Hi, all
Sorry for null subject last mail.
I use Ubuntu 12.04.01 TLS and install asterisk 11.7.0 (tar.gz downloaded
from asterisk.org). We named it Asterisk11.
I want to generate a call file to /var/spool/asterisk/outgoing. This call
will dial out to Local Channel and return to some Extens.
Then
You could save the call file initially to /var/spool/asterisk/tmp, then adjust the permissions
as needed and necessary. Finally copy the call file into the outgoing directory. This also
minimizes the chance that Asterisk tries to execute a partial file, although I don't know
whether one still
...@lists.digium.com] On Behalf Of Rizwan Hisham
Sent: Thursday, November 21, 2013 11:05 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Call files without permission for asterisk to read
Hi all,
I am syncing call files on my secondary asterisk server but without
On Thu, 21 Nov 2013, Rizwan Hisham wrote:
Hi all,I am syncing call files on my secondary asterisk server but
without permission to read for asterisk. So they should be executed when
I grant the right permissions (thats when my primary asterisk server
crashes or shutsdown somehow). But
Have you tried to restart asterisk after setting the correct permissions?
HTH,
Ioan
On Thu, Nov 21, 2013 at 6:04 PM, Rizwan Hisham rizwanhas...@gmail.comwrote:
Hi all,
I am syncing call files on my secondary asterisk server but without
permission to read for asterisk. So they should be
Looking at Eric Wieling's response and the wiki entry he mentioned, the precaution is still
necessary.
jg
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live
Hi all,
I am syncing call files on my secondary asterisk server but without
permission to read for asterisk. So they should be executed when I grant
the right permissions (thats when my primary asterisk server crashes or
shutsdown somehow). But asterisk only tries to read the file at the time of
On Thu, 21 Nov 2013, jg wrote:
Finally copy the call file into the outgoing directory. This also
minimizes the chance that Asterisk tries to execute a partial file...
'mv' not 'cp'
Also, create the file on the same filesystem as the spool directory so
'mv' isn't silently 'promoted' to 'cp.'
Thanks for the responses.
Touching a file after setting permissions does not work. Asterisk only
looks at the new file only, not all the files in the directory.
Restarting asterisk does work, but dont want to do this.
Best way i think would be, as suggested by JG, to sync in a tmp directory
and
As far as I know the linux kernel uses inotify to give Asterisk a hint,
that a new call file is available. Does inotify work in your environment
(external storage device) at all?
Am 18.11.2011 11:29, schrieb Ishfaq Malik:
We have a number of asterisk servers that share a spool directory on an
We have a number of asterisk servers that share a spool directory on an
external storage device (for call recording).
We don't use call files but now are about to just purely for our own
reporting purposes.
Has anyone got any experience on the behaviour of using call files when
several asterisk
On Fri, Aug 12, 2011 at 04:32:09PM +0100, Roger Burton West wrote:
On Fri, Aug 12, 2011 at 12:27:45PM -0300, equis software wrote:
Yes, same server, same filesystem...
I don't do Python, but a web search for shutil.move suggests that it
doesn't reliably use the rename syscall. Might be worth
Hi !
I have a python script that create and move .call files to
/var/spool/asterisk/outgoing
Sometimes...(in this case after 500 successfull calls) Asterisk don´t make
the calls and the .call files are in the outgoing forever...
Any Ideas?
I'm using Asterisk 1.4.22 (in 1.4.36 was the same
On Fri, Aug 12, 2011 at 12:23:22PM -0300, equis software wrote:
shutil.move('/var/tmp/1.call','/var/spool/asterisk/outgoing/1.call')
Are both /var/tmp and /var/spool/asterisk/outgoing on the same
filesystem?
--
_
-- Bandwidth
Yes, same server, same filesystem...
On Fri, Aug 12, 2011 at 12:26 PM, Roger Burton West ro...@firedrake.orgwrote:
On Fri, Aug 12, 2011 at 12:23:22PM -0300, equis software wrote:
shutil.move('/var/tmp/1.call','/var/spool/asterisk/outgoing/1.call')
Are both /var/tmp and
On Fri, Aug 12, 2011 at 12:27:45PM -0300, equis software wrote:
Yes, same server, same filesystem...
I don't do Python, but a web search for shutil.move suggests that it
doesn't reliably use the rename syscall. Might be worth shelling out
to your system's mv command.
