Just to answer your side issue:
On Sun, Sep 27, 2009 at 04:05:30PM -0500, Andy Howell wrote:
The only Warning or Error I see is when asterisk first starts a new call.
logger.c: -- Starting simple switch on 'DAHDI/1-1'
[Sep 27 15:55:50] WARNING[4199] chan_dahdi.c: Unable to enable echo
On Mon, Sep 28, 2009 at 01:38:02AM +0300, Andrei Verovski (aka MacGuru) wrote:
Hi!
I have Asterisk 1.6.1 installed on OpenSuSE 11.0 running with choppy sound.
One specialist on the forums asked me if I have DAHDI configured, he assumed
that this could be cause of choppy sound problem.
Hi!
I had the same problem...
dahdi_transcode command solved the problem!
CheerS!
Ismael.
On Mon, Sep 28, 2009 at 1:25 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:
Just to answer your side issue:
On Sun, Sep 27, 2009 at 04:05:30PM -0500, Andy Howell wrote:
The only Warning or Error
Hello,
now I can reset and load the firmware, but I do not see the entries
in /proc and the genzaptel command ignores the astribank. The dual PRI
card works fine.
r...@pbx1:~# zaptel_hardware
usb:002/002 xpp_usb- e4e4:1161 Astribank-modular
USB-firmware
pci::01:05.0 wct4xxp+
Hi All,
I have a TE121P card installed and since connected it to the PRI I keep
getting the Current Alarm as continually changing from Blue
Alarm/Recovering and Recovering.
The config I have is:
/etc/dahdi/system.conf
bchan=1-15,17-31
dchan=16
echocanceller=mg2,1-15,17-31
Benny Amorsen schrieb:
Jared Smith jsm...@digium.com writes:
Again, the emphasis on the dCAP exam is real-world knowledge of how to
build a simple small-business PBX with Asterisk. If you've used
Asterisk in a professional capacity, it should be very straightforward
to pass the practical
If you need this from the moment that the agent connected you can measure
the length of the leg that goes with the agent.
If you need to measure this from the moment the call was answered, you can
measure the length of the 'main' cal of the leg (the one that called the
command queue())
If you
Dear All,
I have a perl script running on my asterisk server...This script is running
for all incoming calls...It checks if a user is registered on openSIPS
server...Else it return a busy tone...
I would like to ask you please about the syntax to call a dialplan defined
on extensions.conf for a
On Mon, Sep 28, 2009 at 08:55:17AM +0200, Loic Didelot wrote:
Hello,
now I can reset and load the firmware, but I do not see the entries
in /proc and the genzaptel command ignores the astribank. The dual PRI
card works fine.
r...@pbx1:~# zaptel_hardware
usb:002/002 xpp_usb-
Hello,
iam searching for an Firefox plugin which can make an sip Invite and
Redirect after 200 OK, so i dont have to use a softphone, just to
initialise a call by clicking on a number
i've found some plugins which only works with a softphone installed on
the system but nothing which works good
Hello,
I wrote a AGI script under Asterisk in php, and I want to stress this
scripts by simulting call, is there a tool or a way on asterisk to simulate
answers to call, and also if possible DTMF?
NB: I am not using SIP phone but only analog lines.
Regards
Mickael
I have a similar problem with DAHDI. If my server gets rebooted, I can't make
any calls until the a call come in from outside. From there I can answer the
call and DAHDI works fine afterwards.
--- On Mon, 9/28/09, Tzafrir Cohen tzafrir.co...@xorcom.com wrote:
From: Tzafrir Cohen
On Mon, Sep 28, 2009 at 04:56:01AM -0700, Landy Landy wrote:
I have a similar problem with DAHDI. If my server gets rebooted, I can't make
any calls until the a call come in from outside. From there I can answer the
call and DAHDI works fine afterwards.
In your case: is the problem reset by
We have a customer who connects PBX boxes (Avaya etc.) to our asterisk
server (1.4.17) as a SIP extension. This customer needs the dialled
number sent to the PBX as well as number that the call is originating
from so he can set up his own routing from his PBX box.
I have tried setting both
The routine ast_extension_states in main/pbx.c is set up (in 1.4.26.1) to
return these values:
switch (devstate) {
case AST_DEVICE_ONHOLD:
return AST_EXTENSION_ONHOLD;
case AST_DEVICE_BUSY:
return AST_EXTENSION_BUSY;
case
It is probably as simple as doing a manual modprobe wctdm.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Angus Asterisk
Sent: Sunday, September 27, 2009 11:18 AM
To: asterisk-users@lists.digium.com
Subject:
Hi all,
I need a goto If statement syntax that check if a variable is not null then
go to dialplan 1 else go to dialplan2
Regards
___
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AstriCon 2009 - October 13 - 15 Phoenix,
michel freiha wrote:
Hi all,
I need a goto If statement syntax that check if a variable is not
null then go to dialplan 1 else go to dialplan2
exten = s,n,GotoIf($[${conference.room} != ]?s-process,1:s-notexist,1)
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty
michel freiha escribió:
Hi all,
I need a goto If statement syntax that check if a variable is not
null then go to dialplan 1 else go to dialplan2
Regards
Hi,
Use the function EXISTS() inside your GotoIf.
http://www.voip-info.org/wiki/index.php?page=Asterisk+func+exists
Cheers,
--
Ing.
