Hi, is there a way of controlling the volume of a call? I'm bridging calls with a JS script. Sometimes the people getting the calls complain the volume is too low. I've recorded a few of the calls and most of the times, while playing the recorded wav files, the volume of LegB (second leg of the bridge) is pretty hard to hear, even with the computer and player volume to the max (ok, it's a laptop, but even with headphones). I saw there are some volume control options for conferences, but I couldn't find anything for regular "originate calls" or bridges. I am doing transcoding, so that might help (?). Thank you very much for your help. Best, Nicolas
The volume should really be set on the devices who are originally encoding the audio (the phone or analog card) Digital audio never changes so the server is not the right place to mess with the volume because you will have to actually manipulate the digital signal to do it. We have a way but I recommend you find the real source of your problem. <action application="set_audio_level" data="read 1"/> change read to write if you want to do it going the other way On Mon, Jan 4, 2010 at 12:45 AM, Nicolas Brenner <nicolas@medularis.com>wrote:
Hi, is there a way of controlling the volume of a call? I'm bridging calls with a JS script. Sometimes the people getting the calls complain the volume is too low. I've recorded a few of the calls and most of the times, while playing the recorded wav files, the volume of LegB (second leg of the bridge) is pretty hard to hear, even with the computer and player volume to the max (ok, it's a laptop, but even with headphones). I saw there are some volume control options for conferences, but I couldn't find anything for regular "originate calls" or bridges. I am doing transcoding, so that might help (?).
Thank you very much for your help.
Best,
Nicolas
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
Great! Thanks! I'll play around with those setting to see how it goes. Can I set that variable "per leg"? (instead of globally for a call). The devices getting the calls are regular phones and the termination service is provided by a few different VoIP companies. I'd say the volume problem has to do with bad or poorly configured GSM gateways (that's how they make calls to cellphones), plus maybe some problems in the GSM network relating to poor signal or something like that. I can't really control the PSTN or the GSM network and have almost zero influence with the VoIP companies, so my best bet now is to mess with the transcoding. Thank you very much, we'll see how it goes. Nicolas On Mon, Jan 4, 2010 at 1:30 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
The volume should really be set on the devices who are originally encoding the audio (the phone or analog card) Digital audio never changes so the server is not the right place to mess with the volume because you will have to actually manipulate the digital signal to do it. We have a way but I recommend you find the real source of your problem.
<action application="set_audio_level" data="read 1"/>
change read to write if you want to do it going the other way
yes the example is per_leg per_direction On Mon, Jan 4, 2010 at 12:15 PM, Nicolas Brenner <nicolas@medularis.com>wrote:
Great! Thanks! I'll play around with those setting to see how it goes. Can I set that variable "per leg"? (instead of globally for a call). The devices getting the calls are regular phones and the termination service is provided by a few different VoIP companies. I'd say the volume problem has to do with bad or poorly configured GSM gateways (that's how they make calls to cellphones), plus maybe some problems in the GSM network relating to poor signal or something like that. I can't really control the PSTN or the GSM network and have almost zero influence with the VoIP companies, so my best bet now is to mess with the transcoding.
Thank you very much, we'll see how it goes.
Nicolas
On Mon, Jan 4, 2010 at 1:30 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
The volume should really be set on the devices who are originally encoding the audio (the phone or analog card) Digital audio never changes so the server is not the right place to mess with the volume because you will have to actually manipulate the digital signal to do it. We have a way but I recommend you find the real source of your problem.
<action application="set_audio_level" data="read 1"/>
change read to write if you want to do it going the other way
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
FYI, I just wikified this dp app: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_set_audio_level Feel free to add/edit as needed. -MC On Mon, Jan 4, 2010 at 11:16 AM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
yes the example is per_leg per_direction
On Mon, Jan 4, 2010 at 12:15 PM, Nicolas Brenner <nicolas@medularis.com>wrote:
Great! Thanks! I'll play around with those setting to see how it goes. Can I set that variable "per leg"? (instead of globally for a call). The devices getting the calls are regular phones and the termination service is provided by a few different VoIP companies. I'd say the volume problem has to do with bad or poorly configured GSM gateways (that's how they make calls to cellphones), plus maybe some problems in the GSM network relating to poor signal or something like that. I can't really control the PSTN or the GSM network and have almost zero influence with the VoIP companies, so my best bet now is to mess with the transcoding.
Thank you very much, we'll see how it goes.
Nicolas
On Mon, Jan 4, 2010 at 1:30 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
The volume should really be set on the devices who are originally encoding the audio (the phone or analog card) Digital audio never changes so the server is not the right place to mess with the volume because you will have to actually manipulate the digital signal to do it. We have a way but I recommend you find the real source of your problem.
<action application="set_audio_level" data="read 1"/>
change read to write if you want to do it going the other way
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
participants (3)
-
Anthony Minessale
-
Michael Collins
-
Nicolas Brenner