Hello,
Check if file is owned by asterisk user.
Also, don't directly create in to /var/spool/asterisk/outgoing/
Create in somewhere else first and then move file to outgoing folder.
Good luck.
On Fri, Aug 12, 2011 at 7:09 PM, Danny Nicholas da...@debsinc.com wrote:
Another thought – when a
Selam Yavuzhan,
Backup paneli yerine, elle backup almayi deneyebilirsin.
Ayrica Trixbox icin buradan destek alabilecegini zannetmem.
Trixbox'in kendi forumlarina yaz derim.
Kolay gelsin.
On Wed, Dec 9, 2009 at 10:11 AM, Yavuzhan Canli yca...@tekfen.com.trwrote:
Hi all,
I have installed
/asterisk-users/attachments/20091204/9a953328/attachment-0003.htm
--
Message: 25
Date: Fri, 4 Dec 2009 09:29:35 +0200
From: Hakan C ella4e...@gmail.com
Subject: Re: [asterisk-users] hey please help me my 3rd email of how
to change From fileld username
exten = 111,1,Set(CallerID(num)=123456)
On Fri, Dec 4, 2009 at 8:32 AM, Masood Ahmed masoo...@gmail.com wrote:
hy
Hope everyone is fine, I have one issue coming in asterisk , What i am
doing
is i am generating a callback if some one calls at a specif access number
on
asterisk,
Asterisk
It does nothing on hardware channels.
SendText is just works on SIP channels.
Purpose of SendText is showing text messages on user phone screen.
show application SendText
-= Info about application 'SendText' =-
[Synopsis]
Send a Text Message
[Description]
SendText(text[|options]): Sends
Hello,
Take a look to Switch Converter: http://www.nch.com.au/switch/
Good luck.
On Fri, May 1, 2009 at 9:39 AM, Sai P. Varanasi saiprabhak...@yahoo.comwrote:
Hi,
I am recording 16 bit 48kbps PC audio file in wav format and trying to
convert it to g722 using sox to play in asterisk. But,
Hello,
I had same problem.
Try: ftp://ftp.sangoma.com/linux/custom/Marc/wanpipe-3.3.16.22.1.tgz.
Good luck :)
On Thu, Apr 30, 2009 at 7:10 PM, Jeremy Mann jm...@txhmg.com wrote:
Newest wanpipe (3.3.16) beta drivers do not compile against dahdi-linux
2.2.0-rc2 which is what you get when you
Hello Jonathan,
I think the best way is:
1- Zaptel / DAHDI
2- LibPRI
3- Asterisk
On Thu, Apr 30, 2009 at 7:37 PM, Jonathan Moore supermegat...@gmail.comwrote:
I've read a lot of conflicting information on this around the web, and
wanted to see if I could get some thoughts for any of you..
Dear Aman,
You may create your own dialplan and you can try with it.
Put something like that:
exten = 611,1,Dial(ZAP/port/0114454212111|60)
then asterisk -rx 'dialplan reload', then call 611.
I have no idea about Trixbox, it's Asterisk :)
On Thu, Apr 30, 2009 at 2:50 PM, Aman Dhally
Hello Justin,
You can try with a softphone first.
Good luck.
On Thu, Apr 30, 2009 at 6:37 PM, Justin Piszcz jpis...@lucidpixels.comwrote:
Hello,
I am using an SPA3102, all is working with asterisk 1.4, voice mail,
outbound calling etc, and it even passes the cid name/num to my analog
Hello.
If you configure and install Asterisk first, how it detects if Zaptel or
DAHDI installed?
Thanks.
On Thu, Apr 30, 2009 at 8:15 PM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:
On Thu, Apr 30, 2009 at 11:37:32AM -0500, Jonathan Moore wrote:
I've read a lot of conflicting information
Hi.
Probably you should use SS7.
It depends on your hardware.
On Wed, Nov 19, 2008 at 8:44 AM, mark morreny [EMAIL PROTECTED] wrote:
Hi Andrew,
Thank you for your info. I am actually looking for connecting mobile base
station with asterisk via E1.
Any idea on where I should start
Hello Jon,
Maybe you can think about SNMP support in Asterisk.
Also you can develop custom applications in many languages or take a look to
Nagios (http://www.nagios.org/)
Try that command on your Asterisk box:
asterisk -rx 'pri show spans', it returns PRI status.
Good lucks
On Wed, Nov 19,
Hello.
Enable verbosity first.
core set verbosity 10
and then create a test extension:
exten = _111,1,Answer
exten = _111,n,MusicOnHold
exten = _111,n,Hangup
then try to dial 111 and hangup phone after 10 secs.
and post your CDR configurations if you mind.
Thanks
On Wed, Nov 19, 2008 at 7:51
Hello.
I've never used BRI but you can take a look to wiki.sangoma.com
You can kindly ask to them for support.
