Re: [asterisk-users] colors in the console
Lacy Moore - Aspendora wrote: I'm wondering if anyone else has experienced this. Up until a few days ago, when accessing the CLI from my terminal program (Private Shell), the output was in color. I haven't upgraded, rebuilt, or to my knowledge, changed anything in Asterisk that would change this. My terminal settings were the same as well. I have two computers that I access the CLI regularly on, and neither show color anymore. When I disconnect, Private Shell shows the disconnect in red, just like before. This tells me that Private Shell is still doing color. What controls the color coding in the CLI? I found something in the source about it, but again, since it has been recompiled, this should not have changed. Is there a config file somewhere that I'm too blind to find? Thanks! -- Lacy Moore Somewhere I wish I wasn't I believe that only the CLI console provides color: e.g. asterisk -c. Connecting to an already-running asterisk process will not provide color: e.g. asterisk -r. Earle ___ --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] problem with safe_asterisk
Gordon Henderson wrote: On Tue, 13 Feb 2007, Tzafrir Cohen wrote: On Tue, Feb 13, 2007 at 01:35:50PM +0100, Andrea De Vita wrote: Hi all, I have installed some Asterisk machine, all with the same problem. My typical configuration is: - Asterisk 1.2.14 (or 1.4.0beta3) - CentOS 4.4 server. The problem is this: When I start Asterisk with the default init script (/etc/init.d/asterisk start) distributed with the source, and kill (or kill -9) Asterisk-pid, then safe_asterisk doesn't correctly work (it dies and not restart Asterisk). Instead, if I start Asterisk with safe_asterisk command from shell, after "kill Asterisk-pid", safe_asterisk restart Asterisk correctly. I would use the init script because I like to use Linux-HA that require this. Edit that init.d script lightly not to use safe_asterisk. safe_asterisk is not close to robust anyway, and thus will only complicate things. Seconded. Have a look at this: http://www.drogon.net/init.d.asterisk Gordon, What is the point of reloading extensions after starting asterisk in the start section of your case statement? Earle ___ --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
[asterisk-users] chan_cellphone won't pair with phone
I'm running chan_cellphone version 13 on the latest svn trunk (as
root). I believe I have chan_cellphone set up correctly (bt addr and
port retrieved from the "cell search" CLI command). When I load the
chan_cellphone module, my Motorola V3m asks if I want to allow "Asterisk
PBX", I say yes and enter the for the pin, then my phone tells me
the pin is invalid. Here is the log entry generated by chan_cellphone:
*CLI> [Mar 3 22:53:12] DEBUG[16888]: chan_cellphone.c:682
rfcomm_connect: connect() failed (111).
*CLI> cell show devices
ID Address Connected State
razrxx:xx:xx:xx:xx:xx NoInit
I'm running:
- Fedora Core 6 - kernel 2.6.19-1.2911.6.4.fc6
- bluez-libs (libbluetooth) 3.7-1
- bluez-utils 3.7-2
I'm using the hcid.conf and pinhelper script from contrib/bluetooth
(which I moved to /etc/bluetooth).
I'm not sure how to debug this further. Any ideas?
Earle
---
Here's the directory listing and the contents of the files (comments
removed for brevity):
[EMAIL PROTECTED] ~]# ls -l /etc/bluetooth/
total 12
-rw-r--r-- 1 root root 1428 Mar 3 18:36 hcid.conf
-rwxr-xr-x 1 root root 27 Mar 3 18:36 pinhelper
-rw-r--r-- 1 root root 297 Oct 2 18:40 rfcomm.conf
--- pinhelper ---
#!/bin/sh
echo "PIN:"
--- hcid.conf ---
options {
autoinit yes;
security auto;
pairing multi;
pin_helper /etc/bluetooth/pinhelper;
}
device {
name "Asterisk PBX";
class 0x3e0100;
iscan enable; pscan enable;
lm accept;
lp rswitch,hold,sniff,park;
auth enable;
encrypt enable;
}
--- cellphone.conf ---
[general]
interval=60; Number of seconds between trying to connect to
devices.
[razr]
address=xx:xx:xx:xx:xx:xx ; retrieved from "cell search" CLI command
port=4
context=incoming-mobile
___
--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
[asterisk-users] app_swift: Failed to set voice
I'm trying to get app_swift (v0.9.1 from http://www.loopfree.net/app_swift/)
working, but it's having issues (see below). I'm running 1.4.0beta3 on FC6.
