Re: [asterisk-users] Native music on hold not playing on incoming calls
Giorgio Incantalupo schrieb: Hi, I'm trying to make native music on hold work on my Asterisk 1.2.9.1 server with a Sangoma PRI card. If I use a IAX phone connected to the PBX, I hear the music, but if I make a call from outside I hear nothing even if Asterisk console says music has started... it seems something related to zapata.conf but I cannot understand what's wrong. I also put musiconhold=native for every channel inside zapata.conf without success. Is there anybody who can help me, please? TIA Giorgio ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Did you Answer() the channel before playing the music? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: SV: [asterisk-users] Manage 'full' log file
[EMAIL PROTECTED] schrieb:
Thanks for the quick response!
I read about logrotate at voip-info.org but I didn't quite understand it. I'm
no asterisk/linux expert unfortunately.
First of all. What exactly does happen when I run:
/usr/sbin/asterisk -rx 'logger rotate'
Does it clear the file and create a new one? Can I run this manually without
any interruption in the system?
And what does the script do? I understand it rotates the logs. But does it
delete the old files? Where do I put the script? How do I run it? As you can see I'm
really a newbie on this. Unfortunately the docs for asterisk are often with the
expectation that you know everything... :)
Regards,
Jan
-Ursprungligt meddelande-
Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För Ex Vitorino
Skickat: den 8 januari 2007 13:10
Till: Asterisk Users Mailing List - Non-Commercial Discussion
Ämne: Re: [asterisk-users] Manage 'full' log file
We've been using logrotate without any issue... We're using
the below quoted configuration. Notice the invocation of
Asterisk's CLI logger reload command so as to close the
old files and open new ones.
Cheers,
--
Ex Vito
/var/log/asterisk/messages /var/log/asterisk/queue_log
/var/log/asterisk/event_log {
weekly
rotate 52
dateext
compress
delaycompress
nocreate
missingok
sharedscripts
postrotate
/usr/sbin/asterisk -rx logger reload
endscript
}
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This script goes into /etc/logrotate.d in a seperate file. It will
compress the log weekly and store it in the same directory the original
log was in.
-- F. Foerster
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Re: [asterisk-users] AOC-D or similar
Ale wrote: hi all, I'm trying to send text messages to Snom 300 to show the credit remaining during the call... Sending a MESSAGE directly to the phone via udp i'm able to update the text on the display... but not during the conversation. I read about AOC, but i can't find any documentation about Asterisk + SIP + AOC Have you any experience, docs or workaround to suggest? Thx Ale ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users AFAIK the problem is that AOC is simply not implemented in Asterisk at the time. There are patches for Zap channels, but not everybody seems to like them. - Fabian Foerster ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] long busy()
Christophorus Laube schrieb:
[Description]
Busy([timeout]): This application will indicate the busy condition to
the calling channel. If the optional timeout is specified, the calling
channel
will be hung up after the specified number of seconds. Otherwise, this
application will wait until the calling channel hangs up.
This is what I found when I typed show application busy in the CLI.
Did I interpret it wrong?
regards, Christophorus
Mailinglisten schrieb:
Christophorus Laube schrieb:
hi list,
I set up a new asterisk machine with asterisk 1.2.13 and misdn
0.3.1rc27.
I use an e1 card with sip clients. My extensions look like this:
[E1]
snip...snip
exten = 33006733,1,Set(CALLED=${EXTEN})
exten = 33006733,2,Dial(SIP/[EMAIL PROTECTED])
exten = 33006733-ANSWER,3,Answer()
[SIP]
exten = _X.,1,Noop()
exten = _X.,2,SetCallerPres(allowed_passed_screen)
exten = _X.,3,Dial(mISDN/g:E1/${EXTEN},40)
exten = _X.-BUSY,4,Busy(1)
But whenever a sip client calls to an exten that is busy through e1 I
get busy tones for 10s before I get disconnected. But I want to have
it only for 1s.
Does anyone know how to fix that?
regards, Christophorus
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AFAIK the BUSY() command has nothing to do with the busy indication.
You can't pass anything to this command.
Check: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Busy
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I think that is something that should be pointed out on the website,
too, then. I did not run that command on the CLI before, sorry.
Is there any output on the CLI that proves the BUSY command is run at
all? Because I don't really know if
exten = _X.-BUSY,4,Busy(1)
is gonna work. I would say something like:
exten = _X.,3,Dial(mISDN/g:E1/${EXTEN},40,j)
exten = _X.,n+101,Busy(1)
should work if setting the timeout really works that way. Note that the
Dial command has the switch j set which will go to priority n+101 if the
channel is busy.
- Fabian Foerster
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Re: [asterisk-users] Re:Re: outgoing call on ISDN PRI
Michel wrote:
Sorry, sorry !!! I was mixed with another config when I wrote my
first email !!
In fact, User A is registered on Asterisk and user B has a public
phone number (no link with Asterisk).
Our test is : User A calls asterisk server via SIP. As User A context
has a DIAL('user B phone number'),
Asterisk calls user B via ISDN line. Then, user B phone rings and we
can see the caller phone number on
user B phone screen. This caller number is our ISDN line number. What
we would like to do is to hide the caller number (our ISDN line number).
We tried usecallerid, callerid, hidecallerid, restrictcid,
usecallingpres in zapata.conf but it doesn't work.
Do you or anyone know how to hide it?
Thanks you!
--
Message: 4
Date: Tue, 12 Dec 2006 19:04:44 +
From: Tim Panton [EMAIL PROTECTED]
Subject: Re: [asterisk-users] outgoing call on ISDN PRI
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=US-ASCII; delsp=yes; format=flowed
On 12 Dec 2006, at 15:11, Michel wrote:
HEllo list !
When user A calls user B via Asterisk (Users A and B are registered
on the same Asterisk server ) and an ISDN PRI, user B phone
always shows Asterisk server telephone number. How to hide it and
how to forward user A number ?
We tried usecallerid, callerid, hidecallerid, restrictcid,
usecallingpres in zapata.conf but we always see Asterisk server
telephone number !
I'm not getting a clear picture of how the ISDN PRI gets into it if
both users are registered (SIP I assume)
to the same asterisk.
If the call actually goes out via a Public ISDN line, you have to
get the provider to agree to let
you set the outgoing number. Normally they will only let you set it
to one of the inbound numbers
that you have bought from them :-)
If that doesn't help,
please re-phrase the question...
Tim Panton
www.mexuar.net
www.westhawk.co.uk/
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I think your telco is adding the main number assigned to you if you
try not to send one at all or if you send something as a caller ID the
telco didn't allow you to send. IMHO the only thing you can do is ask
the telco not to present the caller ID to the other end. I'm pretty sure
that there is an option to do so in Asterisk, but of course your telco
must support that. Here in Germany this is not a standard feature.
- Fabian Foerster
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Re: [asterisk-users] Re:Re: outgoing call on ISDN PRI
I forgot to mention that the feature in question is called CLIR, or Calling Line Identification Restriction. With that, you can always hide the presentation of your caller ID or do that on a per-call basis. You might want to ask your telco about that. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Playing a sound file on handset pickup
John French wrote: I've added the ability for a user to record a custom message associated with a special IVR menu for occasions when business will be closed for some non-standard amount of time (Maybe 4 days at Christmas...) They just dial 800, record the message then hang up and dial 801 to enable it. Presumably, when they return after the holiday, they should dial 802 to disable it and return to the normally scheduled menus. But they will most likely forget so I'd like to set up some type of reminder functionality; perhaps playing a message back to them stating that the custom message is still enabled before giving them dialtone or something to the same effect. Is this possible and can anyone offer recommendations? Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Why not just add that functionality to the s extension? If no extension is given, they will end up there, won't they? So if that I'm not here message is set up, and the client picks up the phone, we assume that he/she is back and thus delete the notification without notice. - Fabian Foerster ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] long busy()
Christophorus Laube schrieb:
hi list,
I set up a new asterisk machine with asterisk 1.2.13 and misdn 0.3.1rc27.
I use an e1 card with sip clients. My extensions look like this:
[E1]
snip...snip
exten = 33006733,1,Set(CALLED=${EXTEN})
exten = 33006733,2,Dial(SIP/[EMAIL PROTECTED])
exten = 33006733-ANSWER,3,Answer()
[SIP]
exten = _X.,1,Noop()
exten = _X.,2,SetCallerPres(allowed_passed_screen)
exten = _X.,3,Dial(mISDN/g:E1/${EXTEN},40)
exten = _X.-BUSY,4,Busy(1)
But whenever a sip client calls to an exten that is busy through e1 I get busy
tones for 10s before I get disconnected. But I want to have it only for 1s.
Does anyone know how to fix that?
regards, Christophorus
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AFAIK the BUSY() command has nothing to do with the busy indication. You
can't pass anything to this command.
Check: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Busy
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Re: [asterisk-users] How to stop Asterisk to pick up incoming PSTN signal
Gidean Chan schrieb: Hi, How to stop Asterisk to pick up incoming PSTN signal but keep the functionality to make the call out? Thanks Gidean ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Disabling incoming PSTN calls works like that in the dialplan: [from-pstn] exten = _X.,1,NoOp() Change the context to your needs. -- F. Foerster ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
