Displaying 20 results from an estimated 700 matches similar to: "Echo() app doesn't work"
2010 Dec 04
3
samba4 AD controller, production
Hello, list,
Anyone running a recent version of samba4 as the only one AD controller in the domain, *in production*? If so, what is your overall experience?
I am going to make such an install for a site with some 50 users (and frankly speaking, I am frightened ... so give me some courage!)
Would you recommend me to go ahead and do it? Latest release?
Any feedback is more that appreciated
2010 Nov 08
1
Add text to a stacked barplot
Hi All,
I need some help in putting text in a stacked barplot. The barplot is filled
with 5 levels and now I would like to put text to each level in the stacked
barplot. However, it seems that the code that I am using is not placing the
text at the correct hight (centered at each fill) in the barplot. Any
suggestions to improve the code to make it work?
barchart(FREQ ~ VISIT
2010 Dec 10
0
Fwd: Re: samba4 AD controller, production
Copied the list as I guess it will be more interesting for them
On Friday 10/12/2010 at 9:11 am, Matthieu Patou wrote:
> On 04/12/2010 19:52, Yassen Damyanov wrote:
>>
>> Hello, list,
>>
>> Anyone running a recent version of samba4 as the only one AD
>> controller in the domain, *in production*? If so, what is your overall
>> experience?
>>
2008 Jul 11
0
Outgoing calls but no incoming calls with X100P
Hi all,
I have a problem with my asterisk box and an X100P FXO card. I am able to
place outgoing calls from my SIP phone (Cisco 7940) to any external number
using my PSTN line, but when I call my PSTN line from my cell phone, the
Cisco doesn't ring (and no message appears in the Asterisk CLI).
Here are my config files:
zaptel.conf
fxsks=1
loadzone = be
defaultzone = be
2010 Mar 16
1
Asterisk hangup all incoming calls after 10 seconds
Hello Gentleman,
I'm new to asterisk, this is my first instalation, first post... so I'd like
to apologize if this question has already been made. I googled but I
couldn't find nothing similar.
Here's the thing.
I'm migrating from ATA to Asterisk one of my client's office and I have a
very simple setup.
A Linux PC running Debian Lenny + Asterisk 1.4.21. It's a
2011 Apr 20
1
[IAX] Everyone is busy/congested at this time (1:0/0/1)
Hi,
I have a problem with IAX accounts...
I set up a few months ago an Asterisk server, with mysql support to load iax
accounts.
Settings seems fine because apparently the system works as expected.
Yesterday I tried to add another iax account in the iax.conf directly. And I
have a problem with this new account (named 444).
I can authenticate from 444 to the server, and I can receive calls from
2009 Nov 21
3
Connect two Asterisk Server in IAX ?
Hi
My first post get no answer :=<, i post new with new elements.
I have two Asterisk server, running on Asterisk 1.6:
SRV1 = 192.168.0.5 on Asterisk 1.6.1.4
SRV2 = 192.168.0.20 on Asterisk 1.6.1.8
I want create a link for exchange call.
on Srv1:
iax.conf:
[general]
bindport=4569
bindaddr=0.0.0.0
language=fr
bandwidth=low
jitterbuffer=no
forcejitterbuffer=no
autokill=yes
2006 Dec 31
0
IAX & WaitExten
Hello list,
I've got a problem (maybe only a problem of understanding how * works) with IAX and WaitExten.
To simplify the problem I've brought it down to the following scenario:
- 3 Asterisk Server A,B and C (central).
- A and B both register with C.
Now I want to be able to dial an extension at A to become connected to C and there I want to dial an extension to become connected to B.
2006 Nov 29
2
Trouble using 2 IAX2 DiDs provided by different ITSPs
Asterisk 1.2.7
Redhat 9
I have DiDs from two different ITSP both set up as IAX2. Each one
works when it's the only one in my iax.conf, but when I have them both
defined in iax.conf at the same time, only one will work. My iax.conf
is provided below.
Any ideas how to fix? I'd like to use both DiDs!
Thanks,
H
My iax.conf is below. When I dial the DiD provided by ITSP_B, the
other
2006 Nov 01
5
DTMF over IAX
Ok sorry for not being specific. I am having a problem when people
outside call in to my number which terminates at VoicePluse then The
send IAX to me and I do not get any tones. People press buttons but it
just goes to the next dialplan fall through. It happens 60-70% of the time.
extentions.conf
[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
priorityjumping=no
2015 Feb 05
1
IAX2 problem for WAN connections
> On Thursday 05 Feb 2015, jg wrote:
>> Calling from ServerB to ServerA works, but not vice versa. The only odd
>> thing that appears to me is the different perceived port on ServerA.