I suspect the issue I'm having is more specific to the MeetMe app but
that's just a guess. app_meetme.c (in 1.6.1.6) calls
ast_devstate_changed(AST_DEVICE_INUSE, meetme:%s, conf-confno)
when the first caller enters a conference and
ast_devstate_changed(AST_DEVICE_NOT_INUSE, meetme:%s,
extension.ael:
if (0!=${MYVARIABLE}) {
...
}
or test for empty/unset variables, use:
${EXISTS()} or
${ISNULL()}
regards
klaus
michel freiha schrieb:
Hi all,
I need a goto If statement syntax that check if a variable is not null
then go to dialplan 1 else go to dialplan2
Regards
Dear Doug,
s-process and s-notexist are contexts or what because I'm getting the below
erroe:
[Sep 28 16:35:07] WARNING[1530]: pbx.c:2470 __ast_pbx_run: Channel
'SIP/voxbone.com-092392e0' sent into invalid extension 's-process' in
context 'a2billing4', but no invalid handler
regards
On Mon, Sep
michel freiha wrote:
Dear Doug,
s-process and s-notexist are contexts or what because I'm getting the
below erroe:
This was just an example of what I've use in my dial plan. You'll need
to modify it to fit your dial plan.
Doug
--
Ben Franklin quote:
Those who would give up Essential
michel freiha wrote:
Dear Doug,
s-process and s-notexist are contexts or what because I'm getting the
below erroe:
I guess need to slow down a bit and read.
Yes, they are contexts.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
I am looking to configure the asterisk voicemail system to stop asking for
the folder (work, personal, etc) in which to save messages when I do
save them.
Is there any configuration to do this?
Mike
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In your case: is the problem reset by restarting asterisk?
'dahdi
resstart'?
The problem does not reset by restarting asterisk.
I've noticed that I can call other sip phones but, when trying to call out, I
get the same (Busy/Congested/Not-Available) congested messege.
Hi,
I'm seeing a very strange error when dealing with Diversions. If a
call setup to a number comes to an Asterisk server, that server sends
a request to a third proxy, that proxy sends the call back with a
Diversion flag, Asterisk complains about the host not existing (and
the host is the
sean darcy wrote:
Martin wrote:
u don't change the ${uniquefile} for the second System/Originate
try to add a string to the ${uniquefile} ...
eg
${uniquefile}0
Martin
But I generate another unique file in [fax-tx] just before I try to send
the confirm.
Here's the first call:
On Mon, Sep 28, 2009 at 12:30 PM, sean darcy seandar...@gmail.com wrote:
Well one of the problems it that SendFax doesn't like the tiff file(BTW,
SendFax from app_fax.so gives you clue what the problem is). It requires
a special sort of tiff file. I never found any way to generate a
compliant
Hi, I need to eneble or disable cdr registration dynamicaly in my dialplan.
Is there any way to do this?
Thanks
___
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AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now:
Hi All,
I'm running into a reoccurring nat issue, phones (Linksys, Polycom or
Cisco) behind a cisco router, 2600 series, ios 12.4. Sometimes 2 or
more phones will have the same outside port number and will register
to Asterisk with the same port number. As far as I can tell this is
only
NoCDR
On Mon, Sep 28, 2009 at 1:10 PM, equis software equissoftw...@gmail.com wrote:
Hi, I need to eneble or disable cdr registration dynamicaly in my dialplan.
Is there any way to do this?
Thanks
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Hi, Tzafrir!
I have installed DAHDI kernel modules and dahdi-tools
/etc/init.d./dahdi running
$ ls /usr/lib/asterisk/modules/res_timing_*
results ==
/usr/lib/asterisk/modules/res_timing_dahdi.so
/usr/lib/asterisk/modules/res_timing_pthread.so
$ dahdi_test
results ==
99.999% 99.990% 99.994%
On Mon, Sep 28, 2009 at 02:26:29PM +0200, Tzafrir Cohen wrote:
On Mon, Sep 28, 2009 at 04:56:01AM -0700, Landy Landy wrote:
I have a similar problem with DAHDI. If my server gets rebooted, I can't
make any calls until the a call come in from outside. From there I can
answer the call and
Thanks!
On Mon, Sep 28, 2009 at 2:28 PM, Paul Dugas p...@dugasenterprises.comwrote:
NoCDR
On Mon, Sep 28, 2009 at 1:10 PM, equis software equissoftw...@gmail.com
wrote:
Hi, I need to eneble or disable cdr registration dynamicaly in my
dialplan.
Is there any way to do this?
Thanks
I've read about this problem a few times with DAHDI. Since it only takes
about 5 minutes to remake a release of DAHDI, I would keep a release like
2.1.0.4 on hand to fall back on in case this isn't some minor mistake.