R
--
...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Roger Burton
West
Sent: Friday, August 12, 2011 10:32 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] .call files in /var/spool/asterisk/outgoing
On Fri, Aug 12, 2011 at 12:27:45PM -0300, equis software wrote
-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Roger Burton
West
Sent: Friday, August 12, 2011 10:32 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] .call files in /var/spool/asterisk/outgoing
On Fri, Aug 12, 2011
...@lists.digium.com] On Behalf Of equis software
Sent: Friday, August 12, 2011 11:06 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] .call files in /var/spool/asterisk/outgoing
I made 500 calls but not simultaneously. My script checks that there are no
more than 3
] *On Behalf Of *equis software
*Sent:* Friday, August 12, 2011 11:06 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] .call files in
/var/spool/asterisk/outgoing
** **
I made 500 calls but not simultaneously. My script checks
On Sun, May 22, 2011 at 07:05:45PM -0400, Thomas Perron wrote:
This may be an obvious reflection of my Asterisk/Linux/Windows weaknesses
but I want to know in any case!
Can a vb script run somehow on a Linux machine or does it only work on
Windows?
Only on Windows (practically).
If I
I would rather write a new bash script for text and file handing.
I think you can install MONO and run windows stuff... from .net to vbs
On 23 May 2011 08:09, Tzafrir Cohen tzafrir.co...@xorcom.com wrote:
On Sun, May 22, 2011 at 07:05:45PM -0400, Thomas Perron wrote:
This may be an obvious
On Monday 23 May 2011, Thomas Perron wrote:
This may be an obvious reflection of my Asterisk/Linux/Windows weaknesses
but I want to know in any case!
Can a vb script run somehow on a Linux machine or does it only work on
Windows?
AFAIK there is no Linux interpreter for VBS :( But the
This may be an obvious reflection of my Asterisk/Linux/Windows weaknesses
but I want to know in any case!
Can a vb script run somehow on a Linux machine or does it only work on
Windows?
If I were to build a call file script (described in this link
Thomas Perron wrote:
Can a vb script run somehow on a Linux machine or does it only work on
Windows?
Visual Basic is Windows specific.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary Safety,
deserve neither Liberty nor Safety.
--
Hi Doug,
Yes. I have sorted that part out. Also, it seems like the pscp function is
the way that I can tie together the vb script with the logic of the Asterisk
call files learning curve!!
Thanks
On Sun, May 22, 2011 at 8:37 PM, Doug Lytle supp...@drdos.info wrote:
Thomas Perron
On Sun, 22 May 2011, Thomas Perron wrote:
Can a vb script run somehow on a Linux machine or does it only work on Windows?
Virtual machines or Wine may have some possibilities.
I simply want to execute a script that helps me automate the voice
broadcasting/IVR of up to 1 phone numbers.
On Sun, 22 May 2011, Thomas Perron wrote:
Also, it seems like the pscp function is the way that I can tie together
the vb script with the logic of the Asterisk call files learning
curve!!
pscp is a program, not a function. Part of or related to putty as I
remember.
Not a good
I'm the original author of said VB Script.
Steve is right, I had lots of errors - related to the fact that
asterisk watches it too closely and reads the files even before they
are complete - and have since updated it that it first dumps it to a
temp directory, then use a bash script on the linux
Hello,
Thanks for replying.
Answers below:
On 23 April 2011 18:29, Sherwood McGowan sherwood.mcgo...@gmail.com wrote:
On Sat, Apr 23, 2011 at 11:20 AM, Tiago Geada tiago.ge...@gmail.comwrote:
Hi.
Im having trouble setting variables in channel dialplan and re-using them
in Extension
Hi.
Im having trouble setting variables in channel dialplan and re-using them in
Extension dialplan...
Im using the following call file:
Channel: Local/210332450@ZonNew-Outbound
CallerID: ZonNew-Outbound:49:210332450:
MaxRetries: 5
RetryTime: 10
WaitTime: 60
Account: Outbound210332450
Context:
Hi,
Using DumpChan(); Seems that Channel (where the call goes first) is a
sub-channel of Context/Extension (where the call goes on CONNECT) ??
first I have:
Dumping Info For Channel: Local/210332450@ZonNew-Outbound-66c7;2:
Then after:
Dumping Info For Channel:
On Sat, Apr 23, 2011 at 11:20 AM, Tiago Geada tiago.ge...@gmail.com wrote:
Hi.
