Displaying 20 results from an estimated 25 matches for "adontendev".
2006 Apr 19
1
How to read ActiveRecord errors
How can ActiveRecord errors be read within the model itself?
I am deliberately putting error records in the form, however after the save
command, the error messages show up in the browser.
How can the ActiveRecord messages be retrieved in the code itself, either in the
model or the controller code, something like a try, except, catch ?
2006 Apr 21
1
Date edit control
Does rails have a datetime control, one of Javascript calendar types.
2005 Jun 10
1
Wildly inaccurate CDR records
My CDR is displaying wildly inaccurate results.
When I make a call the CDR records the time between connecting into the
server and hanging up, instead of recording the time between dialling
from the server to the PSTN destination via VOIP termination.
It is alright to log the duration of the connection to the server, but
why it does not log calls for termination via voip provider is the main
2005 Oct 08
1
Cannot dial SIP via asterisk
I have been trying to connect via sip and things don't seem to work. What do
messages like this mean?
Oct 9 00:33:57 WARNING[22849]: chan_sip.c:611 __sip_xmit: sip_xmit of 0x81ab834
(len 361) to 216.127.66.119 returned -1: Invalid argument
Oct 9 00:33:58 WARNING[22849]: chan_sip.c:694 retrans_pkt: Maximum retries
exceeded on call 000638cf3adb579455c0d20b2051ba1d@127.0.0.1 for seqno 102
2006 Jan 17
1
Is there a key sequence to stop a call as its ringing?
Is there a key sequence to stop a call as its ringing, before the call is
answered?
The * key stops a call after it is answered, but I'd like a way to cancel the
call during the ringing phase.
/Obelix
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2006 Apr 12
2
How to terminate ringing call before it is answered
Is there a way to terminate a ringing call before it is answered?
I am speaking of prepaid card application in which you want to make another
call, because you current number it is not being answered, and you don't want
to hangup before dialling again.
/Obelix
2006 Apr 13
2
How to terminate ringing call before it is answered?
Is there a way to terminate a ringing call before it is answered?
I am speaking of prepaid card application in which you want to make another
call, because the current number it is not being answered, and you don't want
to hangup before dialling another number.
/Obelix
2006 May 21
1
Events offered by
Which Actions and events to the read/write options in manager.conf give access
to, ie the options below.
read = system,call,log,verbose,command,agent,user
write = system,call,log,verbose,command,agent,user
Are they documented somewhere?
/Obelix
2005 Jun 06
2
Variables and status problems in AGI application
I am running a prepaid application with Asterisk. When authentication has to be
done by DTMF everything works fine. However when the user is authenticated
directly from the sip phone, the channel variables seems to disappear.
Trying to retrieve the channel status always returns -1 instead of the 6 that
happens normally. It also seems to affected the DIALSTATUS and ANSWEREDTIME
variables.
The
2006 Apr 19
0
My database model''s connection appears messed up.
I am trying to add data from a form, and the rails doesn''t appear to be making
any effort at all. I''ve put deliberately erroneous data, but still nothing.
All the development log shows is this.
This is the code in use - based on the example from Rails book
def add_customer_admin
if request.get?
@login = CustomerAdmin.new
else
@login =
2005 Jun 25
1
Looking for link.exe to compile G729 codec
I want to compile the G729 codec to try it out with firefly.
I don't have Visual C++ 6 compiler. Is there a way I can obtain the link.exe
alone for use with cygwin, or a substitute program?
I don't look forward to installing the whole Visual C++ just for the link.exe
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2005 Jul 03
0
how to configure asterisk user and group rights
I'd like to these three things about asterisk:
1. How the asterisk program can be configured to run as a different user from
root.
2. what directories and files it must have read and right access to
3. Setup an asterisk group, which also has some of the rights the asterisk user
has rights to, and what else it can be used for.
Are these some info sources which go into these areas in depth?
2005 Jul 03
1
Repost: how to configure asterisk user and group rights
I'd like to these three things about asterisk:
1. How the asterisk program can be configured to run as a different user from
root.
2. what directories and files it must have read and right access to
3. Setup an asterisk group, which also has some of the rights the asterisk user
has rights to, and what else it can be used for.
Are these some info sources which go into these areas in depth?
2005 Jul 10
0
Problems with firefly connection via SIP
My firefly softphone is having problems connecting via SIP.
When I set it up, one provider does appears to connec, but trying to call
results in a 'Couldn't start call'
The other responses with a 401 failure code.
Xten connects okay via SIP.
Is there something about Firefly SIP configuration that I don't know about?
IAX connects okay
/
Obelix
2005 Aug 02
1
How does TDM work?
How does TDM work, how do you connect to it?
I have the impression it can't be routed like ethernet, but a cable from your
switch has to be plugged into the providers equipment.
I have seen the Asterisk info about TDMoE - does this mean that the Asterisk
card will modulate the signal on the Ethernet cable to allow it plug directly
into a proper TDM connection?
Will someone please enlighten
2005 Oct 01
0
how to backup asterisk installation for upgrade
I want to backup Asterisk based on 1.07 to install 1.09 and 1.2beta on another
server.
Which files and folders do I have to backup, in order to restore if things don't
work right?
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2005 Oct 08
1
How to check what codec translations are in use in a call?
How does one check what codec translations are in use in a call?
I am connecting to sip system which says 488 "4XX Not Acceptable Here". I don't
know what is stopping the call from being accepted and I'd like to know if
there are codec issues involved.
/Obelix
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2005 Oct 14
0
DTMF tones not working with SIP
My Asterisk PBX seems unable to receive DTMF information via SIP. I have tried
all the various methods, rfc2833, inband and info and they all don't seem to
work. IAX2 works fine. Is there something I must be missing
?
/Obelix
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2005 Oct 15
1
Looking for Info on OH323
I have compiled the OH323 module for my system.
When can I find some info on how to properly configure it?
I haven't read any info for its configuration, and I need some starting info.
Were do I start?
Obelix
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2006 May 24
2
What and When is the next version of Asterisk?
What and When is the next version of Asterisk?
/Obelix