[asterisk-users] What web interfaces are available today for debian based Asterisk installation?

are welcome. Thank you, Maxim. -- Cheers, Maxim Vexler "Free as in Freedom" - Do u GNU ? ___ --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] PSTN disconnect tones on voicemail messages

an solve this by setting the following in voicemail.conf <<< ; How many seconds of silence before we end the recording maxsilence=5 ; Silence threshold (what we consider silence, the lower, the more sensitive) silencethreshold=512 HTH -- Cheers, Maxim Vexler "

Re: [asterisk-users] Zap channel faxing in or out fails but phone calls work.

ynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Actually it's quite rational. Check the paragraph on ECM at http://www.voip-info.org/wiki-Asterisk+fax -- Cheers, Maxim Vexler "Free as in

Re: [asterisk-users] Setting a threshold for asterisk to take ZAP line off hook ?

On 7/19/06, Maxim Vexler <[EMAIL PROTECTED]> wrote: On 7/18/06, Mojo with Horan & Company, LLC <[EMAIL PROTECTED]> wrote: > I think when a PSTN line says 'Ring' it's simply for aesthetics... The > line is 'answered' the instant * connects to it fo

Re: [asterisk-users] Setting a threshold for asterisk to take ZAP line off hook ?

tion process. When you are hearing ringing from the PSTN through a zap card, the rings are coming from the phone company and are just sound. * doesn't decode that and act on it yet.) Maxim Vexler wrote: > On 7/16/06, Martin Joseph <[EMAIL PROTECTED]> wrote: >> On Jul 16, 2

Re: [asterisk-users] Setting a threshold for asterisk to take ZAP line off hook ?

On 7/16/06, Martin Joseph <[EMAIL PROTECTED]> wrote: On Jul 16, 2006, at 11:36 AM, Maxim Vexler wrote: > Hello list > > I'm trying to setup asterisk as an answering machine. > > How can I set asterisk to Answer() incoming call ONLY after specified > count of rin

[asterisk-users] Setting a threshold for asterisk to take ZAP line off hook ?

s incoming call IVR after the specified(where?) number of seconds. Thank you. -- Cheers, Maxim Vexler "Free as in Freedom" - Do u GNU ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or

Re: [asterisk-users] How to create or test tone configuration to include them in zaptel

ailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Cheers, Maxim Vexler "Free as in Freedom" - Do u GNU ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users

Re: [asterisk-users] Asterisk + fax

med "fax" in your "default" context. Also look at http://www.voip-info.org/wiki-Asterisk+fax for helping info. -- Cheers, Maxim Vexler "Free as in Freedom" - Do u GNU ? ___ --Bandwidth and Colocation provided by Easynews.com

Re: 'cpmv' instead of 'cp' Re: [asterisk-users] [tip]semicolon trouble:

RIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Is the above claim tested ? If so then this is a great tip! thank you! Also, http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out could use this tip in a

Re: [asterisk-users] [tip]semicolon trouble: System($(sleep 4; cp 1.call out)&) not working, but System($( sleep 4 && cp 1.call out)&) ; )

mp/file1 \; touch /tmp/file2) works for me. Note that && and || are logical operators, so that if the first command fails (speaking bashisem: return non zero value) and you used the && operator then the second command won't run. where as ';' simply means "run the

Re: [asterisk-users] prob with debian and chan_zap

--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 A bit of information would help Please post the output of : ls -lAr /dev/zap/ and ps -ef | grep asteris

Re: [asterisk-users] Tired of fax calls... :-/

IBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users NVFaxDetect does just that ;) Any why, you might find it more useful to actually receive the fax : http://www.voip-info.org/wiki/view/NVFaxDetect -- Cheers, Maxim Vexler "Free as in Freedom&

Re: [Asterisk-Users] s / i extension difficulty

/lists.digium.com/mailman/listinfo/asterisk-users You might want to check pattern matching : http://www.voip-info.org/wiki/index.php?page=Asterisk+Dialplan+Patterns -- Cheers, Maxim Vexler "Free as in Freedom" - Do u GNU ? ___ --Bandwidth

Re: [Asterisk-Users] IVR menus on different DIDs

t To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No it is not. Please attach the relevant part of the log from /var/log/asterisk/full. Also, as a reference, you might wish to check FreePBX project can generate IVR dial logic using a simple web manag

Re: [Asterisk-Users] Changing standard Voicemail behavior

, I no answer caller is sent to user custom IVR. If user press 9 he is sent back to Main IVR otherwise he gets user to record a message in our user's Foo voicemail box. -- Cheers, Maxim Vexler "Free as in Freedom" - Do u GNU ? ___ --Bandwi

Re: [Asterisk-Users] ASTCC: How to reset periodically all "card in use" flag back?

Maxim. ___ --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 -- Cheers, Maxim Vexler "Free as in Freedom" - Do u GNU ? ___

Re: [Asterisk-Users] Include Text file in Dial Plan

HTH Maxim. ___ --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 -- Cheers, Maxim Vexler "Free as in Freedom" - Do u GNU ? __

Re: [Asterisk-Users] Placing call files in/var/spool/asterisk/outgoing/ does not work

ar/spool/asterisk/outgoing/max.call" # echo "$CALLFILE" >> "/tmp/calltomake.txt" I call it by doing a simple ./makecall.sh SOMEPHONE The only problem I'm having now is that I must run the script as user asterisk,

[Asterisk-Users] Placing call files in /var/spool/asterisk/outgoing/ does not work

/var/spool/asterisk/outgoing/max.call: Permission denied, deleting May 24 08:57:27 WARNING[10618]: pbx_spool.c:389 scan_thread: Failed to scan service '/var/spool/asterisk/outgoing/max.call' What am I doing wrong ? Thank you. -- Cheers, Maxim V

[Asterisk-Users] Configuration call hijack for users in a hunting group ?

is for the 2.6 setup ? Can it be done from the AMP web managment portal ? Our setup uses the zapata.conf file for phone configurations, and the call's are coming from PSTN TDM400 FXO card. Thank you. - Cheers, Maxim Vexler (hq4ever). Do u GNU ?

[Asterisk-Users] how to configure my [EMAIL PROTECTED] 1.0.9 to do call forwarding ?

? Can it be done from the user's phone (Sipura 841) ? Can I as the wwwadmin user set it ? Thank you. My knowledge in dialing rules is rather limited (i.e. null...) -- Cheers, Maxim Vexler (hq4ever). Do u GNU ? ___ --Bandwidth and Colocation pro