Im having trouble setting variables in channel dialplan and re-using them
in Extension dialplan...
Im using the following call file:
Channel: Local/210332450@ZonNew-Outbound
CallerID:
Hello Guys,
In the case of a multiserver environment for outbound
automatic calls, can you share you experience and preference between call
files and ami originate ?
thanks
--
*Adolphe CHER-AIME
Network / VoIP Engineer
CCNA, CCNA VOICE, Global VSAT Forum Certified
(509)
On 11-04-20 12:20 PM, Adolphe Cher-Aime wrote:
In the case of a multiserver environment for outbound
automatic calls, can you share you experience and preference between call
files and ami originate ?
I prefer using the AMI as I have better call control. I also get to
Thank you for your answer. I also prefer AMI for its flexibility.
However, i have an application developped in PHP used to make more than
10 calls a day by group of 120 concurrent calls. My problem with AMI is
that client keeps disconnected to AMI server. I use astmanproxy as proxy
On 11-04-20 04:44 PM, Adolphe Cher-Aime wrote:
Thank you for your answer. I also prefer AMI for its flexibility.
However, i have an application developped in PHP used to make more than
10 calls a day by group of 120 concurrent calls. My problem with AMI is
that client keeps
Thanks Paul,
I will take a look at twisted i will let you know.
Regards
On Wed, Apr 20, 2011 at 5:38 PM, Paul Belanger pabelan...@digium.comwrote:
On 11-04-20 04:44 PM, Adolphe Cher-Aime wrote:
Thank you for your answer. I also prefer AMI for its flexibility.
Try Set instead of SetVar.
On Sat, Feb 12, 2011 at 9:59 PM, Dan Dan dani.mani...@gmail.com wrote:
Hi,
I am having trouble passing variables via the call files, here is my call
file via the php:
fputs($oSocket, Action: login\r\n);
fputs($oSocket, Events: off\r\n);
Hi,
I am having trouble passing variables via the call files, here is my call
file via the php:
fputs($oSocket, Action: login\r\n);
fputs($oSocket, Events: off\r\n);
fputs($oSocket, Username: $strUser\r\n);
fputs($oSocket, Secret: $strSecret\r\n\r\n);
fputs($oSocket, Action:
@lists.digium.com
Subject: Re: [asterisk-users] Call files error
How can I do that, and do it with LCR?
2011/2/8 fai...@vopium.com
Why don't you use single callfile and set CLI and other perameters in
dial-plan as unique as you need?
-Original Message-
From: Tamás Dajka tda...@gmail.com
Hi All,
I'm having some troubles with using call files.
I'm trying to establish the following:
- want to use call files to connect two (outside) extensions
- want to use the outbound routes set in FreePBX
- want to set the outgoing callerid for both calls
- want to set a custom CDR field in
Why don't you use single callfile and set CLI and other perameters in dial-plan
as unique as you need?
-Original Message-
From: Tamás Dajka tda...@gmail.com
Sent: Tuesday, February 8, 2011 7:45am
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Call files error
Hi All
However the two calls are placed, the CDRs and the callerids are set
correctly, we can't hear each other. As I saw in the logs, the problem is
that the calls are placed in the same context, and not being connected (
like one call, but with the variable EXTEN changed ).
I'm really confused
This is obvious for the first Channel ( Channel:
Local/0036701234567@CustomCallOut-1/n ), but how to set on the other party?
I tried with Context: CustomCallOut-2/n but didn't worked.
2011/2/8 Sherwood McGowan sherwood.mcgo...@gmail.com
However the two calls are placed, the CDRs and the
-users@lists.digium.com
Subject: [asterisk-users] Call files error
Hi All,
I'm having some troubles with using call files.
I'm trying to establish the following:
- want to use call files to connect two (outside) extensions
- want to use the outbound routes set in FreePBX
- want to set
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Call files error
How can I do that, and do it with LCR?
2011/2/8 [mailto:fai...@vopium.com] fai...@vopium.com
Why don't you use single callfile and set CLI and other perameters
Hi,
The exact problem that I'm experiencing is described at
http://www.spinics.net/lists/asterisk/msg122364.html in an earlier
posting to the mailing list, but I could find no replies to it.
