Its now up at
http://www.voip-info.org/tiki-index.php?page=Asterisk%20gui%20phpconfig
I would be interested in any feedback. Hope it helps.
I've checked with IE and the numbers on the right point to extensions
defined in the file you are editing. That's a pretty nice feature.
The problem I
On Sun, 27 Feb 2005 12:57:37 -0600, Anton Krall
[EMAIL PROTECTED] wrote:
Guys.. which free softphone is the best,grandest,most recommended one out
there? based on your own experiences..
Shameless plug : http://www.marccharbonneau.com/asterisk/mediaxphone.php ;)
But it only works on Windows for
Anybody knows what one should do to listen to GSM files?
I know QuickTime can play gsm files. Maybe your users that succeeded
had it installed.
hth
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I have been doing various testing with asterisk and its been going great.
However I am a bit feedup of using vi for editing configs, and would rather
do it from any machine on my LAN. I am running debian and * via xorcom rapid
on a test PC at the minute.
I had the same problem. So I did a
Is there any reason to avoid * on Fedora Core 3 at this time?
Have most/all of the issues been resolved now?
I don't know about the issues on FC3, but I wouldn't want to use a
testing distro on a production server.
If you are looking for a stable distro that cost nothing, have a look
at CentOS
Some time ago, I had the same probs with phpconfig and had to search and
google quite a long time to get it running. Since our systems are now
running fine with phpconfig, I simply forgot the above fgetc/fgets issue.
Therefore...
A wonderful place for all this would be the wiki ;-)
Better
I have yet to discover a software package that would both register and
have ulaw codec. The SIP communicator (Java) came closest to usable,
but didn't have the ulaw codec working. What do you use for
communications?
for SIP you can use X-Lite :
How do I config Asterisk so when the directory cmd is used, the name of
the found entry comes from a pre-record gsm file instead of being spelled
letter by letter?
If the user as recorded is name, this file will be used. When it's not
recorded, * will spell it.
Dial to your voicemail and
1) when Asterisk try to build the mailbox directory under the path :
/var/spool/asterisk/...
Don't know about realtime, but in standard version, the directory is
built the first time you leave a message to this mailbox
hth
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In either case (background or backgrounddetect) when I hit 1 or 2 there
is a 5 to 8 second delay AFTER I hit the button before it goes to the
menu.
I think setting the digit timeout would help.
DigitTimeout(seconds) : Set maximum timeout between digits
exten = s,1,Wait,1
Hello
I am using asterisk 1.0.0, here i am facing one
problem that the email-aatchment setting for each
extesion is not working individually.
When globally attach=yes is set the voicemail will be
sent as attachment no matter for any extension if
attach=no is set for it.
Same in
How do I route all the outgoing calls through a SIP gateway, this should
send more than one outgoing call to the sip gateway at once. please show me
the sample configurations on how to do this.
my SIP gatway can accecpt direct IP traffic or SIP proxy traffc.
Do you AT LEAST try to find
My question is how come it was working fine before and after a reboot it
stopped working!? I had the server running for weeks without any problem...
Maybe you updated your Linux, like running yum update, etc
Just a wild guess
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down on a number of unecessary postings, and their assosciated replies from Mr
Critchfield.
Well, that would be bad, because sometimes the answers from Mr
Critchfield just make me roll on the floor laughing.
Keep up the good work Steven ;)
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