Viktor,
Thank you
I have tried as advised but the telco is still seen all the E1 channels as “externally
blocked” (extblock) channel?
And Until that “extblock” is cleared on the telco
side, the call may not go thru.
I will be glad, if I can have a sample
config from anybody
Hi everyone,
i am face with an asterisk use management interface, at the pressent, i am using AMP (asterisk Management Portal:
http://coalescentsystems.ca/index.php?option=com_content&task=view&id=31&Itemid=57 ). Does anyone know a better and more documented management interface for * ?
Thanks
Tomislav Parčina wrote:
Does anybody use UTStarcom F1000G Wi-FI VoIP phone?
http://www.utstar.com/Solutions/Handsets/WiFi/
I'm planning to buy one and I need to know did you have any problems with
phone. What is the sound quality? How close you need to be to the access point?
Please, any infor
Can you try not recording for a bit and see if that helps?
--
Cheers,
Matt Riddell
___
http://www.sineapps.com/news.php (Daily Asterisk News - html)
http://freevoip.gedameurope.com (Free Asterisk Voip Community)
http://www.sineapps.com/rssfeed.php (Da
Hello,
I'm using these phones and I have also flashed them with new images. But, I
use them with IAX2 ... If that helps you contact me off-list, pls. Settings
are like any other PA168/PA1688 based phone ..
Jürgen
Good Evening from Toronto.
I have just installed [EMAIL PROTECTED] 2.6 and
Ditto on the volume issue, it's EXTREMELY low, but the latest firmware does
allow login to web-portal protected billed wifi systems.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vahan Yerkanian
Sent: Thursday, March 02, 2006 12:03 AM
To: Asterisk Use
Vladyslav wrote:
Just my couple notes on spa3000 and PSTN DTMFs.
Such schema:
PSTN <-> SPA3000 <-> Asterisk
Have problems with DTMF detection on incoming calls
when call comes from cell phone. Once per 4 times it
misdetect some ditigs (whether first digit will be
doubled or unrecognized at all
Hello Tomislav,
I borrowed F1000 from my friend for testing. I am not sure if that is different
from F1000G, but I am experiencing the following issues:
1. As a user, it is not easy to get a firmware update as I need to have a
service contract.
2. Even with the latest firmware I got from sipgate
wendell hamilton wrote:
Ditto on the volume issue, it's EXTREMELY low, but the latest firmware does allow login to web-portal protected billed wifi systems.
You don't want web-portal based auth schemes for your metropolitan wifi
network, as someone can use your network as a transport with two
sox -V foo.mp3 -t au -r 8000 -U -b -c 1 foo.ulaw resample -ql
Chris
- Original Message -
From: "Tomislav Parčina" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Wednesday, March 01, 2006 11:58 AM
Subject: [Asterisk-Users] MOH native files
W
Title: HDLC error
Can anyone help and point me in a useful direction. I'm using * 1.2.4 with Zaptel 1.2.4. I have a TE110P card and it’s a Supermicro P8SCT mobo. If I run the PRI card in the PCI-X slot it shares an interupt with eth0 but I don't get problems. I've been trying to move it o
On Thu, 2 Mar 2006, Vladyslav wrote:
Just my couple notes on spa3000 and PSTN DTMFs.
Such schema:
PSTN <-> SPA3000 <-> Asterisk
Have problems with DTMF detection on incoming calls
when call comes from cell phone. Once per 4 times it
misdetect some ditigs (whether first digit will be
doubled or u
Yep, I tried it and indeed, it lowers cpu usage, so I switched from wav to
gsm format and Im thinking about doing the ramdisk solution for recording...
Sounds like a good move?
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Matt Riddell [NZ]
|Sent:
For most of my everyday needs I ended up coding my own
small one in MySQL and PHP, does the job I need but its far from complete.. to
me, AMP is still the king :)
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dumpolid
ExeplishSent: Thursday, March 02, 2006 1:59
Bayrouni wrote:
Can I run asterisk under normal user?
Or any other user than root?
