Displaying 20 results from an estimated 2000 matches similar to: "DBget and DBput in extensions.conf"
2005 May 15
1
Old DBGet/DBPut vs. new Set(var=${DB(...
Hello
I upgraded to CVS head yesterday (due to the lack of zaptel drivers
working with 2.6.10)
And noticed that now DBGet and DBPut have been deprecated in favour
of the new Set/DB one.
In the UPGRADING.txt in Asterisk it says:
* The applications DBGet and DBPut have been deprecated in favor of
functions. Here is a table of their replacements:
DBGet(foo=family/key)
2003 Sep 09
2
DBPut and DBGet performance
hi,
This question is about DBPut and DBGet,
Can i put about 1000 keys in a single family, (only once for the lifetime)
for ex.
exten => _X.,5,DBput(family/key1=${val})
...
exten => _X.,5,DBput(family/key1000=${val})
like above and if i later retrieve it, randomely, with inbound calls,
will it affect performance?
Surajee
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2003 Jul 08
0
dbget & dbput
Hi,
do i need some other software than asterisk to use database commands -
dbput and dbget in asterisk ?
regards
Marian
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Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic
Tel: +421-46-5430 754 # Fax: +421-46-5439 144
http://www.sunteq.sk/
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A mind is like a parachute... it only works when it's open.
2005 Jul 13
1
DBput from the web?
Does anybody has a php code for using DBput (DBget, DBdel) from a web
interface, which database is used for astrisk?
bye
Ronald
2003 Sep 25
2
FW: RE: AntiSpam UOL
Every time I send an e-mail to the * list, I receive this "AntiSpam UOL"
E-mail. is anybody else experiencing the same?
How can I get rid of it?
Uriel
-----Original Message-----
From: AntiSpam UOL [mailto:andersoncbr.sspam@uol.com.br]
Sent: Wednesday, September 24, 2003 11:51 PM
To: uriel@adelphia.net
Subject: RE:RE: [Asterisk-Users] SIP / GrandStream Configuration
Ol?,
2004 Jun 22
6
*69
Hello,
I've managed to build in the "last number repeat" outlined at
http://www.voip-info.org/wiki-Asterisk+last+number+repeat to call back
the last person _I_ called from a particular phone, and now I'd like to
try to do something similar for the common *69 -- call back the last
number that called me. I assume I'll do part of this in my standard
extension macro --
2003 Sep 01
6
Change include contexts runtime
Hi there
How do I change the dialplan runtime, if I for example wants all calls on
the main number to be answered by a voicemail (when it is out-of-office
hours).
I want to be able to change the configuration by pressing a DTMF combination
e.g. *82. Can't figure out whether it is necessary to change contexts or how
to do it.
I have read a lot of examples and config documentation, but I
2003 Jul 28
1
Call Forwarding and DND conf
I have put together this call forwarding and dnd config:
I'm sure it can be dome with macro's but I couldn't figure that out...
anyone care to input.
74 Turns DND on my phone will not ring, drops caller to voicemail...
73 Turns DND off
72+ext forward your extension to another extension and voicemail is left
at the forwarded extension.
71 turns off call forwarding.
; dnd Could
2004 Aug 10
5
Blocking the 'Do Not Call" List
Anybody have any experience with blocking numbers in the U.S's Do Not
Call list?
We have a customer that will be getting their own Asterisk server from
us, and they want it to be check outbound numbers against the do not
call list; this is for a backup, in case there's a slip up and one of
their people try to dial somebody on the do not call list.
The list has millions of numbers, and
2006 Jan 23
2
Fw: setting outgoing caller ID by the queue an extension is logged into
Greetings fellow list members,
I am trying to add some tricky functionality to Asterisk dialplan and I
was curious if anyone else has come up with a solution to something like
this.
Basically I have phone representatives that log into one of several
queues (not using chan Agent, we log in by the extension), and
frequently these agents have to make attended transfer calls to outside
numbers.
2003 Jul 15
3
Conditional Contexts
I was wondering if the following was possible:
2 separate incoming contexts. The first will be used when
there is a secretary present. The second will be used when there is
no secretary.
I know that this can be done using includes and specifying the time
in which each separate context would be included. However, I would
like to be able to switch them from the reception telephone.
For
2006 Mar 03
2
Does an entry in AstDB stay after reboot?
I set up a call forwarding script in extensions.conf which uses the
AstDB but I'm wondering if I reboot the server, will the entry in
AstDB still reside?
What the script does is when a call comes in, it check to see if there
is a null value or a call forward number. If null, it will call the
local office connections. If there is a number, it calls that. Now I
just need to know if I reboot
2004 Apr 09
5
vm e-mail notification stopped
After rebooting my asteriks server, e-mail notifications are no longer being
sent after a voice-mail is left.
I can see the messages in /var/spool/asterisk/vm.
has anybody had the same experience? how was it resolved?
Uri
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2003 Apr 26
1
Dynamic IP Addrress Work Around
I have one of my asterisks off an xDSL modem with a dynamic IP address. The
IP address keeps on changing so I am curious as to what other people are
doing to work around this issue. I am contemplating to register for a
dynamic DNS service but then my communications become dependent on the
availability of this service. The "host=dynamic" for SIP seems interesting.
Any suggestions?
2003 Dec 15
3
Outgoing calls for a fancy address book app
Hi !
I'd like to connect phpgroupware to asterisk: when a user click on a phone
number, his phone rings and he gets connected to the number he just clicked.
I've tried by putting various files in /var/spool/asterisk/outgoing, without
results (we are using SIP phones + CAPI channels).
Is there a way to do that ?
(If it's impossible (something impossible in *, LOL ?!?) I will
create
2009 Apr 21
4
Asterisk Database
My setup : Trixbox 2.6.1 & TE410P running well .:
1. I need to store the CallerId of the PSTN caller with his language preference so that next time he is played the prompt in his language that he chose the first time.What would be better - storing his number in the Asterisk DB and using Dbput and DBget ? or storing it in MySQL from the dial plan and quering it everytime to see the callers
2003 May 14
20
Call forwarding
Yo,
Inspired by the example in the tips & tricks-section of
"http://www.junghanns.net/asterisk/", I built a more elaborate
call divert-feature. This one validates if the extension a call-forward
is to be set to is actually valid for the current context and
additionally saves this context into the DB and always uses it to
originate the divert from, as you can't expect the
2005 May 16
2
Pass variable to Authenticate?
I'm trying to figure out a way to make my own agent login, because I
don't like how the default works.
I have the login and logout working fine using the dynamic add and
remove commands, but I need to be able to create a list of users and
passwords.
I thought of a way to do it using a list of passwords, but the agent
would only ever be prompted for their password. I won't want that.
2003 May 07
2
vmail.cgi cannot read/delete messages
vmail.cgi rocks (if I can borrow the expression for Mark Street). As Mark
pointed out, the /vm/INBOX messages are created with 0700 security and
vmail.cgi is not happy. Apache/cgi/vmail.cgi cannot play them unless I fool
around with the Apache wrapper or chmod 755 *.* thefiles myself. (Tedious,
that is why I like computers).
Obviously this is not acceptable. I took a trip to the
2003 Sep 11
3
SIP busy
Hi,
I would like * to treat a SIP extension as a normal extension, when it
comes to the busy functionality. In other words, if someone tries to
call the SIP phone and there is already an ongoing conversation, the new
caller should get a busy message/tone
Is there any parameter that I can set? Is this something that should be
configured at my softphone?
Best,
PHM