Displaying 20 results from an estimated 4000 matches similar to: "bare minimums"
2003 Jul 05
3
Activate MySQL logging
<P>hi,</P>
<P>Can anybody pls tell me how to activate loggin CDR on mysql db. I tried editing the /etc/asterisk/cdr_mysql.conf file and recompiling asterisk, it didn't work. Normal loggin on Master.csv file works fine.</P>
<P>Thank you inadvance,</P>
<P>Surajee</P>
<P> </P><br>
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2003 Jul 14
3
New budgetone firmware
Hi.
Has anyone experienced with the new firmware .77 ?
There's Day Light Saving time now, but haven't
time to play with it, till now.
Matteo.
--
Matteo Brancaleoni
Espia System Administrator - IT services
Website : http://www.espia.it
Email : mbrancaleoni@espia.it
2004 Jan 13
2
Nufone.net net wackiness?
I can't send mail to any addresses in nufone.net; they all get rejected
by a spam blocker.
And their website is gone, too!! The URL leads to a "parking site."
My accounts still seem to work, but I wonder WTH is going on?
Thx.
B.
2004 Jul 05
4
IAX Call Pickup
I've looked in the obvious places but haven't found a definitive
answer to the following: can an IAX extension (an Iaxy phone, for
instance) do call pickup via *8?
Adolfo
2003 Nov 24
4
One voicemail -> multiple recipients?
The subject pretty much says it all. I have a customer who would like
to have an option where a caller can leave a voicemail in such a fashion
that it would be simultaneously delivered to a set of mailboxes all at
once--the idea is "trouble ticket" type operation where multiple
technicians will *each* get the vm.
He prefers that, if we can do it, to a "shared mailbox"
2004 May 07
4
SIP Wokflow diagram
Hi everybody,
I would like to create SIP call flow Diagram under Windows. Is anybody
know a program to perform it? I have already Ethereal and I would like
an explicit diagram just to show where something have problems...
Thanks
Ignace
2003 Jun 13
3
Call queues for phone operator
Hi.
I was wondering how can I make incoming calls to wait if the phone
operator is busy. I've 8 incoming lines, with 30 extensions.
What I need is if the operator is busy with call nr #1 , the new
incoming call waits until the op. is free.
Looking into app_queue seems the way to go.
So I want to ask if I'm right or wrong:
I set up only a queue , is to say operatorq, where
the only member
2004 Jul 21
2
fonction Getvar
Hia ....
i try to use the fonction Getvar of asterisk to get a variable myDNIS
that i have define. i use it as follow
Action: Getvar
Channel: SIP...
Variable: myDNIS
but asterisk don't know it .i have the response as follow
Response: Error
Message: Invalid/unknown command
does everybody meet this problem . i try all possible combination and
nothing
help please ..!! :-(
thanks in advance
2003 Jul 22
3
busydetect and random hangups
Hi,
I'm having random hangup problems with zap channels.
If I turn busydetect off in Zapata.conf, * fails completely to detect a
user hangup in the middle of a script.
On the other hand, if I turn it on, everything works much better, but
long calls tend to be hung up without a motive.
Any other parameter that I can try? Any #define that I can tweak and
recompile?
Will callprogress
2003 Jul 23
2
SIP info
I was wondering what are the values for sending dmtf
via sip info.
I mean, when I use dtmf relay via sip info, the sip/sdp message
contains a Signal=X where X is the dmtf.
That's ok for dtmf 0-9 . but what when dtmf is * or # ?
we must send signal=# ?
I ask that because I noticed that budgetones phone sends out
* as signal=10 and # as signal=11 . but asterisk
don't detect them, 'cause
2003 Apr 08
1
Wiki for the * community.
Hi 2 all.
I was thinking to start a little web site with phpwiki,
to let the * community build a sort of shared
documentation 'bout * & related.
That because in a wiki "place" all grows faster,
and is also the right place to share experiences.
For example it's right to have documentation
about * installations, ie who has done what with asterisk
Till now we don't know
2004 Jan 16
7
CAPI not installed, after changed from i4l to CAPI
I had unexpected hangups from my asterix box using the i4l driver. (SIP
<-> SIP calls worked execellent, but SIP<->ISDN didn't.)
Then I changed the i4l driver in modem.conf with the chan_capi from
jungham. (http://www.junghanns.net/asterisk)
I followed his instructions in the INSTALL file, and first encountered
some errors compiling it. It help by deinstalling several
2004 Jan 14
3
NAT friendly TFTP Server
Hello,
For those interested in overcoming the problem with some NATs and Firewalls in regards to tftp. I found a nice little tftp server here:
http://freshmeat.net/projects/jtftp/?topic_id=87
I tried it and it works great.
Regards,
Andres.
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2004 Jan 08
5
AbsoluteTimeout Users Messages
Hi, All
Is there a provision for "AbsoluteTimeout" application to notify
called and calling party of the reason why the call suddenly ended?
This way, the parties will be much better informed, hence they
will/should not think that
their VOIP/telco provider(s) are providing bad service.
Ta
SJ
2003 Apr 18
1
Account code on SIP
I was wondering if the accountcode flag works
with sip channels. I was looking into the
debug and ,even if I have the line accountcode=XXX
into the users sections of my sip.conf, I don't see
it logged into the cdr.
Matteo Brancaleoni
mbrancaleoni@espia.it
Emmegi System Administrator
EspiA - EMMEGI Srl - e*solution provider
Uffici: Via Pascoli, 37
20129 Milano - Italy
Sede Legale: Corso
2003 May 25
2
Message Waiting and VoiceMail 2
Hi.
I noticed that if new messages are recorded
with voicemail2 , they're not detected by
the message waiting indicator, so
the mailbox=XXXX param has no effect, and
no message waiting is sent to the phone
(sip & adsi, or stutter dialtone)
Any hint?
--
Brancaleoni Matteo <mbrancaleoni@espia.it>
Espia - Emmgi Srl
2003 May 01
2
Max number of connection in IAX ?
Hi.
I was wondering if there's a parameter to limit
the number of concurrent sessions in IAX, globally or
on a per-user basis.
That could be needed for security purposes
(to prevent dos attacks), to limit bandwidth / cpu usage, or
to not allow more than N guest connections, for example.
Any other VoIP channel support that?
(like SIP, MGCP)
Matteo.
--
Brancaleoni Matteo
2003 Sep 24
3
RedHat 9.0 and 100 percent CPU utilization
Please, don't hate me because I use Redhat. I am
aware that I am asking for problems in running
Asterisk on Redhat. I recently aquired a nifty
server, moved my digium cards, and installed asterisk.
I noticed that one of the four processors was being
used at 100% and nothing was working. I tracked CPU
utilization back to the Asterisk process. Please,
help.
James
2004 Jul 16
8
Asterisk-1.0 RC1
We have officially made the first release candidate of Zaptel, Libpri,
Asterisk and Gastman available. While there are still open major bugs,
they are relatively limited, and it was time to go ahead and get the 1.0
ball rolling in earnest.
ftp://ftp.digium.com/pub/asterisk
Enjoy the code. Special thanks to all the bug marshals and contributers
and to everyone who has supported Asterisk
2003 Oct 12
2
INFO method and DTMF translation
Hello guys,
I have searched high and low, but not found any information about
rules of using DTMF in SIP INFO method. Cisco has described something with
Signal=, but it look like this feature is dependent on implementors?
The problem is chan_sip.c cannot correctly translate received DTMF
digits, especially #,*. At least with my Antek EGW-804 gateway.
Looking into chan_sip.c, I found this code: