similar to: Indications.conf: change volume

Can anyone tell me if there is a way to vary the output levels (dB) of the tones generated in indications.conf? I generate a few custom tones and sometimes people tell me they are a little too loud. Thanks Lee -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090522/7ae8c75d/attachment.htm

Hi, I have defined a dialplan in lua and now would like to use "dial" with the macro M to implement some logic, when the callee-channel gets created. Working old style would be (extensions.conf) [default] exten => _X,1,dial(SIP/1,,M(mymacro^parameter)) [macro-mymacro] exten => s,1,verbose(${ARG1}) How to implement the same functionality using pbx_lua? Details: Asterisk 11.7

I am wondering if anyone could help me figure out how to turn down the volume on all the dial tones, indications, etc.. and especially the call-waiting CHIRP! I don't want to change the txgain and rxgain because they are working at levels that I would like. However, when voice conversations and voicemail recordings are at good levels then the dial tones, busy tones, etc are way too loud.

Hey, I am running default Asterisk 11.16.0 on a FreeBSD-Machine. I need to register to several other SIP-Services (actually 3): short sip.conf register => XX at a register => XX at b register => XX at c If I remember correctly this worked quite well, but I now checked the system again and it is only obeying the first register statement. "sip show registry" only reports the

Anyone noticed the snom 360 indications are incorrect for US zone? menu->preferences->tone scheme->usa indications.conf: [general] country=us extensions.conf: exten => 1111,1,Answer exten => 1111,n,Playtones(dial) exten => 1111,n,Wait(30) exten => 2222,1,Busy exten => 3333,1,Answer exten => 3333,n,Playtones(busy) exten => 3333,n,Wait(30) hit speakerphone on the

Hi all, i am building an authentification system based upon MIT kerberos. I set up a realm in which I can use the ftp and telnet server/clients infrastructure for the purpose of authentification. To backup my server i would like to use rsync, but instead of a ssh/rsh/stunnel with an encrypted kerberized telnet. My problem is, that i could connect manually via telnet.krb5, but if like to use it

I've noticed that native music on hold volume seems to be very loud sometimes. Is there anyway to turn this down? I know when using mpg123 I can set "quietmp3" but what about when using native?

We generally get our voicemails emailed to us from asterisk, but some people's messages are extraordinarily loud or quiet. I don't suppose there is any feature to even out the volume level is there?

Hi, Just wondering if someone could better explain how the indications.conf file actually affects Asterisk? I am using a Cisco 7940 from my Asterisk system, and have set in indications.conf "country=au" thinking that this would make the dialtones/call progress sound like the familiar Australian tones? However when I call another extension on my system, it still sounds like

Hi all, I' m currently testing a Wifi UT Starcom F1000 and for my taste even with the audio volume level even on max. setting, the speaker is not loud enogh. Is there a way to raise the audio-level specifically for a concrete SIP device via a sip.conf setting or something like that ?? TIA, Bruno

Hi, I have Asterisk 1.2 + Adit600 Channel bank(which gives analog output and also takes PSTN lines into the Asterisk system). Conversations recorded by the ASTERISK comes in two separate Files: xxxxxx.0-in (GSM Audio) for the Asterisk Extension Side of the conversation; xxxxxxx.0-out (GSM Audio) for the Caller's side of the conversation. I have Quick Time Player to playback the

Is there a way to decrease the volume on the native files version of MOH in 1.4? I've had several people complain that it is too loud. Peder

Hello, i have succesfullu setup asterisk with Sangoma E1 card, evrything works well but i don't know how to pass indications from telco switch to the user - when users call bad number telco switch shuld talk "unallocated number" but its only send PRI_CAUSE 1. How to pass voice indications thru asterisk to clients? My /etc/zaptel.conf: span=1,0,0,CCS,HDB3,CRC4 dchan=16

You have the point. "I get very distorted speech when using libspeex in my application. What¡¯s wrong? .... If the input speech has an amplitude close to ¡¾2^15, it is possible that at decoding, the amplitude be a bit higher than that, causing clipping when saving as 16-bit PCM." This is one of FAQs in 'speex-manual' you wrote. I guess amplitudes close to 2^15 bother me. Do

Have a problem which can't find solution to on WIKI.. Trying to get * to use UK based indication tones. i.e. british ring, dial tone, busy signal. Have changed the indications.conf file to default to UK. However this seems to have no affect. What am i missing. Am using 1.0.3 stable. Many thanks Andrew. ---------------------- indications.conf [general] country=uk [uk] description =

Apologies if this is an old question; I've searched the list and the wiki but have not been able to find a definitive answer. I have an Aastra 480i phone registered with * 1.2.4; I want to generate UK ringback tones when the handset dials another internal extension. On my Zap channels, I have this in place by editing /etc/zaptel.conf; however I've had no luck with the Sip handset (I have

Ok, Here's another bizarre one (no strange curve balls to throw this time :P). I have several mp3 files of some easy listening music that I pulled off some CDs we have. They sounds fine and are at a nice volume level. When I run this script I wrote: (I run it by doing ./script < filename) mpg123 -s --rate 44100 --mono $1.mp3 > $1.raw sox -r 44100 -w -s -c 1 $1.raw -r 8000 -c 1 $1.wav

Hi All I've got an asterisk system, using a couple of Xorcom Astribanks to provide FXS ports. (I'm using the zaptel 1.2 branch, if that matters) I've noticed that the ringing volume is a lot louder than on our old phone system, and people are starting to complain it's too loud. (This is the noise the phone makes when it rings, not the noise in your handset when you ring

Hi, I?m trying to make * work with Brazilian analog signalling.. I?m using the following in indications.conf file... [br] description = Brasil ringcadence = 1000,4000 dial = 425 busy = 425/250,0/250 ring = 425/1000,0/4000 callwaiting = 425/60,0/250,425/60,0/5000 I changed zaptel.conf to loadzone=br #loadzone=fr #loadzone=de #loadzone=uk #loadzone=fi #loadzone=jp #loadzone=sp #loadzone=no

Hi, Just noticed Asterisk is not playing 'ring' tone as defined in indications.conf when Dial command is used with 'r' option. For example: [test] exten => 123,1,PlayTones(ring) exten => 123,n,Wait(5) exten => 123,n,Playback(demo-congrats) exten => 123,n,Hangup() exten => 321,1,Dail(LOCAL/123 at test/n,60,r) When I now dial with a SIP phone - 123 I can hear nice