DAHDI is still relatively new (about 18 months) and this rears it's head
every
On Mon, Sep 28, 2009 at 1:57 PM, Andrei Verovski (aka MacGuru)
andre...@starlett.lv wrote:
Sound still choppy. Probably I'm missing something very very simple. There are
2 Ethernet interfaces on the server, ,I do not have any actual Zaptel
devices, but as I understood, dahdi is still need for
Hi
Right now during test phase I'm using several XTel clients and 1.6.1 server
all on local network. Pings accross local network are stable 0.19xx ms.
The same Xten clients connected to external SIP server (not mine) working
fine.
Choppy means repeating fragments of sound are badly
On Mon, Sep 28, 2009 at 2:47 PM, Andrei Verovski (aka MacGuru)
andre...@starlett.lv wrote:
Right now during test phase I'm using several XTel clients and 1.6.1 server
all on local network. Pings accross local network are stable 0.19xx ms.
I've never heard of XTel, but go looking for settings
Hello there!
I'm working on some modifications on Asterisk to adapt it to our needs
considering some particular demandings of the infraestructure we want to
provide.
Two of these modifications are:
1- A proprietary configuration driver that will communicate with a
server that will be the
Hello there!
I really hate when this happens, but...
It seems channel variable SIPCALLID will have the info I need, so the
changes will be reduced to propagate the channel to ARA driver method
realtime_var_get.
If someone have any additional info, or can indicate some problem on
using this
Funny. The first thing I always do after a reboot is call in from my
cell to make sure things work. But last night I rebooted and immediately
tried dialing out (with a TDM842B) and got:
WARNING[2720]: app_dial.c:1721 dial_exec_full: Unable to create channel of
type 'DAHDI' (cause 34 -
I also found this weird, I thought my equipment was the problem. Good to know
about this issue so, Digium takes care of the problem.
I'm running:
asterisk-1.6.1.5
dahdi-linux-2.2.0.2
libpri-1.4.10.1
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On 09/28/2009 01:06 PM, Danny Nicholas wrote:
Funny. The first thing I always do after a reboot is call in from my
cell to make sure things work. But last night I rebooted and immediately
tried dialing out (with a TDM842B) and got:
WARNING[2720]: app_dial.c:1721 dial_exec_full: Unable to
Jeff LaCoursiere wrote:
It would be really cool if iaxmodem would actually answer an incoming
modem call and pass traffic to something like pppd. For those times when
the pstn link is up, but something is wrong with the 'net connection...
I have a working setup of what you describe.
I'd appreciate it if someone was able to assist.
Running the command:
dahdi_hardware:
pci::08:08.0 wcte12xp+d161:8000 Wildcard TE121
dahdi_scan:
[1]
active=yes
alarms=REC
description=Wildcard TE121 Card 0
name=WCT1/0
manufacturer=Digium
devicetype=Wildcard TE121 with
http://www.noojee.com.au/Page/NoojeeClick
Built for firefox.
PaulH
Stefan Schmidt wrote:
Hello,
iam searching for an Firefox plugin which can make an sip Invite and
Redirect after 200 OK, so i dont have to use a softphone, just to
initialise a call by clicking on a number
i've found
1) I have not seen a blue light (usually red/yellow) before on a Digium card
and so don't really know what it means.
2) Try to see if you can see any messages coming up from the Asterisk box
itself (not thru putty or other remote console). You should see a steady
stream of error messages
Many thanks John of Sydney.
I removed the CRC4 and it worked straight away. Can you recommend a CRC type
at all or would it be best to leave it as nothing?
David of Brisbane.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Many thanks John of Sydney.
My pleasure my fellow Asterisker!
I removed the CRC4 and it worked straight away. Can you recommend a CRC
type at all or would it be best to leave it as nothing?
Just leave it blank would do.
-Original Message-
From:
On Mon, Sep 28, 2009 at 04:09:43PM -0500, Shaun Ruffell wrote:
On 09/28/2009 01:06 PM, Danny Nicholas wrote:
Funny. The first thing I always do after a reboot is call in from my
cell to make sure things work. But last night I rebooted and immediately
tried dialing out (with a TDM842B) and
2009/9/26 Alec Davis siva...@paradise.net.nz
When did that happen? Added to libpri, someone beat me to it.
Hello Alec,
In this question, I was refering to your ongoing work
I'm sorry if this question let anyone believe this ISDN subaddresswas
already done and pushed into libpri trunk.
On Mon, Sep 28, 2009 at 1:09 PM, David Backeberg dbackeb...@gmail.com wrote:
On Mon, Sep 28, 2009 at 12:30 PM, sean darcy seandar...@gmail.com wrote:
Well one of the problems it that SendFax doesn't like the tiff file(BTW,
SendFax from app_fax.so gives you clue what the problem is). It requires
I'm interested, and I expect others will be on how you might use it.
Our use on the mantis bug, is to allow 3 ISDN connected sites (no reliable
internet) each running asterisk, to dial other staff members in the other
branches.
The key to this working is the subaddress being populated with the
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