Displaying 20 results from an estimated 2000 matches similar to: "*69"
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
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.
If Then Else Statements - Outbound Dialling on ISDN using CAPI -Individual Dial out Plans using msns
2004 May 04
2
If Then Else Statements - Outbound Dialling on ISDN using CAPI -Individual Dial out Plans using msns
Hi All,
Many thanks to Marc who helped me with a previous Capi Dialout plan -
however.....
What I now would like to be able to do is: -
We have 8 msn's 383590, 383591 383592 etc.
What I would like to do is set up an If Then Else type statement along the
following lines: -
If extension 7957 Then
Dialout on Capi msn 383590
ElseIf extension 7958 Then
Dialout on Capi msn 383591
ElseIf
2005 May 15
1
Old DBGet/DBPut vs. new Set(var=${DB(...
Hello
I upgraded to CVS head yesterday (due to the lack of zaptel drivers
working with 2.6.10)
And noticed that now DBGet and DBPut have been deprecated in favour
of the new Set/DB one.
In the UPGRADING.txt in Asterisk it says:
* The applications DBGet and DBPut have been deprecated in favor of
functions. Here is a table of their replacements:
DBGet(foo=family/key)
2003 May 08
3
DBget and DBput in extensions.conf
Where can I learn the syntax for DBput and DBget?
is it working with MySQL? do I need to set up tables?
URiel
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2006 Mar 10
3
RFC Follow Me Find Me script
This is a follow/find me script that I can't quite get to work, asterisk
wont save forward/${calleridnum} to AstDB... any comments or thoughts on
how to make this work or change it to work differently are appreciated.
The voice prompts to go with all playback/background extensions are
commented appropriately. I hope this code is of use to some of you and
any help with a perfected
2004 Aug 20
6
Asterisk PBX Functions via SIP phone
Hi All,
I am using a Grandstream BT100 and I have been trying to get the PBX features to work for DND, call foward, etc. These functions do work when I use my POTS phones hooked up to my Zap cards. But I cannot get the PBX functions (ie *78, *79) to work using my SIP phones. Is there a feature that has to be enabled to do this? I know these functions are available within the GS phone but all of
2005 Jun 02
1
Newbie :Call Forwarding problem
Dear All,
I was trying to enable call forwarding, following the steps of the link
on voip.org regarding this issue it doesn't work and the phone I am
trying to implement on is still ringing. below is my conf in
extensions.conf and the CLI output during the process.
My configuration is :
exten => _*5X.,1,DBput(CF/${CALLERIDNUM}=${EXTEN:2})
exten => _*5X.,2,Hangup
exten =>
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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2005 Jan 31
1
A neat "hot seating" mplementation
Has anyone implemented "hot seating" in any neat way? This where
people can log in to any phone in the company and have their
calls/voicemail come to that particular handset.....
2005 Jul 11
1
ASterisk@home + Broadvoice = Almost working installation...
Hello Guys,
I'm somewhat of a newbie and am desperately seeking for some help...
I've managed to get asterisk up and running on my server, and signed up with a
broadvoice account...
I'm having no problem dialing and communicating between extensions, but whenever
anyone tries to call my broadvoice account, they are greeted by no ring or
anything, but rather simply a direct to
2005 Aug 02
12
WHat does it take
How many times do you ask for help here before getting a respone? Every
single thing I do No matter what I get busy extensions. I am willing to pay
someone to help here. Anybody got a clue?
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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
2005 Jun 01
2
IVR Load
Hi,
Thinking about an IVR application and trying to get a handle on the best
way to structure it so that the maximum number of concurrent calls can
be achieved..
If the voice prompts were stored in a GSM format and were being played
out through an IAX trunk that uses GSM compression would asterisk do a
decompress/compress on the audio or would it simply pass through the GSM
encoding?
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 Sep 27
2
Auto CallBack on busy
Auto Callback on Busy
Register on Busy
I have implemented it as
1- I store Caller and Called party numbers in database when Called part is busy
2- I retrieve it from database and Caller is called by called party when Called party hangs up
It is working fine with all kind of SIP phones I have with me
basic configuration for extensions.conf is given and can be accommodated according to
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 Jan 02
1
Asterisk Gotoif / last called
Hi guys
Ive been trying to get this to work for ages now, basicaly im trying to do if ${woteva} = "" (nothing), or its none existenant then do label 1, else label 2. for my last called function, so it will play a different message if theres no last call in the system or it was anonymous.
ive tried
exten => 1000,1,GotoIf($[${last-call${CALLERIDNUM}} = ""]?4:5)
and heaps of
2005 Mar 24
1
Can I use my callscreen macro w/ sip?
(Sorry if this has hit the list before. I sent it yesterday, but never
saw it come through)
I actually got rid of my phone service.. no more pots line in my house.
But, I miss my call screen macro. Any way to do this with a SIP
channel? (Obviously the parking isn't the problem, but rather recording
their name). I set it up so they should only have to record their name
once provided they