Try this: http://www.voip-info.org/wiki/view/Asterisk+non-root
--
Best regards,
Bartosz Piec
___
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Asterisk-Users
Anton Krall wrote:
> For most of my everyday needs I ended up coding my own small one in
> MySQL and PHP, does the job I need but its far from complete.. to me,
> AMP is still the king :)
>
> Hi everyone,
> i am face with an asterisk use management interface, at the pressent,
> i am using AMP (ast
On Wed, 1 Mar 2006, Marcus Hofbauer wrote:
> Which chan_capi Version should I apply this patch to?
>
> I'm using chan_capi-cm-0.6.4. If I try to patch I get this message ...
>
> patching file chan_capi.c
> Hunk #1 FAILED at 3354.
> 1 out of 1 hunk FAILED -- saving rejects to file chan_capi.c.rej
On Thu, 2006-03-02 at 10:59, [EMAIL PROTECTED] wrote:
> On Thu, 2 Mar 2006, Vladyslav wrote:
> > Just my couple notes on spa3000 and PSTN DTMFs.
> > Such schema:
> > PSTN <-> SPA3000 <-> Asterisk
> >
> > Have problems with DTMF detection on incoming calls
> > when call comes from cell phone. Once p
Hi,
what about the Asterisk PBX Manager:
http://www.thirdlane.com/opensource.htm#manager
It's based on webmin and well documented.
Stefan
Am Donnerstag, 2. März 2006 09:55 schrieb
[EMAIL PROTECTED]:
> Message: 9
> Date: Thu, 2 Mar 2006 08:59:23 +0100
> From: "Dumpolid Exeplish" <[EMAIL PROTEC
I am sorry, but I don't understand the answer.
At least in Italy Human resources department doesn't undertand a bit about
Hardware supported by asterisk.
We are moving a medium factory from a traditional pbx to an asterisk
solution.
The Human operator now has a kind of hardware.
I would like to k
thanks for this info, I have some doubts
If I had already installed AMP , but I want to have
PBX Manger installed , so that I can use both of them
and compare each other
will it cause problem if I install PBX manager
, if there is already AMP installed
Thanks
Jos
On Thu March 2 2006 19:22, "Stefan-Michael. Guenther (in-put GbR)"
<[EMAIL PROTECTED]> wrote:
> Hi,
>
> what about the Asterisk PBX Manager:
>
> http://www.thirdlane.com/opensource.htm#manager
>
> It's based on webmin and well documented.
>
> Stefan
Hi Stefan. What documentation have you found f
On Thu March 2 2006 19:32, John Joseph <[EMAIL PROTECTED]> wrote:
> thanks for this info, I have some doubts
> If I had already installed AMP , but I want to have
> PBX Manger installed , so that I can use both of them
> and compare each other
>will it cause problem if I install PBX manage
Good Day
Everybody,
I am in the process of planning a
phone system for a small business (15 extensions - 4 PSTN lines [to be connected
by ATAs]). The plan is to install an IP Phone everywhere there is an existing
computer workstation - using the same LAN for phones and
computers.
The l
Hi Nick,
the english documentation is included in the webmin module, you can access it,
when you click on the "help" link in the upper left corner.
If you prefer a german documentation:
http://www.in-put.de/voice-over-ip/PBX-Manager_4-0_de.pdf
Stefan
Am Donnerstag, 2. März 2006 10:29 schrieb
Dan Elder wrote:
Hey all, I've been seeing this repeatedly and am wondering if anyone has a clue
what's causing it.. at least once a day I see two zap fxo channels being
bridged, and hanging..now, these two channels should never bridge, but they
keep doing it.. any leads on where to look for w
Hi,
Am Donnerstag, 2. März 2006 10:32 schrieb John Joseph:
> thanks for this info, I have some doubts
> If I had already installed AMP , but I want to have
> PBX Manger installed , so that I can use both of them
> and compare each other
>will it cause problem if I install PBX manager
> ,
You don't seem to have disconnect supervision enabled.
Julian.
On 3/2/06, Warren Burstein <[EMAIL PROTECTED]> wrote:
[...]
> One additional mystery is that I don't know why these calls persist.
> When I hang up either of the bridged extension on my test system, the
> bridged call ends. When a si
Hello list,can anyone give me sample conf file for TE406Any help will be appreciated.ThanksLito
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To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mail
In article <[EMAIL PROTECTED]>, [EMAIL PROTECTED] says...