Any thoughts?
*CLI> -- Executing [EMAIL PROTECTED]:1] Answer("SIP/spa3k-fxs-08e884b0",
"") in new stack
-- Executing [EMAIL PROTECTED]:2] Swift("SIP/spa3k-fxs-08e884b0",
"Diane^your text here!") in new stack
[Nov 10 23:40:43] ERROR[21132]: app_swift.c:240 swift_exec: Failed to set voice.
-- Executing [EMAIL PROTECTED]:3] Hangup("SIP/spa3k-fxs-08e884b0", "") in
new stack
== Spawn extension (internal, 100, 3) exited non-zero on
'SIP/spa3k-fxs-08e884b0'
Sent via the WebMail system at mail.valcom.com
___
--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
[asterisk-users] Survey: In what ways do you use Asterisk at your house?
All, I'm starting to tinker with Asterisk for use in my home. Here's my current setup: Cox broadband telephone <--> spa3k-fxo analog phones + answering machine (all on one line) <--> spa3k-fxs I can pick up a phone in my house, dial a certain extension, and the spa3k will connect me to Asterisk, which currently plays a message and hangs up (not particularly useful). If I dial any other number, the spa3k dials that number out on the fxo. The main thing I'm trying to do right now is replace my answering machine with *-based voicemail. I want to retain the ability to screen calls (listen on a speaker while a person is leaving a message), but I'm not sure of the best way to go about this. Recommendations are welcome. Note that my * box and answering machine are on two different floors in my house, so running a speaker in the kitchen (answering machine location) from the sound card on the * box is doable, but not desirable. Also of note: I only have basic no-frills phone service (no caller id, no call waiting, etc), though I am open to adding options if there's a good reason. The main reason for this e-mail is to see what other people are doing. - What service provider/technology do you use for origination/termination? - What hardware/software do you use and how does it all tie together? - What tasks do you use * to accomplish? - Any other pertinent info. I'm trying to find practical uses for *, but I'd like to throw in some fun/pointless stuff as well. Thanks for your time. Earle ___ --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] Getting app_cepstral to work with Asterisk 1.4.0-beta3
Hall, Eric M. wrote: Using this link http://www.oldskoolphreak.com/tfiles/voip/installing_app_cepstral.txt This is a Dell PowerEdge 1950 running Whitbox 4 and Asterisk 1.4.0-beta3 I get the following errors on make install Any help would be GREAT! Thanks Eric, I had similar compilation issues when trying to use app_cepstral. This doesn't answer your question, but I've had good success using app_swift. http://www.loopfree.net/app_swift/ Earle ___ --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] strange caller display
Rilawich Ango wrote: Hi all, I am using asterisk1.2.14,realtime and I find there is a strange case in the receiver's display. I have a dial plan to route a call to the destination. I haven't set the callerid(num) for the caller. In the receive ends, it's display shows "asterisk" when I make a call to the receiver. I wonder why "asterisk" shows in the display as I haven't set any word - asterisk in any configuration file. How to remove that word from the receive end if it is a default word? Below is the log dump from ngrep. There is no "asterisk" in the from header except the option message. I wonder why "asterisk" will be shown in the receiver end's screen. ango callerid on SIP channels defaults to "asterisk". The only way to override it is to set it in your sip.conf. Though I think that if you get callerid from the calling device, that would override the default as well. I think the order of precedence is (highest to lowest): sip.conf callerid for calling device calling device's transmitted callerid default - "asterisk" Earle ___ --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
[asterisk-users] Looped message playback
Hello, I'm trying to play an audio file to a phone an arbitrary number of times. The audio is a five-second segment of a sine wave. I need this to be played repeatedly without gaps between playbacks. I've tried doing this in the dial plan, e.g.: exten => s,1,Playback(tonefile) exten => s,2,Goto(1) but there is too long of a gap between the playbacks. Does anyone know of a way to achieve this? Thanks, Earle ___ --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] Looped message playback
Eric "ManxPower" Wieling wrote: Earle Clubb wrote: Hello, I'm trying to play an audio file to a phone an arbitrary number of times. The audio is a five-second segment of a sine wave. I need this to be played repeatedly without gaps between playbacks. I've tried doing this in the dial plan, e.g.: exten => s,1,Playback(tonefile) exten => s,2,Goto(1) but there is too long of a gap between the playbacks. Does anyone know of a way to achieve this? You have a long gap in your tone file. ___ --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 Eric, Thanks for the reply. There is no gap in the tone file. The file begins with the sine wave going positive from the zero-crossing and ends with the wave at the zero-crossing from negative. Also, I can loop the file on my PC and there are no gaps in the audio. Earle ___ --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] Looped message playback
