Hi João.
We made up a script that sends received faxes trough a smtp server as an
attachment.
the FAX.ael
context FAX
{
s = {
Answer();
Set(TIMEOUT(absolute)=600); // 10 min
Wait(3);
if(${CALLERID(num)}=) {
I downloaded spandsp0.0.6pre17
I download http://sf.net/projects/agx-ast-addons for app_txfax and found
trunk/app_fax to be newer so I used that.
spandsp compiled fine.
app_fax compiled
when loading I get:
[Apr 5 08:55:54] ^[[1;31;40mWARNING^[[0;37;40m[7505]:
Hi Guys,
i have a small issue but bothering me, after restarting asterisk (version
1.4 running on centos) i have the following message that comes repeatedly
when i am connected to the CLI:
-- Remote UNIX connection
-- Remote UNIX connection disconnected
-- Remote UNIX connection
--
In rtp.conf, there is rtcpinterval = .
If I set this like rtcpinterval = 6 , it make enough intervals
60 seconds .
Default is every 5 seconds.
Is this right ?
Even if I can't stop sending RTCP, delaying the transmission of RTCP will be
good.
; rtcpinterval = 5000 ;
It's not a sign of anything. You get this on the CLI anytime an asterisk
-rx command is executed if your verbose level is greater than 2. core set
verbose 1 or core set verbose 0 makes this go away.
_
From: asterisk-users-boun...@lists.digium.com
Is this Trixbox or some similar system?
Zeeshan A Zakaria
--
Sent from my Android phone with K-9 Mail.
On 2010-04-05 10:36 AM, khalid touati khalidtou...@gmail.com wrote:
Hi Guys,
i have a small issue but bothering me, after restarting asterisk (version
1.4 running on centos) i have the
You probably have a cron job running that executes 'asterisk -rx'
-Dave
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of khalid touati
Sent: Monday, April 05, 2010 10:29 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Thank you Danny, i did set verbosity to 1 and it's gone, thanks to all of
you guys!!
2010/4/5 David Gibbons d...@videon-central.com
You probably have a cron job running that executes ‘asterisk –rx’
-Dave
*From:* asterisk-users-boun...@lists.digium.com [mailto:
can anyone help me out in this, a big number of my faxes are lost everyday!
i would really appreciate any help on how i can tweak asterisk (rxfax) to
receive all faxes!
2010/4/2 khalid touati khalidtou...@gmail.com
i went ahead and i used this line:exten = 3772,n,rxfax(${FAXFILE}|debug)
as it
Setting verbosity to 0 doesn't make it go away, just stops displaying it on
the CLI, and so does it stops displaying a lot of other useful information
which you might actually need to see in your CLI. So first try to figure out
which computer on your network is running asterisk -rx command, and
Thank you so much Zakaria for this valuable infos, i am not using freePBX
but i do think that happened because i did restart asterisk while it was
connected to CLI using another session of putty. but now i have just one
session, so you think maybe if i try to find and kill that sshd process,
that
Pablo Ruiz wrote:
Hello,
Does anyone know when we will see asterisk 1.6.1 (and/or 1.6.2) binary
packages at packages.asterisk.org http://packages.asterisk.org?
Greets.
Packages for 1.6.2 will be available Real Soon Now. It's near the top of my
short list.
They exist, and are sitting
How is your system configured?
Debug output of faild faxes?
This kind of information is needed to help you!
Regards,
Juan
khalid touati wrote:
can anyone help me out in this, a big number of my faxes
are lost everyday! i would really appreciate any help on how i can
tweak asterisk (rxfax)
Unless asterisk -rx is running through this sshd session, it won't help.
Look for what process is doing it. Do you have Munin installed, or some
other monitoring tool to monitor asterisk, or maybe some cron job?
Zeeshan A Zakaria
--
Sent from my Android phone with K-9 Mail.
On 2010-04-05 11:45
Hi Juan,
my system is an asterisk 1.2 on gentoo, it is configured to receive faxes
through rxfax and then to use fax2email to convert the tiff to pdf and send
it to front desk:
exten =
3772,1,Set(FAXFILE=/var/spool/asterisk/fax/${STRFTIME(${EPOCH},GMT-5,%F_%T_${CALLERIDNUM}.tif)})
exten =
I just switched from 1.4.30 to 1.6.2
I initiated a call file - same way in 1.4.30 and nothing happened.