I installed Asterisk using Ubuntu's apt-get and then fixed the mysql
conf (which doesn't load if you
You often don't get cdrs or at least useful ones unless you run the call files
through a Local channel
You maybe already doing this
Can you check the Master.csv and see if it also is recorded incorrectly there.
Is this just an issue with mysql cdrs or something else. In my setups which use
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=
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
--
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
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,
Hello,
I'm trying to call different SIP-accounts to connect them to a
conference.
This is my call-file :
Channel: SIP/test3SIP/test1
MaxRetries: 2
RetryTime: 60
WaitTime: 30
Context: from-conf
Extension: 1000
I get the following in the CLI :
[Mar 22 14:40:26] -- Attempting call on
Not too long ago I needed to do the same thing but apparently you need to
have a separate call file for every call. The dial command didn't work with
an '' separating multiple destinations. I did it through a php script
running via agi.
On 2010-03-22 9:56 AM, jonas kellens
Hi,
Using a 1.4 system in which dialplan is written using extensions.conf, I can
use a custom .call file.
On another system in which dialplan is written using extensions.ael, I can't
use any custom .call file : system keeps replying :
apply_outgoing: At least one of app or extension (or keyword
Local channel will help you send your call through the dialplan.
You can make all your decision there.
If it answers, then the specified application will be execute.
Check this example
http://www.astblog.com/2008/09/18/use-the-power-of-local-channels/
David Klaverstyn wrote:
I have
I have successfully created call files and I can get Asterisk to make
calls based on those files. The problem I have is that it seems you
need to use a Channel for the first leg of the call file. This means I
have to use either a ZAP, SIP or IAX2 channel. What I would prefer to
do is send the
On 6/11/2008 3:01 p.m., David Klaverstyn wrote:
Is it possible to send the first leg of a call file to DUNDi and if not
aviable send over IAX2 or then ZAP?
The call files seem to be limited to a channel and not allow the first
leg of the call to be decided by the path of a context, extension.
On 14 Oct 2008, at 18:05, Christian Victor wrote:
Steven Howes schrieb:
Have created a system that involves using call files in the outgoing
spool folder. On some occasions it retries which is fine is there
any way to view calls waiting retries from the CLI? Using 1.4 btw.
Have googled to
Hi All,
Have created a system that involves using call files in the outgoing
spool folder. On some occasions it retries which is fine is there
any way to view calls waiting retries from the CLI? Using 1.4 btw.
Have googled to no avail (although it is near the end of the day so I
might
Steven Howes schrieb:
Have created a system that involves using call files in the outgoing
spool folder. On some occasions it retries which is fine is there
any way to view calls waiting retries from the CLI? Using 1.4 btw.
Have googled to no avail (although it is near the end of the
Hello again..
I am working on using call files to have a form of ringback - eg if an
extension is busy, the caller can dial a number and when the callee is
free, the call gets made.
I am trying to use a call file, which kind of works okay, however, if
users have voicemail, it connects to
Is there a way to set a call timer on calls created with call files? I'm
looking specifically at having Asterisk hang up the call after a certain
period of connection.
Obviously, when I try passing an |S(time) on the channel line, I get an
invalid call file... so I'm wondering if there's
On Jul 25, 2008, at 11:18 AM, SIP wrote:
Is there a way to set a call timer on calls created with call files?
I'm
looking specifically at having Asterisk hang up the call after a
certain
period of connection.
Obviously, when I try passing an |S(time) on the channel line, I
get an
SIP wrote:
Is there a way to set a call timer on calls created with call files? I'm
looking specifically at having Asterisk hang up the call after a certain
period of connection.
Obviously, when I try passing an |S(time) on the channel line, I get an
invalid call file... so I'm
Mark Michelson wrote:
SIP wrote:
Is there a way to set a call timer on calls created with call files? I'm
looking specifically at having Asterisk hang up the call after a certain
period of connection.
Obviously, when I try passing an |S(time) on the channel line, I get an
invalid call
That worked beautifully. Thanks, Mark.
N.
Mark Michelson wrote:
Mark Michelson wrote:
SIP wrote:
Is there a way to set a call timer on calls created with call files? I'm
looking specifically at having Asterisk hang up the call after a certain
period of connection.
I am trying to use call files that dial and play a wave file
on 3 asterisk boxes console dsp.
This is working.