> You need to use mpg123 to convert the mp3 files to wav files first.
>
> mpg123 -w out.wav in.mp3
This one works. Thank you!
> sox out.wav -r 8000 out.gsm
I have problem with this command. It runs fine, but when I play that file it is
In article <[EMAIL PROTECTED]>, [EMAIL PROTECTED] says...
> sox -V foo.mp3 -t au -r 8000 -U -b -c 1 foo.ulaw resample -ql
>
> Chris
This is what happens.
[EMAIL PROTECTED] mohmp3]# ls
fpm-calm-river.mp3 fpm-sunshine.mp3 fpm-world-mix.mp3
[EMAIL PROTECTED] mohmp3]# sox -V fpm-calm-river.mp3 -
Hi,
some days ago we discused here the need for an analyzer
for the 1000 Hz tone, as opposite application to Milliwatt.
Here it is: Mwanalyze
http://planinternet.net/download/voip/asterisk/app_mwanalyze.c
It performs a Fourier analysis for a fixed frequency
and tells the amplitude.
The freque
Hi all,
I am
new to asterisk.How can we direct an incoming call to a specific
IP:port.
I need to redirect some specfi calls to one device , rest should be routed normally.
Thanks in advance
regards
Vikram
___
--Bandwidth and Colocation prov
On Thu, Mar 02, 2006 at 10:22:29AM +0100, Stefan-Michael. Guenther (in-put GbR)
wrote:
> Hi,
>
> what about the Asterisk PBX Manager:
>
> http://www.thirdlane.com/opensource.htm#manager
>
> It's based on webmin and well documented.
As it was not mentioned explicitly, and the URL is a bit misle
You need to install either libmad or libmp3lame.
Sox checks for this on startup.
Chris
- Original Message -
From: "Tomislav Parčina" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Thursday, March 02, 2006 10:46 AM
Subject: [Asterisk-Users]
I have tried to use native music on hold. In dir /var/lib/asterisk/moh-native/
I have some wav files (with 755 permission). In musiconhold.conf I have
[native]
mode=files
directory=/var/lib/asterisk/moh-native
And in sip.conf I have
musicclass=native
When I put call on hold this is what I get a
Hi!
This is the debug info what I get now:
CONNECT_IND ID=001 #0x1e87 LEN=0041
Controller/PLCI/NCCI= 0x102
CIPValue= 0x10
CalledPartyNumber = default
CallingPartyNumber = <21 83>69910204082
CalledPartySubaddress =
Hi Group,
I was able to install Asterisk and its addons successfully. Now I want to
eliminate sip.conf and extensions.conf and use everything from Mysql DB, Is
this possible? I have seen this page
http://www.voip-info.org/wiki/index.php?page=Asterisk%20extensions%20from%20mysql
and learnt that w
In article <[EMAIL PROTECTED]>, [EMAIL PROTECTED] says...
> But when a call enters queue_1 or queue_2 it allways rings everyone directly
> without checking if Agent1 is available or not. It should distribute the
> calls from queue_1 to the other agents only when agent/1 is unavailable and
> agent
I guess it doesn’t work by default
on my phone. You still need to press hash to transfer calls. The transfer button
doesn’t work. Where do I set it?
Regards
Mark Brooker
T: 02 4959 8670
M: 0415 846 865
F: 02 9882 0947
E: [EMAIL PROTECTED]
W: http://www.mbit.com.au
what do u mean by solution ? please define what you
want.
--- ram <[EMAIL PROTECTED]> wrote:
> Hi
>
> did you got any solution
>
> iam also looking the same solution
>
> if you find kindly tell me which solution works for
> u
>
> ram
>
>
> On 2/27/06, Micke Andersson <[EMAIL PROTECTED]>
>
One thing to keep in mind when someone says "Asterisk does that by default" is that a lot of people have AMP installed, and an AMP installation includes extra configuration and features as well as the web interface. It may be that there is phone-specific config installed with AMP that is not instal
Did you set immediate=yes in capi.conf?
On Thu, 2 Mar 2006, Marcus Hofbauer wrote:
> Hi!