Good luck.
On Tue, Nov 18, 2008 at 5:45 PM, Claus Herwig [EMAIL PROTECTED] wrote:
Hello,
there has been a post to this list somewhere arount april which said
that it is possible
?
It doesnt need write something huge.
Hope it helps.
Thanks.
On Wed, Nov 19, 2008 at 2:29 PM, Jon Weisman [EMAIL PROTECTED] wrote:
Thanks Hakan,
I was kind of hoping I wouldn't have to write anything. Anybody else got
something I could just use?
- Original Message -
*From:* Hakan C
Hey,
Yeah, its possible.
You just need a PC with network card and Asterisk.
Read the Asterisk book, http://voipspeak.net/index.php?/content/view/33/2/
Good luck
On Wed, Nov 19, 2008 at 1:10 PM, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hello List,
i would like to set up the following
Hello,
First step:
Stop and Uninstall DAHDI, well theres no uninstall script in DAHDI source,
so just stop it and remove kernel modules.
/etc/init.d/dahdi stop
then go to your /usr/src
dont forget to install your kernel headers and sources, and these packages
are necessary:
gcc
g++
make
Hi,
Did you try relaxdtmf = yes in your Zaptel/DAHDI conf?
On Wed, Nov 19, 2008 at 11:46 AM, Mikel Lindsaar [EMAIL PROTECTED] wrote:
I plug the NEC back straight to the Telco and all works well again.
I just got on the phone to Digium and we've raised a ticket with some pri
intense
Hello.
Set your verbosity to 10 with 'core set verbosity 10' and put a test call,
paste your outputs.
Thanks.
On Tue, Nov 18, 2008 at 3:00 PM, Steve Totaro
[EMAIL PROTECTED] wrote:
On Mon, Nov 17, 2008 at 6:04 PM, Juan Carlos Castro y Castro
[EMAIL PROTECTED] wrote:
Weird thing happening
Hello.
Use 'core show application MeetMe' and see how meetme works, may its makes a
sense.
On Mon, Nov 17, 2008 at 4:43 PM, Giedrius Augys [EMAIL PROTECTED] wrote:
Hello,
I've asterisk 1.4.22. I need to that the first conference user hears
You're the only conference user... . When the
Post your /etc/dahdi/system.conf and /etc/asterisk/chan_dahdi.conf
On Wed, Nov 19, 2008 at 3:16 PM, Jerry Geis [EMAIL PROTECTED] wrote:
/
//
// dahdi_dummy loads as shown.
//
// When compiling asterisk 1.4.22 it compiles fine.
//
// when running I get the message:
// ]
Hey,
I think its normal.
Talk to Sangoma support.
Good luck
On Thu, Nov 20, 2008 at 1:14 AM, Andres [EMAIL PROTECTED] wrote:
Hi,
Does anybody have an idea as to why dahdi_test results drop to
unacceptable levels after doing a wanrouter stop/start using a Sangoma
card? See below that it
Use AGI,
in PHP, something like that :)
$validExtensions = array(320, 302, 314);
$agi-exec(ZapBarge, $validExtensions[1]);
Thats just an example with phpAGI, you can modify it as you wish.
I am using it.
So, limit user input, they can only Barge valid extensions in array.
Also, you can use g
Can you post your dialplan?
On Mon, Nov 17, 2008 at 4:41 AM, Jerry Geis [EMAIL PROTECTED] wrote:
When I upgraded from 1.4 to 1.6 it seems my dialplan extensions.conf
is not loaded. in UPGRADE.txt I dont see any reason why.
^[[1;30m == ^[[0mParsing '/etc/asterisk/extensions.conf': ^[[1;30m
Hello.
Go to Zaptel dir and type
make uninstall
make uninstall all
make remove
Before removing Zaptel, be sure Zaptel is stopped.
/etc/init.d/zaptel stop
There are some files which not removed by make.
If necessary, you can delete these files manually.
But if Zaptel is not loaded, it's not
Hello.
Have you ever tried updating your GCC version?
Thanks.
On Thu, Oct 2, 2008 at 8:30 PM, Satish Patel [EMAIL PROTECTED] wrote:
Regards,
Satish Patel
Quoting Tzafrir Cohen [EMAIL PROTECTED]:
On Thu, Oct 02, 2008 at 11:33:01AM -0400, Satish Patel wrote:
Regards,
Satish Patel
Hello.
I've just installed
asterisk-1.4.21.2
zaptel-1.4.12.1
chan_ss7-1.0.10
libpri-1.4.7
I am using Sangoma A104 card with wanpipe-3.2.7.1 drivers.
My OS: Ubuntu 8.04 Server
Kernel: 2.6.24-16-server
I am getting a choppy GSM playback and too many defunct AGI processes when
channel closes.
i am
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