Displaying 20 results from an estimated 11000 matches similar to: "Asterisk Not hanging up DS0 when number called is busy."
2005 Jan 31
0
PRI not hanging up the channel after Executing Hangup when dialing busy number.
I have a PRI that if you dial a number that is busy, the channel does
not hang up, it then sends "h|1"to the phone company which will then
plays back to the end sip user "You don't need to dial a one or zero"
I am running stable CVS-v1-0-01/20/05-02:45:17. I have pasted the
important bit from the exten and sip configs below simplest possible
example that will show the
2004 Feb 03
2
busy tones
Hi
When I call a phone with CAPI if the phone available I hear ringing ok
but if the phone is busy I don't hear anything at all.
Also, when I call a mobile phone and it is turned off I don't hear the
operator voice answer me telling me that the request phone is turned off
or unavailable.
Any ideas?
m
2004 Nov 22
1
wiki down ?
im getting:
Fatal error: Unknown function: mssql_get_last_message() in
/var/www/html/tikiwiki-1.8.2/lib/adodb/drivers/adodb-mssql.inc.php on
line 415
to the wiki..
Jason
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
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 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
2004 Feb 03
2
Playing announcement to called user prior toConfirmation
I wish 'A(x)' was available with AgentCallBackLogin!! :(
-----Original Message-----
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Matteo
Brancaleoni
Sent: Tuesday, February 03, 2004 6:48 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Playing announcement to called user prior
toConfirmation
show application
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 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
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
2011 Jun 13
5
No audio after a reinvite changing codec
Hi all,
we have a problem with a reinvite sent by our SIP provider to change audio
codec due to the recognition of a fax tone.
After that the SIP call session has been established (INVITE and 200 OK) we
have the following codec situation:
UAC ASTERISK UAS | ASTERISK UAC
PROVIDER
g711 <----------------------> g711
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
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 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
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>
--------------This mail sent
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
2003 Jun 11
2
Configuring zhone zplex to 24 fxs ports
Hi.
I was wondering if the zplex in the dev kit could
be configured to have all fxs ports, instead
of the standard 8 fxo + 16 fxs.
If so, anyone managed to do that?
Matteo.
--
Matteo Brancaleoni
Powered by RedHat Linux 8.0
Linux User #153521
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G e
2003 Oct 07
1
[PATCH] allow announcements in app_dial
Hi.
Since a customer requested us that feature, I wrote this
little patch for app_dial to allow to play an
announcement to the called party, as soon he answers.
you can define the file to play in the dial() option,
using A(filename).
for example:
exten => blah,1,Dial(Zap/blah,30,rA(/my/own/announce)Tt)
that doesn't break anything ...
feel free to blame me for anything bad this patch
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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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