Well, it _was_ up again Friday, and now it's down again Monday! :(
Moises Silva wrote:
Hum, I know Stefan, he is an asterisk-java dev, but he is not online
right now, I will let him know ASAP. Thanks!
On 4/12/07, Doug Garstang [EMAIL PROTECTED] wrote:
Does anyone know who maintains
Err, what happens if someone transfers a call and the new call leg gets
routed through a different asterisk server because the dns changed?
Andrew Latham wrote:
Use round robin on DNS with a replicated DB on each server
On 4/16/07, J. Oquendo [EMAIL PROTECTED] wrote:
Without using Dundi
That used to happen to us _ALL_ the time. Sometimes you'd just have to
press the 'Directory' key and the phone would instantly reboot. It was
very easy to reproduce and Polycom where useless at admitting it might
be a problem. It occurred on several phones. Funnily enough, the phone
it was
I don't know if this is a recent issue... When the read() application is
given a file that does not exist, it aborts the ENTIRE dial plan. That
can't be right. Playback() and Background() don't do this. Couldn't find
a bug in mantis for it...
[Apr 26 17:20:12] VERBOSE[14611] logger.c: --
What a cool idea!
J. Oquendo wrote:
http://etel.wiki.oreilly.com/wiki/index.php/Main_Page
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asterisk-users mailing
No, you can get Asterisk and NFS to work fine together. It was in my
past job, so I can't remember the exact settings, but there was some
magic combination of NFS client mount settings that would cause Asterisk
to return immediately, rather than hang, if there was an NFS
communications
Well, you should be able to leave it open. However, I don't know what
would happen if MySQL times out and disconnects the connection because
it considers it stale. I don't know if you can check that error and
reconnect.
Yehavi Bourvine +972-8-9489444 wrote:
Hello,
I would like to
I remember an app called 'vomit' that could allegedly reconstruct audio
files from tcpdump pcap files.
Salvatore Giudice wrote:
I think you want:
tcpdump -C 100 -W 10 -w /tmp/tcpdump -i eth1 -s 0 udp dst portrange
5060-65534
dst port port
True if the packet is ip/tcp, ip/udp, ip6/tcp or
I have a large dial plan here with over 3000 lines, and several dozen
macros. As it grew, it became apparent that there was some problems.
1. When you pass arguments to a macro in the form of $ARG1, $ARG2 etc,
if that macro calls another macro, and passes arguments like this as
well, you lose
Philipp Kempgen wrote:
Doug Garstang wrote:
I have a large dial plan here with over 3000 lines, and several dozen
macros. As it grew, it became apparent that there was some problems.
1. When you pass arguments to a macro in the form of $ARG1, $ARG2 etc,
if that macro calls another macro
Andreas Sikkema wrote:
You're so right!
I thought about having just a catchall _. extension in the
dialplan and doing everything else in a real language via AGI -
PHP, Perl, ... whichever you like. It would make the programming
part much easier as the scope of variables is just as you
expect it
The polycom lets you do either attended or unattended transfers. If you
want an unattended transfer, you press the 'blind' soft key. It's been a
few months since I've looked at this, so a bit fuzzy on the details.
Jason Adams wrote:
Isn’t that the function of an attended transfer? User3
ChanAvail()
[EMAIL PROTECTED] wrote:
Hello everybody,
Is there a possiblity to check in the dialplan whether a SIP user is
registred?
Something like :
exten = user1,1,GotoIf(isRegistred(user1)? context1, context2, 1)
Thanx,
Kalle
___
Does anyone know how I could get the SayUnixTime application to say
files from a different sound directory?
It looks like it uses the language as a base to determine where to play
sound files from. I need to override that.
Thanks,
Doug.
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What's wrong with this?
exten = s,n,Set(CIDNUM=16505551212)
exten = s,n,Set(foo=${REGEX(^[0-9]+$ CIDNUM)})
This always returns 0, false. That isn't correct.
