Re:[Asterisk-Dev] how to detect busy tone ?

2004-11-09 Thread dev2003
I have set busydetect =yes and busycount=4 , but x100p still can not detect busy tone. where define busy_detect algorithms detect 450Hz ? what means goertzels ? /* Number of goertzels for progress detect */ #define GSAMP_SIZE_NA 183 /* North America - 350, 440, 480, 620, 95

[Asterisk-Dev] Asterisk and RADIUS status

2004-11-09 Thread Daniel Pocock
I've just been looking at: http://appradius.minitelecom.org/ It appears this module is based on the user entering their code via DTMF. Has anyone done (or is contemplating doing) support for IAX and SIP user/incoming request authentication based on a RADIUS server? ___

[Asterisk-Dev] Monitor on Asterisk´s Manager API

2004-11-09 Thread Francisco Dias
Do you know what are the options that one can put when using:   Action: Monitor Channel:   to the asterisk´s manager API.Because i would like to set the extension(wav,etc..) , the file and path to where it would be recorded and the multplexing option of the input file and output file of the

Re: [Asterisk-Dev] how to detect busy tone ?

2004-11-09 Thread C. Maj
On Tue, 9 Nov 2004, Soren Rathje waxed: > [EMAIL PROTECTED] wrote: > > [EMAIL PROTECTED] > > > > how to detect busy tone 450Mhz 0.35s 0.35 > and then hangup? > > > > modify dsp.c or chan_zap.c ? dsp.c, but also look at the BUSY* compile options in the Makefile in the top level asterisk directory

[Asterisk-Dev] What's this sugar business about?

2004-11-09 Thread Benjamin on Asterisk Mailing Lists
trying to build today's CVS I get this error ... bison ast_expr.y --name-prefix=ast_yy -o ast_expr.c /usr/bin/gm4: m4sugar/m4sugar.m4: No such file or directory checked on many other boxes (Linux, BSD, OSX) and this file is not on any of them. is this a new pre-requisite? thanks rgds benjk --

Re: [Asterisk-Dev] UK BT Caller ID - work on new patch

2004-11-09 Thread Steve Underwood
Marc McLaughlin (LUSYN) wrote: Hi all, The original patch developed by Tony Hoyle was reported in Bug ID 1719. Modified versions of these are now available which work with v1 of * (download from www.lusyn.com/asterisk/patches.html). I'm trying to come up with a new patch for UK BT Caller ID on X10

[Asterisk-Dev] UK BT Caller ID - work on new patch

2004-11-09 Thread Marc McLaughlin (LUSYN)
Hi all, The original patch developed by Tony Hoyle was reported in Bug ID 1719. Modified versions of these are now available which work with v1 of * (download from www.lusyn.com/asterisk/patches.html). I'm trying to come up with a new patch for UK BT Caller ID on X100P with the aim of it being ac

Re: [Asterisk-Dev] how to detect busy tone ?

2004-11-09 Thread Soren Rathje
[EMAIL PROTECTED] wrote: > [EMAIL PROTECTED] > > how to detect busy tone 450Mhz 0.35s 0.35 and then hangup? > > modify dsp.c or chan_zap.c ? > ___ > Asterisk-Dev mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-dev > T

RE: [Asterisk-Dev] how to detect busy tone ?

2004-11-09 Thread Whisker, Peter
I assume you mean 450Hz! The standard busy_detect algorithms should pick up a 50% duty-cycle tone at 450Hz. It picks up the UK 400Hz tones with the same timing. The parameters in dsp.c look OK to pick it up without need for change: #define BUSY_PERCENT10 /* The percentage diffr

[Asterisk-Dev] how to detect busy tone ?

2004-11-09 Thread dev2003
[EMAIL PROTECTED] how to detect busy tone 450Mhz 0.35s 0.35http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev

Re: [Asterisk-Dev] sip trustrpid

2004-11-09 Thread Olle E. Johansson
Richard wrote: Hi, I have sip trustrpid enabled. If I make a call from sip phone to pstn, the call pulls the right caller-id from “Remote-Party-ID” and sends out to the carrier. It works perfectly. I also use a SIP provider for long distance. When the sip call is sent to the provider, * ignores

RE: [Asterisk-Dev] MeetMe - Manager Commands?

2004-11-09 Thread Ben Merrills
I think commands like that would be excellent! I'd be very interested in seeing their development. Also, commands to change the status of a user, I.e. make them admin etc. Ben -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Boehm Sent: 08 Novembe

[Asterisk-Dev] sip trustrpid

2004-11-09 Thread Richard
Title: sip trustrpid Hi, I have sip trustrpid enabled. If I make a call from sip phone to pstn, the call pulls the right caller-id from “Remote-Party-ID” and sends out to the carrier. It works perfectly. I also use a SIP provider for long distance. When the sip call is sent to the provider,