>
> This is the debug info what I get now:
>
>
> CONNECT_IND ID=001 #0x1e87 LEN=0041
> Controller/PLCI/NCCI= 0x102
> CIPValue= 0x10
> CalledPartyNumber
If you want to try and get it working try reading through
http://lists.digium.com/pipermail/asterisk-users/2006-February/147198.html.
I had a tough time stamping out echo to far-end analog pstn connections.
Stagg Shelton
www.oneringnetworks.net
Imran Ahmed wrote:
Use:
modprobe wct4xxp vpm
In article <[EMAIL PROTECTED]>, [EMAIL PROTECTED] says...
> Hello Tomislav,
>
> I borrowed F1000 from my friend for testing. I am not sure if that is
> different from F1000G, but I am experiencing the following issues:
> 1. As a user, it is not easy to get a firmware update as I need to have a
>
AMP is being run but it seems the transfer
needs to be configured in the phone somewhere so when you press the transfer
button its like hitting #.
-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark Aufflick
Sent: Thursday, 2 March 2006 10:5
>Actually,
>
>I believe that something is wrong with the way
asterisk >implements the
>whole rfc2833 in rtp.c , moreover, the default value
of >100ms in
>dtmf_tones[] in do_senddigit() inchannel.c is to
short >to be detected
>for lots of commercially available fxo gateways.
I can't agree on thi
Hi Group,
Please read my previous message below, I want to configure Asterisk with Mysql
and make Asterisk dynamic so that Asterisk will read everything from Mysql and
we can make changes to mysql data directly. Please tell how can we do this and
point me to related documentation.
Thanks for you
Hi,
I am using 1.2.3, and sometimes I can see the
following message:
Mar 2 08:42:42 WARNING[25937] ast_expr2.fl:
ast_yyerror(): syntax error: syntax error, unexpected TOK_PLUS, expecting
TOK_MINUS or TOK_COMPL or TOK_LP or TOKEN; Input: +
1 ^Mar 2 08:42:42 WARNING[25937] ast_expr2.fl:
On 09:44, Thu 02 Mar 06, Dov Bigio wrote:
> Hi,
>
> I am using 1.2.3, and sometimes I can see the following message:
>
> Mar 2 08:42:42 WARNING[25937] ast_expr2.fl: ast_yyerror(): syntax error:
> syntax error, unexpected TOK_PLUS, expecting TOK_MINUS or TOK_COMPL or TOK_LP
> or TOKEN; Input:
>
Hi,
when I get a SIP call from an unknown user, I can
see the IP address in the channel name.
When the call comes from a known user (sip friend),
I can see only the username in the channel name.
Ok, most users will use the IP address, which they also
register, thus can be lookup up in the regis
On Wed, Mar 01, 2006 at 10:39:36PM +0100, Rene Kluwen wrote:
> I have the same "problem".
> My solution is differentiate in extensions.conf, since all calls are
> terminated to different MSISDN's.
>
> So in extensions.conf I have something like:
>
> [incoming]
> exten => 9995551212,1,Goto(company
Hi
I'm browsing around the internet looking for signs that the IAX client
library and app_meetme support video.
I stumbled across this post by SteveK on the 27th of Feb 2006.
"My company is looking to hire a full-time developer, who will be working
about 25-50% of the time on iaxclient; in pa
On Thursday 02 March 2006 11:46, Dana Harding wrote:
> Good Day Everybody,
Hi.
> I'm convinced that a QoS-based approach is the way to
> ensure file transfers do not interrupt VoIP traffic.
> (not to mention the fact that I don't want to have to
> dig a ditch in the ground to run another cable
Michiel van Baak wrote:
On 09:44, Thu 02 Mar 06, Dov Bigio wrote
Mar 2 08:42:42 WARNING[25937] ast_expr2.fl: ast_yyerror(): syntax error:
syntax error, unexpected TOK_PLUS, expecting TOK_MINUS or TOK_COMPL or TOK_LP
or TOKEN; Input:
+ 1
^
Mar 2 08:42:42 WARNING[25937] ast_expr2.fl: If you
Hallo!
I had ordered a Sipura SPA-3000 in the UK, but the supplier turned out
to be unreliable and never shipped.