I was not aware of changes in the call file to 1.6.2?
I was watching the cli and no error showed or anything.
In the manager.conf I have things setup.
[MyDial]
secret=
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of khalid
touati
Sent: Monday, April 05, 2010 10:29 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
you did it Bradley! that was two instances running (i don't know when and
how), i gotta confess i'm not that celever to check basic things first.
Zakaria thank you for your help, i'm not using Munin or any cron job when i
looked for sshd sessions i could find just the ones i am using, so that was
Jerry Geis wrote:
I just switched from 1.4.30 to 1.6.2
I initiated a call file - same way in 1.4.30 and nothing happened.
I was not aware of changes in the call file to 1.6.2?
I was watching the cli and no error showed or anything.
In the manager.conf I have things setup.
[MyDial]
I noticed the same thing - i think something about the permissions has
changed, because when I set it to read=all, write=all, it started
working again. Haven't dug around enough to find out exactly what's up
though.
Thanks that works for me again also.
jerry
--
Hi,
Wondering if there is any way to set up an external extension in * where the
external number requires an extension?
So if I have ext. 250 Local/5556667...@outbound, can I add an extension
number to that to be automatically dialed somehow?
Thanks,
Kenny
--
On Monday 05 April 2010 11:31:04 Jonathan Addleman wrote:
Jerry Geis wrote:
I just switched from 1.4.30 to 1.6.2
I initiated a call file - same way in 1.4.30 and nothing happened.
I was not aware of changes in the call file to 1.6.2?
I was watching the cli and no error showed or
Thanks for the update Jason,
How do the upgrades work if v1.6.0 is already install and one wants to
upgrade to 1.6.2 (once it's available)?
yum upgrade asterisk*
???
Thanks
On Mon, Apr 5, 2010 at 11:37 AM, Jason Parker jpar...@digium.com wrote:
Pablo Ruiz wrote:
Hello,
Does anyone
You might be able to do Local/555666,,2...@outbound or
Local/555666www...@outbound.
Untested - give it a shot.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kenneth
Noisewater
Sent: Monday, April 05, 2010
bruce bruce wrote:
Thanks for the update Jason,
How do the upgrades work if v1.6.0 is already install and one wants to
upgrade to 1.6.2 (once it's available)?
yum upgrade asterisk*
???
Thanks
It should be as easy as a `yum update`. That's the goal, anyways.
--
Yes, so this works (maybe safer than read=all and write=all):
read = system,call,command,agent,user,*originate*
write = system,call,command,agent,user,*originate*
I wasted probably a week on this - thanks to no documentation back in the
days with v1.6.
-Bruce
On Mon, Apr 5, 2010 at 1:50 PM,
Hi,
Is it possible to configure Asterisk to fetch for files from the spool
directory in different directories? For example, fetch voicemail files in
/abc/voicemail and call files in /cde/outgoing ?. And is it possible to
configure the filename that Asterisk gives to files, like voicemail
I have a special requirement that insist an Asterisk server, 1.6.1.x, is
used.? I will have 2 SIP trunks coming into the server and I will have
to send calls to these SIP trunks with a round robin distribution
pattern.? I was thinking of using a group count function, if call count
is even
Thanks Andrew,
I have some doubts regarding this issue, firstly, i'm using Asterisk 1.4.21
not 1.6 like the issue you showed to me (
https://issues.asterisk.org/view.php?id=16887) other thing is that i have
many other asterisk servers working good and i never made this change
By the way
On Mon, 5 Apr 2010, Ricardo Coelho wrote:
Is it possible to configure Asterisk to fetch for files from the spool
directory in different directories? For example, fetch voicemail files
in /abc/voicemail and call files in /cde/outgoing ?. And is it possible
to configure the filename that
First off, I also posted this on the digium forums so if anyone here
also reads those, sorry for the cross-post.
When I place an outbound call using SIP to my cell phone, asterisk
immediately starts processing the dialplan without waiting for the call
to be answered. We could handle this on DAHDI
Jerry Geis wrote:
I downloaded spandsp0.0.6pre17
I download http://sf.net/projects/agx-ast-addons for app_txfax and found
trunk/app_fax to be newer so I used that.