The 3 boxes are noticeably out of sync. From using 3 different call files
(time to process) I'm sure is the time delay.
Is there a way to get these audios more in sync?
Jerry
Sync the clocks on your asterisk boxen using NTP or whatever, and then
'touch' the call files into the future so each asterisk waits before
processing it...? Might get them closer.
Another option is get all three boxes into the same meetme room, waiting
a few seconds for them to be ready if
I found the decision in using
Channel: Local/[EMAIL PROTECTED]/n
Denis V. Gudtsov пишет:
Hello, All!
How to specify the context in call file section Channel? Is it possible?
I want to dial external number (12345) and connect it to context
notify, which consist of playback() command:
Hello, All!
How to specify the context in call file section Channel? Is it possible?
I want to dial external number (12345) and connect it to context
notify, which consist of playback() command:
Channel: SIP/12345
Callerid: auto 12345
MaxRetries: 3
RetryTime: 40
WaitTime: 50
Context: notify
hi all,
i have the following .call file:
Channel: IAX2/[EMAIL PROTECTED]/myPOTSline
MaxRetries: 2
RetryTime: 60
WaitTime: 30
#
# Assuming that your local extensions are kept in the
# context called [extensions]
#
Context: default
Extension: 156
Priority: 1
when i drop the .call file into the
I've got a curious one: all of a sudden my .call files and my manager
API 'Originate' actions are no longer producing a CSV file. The call
still generates just fine, and Master.csv is updated. However, I don't
get the usual CSV file in the form of xx.csv where xx=account
number.
I
Hi,
I was trying out call file just to see how they worked and my system
does not seem to do anything with them, although asterisk *is* deleting
the files that I put into /var/spool/asterisk/outgoing.
1. I nano'd a quick call file like so:
Channel: SIP/axVoice/910555
CallerID : Leebo
On Tue, Dec 19, 2006 at 10:17:51AM -0500, Lee wrote:
Hi,
I was trying out call file just to see how they worked and my system
does not seem to do anything with them, although asterisk *is* deleting
the files that I put into /var/spool/asterisk/outgoing.
1. I nano'd a quick call file
In the CLI:
sip show peer axVoice
show dialplan main_menu
set verbose 3
Then drop the call file
What is the CLI trace of the above?
Hi, thanks for responding. Please see the output below.
Please note that moving a call file into /var/spool/asterisk/outgoing
did not produce any CLI
On Tue, Dec 19, 2006 at 12:44:39PM -0500, Lee wrote:
In the CLI:
sip show peer axVoice
show dialplan main_menu
set verbose 3
Then drop the call file
What is the CLI trace of the above?
Hi, thanks for responding. Please see the output below.
Please note that moving a call
Tzafrir Cohen wrote:
On Tue, Dec 19, 2006 at 12:44:39PM -0500, Lee wrote:
In the CLI:
sip show peer axVoice
show dialplan main_menu
set verbose 3
Then drop the call file
What is the CLI trace of the above?
Hi, thanks for responding. Please see the output below.
Please note that moving a
Lee wrote:
As I mentioned above, the action of dropping a .call into the
/outgoing directory did not produce any CLI output. I did this
through 2 putty sessions. The first, we setup to watch the CLI output
and the second was to use the commandline to move the .call into the
/outgoing
: Re: [asterisk-users] .Call files do not seem to
work
Tzafrir Cohen wrote:
On Tue, Dec 19, 2006 at 12:44:39PM -0500, Lee wrote:
In the CLI:
sip show peer axVoice
show dialplan main_menu
set verbose 3
Then drop the call file
What is the CLI trace of the above?
Hi, thanks for responding
Colin Anderson wrote:
If you are using Windows to generate the .call files, make sure they are in
Unix format (LF only at EOL, not CR+LF) - Notepad makes bad Unix files. Use
Crimson Editor www.crimsoneditor.com to make the file, and click Document
File Format Unix Format.
I ran into this
Message-
From: Lee [mailto:[EMAIL PROTECTED]
Sent: Tuesday, December 19, 2006 2:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: SPAM-LOW: Re: [asterisk-users] .Call files do not seem to
wo rk
Colin Anderson wrote:
If you are using Windows to generate the .call files
Colin Anderson wrote:
The only other thing that comes to mind is that .call files are very
sensitive to whitespace; you may have unintentially padded the .call file
with whitespace or tabs that it does not like.