Yesterday I went looking for alternative suppliers and found the Linksys
SPA3000 device. It's a different box, but the specs look very similar.
Is this the same device? Has anyone u
Is there a free linux tool which can test voip call
quality between two Asterisk PBX.
It will help me to test the WAN network between them.
I have only found commercials ones, so if you know a
free one, let me know.
___
--Bandwidth and Col
Hi,
The documentation for GoToIfTime(times,days_of_week,...) says "times is
a list of one or more time ranges". I cannot actually see how I can set
the list of time ranges for this application. In fact I think I cannot
put more than one time range to specify morning and afternoon hours.
Check
Hi Frank,
Linksys bought Sipura some time ago (and Cisco owns Linksys btw).
I'd say it's a pretty safe bet that it's the same box.
I just recieved word from Sipura that the following products
are being 'end-of-life'd ':
* SPA841, EOL per December 2005
* SPA2002, Limited supply till Mid April an
On 08:32, Thu 02 Mar 06, Doug Lytle wrote:
> From what has been discussed in the last month, it would indicates that
> it's a variable that hasn't been defined before doing a math function.
> (i.e. not setting a=0 before doing an a=a+1)
Hhmm, Only thing in my setup that looks like it can go wron
amaury BOSSE schrieb:
Is there a free linux tool which can test voip call quality between two
Asterisk PBX.
It will help me to test the WAN network between them.
I have only found commercials ones, so if you know a free one, let me know.
Hi,
just some hours ago I published in this list:
Try and download the correct sip.cfg for
your boot ROM ver from here and see if it corrects the problem. We use AMP with
these files and we never had an issue with the transfer button not working.
http://www.freedomphones.net/polycom/files/
Make sure that you reboot the phone after
On 02/27/06 19:17 [EMAIL PROTECTED] said the following:
the callgroup/pickupgroup settings are correct...
dialing *8 or *8# on any client (zap/sip/sccp) results in unknown
extension...
i can confirm that this bug exists in 1.2.4 as well. we've managed to fudge
it by dialplan tricks and Pick
hi Vikram,
this is quite simple if the incoming call has DID.
if it does, then
1. in the AMP extension define the IP:port as a SIP extension.
2. using the AMP you go to the INBOUND Routing. enter the DID you want to route to the IP:port.
3. enter the extension numer you have defined in step 1 abov
Great, Thank you for contributing!
They best thing to do would be open up a isue on the bugtracker.
http://bugs.digium.com
Go ahead and add this as well as any other info.
I would do this but you as the author need to do it.
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[
I am having an issue when parking calls that its taking sometimes up
to 15 seconds for the annoucing position to come back. The calls come
in through a TDM04B and many times it takes at least 15 seconds for
the call position to be announced.
Thanks
--
Tom Vile
Baldwin Technology Solutions, Inc
C
In article <[EMAIL PROTECTED]>, [EMAIL PROTECTED] says...
> i can confirm that this bug exists in 1.2.4 as well. we've managed to fudge
> it by dialplan tricks and Pickup().
Please report the bug.
In 1.2.1 it works fine.
--
Tomislav Parcina
tparcina#lama.hr
Hi,
we use a Grandstream HandyTone 386 analog adapter.
At port one there is a fax machine connected. If the fax is connected to a
remote fax machine a second incoming call does not get a busy signal.
Does this belong to my dialplan or my sip registration settings?
Thanks and Regards,
Bernd
I got a complaint this morning that no one was able to call out on our
phone system. I went into asterisk and did "show channels" it said, 0
active calls and 13 active channels. How can their be 13 active channels
and no calls? I rebooted the system and everything seems fine now.
Anyone have an
On Mar 2, 2006, at 2:15 AM, Vahan Yerkanian wrote:
Reboot once again and it picks up the new config. Two-step
provisioning takes a couple of reboots to insure the device has
reconfigured itself. Applies to 2100, 3000, 841 and 941 models.
I've had good results on our 942 by setting the resyn
yes ...
2006/3/2, Armin Schindler <[EMAIL PROTECTED]>:
> Did you set immediate=yes in capi.conf?
>
> On Thu, 2 Mar 2006, Marcus Hofbauer wrote:
> > Hi!