Bill Gibbs wrote: Why not just merge the file together a few times using an audio program and make a longer file? Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Earle Clubb Sent: Thursday, September 21, 2006 9:45 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Looped message playback Hello, I'm trying to play an audio file to a phone an arbitrary number of times. The audio is a five-second segment of a sine wave. I need this to be played repeatedly without gaps between playbacks. I've tried doing this in the dial plan, e.g.: exten => s,1,Playback(tonefile) exten => s,2,Goto(1) but there is too long of a gap between the playbacks. Does anyone know of a way to achieve this? Thanks, Earle ___ --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 ___ --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 The duration of the tone can vary at runtime and I have no way of knowing beforehand what the duration will be. Earle ___ --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] Looped message playback
Mojo with Horan & Company, LLC wrote:
Depending on the format of your audio file, you could generate a
one-second sample of audio and then something like the following
#!/bin/bash
NUM=$1
CUR=0
rm -f bigtonefile;
[ $CUR != $NUM ] && {
cat tonefile >> bigtonefile;
CUR = $CUR+1;
}
System(generator 7)
Then Playback(bigtonefile) ; for seven seconds of audio
You could use half-second tones or less if you wanted finer granularity.
Just a suggestion, think outside the box, they say :)
Earle Clubb wrote:
Bill Gibbs wrote:
Why not just merge the file together a few times using an audio program
and make a longer file?
Bill
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Earle
Clubb
Sent: Thursday, September 21, 2006 9:45 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Looped message playback
Hello,
I'm trying to play an audio file to a phone an arbitrary number of
times. The audio is a five-second segment of a sine wave. I need
this to be played repeatedly without gaps between playbacks. I've
tried doing this in the dial plan, e.g.:
exten => s,1,Playback(tonefile)
exten => s,2,Goto(1)
but there is too long of a gap between the playbacks. Does anyone
know of a way to achieve this?
Thanks,
Earle
___
Great idea. Unfortunately I may never know the duration of the tone
until after it is turned off. For example if switch is turned on, the
tone should begin. It should keep playing until the switch is turned off.
Earle
___
--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] Looped message playback
John Marvin wrote: Earle Clubb wrote: Hello, I'm trying to play an audio file to a phone an arbitrary number of times. The audio is a five-second segment of a sine wave. I need this to be played repeatedly without gaps between playbacks. I've tried doing this in the dial plan, e.g.: exten => s,1,Playback(tonefile) exten => s,2,Goto(1) but there is too long of a gap between the playbacks. Does anyone know of a way to achieve this? Well, I don't have a solution for the general case (looped playback), but if you are only playing a sine wave, couldn't you use Playtones() instead? It has the ability to play a tone indefinitely until you tell it to stop. John I thought about that. The problem is that I need to be able to play any kind of tone (e.g. warble, etc.). I'm only using a pure tone right now because it's easy to hear the gaps. Earle ___ --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] Looped message playback
Eric "ManxPower" Wieling wrote: I have done looping playback and never experienced significant gaps. Can you give me an example of what worked for you? Did the files contain tones or voice? Earle ___ --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] Looped message playback
Eric "ManxPower" Wieling wrote: Earle Clubb wrote: Hello, I'm trying to play an audio file to a phone an arbitrary number of times. The audio is a five-second segment of a sine wave. I need this to be played repeatedly without gaps between playbacks. I've tried doing this in the dial plan, e.g.: exten => s,1,Playback(tonefile) exten => s,2,Goto(1) but there is too long of a gap between the playbacks. Does anyone know of a way to achieve this? You have a long gap in your tone file. Eric, You were correct. The file had some header information that should not have been there. I manually stripped of the header so there's only audio data and now the above works fine. Thanks. Earle ___ --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
