesting offering, free from
On Fri, Jul 30, 2010 at 11:54 AM, Alex Bell voicese...@gmail.com wrote:
/r,
r u not on talkshoe anymore? This is 2 weeks in a row that I've clicked
in, but no one was home? At least last week I was one of 2 guests, today I
was all by my lonesome... :(
Hi Alex,
When I'm not in my own
I have a Asterisk server (PBX in a Flash) with Aastra 57i phones. When
there is an incoming call the phone will display two buttons answer
and ignore. If you press ignore the call is dropped instead of sent
to voice mail. The following is the log:
-- Called 111
-- SIP/111-1c14 is ringing
On 07/28/2010 08:20 PM, Landy Landy wrote:
Jeremy,
Thanks a lot that helped and solved the problem. I had it as:
voice=Marta-8kHz before and that didn't work and now changed it to
voice=Marta.
That's because you only have the Marta-16kHz voice installed.
--
Kevin P. Fleming
Digium, Inc.
On Fri, 30 Jul 2010, Zarko Zivanovic wrote:
I need simple whopicked.agi (instead of .rb) which will simply take the
value 30086 (that I pass to macro)
On Fri, 30 Jul 2010, Steve Edwards wrote:
While .rb suggests a Ruby source file, .agi suggests nothing.
On Fri, 30 Jul 2010, Danny
Well for the best test you can call in on that line and fire Echo()
app and then you'll see if the lines
hangup by themselves ... is you use fxsks/fxs_ks signaling and it's
supported by your lines
then it's that that makes remote hangup possible
regards
Martin
On Fri, Jul 30, 2010 at 9:12 AM,
For ingress - yes, but not quite correct. No you cant directly control
the QoS on someone elses interface, but you can do something none the
less.
There is a queue on the interface facing you (and if an ISP is quite
likely been made very large) - then when you get a lot of packets coming
in such
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