>>
>> Does someone have an idea at what to look in detail?
>
> Look in /etc/asterisk/iax.conf in the first instance.
>
Basically I used the example from the Asterisk book
2015 Feb 05
2
IAX2 problem for WAN connections
Hi,
I am trying to connect two Asterisk servers using IAX2. Everything works fine when I couple them
within a LAN segment, but not when I connect them using WAN connections. I made sure that the
routers' ports are mapped properly and checked this with additional ssh rules.
ServerA is a Raspberry box with the vendor's Asterisk version 1.8.13.1 and ServerB is normal
CentOS 7 box with
2006 Feb 07
1
asterisk to FWD
Hello all,
Here is my problem,
I try to place a call to FWD (free world dialup) trough my asterisk PBX.
my config is as follow:
extensions.conf
----------------
[internal]
exten => 613,1,Dial(IAX2/iaxfwd-outbound/613) (service echo de FWD)
exten => xxxxxx,1,Dial(IAX2/iaxfwd-outbound/xxxxxx) mon numero FWD
exten => yyyyyy,1,Dial(IAX2/iaxfwd-outbound/yyyyyy) celui d'un ami FWD
2006 Aug 28
3
lost packets when bridging zap and iax
We have a machine with a TE410P in it acting as a client to route calls
via iax2 to our central server,
caller --> ( zap -> iax ) ---> ( iax -> whatever ) --> called
client server
often the called can't hear the caller (both machines on public ip)
'iax2 show netstats" on client machine shows more and more dropped
packets on the
2013 Sep 06
1
11.4.0: iax packets lost by amazon ec2
I have 11.4.0 on an Amazon EC2 instance. SIP works fine, but I can't get
iax to work.
I've opened 4569 in the EC2 Security Group.
I'm using the zoiper client. Using tcpdump I can see the zoiper packets
coming in on 4569, but nothing shows on the asterisk cli.
Frame 33: 79 bytes on wire (632 bits), 79 bytes captured (632 bits) on
interface 0
0000 12 31 3b 12 40 84 fe ff ff ff
2009 Nov 15
1
Call IAX2 => "Call rejected, CallToken Support required"
Hi
i have a small problems on two Asterisk Server 1.6.4 :
The first sent the call to the second, and in the second, i have a error :
[Nov 15 15:30:12] ERROR[5113]: chan_iax2.c:4529 handle_call_token: Call
rejected, CallToken Support required. If unexpected, resolve by placing
address IP_FIRST_ASTERISK in the calltokenignore list or setting user
04TELNUMBER requirecalltoken=no
on the second
2007 Jul 27
3
Need help with inbound IAX
I have just started working with Asterisk and have run into a road block
concerning IAX and an inbound DID from callwithus.com. I am getting
nowhere and I don't really know how to isolate the problem. The asterisk
version is 1.2.7 on ubuntu, sits behind a firewall with iptables. I can
connect and make a call to other internal extensions using zoiper and
iax. When I try and use the number,
2006 Oct 27
1
Iax bug ?
Hello,
I'm french, so excuse my poor English.
I'm face to a terrible thing, with has stole a lot of my time.
On the .184 machine, I've the following iax.conf :
[general]
rtcachefriends=yes
bandwidth=high
tos=reliability
jitterbuffer=no
autokill=yes
#include "iax.voip1.conf"
#include "iax.renoir.conf"
The iax.voip1.conf file contains :
[VOIP1]
type=friend
2011 Feb 16
5
Polycom IP335
I am posting here since you guys are my last hope.
I am trying to configure a Polycom Soundpoint IP 335 with MWI.
Is there any way to eliminate the scrolling messages and Msgs softkey?
I am trying to get it where it's just the light that indicates the new
messages.
I don't know if Asterisk has to send a different notification or what have
you.
Thanks,
--Eric
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2011 Feb 21
2
calls are not going thru e1 line
I'm curious as to what versions of everything you are using. Reason
being this line " -- DAHDI/i1/00256312261627-1 is proceeding passing
it to SIP/5000-00000000".
It states "DAHDI/i1/00256312261627-1..." and I don't recall seeing that
before (my 2.4.0 says " -- DAHDI/1-1 is proceeding passing it to
SIP/801-0000000c" [1-1 being the span and channel