You copied app_fax.c from Asterisk SVN trunk?
spandsp compiled fine.
app_fax compiled
when loading I get:
[Apr 5
On Apr 5, 2010, at 8:20 PM, Steve Edwards wrote:
On Mon, 5 Apr 2010, Ricardo Coelho wrote:
Is it possible to configure Asterisk to fetch for files from the spool
directory in different directories? For example, fetch voicemail files
in /abc/voicemail and call files in /cde/outgoing ?.
Hello Friends,
Kind request to you all - *If you would want 6 crore children to have
childhood please sign a petition on
**http://www.indyatweets.com*http://www.indyatweets.com/
* (image on your top right)*
--
With Best Regards,
***
Sarfaraz Chougule
On Mon, 5 Apr 2010, JR Richardson wrote:
I have a special requirement that insist an Asterisk server, 1.6.1.x,
is used.? I will have 2 SIP trunks coming into the server and I will
have to send calls to these SIP trunks with a round robin distribution
pattern.? I was thinking of using a
On Mon, 5 Apr 2010, Ricardo Coelho wrote:
On asterisk.conf it is only possible to specify which directory is going
to be used to store voicemail, call files, etc (astspooldir). What I
would like to know is how to have some directories of the spool in one
path and other directories of the
Hi guys. I am facing this problem here, using a2billing. error: 'Access
denied for user 'a2billinguser'@'localhost' (using password: YES)' I am
following this step by step
http://www.asterisk2billing.org/cgi-bin/trac.cgi/wiki/Installation%20Guideand
wend i get into the point that i have to Create
Fulajtár Pál wrote:
Hi,
I am recently updated my asterisk 1.4.1 to 1.6.2.6 because of fax
service. I have installed spandsp 0.0.5, then 0.0.6 pre 17 as well
because to have support. app_fax is available in menuselect and loaded
into my asterisk. The upgrade went without any problems.
I'm trying to debug a problem with SIP trunks from a provider, but I'm looking
for more detail then what debug is showing me in the console. (I'm running
the Asterisk 1.6.0.10 w/ Trixbox) I'm rejecting incoming calls with a 401
Forbidden, but debug isn't saying why...bad secret, bad domain,
Use tcpdump and wireshark, it'll give you details at packet level. I have
some instructions for it on my blog at www.ilovetovoip.com which may be
helpful to you on how to do it.
Zeeshan A Zakaria
--
Sent from my Android phone with K-9 Mail.
On 2010-04-05 6:15 PM, McCann, Brian
I would suggest you try this. It works:
http://a2billing2asterisk.googlepages.com
On Mon, Apr 5, 2010 at 5:51 PM, Daniel Abreu dlab...@gmail.com wrote:
Hi guys. I am facing this problem here, using a2billing. error: 'Access
denied for user 'a2billinguser'@'localhost' (using password: YES)' I
The problem is due to the wrong password for accessing mysql database.
You can discuss more on asterisk2billing forum
http://forum.asterisk2billing.org
http://forum.asterisk2billing.orgShariq Khan
On Tue, Apr 6, 2010 at 2:51 AM, Daniel Abreu dlab...@gmail.com wrote:
Hi guys. I am facing this
I've been asked for recommendations for a small call centre, an ethernet SIP
deskphone with a wireless headset.
Similar approach would be a mobile phone with bluetooth head set.
Either I've not looked hard enough, or there isn't much on offer.
Alec Davis
--
On Mon, Apr 5, 2010 at 9:37 PM, Alec Davis siva...@paradise.net.nz wrote:
I've been asked for recommendations for a small call centre, an ethernet
SIP deskphone with a wireless headset.
Similar approach would be a mobile phone with bluetooth head set.
Either I've not looked hard enough,
Dear List,
Are there any way of configuring of Asterisk so it'll cache sound files in
memory, and when Asterisk receive a call, instead of loading sound files from
the disk, it will load from the memory and so Asterisk can process much more
call at a time than with faster speed it is not
Are there any way of configuring of Asterisk so it'll cache sound files in
memory,
and when Asterisk receive a call, instead of loading sound files from the disk
Not directly, but it's not really needed. A long as the machine has
enough RAM, the files will be served from RAM by the operating
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