it should work for sip channel too. I recorded the downlink channel in
wav-format. Does the rx or txgain ajusting only work with alaw or ulaw?


2011/3/7 Faisal Hanif <[email protected]>

> This settings are for ISDN configurations I think.
>
>
>
> *From:* [email protected] [mailto:
> [email protected]] *On Behalf Of *Felix Dong
> *Sent:* Monday, March 07, 2011 6:07 PM
>
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] Loudness of recorded wav-audio
>
>
>
> I tried to ajust the tx- and rxgain for the sip peer in sip.conf. And
> restarted the asterisk. But it takes no effect. Any suggestion?
>
> 2011/3/4 Danny Nicholas <[email protected]>
>
> Defaults are 0.0 (leave volume unchanged)  +values make volume louder, -
> softer.
>
>
> ------------------------------
>
> *From:* [email protected] [mailto:
> [email protected]] *On Behalf Of *Felix Dong
> *Sent:* Friday, March 04, 2011 8:55 AM
>
>
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] Loudness of recorded wav-audio
>
>
>
> Could yoz tell me the default value of rxgain or txgain, if there is no
> rxgain or txgain in conf-data defined?
>
> Von meinem iPad gesendet
>
>
> Am 04.03.2011 um 15:34 schrieb "Danny Nicholas" <[email protected]>:
>
> In sip.conf, add rxgain=-4.0 to the peer.  This (feel free to correct)
> should reduce the incoming volume by 4 decibels. You’ll have to do a “sip
> reload” for this to take effect.
>
>
> ------------------------------
>
> *From:* [email protected] [mailto:
> [email protected]] *On Behalf Of *Felix Dong
> *Sent:* Friday, March 04, 2011 8:33 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] Loudness of recorded wav-audio
>
>
>
> Thank you! How can I reduce the RXgain?
>
>
> Am 04.03.2011 um 15:21 schrieb "Danny Nicholas" <[email protected]>:
>
> ------------------------------
>
> *From:* [email protected] [mailto:
> [email protected]] *On Behalf Of *Felix Dong
> *Sent:* Friday, March 04, 2011 2:31 AM
> *To:* [email protected]
> *Subject:* [asterisk-users] Loudness of recorded wav-audio
>
>
>
> Hello,
>
>
>
> I sent a wav-audio to Asterisk though SIP and ISDN channels and recorded it
> in wav-audio at the Asterisk server. I found the loudness level of the
> recorded audio was too high comparing with the orginal audio. How can I
> ajust it, so that there will be no amplifier used for recording.
>
> Thanks a lot.
>
>
> best regards
>
> Felix
>
>
>
> two options are:
>
>    1. reduce RXgain – assuming your are using Record() command
>    2. use sox to reduce the volume;  something like sox –v .8 file1.wav
>    file2.wav
>
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>               http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>               http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>               http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>               http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
               http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Reply via email to