> >
> > This is the debug info what I get now:
> >
> >
> > CONNECT_IND ID=001 #0x1e87 LEN=0041
> > Controller/PLCI/NCCI= 0x102
> >
Also, SATA on a onboard SATA card will eat more CPU then a SCSI system.
Are you running software RAID by chance with your SATA? SCSI or SCSI
Raid will not each CPU near as much since the HBA does all the work and
does tie up the CPU with all its I/O's. We have successfulyl recorded
5+ calls at a ti
On 03/02/06 19:30 Tomislav Parèina said the following:
What have I done wrong? That file IS in that directory.
what are the file permissions/ownership and are they readable by the
asterisk process ?
--
Regards, /\_/\ "All dogs go to heaven."
[EMAIL PROTECTED]
Hi all,
Is there a solution to solve this ?
ASTERISK 1.2.4
||
Internet===SER/OPENSER=Nat==[private net]
|| sip agents
rtpproxy/mediaproxy
Sip agents use SER/OPENSER as an outbound sip proxy
and asterisk as a registar se
Here my capi.conf ...
[EMAIL PROTECTED] ~]# vi /etc/asterisk/capi.conf
;
; CAPI config
;
;
; general section
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
language=de
;ulaw=yes;set this, if you live in u-law world instead of a-law
; interface sections ...
[ISD
Hi,
playing with Asterisk native attended transfer, I noticed that, while
waiting for transferee to respond, I can take again original conversation
only dialing the atxfer sequence many times successively (4-5, in quick
sequence).
Why not I can hit it only one time?
Mimmus
___
Hi, this is the second time that i post this, may be a wasnt clear the
first time.
Im having problems with an incoming peer after i upgraded asterisk from
1.0 to 1.2.4, in 1.0 i used to configure the incoming peers like this:
register => @prepago-in
[prepago-in]
type=friend
host=192.168.10.102
Thanks :) When we were using Mark2 with aggressive suppression, we had
zero problems, but decided to go with the hardware canceler in our new
gateway since hardware's supposed to be better than software...
hopefully this works for us too.
Aaron
Stagg Shelton wrote:
If you want to try and get
A good hardware SATA raid card should use less CPU time as well. I like to use 3ware cards myself.On 3/2/06, Ron McCarthy <
[EMAIL PROTECTED]> wrote:Also, SATA on a onboard SATA card will eat more CPU then a SCSI system.
Are you running software RAID by chance with your SATA? SCSI or SCSI
Raid will
Could I use this to distinguish human voices from
machine beeps and/or ambient noise etc, by (after a few adaptations) calling
it a number of times on the same set of samples with some representative set
of frequencies? Or is there a better, less CPU-torturing way to do
that?> -Original Message
You don't need switchport mode trunk when using switchport voice vlan.. On 3/1/06, Nicholas Kathmann <
[EMAIL PROTECTED]> wrote:Joao Pereira wrote:> Hello to all
> I would like to know If some of you have already configured an Cisco> IP Phone (7940 or 7960) to work in a different VLAN than the PC t
I'm guessing Digium is having issues with their mailing list. I just
received the following:
Your membership in the mailing list Asterisk-Users has been disabled
due to excessive bounces The last bounce received from you was dated
10-Feb-2005. You will not get any more messages from this list
It actually depends on the switch model. Some put the port into
trunking mode automatically with the sw voi command, and some do not.
Hopefully one day Cisco will finally make their own products and become
uniform instead of buying several companies and glue'ing them all
together to get an ethern
Juan Carlos Castro y Castro schrieb:
Could I use this to distinguish human voices from machine beeps and/or
ambient noise etc, by (after a few adaptations) calling it a number of
times on the same set of samples with some representative set of
frequencies? Or is there a better, less CPU-torturi
> In article <[EMAIL PROTECTED]>, [EMAIL PROTECTED] says...
> > i can confirm that this bug exists in 1.2.4 as well. we've managed to
fudge
> > it by dialplan tricks and Pickup().
>
> Please report the bug.
>
> In 1.2.1 it works fine.