The attached .call file works on my 1.0.9 server. Maybe it can give you some
Hi,
if I dial normal with the dial comman I have in my cdr file the peer-name as
source and the CALLERID (number and name) as I have set it in the dialplan.
Now Iam using call files and Iam using in the file for example:
Callerid: name 333
333 will be used for the field src AND the
Hello,
I have a couple of questions:
1) Before heading off for a bit of vacation, I was having a wierd
problem where I was getting more than one call per callfile placed in
the outgoing/ spool. I describe it here:
http://forums.digium.com/viewtopic.php?t=3455
so far, so good - it's not doing
David N. Welton [EMAIL PROTECTED] wrote:
2) app_txfax
I need to know if a fax has gone through or not. My reading of txfax
seems to indicate that it basically just fails, rather than giving me
anything I can work with to try and fail gracefully (letting the user
know that things didn't go
Darren Nickerson wrote:
3) I'm working on a small, simple email-fax system. Just out of
curiosity, what else is out there for Asterisk? I found AsterFax, but
it looks a little bit hairy to set up...
You really should consider HylaFAX - www.hylafax.org. It has what you're
missing - a fully
On Mon, 2006-01-09 at 16:40 +0100, David N. Welton wrote:
Hi,
I thought about using Hylafax, but after looking around a bit, I got the
impression that it's not exactly trivial to integrate it with Asterisk,
and that it will require a dedicated incoming line. Perhaps I'm mistaken?
I thought about using Hylafax, but after looking around a bit, I got the
impression that it's not exactly trivial to integrate it with Asterisk,
and that it will require a dedicated incoming line. Perhaps I'm mistaken?
It isn't that bad basically download compile and install the trick is to
find
Colin Anderson [EMAIL PROTECTED] wrote:
The big weakness in Hylafax is the client. 90% of the time the client will
be under Windows, and your choices are Cypheus, which is pretty and user
friendly but slow and crash-y or WHFC which is ugly and nasty but works
100%
and has slick features like
Hi!
Upgarde to 1.2.1 and try again - 1.2.0 (and maybe the beta) had a bug
concerning .call files and the non-passing on of variables that might
affect you as well.
Cheers, Philipp
Hmmm seems like every dialplan snippet I've seen so far relies on
ResponseTimeout and looping back to s,1. Is
If I generate a .call file to an external callee through my PRI, Asterisk
will not wait to execute the priority in the target context, and instead
will continue on as soon as the channel is dialled. I want it to wait for an
answer, THEN continue. It detects the answer correctly. I have
Colin Anderson wrote:
Weird thing is, I swear this worked the way I wanted it to when I was
running 1.0.9, now it's not under 1.2 beta 1. Am I crazy?
I'm not saying this has been fixed since that point, but why in the
world are you running 1.2.0 beta 1 when 1.2.1 has been released?
-Commercial Discussion
Subject: Re: [Asterisk-Users] .call files on PRI not waiting for answer in
de sired context
Colin Anderson wrote:
Weird thing is, I swear this worked the way I wanted it to when I was
running 1.0.9, now it's not under 1.2 beta 1. Am I crazy?
I'm not saying this has been
, 2005 12:35 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] .call files on PRI not waiting for answer in de
sired context
If I generate a .call file to an external callee through my PRI, Asterisk
will not wait to execute the priority in the target context
First, check your permissions. That seems to be the most common issue
there. Make sure that the asterisk user has full read and right permissions
to the directory and the file.
Second check your logs. After you do this, something like:
grep -i warning /var/log/asterisk/full | tail -n 10
may
Hi all,
Created file 1.call in /var/spool/asterisk/tmp/1.call with.
--
Channel: Local/[EMAIL PROTECTED]
MaxRetries: 2
RetryTime: 60
WaitTime: 30
Context: queue
Extension: 1000
Priority: 2
--
When I move this to /var/spool/asterisk/outgoing/ it doesnt get runned
nothing happens, tried on alot of
You can use (at least in asterisk CVS), this:
Channel: Local/[EMAIL PROTECTED]
then in extensions.conf
[from-internal]
exten = 1234,1,Dial(whatever)
exten = 1234,2,Dial(otherprov)
Not testet though ;)
Julian J. M.
On 4/14/05, Mystery Glitch [EMAIL PROTECTED] wrote:
Can I use the .call files
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