Displaying 20 results from an estimated 11000 matches similar to: "New budgetone firmware"
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 Jul 11
1
Configuring BudgeTone and ringer over TFTP
I noticed that the BudgeTone (I have the 102) with the latest firmware
tries to download a file called cfg.txt (presumably the configuration)
and a file called ring.bin (presumably a ringer) from the tftp server.
The ring-in sound on the budgetone is the same as the ring-out sound and
that is going to be confusing for users. I contacted GrandStream and was
informed that both of these formats are
2003 Jun 13
2
Budgetone Supervised Transfer
Hi.
I was wondering if there's a way to do supervised transfers on
the budgetone 102 . Blind works ok, but can't do supervised.
I thought that with the flash button that could be possible, but
seems I'm wrong ....
matteo.
2003 Jul 15
0
Budgetone Transfer (The answer)
Anyone having problems getting transfer to work here is the answer...
It appears the manual is incorrect..
The manual says:
1)Press "Transfer" button.
2)Dial the target extension.
3)Hangup the phone.
This will disconnect the call.. Here is how it can be done..
Matteo gave this solution.. (thanks)
NOTE:'Use # as Dial Key:' must be set to YES
To trasnfer:
1)Press
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 Jul 08
4
Budgetone and Voicemail
I have a problem with using voicemail on the Budgetone phones. When
entering the mailbox and password, sometimes some keys will register
multiple times (as shown on console when it says no such user in config
file) and sometimes some keys won't even register at all. It seems
totally random. Has anyone seen this problem? Any recommendations
would be greatly appreciated. Thanks.
Brian
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
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 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 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 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
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 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 Nov 07
2
Callgroups and Pickupgroups in Console/dsp
Hi all.
I've made a patch for chan_oss.c to enable
callgroups and pickupgroups in it (since wasn't enabled).
I needed it for a special use of the console (pickup
calls arriving to the console from another phone)
btw, If someone is interested, I can submit a patch
to the bugtracker. I won't do it until
that's usefult for someone... since is a very special
features that probably no
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
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
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 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