>
thank you for the information...
hi
on a server i've installed slackware 10.2 and
Asterisk Version 1.2.4
Zaptel Version 1.2.4
Libpri Version 1.2.2
I'm experiencing problem with moh, the music is correctly started:
Started music on hold, class 'default', on channel 'Zap/1-1'
but i don't hear anything... i heard something like
And about the 802.1x ?
The phones can work as passthrough and force the PC to use 802.1x ?
What configuration do we put in the switches? Do we put the switch as
"access" (with 802.1x) or "trunk" (without 802.1x) ?
Thanks
Joao Pereira
Greg Oliver wrote:
It actually depends on the switch mod
Just in my Inbox:
Original Message
Subject: Over 40 destinations for FREE!
From:"[EMAIL PROTECTED]" <[EMAIL PROTECTED]>
Date:Thu, March 2, 2006 17:40
To:
--
De
I have a Toshiba DK424 connected via T1 E&M to a TE110P card. Intermittently when a user dials a number I am getting 'getdtmf' errors on the Ast server and the calls do not go through. If they dial the number once or twice more, it works fine and I receive no DTMF problems.
On another note, end u
Speaking of provisioning the Sipura using the xml files. I ran into a
problem where I'm trying to include in the dialplan that when 9 is
dialed it should not be sent which requires something like this:
|<9,:>1xx| (I might be wrong on the ecact thing, since I'm
writing this from memory)
Howe
Can you explain why you would want asterisk only thru realtime? and
not thru the /etc/asterisk/ ?
The wiki is located at:
http://www.voip-info.org/
the archives for this list is located at:
http://lists.digium.com/
The asterisk irc channel is at:
irc://irc.freenode.net/#asterisk
Google is located
I need your help
I have a sangoma A104D on my dell server; I got card status ok with no alarm
If I dialed the extension 6210006, it shows the output as stated below, but
there is no ringing from the pstn number nor the iax softphone am using on
my pc.
I will be glad if someone can give me a wor
What phones you using?
On 3/2/06, Julian J. M. <[EMAIL PROTECTED]> wrote:
> You don't seem to have disconnect supervision enabled.
>
> Julian.
>
> On 3/2/06, Warren Burstein <[EMAIL PROTECTED]> wrote:
> [...]
> > One additional mystery is that I don't know why these calls persist.
> > When I hang
Tomislav Parčina wrote:
sox out.wav -r 8000 out.gsm
I have problem with this command. It runs fine, but when I play that file it is
twice long as it should be and double slow as it should be. So wav file that
was 2 min long becomes 4 min long gsm file.
How can I fix that?
Tomislav,
Yikes. Managability! It's a lot easier to manage multiple Asterisk systems
configuration from a single MySQL database then it is to manage .conf files on
several redundant Asterisk boxes. I can't believe you asked that question. I'll
apologise in advance because I must be missing part of this th
I have never used a switchport for .1x to a PC connected through a
phone. I would say it probably will not work since it bypasses the idea
of .1x entirely if it does.
You maybe could use it in 802.11 mode, but the phone would probably not
have access until the PC auths (if it would work at all)..
On Thu, 2006-03-02 at 17:51 +0100, Francesco Peeters (Asterisk) wrote:
> Just in my Inbox:
> >From the makers of Voipbuster: http://www.internetcalls.com
>
> Over 40 FREE destinations, PLUS free VoipIn number AND Call Forwarding!
Finerea has sipdiscount.com which also is offering the same deal.
At 06:00 AM 03/02/2006, you wrote:
exten => s-NOANSWER,1,GotoIf($[${DIALEDTIME} = 0]?3)
Looks like the Dial statement is not setting the $DIALEDTIME
in some cases.
This is the general solution to that.
exten => s-NOANSWER,1,GotoIf($["${DIALEDTIME}" = "0"]?3)
Ira
--
No virus found in this
Roger,
Thank you very much for this valuable contribution. In my opinion, this
is a great candidate for asterisk-addons.
Sincerely,
Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer
Roger Schreiter wrote:
Hi,
some days ago we discused here the need for
Looking for a configuration for the "Dial" application that is simular to
the "g" option, but for outbound, not inbound.
Is there a way to configure an outbound call sequence that will continue
to the next priority in the dialplan when the originator disconnects? The
senerio is an outbound call c
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