I also tried:
exten = s,n,Set(dollar=$)
exten = s,n,Set(foo=${REGEX(^[0-9]+${dollar} CIDNUM)})
and that didn't work either. I am trying
You need operator=yes as well...
John Breen wrote:
Help!
I'm (still) having issues with Asterisk Queues.
I want to implement a queue so that callers get the 'all our staff are
busy at the moment, your call has been placed in a queue and will be
answered by the first available operator. You
I am programming a very large dialplan right now (Asterisk 1.4), and a
couple of things are annoying the heck out of me.
1. When in a macro, background() does not work properly. If you use the
background() app inside a macro, and then press a key, execution returns
back to the calling context
I goofed that up on my dCAP exam. Spent 20 valuable minutes trying to
fix it!
Eric ManxPower Wieling wrote:
This can happen if you have a Digium card (maybe Sangoma too) in the
system that is configured, but has no actual line plugged into it. I
don't know if this applies to analog, but I
Has anyone used talked to astmanproxy with the Asterisk Java Manager
interface? First suspiscions are that it will not work.
Astmanproxy sends a connection banner of 'Asterisk Call Manager
Proxy/1.21' which is not what Asterisk Java is expecting. Also,
astmanproxy preprends the name of the
Ok, so I ain't much of a Java programmer, but...
Can the Asterisk Java API be written with threads? Ie, I need to connect
to multiple Asterisk systems from the one java application. I tried to
make my class which implements ManagerEventListener, also implement
Runnable, but got errors
Stefan Reuter wrote:
Jesus Mogollon wrote:
The best option would be to use AstManProxy and connect your event
manager to it.
why would adding a new system in between be better than directly
connecting to multiple Asterisk servers?
=Stefan
Simple. With the manager proxy in between,
Jesus Mogollon wrote:
The best option would be to use AstManProxy and connect your event
manager to it.
I tried this. Two problems...
The Asterisk Manager Proxy sends out a banner of 'Asterisk Manager
Proxy/1.2' whereas the Asterisk-Java interface expects to see 'Asterisk
Call Manager 1.0'
Yuan LIU wrote:
Does application Read() return a status? Console displays stuff, but
show application read doesn't mention any status variable.
Yuan Liu
I know that read() on a non-existent sound file will cause dial plan
execution to abruptly stop (unlike background())... which is very bad
We used ChanSpy to allow a supervisor to listen in on the calls of their
staff. There was one huge problem with this, which I imagine would
affect whisper as well.
The supervisor typically sat fairly close to the worker, and could hear
both the voice of the worker as they spoke AND the
All,
Is there a dial plan command that can stream uncompressed audio from
another source? I see there's an MP3Player command that can stream, but
I assume that plays MP3's, which means it has to decode them. I'm
looking for something that could play .wav or .ulaw (g711) streams.
Doug.
Oh poo. No one seems to know. :(
Doug Garstang wrote:
All,
Is there a dial plan command that can stream uncompressed audio from
another source? I see there's an MP3Player command that can stream,
but I assume that plays MP3's, which means it has to decode them. I'm
looking for something
Stephen Bosch wrote:
Doug Garstang wrote:
Oh poo. No one seems to know. :(
Your mistake is replying to an existing thread and changing the subject
line instead of starting a new one.
Start a new thread, and people are more likely not only to notice your
message, but reply
Eric ManxPower Wieling wrote:
Doug Garstang wrote:
Oh poo. No one seems to know. :(
Doug Garstang wrote:
All,
Is there a dial plan command that can stream uncompressed audio from
another source? I see there's an MP3Player command that can stream,
but I assume that plays MP3's, which means
Steve, I was hoping for something native to Asterisk, ie something not
requiring a new process.
Steve Totaro wrote:
Madplay
Doug Garstang wrote:
Oh poo. No one seems to know. :(
Doug Garstang wrote:
All,
Is there a dial plan command that can stream uncompressed audio from
another source
Does anyone know who maintains the Asterisk-java web site at
asterisk-java.org? The site seems to have been unavailable for a couple
of days now.
Doug
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