[asterisk-users] Polycom low volume

Doug Doug at NaTel.net
Mon Nov 17 21:57:17 CST 2008 

At 15:26 11/17/2008, hin lee wrote:
 >Anybody know why the volume on calls are so low?  How can I increase
 >the volume?
 >
 >
 >--- On Sat, 11/15/08, hin lee <hin87 at yahoo.com> wrote:
 >
 >> From: hin lee <hin87 at yahoo.com>
 >> Subject: Re: [asterisk-users] Polycom low volume
 >> To: "Asterisk Users" <asterisk-users at lists.digium.com>
 >> Date: Saturday, November 15, 2008, 10:55 PM
 >> Links to my configuration files for the polycom phone.  As
 >> you'll see, majority of my settings are default.  Hope
 >> it will help you to determine where my problem is at.
 >>
 >>
 >> MAC Address cfg file
 >> http://docs.google.com/Doc?id=dggrkn86_2dc3qfdgr&hl=en
 >>
 >> Extension cfg file
 >> http://docs.google.com/Doc?id=dggrkn86_5ss94tkf6&hl=en
 >>
 >> Phone cfg file
 >> http://docs.google.com/Doc?id=dggrkn86_4csdzthf9&hl=en
 >>
 >> Server cfg file
 >> http://docs.google.com/Doc?id=dggrkn86_6gp7hr9fg&hl=en
 >>
 >> SIP cfg file
 >> http://docs.google.com/Doc?id=dggrkn86_3djzb86d7&hl=en

"gains" section of my sip.cfg:

<gains
voice.gain.rx.analog.handset="0"
voice.gain.rx.analog.headset="0"
voice.gain.rx.analog.chassis="0"
voice.gain.rx.analog.chassis.IP_300="-6"
voice.gain.rx.analog.chassis.IP_330="0"
voice.gain.rx.analog.chassis.IP_4000="3"
voice.gain.rx.analog.chassis.IP_430="0"
voice.gain.rx.analog.chassis.IP_650="0"
voice.gain.rx.analog.chassis.IP_601="6"
voice.gain.rx.analog.ringer="0"
voice.gain.rx.analog.ringer.IP_300="-6"
voice.gain.rx.analog.ringer.IP_330="0"
voice.gain.rx.analog.ringer.IP_4000="3"
voice.gain.rx.analog.ringer.IP_430="0"
voice.gain.rx.analog.ringer.IP_650="0"
voice.gain.rx.analog.ringer.IP_601="6"
voice.gain.rx.digital.handset="-15"
voice.gain.rx.digital.headset="-21"
voice.gain.rx.digital.chassis="0"
voice.gain.rx.digital.chassis.IP_4000="0"
voice.gain.rx.digital.chassis.IP_330="6"
voice.gain.rx.digital.chassis.IP_430="0"
voice.gain.rx.digital.chassis.IP_650="6"
voice.gain.rx.digital.chassis.IP_601="0"
voice.gain.rx.digital.ringer="-21"
voice.gain.rx.digital.ringer.IP_4000="-21"
voice.gain.rx.digital.ringer.IP_330="-12"
voice.gain.rx.digital.ringer.IP_430="-21"
voice.gain.rx.digital.ringer.IP_650="-12"
voice.gain.rx.digital.ringer.IP_601="-21"
voice.gain.rx.analog.handset.sidetone="-14"
voice.gain.rx.analog.headset.sidetone="-24"
voice.gain.tx.analog.handset="12"
voice.gain.tx.analog.headset="3"
voice.gain.tx.analog.chassis="3"
voice.gain.tx.analog.chassis.IP_300="0"
voice.gain.tx.analog.chassis.IP_330="36"
voice.gain.tx.analog.chassis.IP_4000="3"
voice.gain.tx.analog.chassis.IP_430="42"
voice.gain.tx.analog.chassis.IP_650="36"
voice.gain.tx.analog.chassis.IP_601="0"
voice.gain.tx.digital.handset="0"
voice.gain.tx.digital.headset="0"
voice.gain.tx.digital.chassis="3"
voice.gain.tx.digital.chassis.IP_4000="0"
voice.gain.tx.digital.chassis.IP_330="0"
voice.gain.tx.digital.chassis.IP_430="-3"
voice.gain.tx.digital.chassis.IP_650="0"
voice.gain.tx.digital.chassis.IP_601="6"
voice.gain.tx.analog.preamp.handset="14"
voice.gain.tx.analog.preamp.headset="23"
voice.gain.tx.analog.preamp.chassis="32" 
voice.gain.tx.analog.preamp.chassis.IP_430="32" 
voice.gain.tx.analog.preamp.chassis.IP_601="32"/>

Your sip.cfg:

<gains
voice.gain.rx.analog.handset="0"
voice.gain.rx.analog.headset="0"
voice.gain.rx.analog.chassis="0"
voice.gain.rx.analog.chassis.IP_300="-6"
voice.gain.rx.analog.chassis.IP_330="0"
voice.gain.rx.analog.chassis.IP_430="0"
voice.gain.rx.analog.chassis.IP_650="0"
voice.gain.rx.analog.chassis.IP_601="6"
voice.gain.rx.analog.chassis.IP_7000="0"
voice.gain.rx.analog.chassis.IP_6000="0"
voice.gain.rx.analog.ringer="0"
voice.gain.rx.analog.ringer.IP_300="-6"
voice.gain.rx.analog.ringer.IP_330="0"
voice.gain.rx.analog.ringer.IP_430="0"
voice.gain.rx.analog.ringer.IP_650="0"
voice.gain.rx.analog.ringer.IP_601="6"
voice.gain.rx.analog.ringer.IP_7000="0"
voice.gain.rx.analog.ringer.IP_6000="0"
voice.gain.rx.digital.handset="-15"
voice.gain.rx.digital.headset="-21"
voice.gain.rx.digital.chassis="0"
voice.gain.rx.digital.chassis.IP_4000="0"
voice.gain.rx.digital.chassis.IP_330="6"
voice.gain.rx.digital.chassis.IP_430="6"
voice.gain.rx.digital.chassis.IP_650="6"
voice.gain.rx.digital.chassis.IP_601="0"
voice.gain.rx.digital.chassis.IP_7000="6"
voice.gain.rx.digital.chassis.IP_6000="6"
voice.gain.rx.digital.ringer="-21"
voice.gain.rx.digital.ringer.IP_4000="-21"
voice.gain.rx.digital.ringer.IP_330="-12"
voice.gain.rx.digital.ringer.IP_430="-12"
voice.gain.rx.digital.ringer.IP_650="-12"
voice.gain.rx.digital.ringer.IP_601="-21"
voice.gain.rx.digital.ringer.IP_7000="-21"
voice.gain.rx.digital.ringer.IP_6000="-21"
voice.gain.rx.analog.handset.sidetone="-20"
voice.gain.rx.analog.headset.sidetone="-24"
voice.gain.tx.analog.handset="6"
voice.gain.tx.analog.headset="3"
voice.gain.tx.analog.chassis="3"
voice.gain.tx.analog.chassis.IP_300="0"
voice.gain.tx.analog.chassis.IP_330="36"
voice.gain.tx.analog.chassis.IP_430="36"
voice.gain.tx.analog.chassis.IP_650="36"
voice.gain.tx.analog.chassis.IP_601="0"
voice.gain.tx.analog.chassis.IP_7000="0"
voice.gain.tx.analog.chassis.IP_6000="0"
voice.gain.tx.digital.handset="0"
voice.gain.tx.digital.handset.IP_330="10"
voice.gain.tx.digital.handset.IP_430="10"
voice.gain.tx.digital.handset.IP_650="6"
voice.gain.tx.digital.headset="0"
voice.gain.tx.digital.headset.IP_330="10"
voice.gain.tx.digital.headset.IP_430="10"
voice.gain.tx.digital.headset.IP_650="6"
voice.gain.tx.digital.chassis="3"
voice.gain.tx.digital.chassis.IP_4000="0"
voice.gain.tx.digital.chassis.IP_330="12"
voice.gain.tx.digital.chassis.IP_430="12"
voice.gain.tx.digital.chassis.IP_650="12"
voice.gain.tx.digital.chassis.IP_601="6"
voice.gain.tx.digital.chassis.IP_7000="6"
voice.gain.tx.digital.chassis.IP_6000="6"
voice.gain.tx.analog.preamp.handset="23"
voice.gain.tx.analog.preamp.headset="23"
voice.gain.tx.analog.preamp.chassis="32"
voice.gain.tx.analog.preamp.chassis.IP_601="32"/>

I didn't take a close look, but you can adjust
gains in this section.  Reboot the phone
so it reads it in, and see what happens.

Also, I like to have my volumes persist:

       <volume

       voice.volume.persist.handset="1"
       voice.volume.persist.headset="1"
       voice.volume.persist.handsfree="1"/>

Hope this helps.



 >>
 >>
 >>
 >> --- On Sat, 11/15/08, hin lee <hin87 at yahoo.com>
 >> wrote:
 >>
 >> > From: hin lee <hin87 at yahoo.com>
 >> > Subject: Re: [asterisk-users] Polycom low volume
 >> > To: "Doug" <Doug at NaTel.net>,
 >> "Asterisk Users"
 >> <asterisk-users at lists.digium.com>
 >> > Date: Saturday, November 15, 2008, 10:40 PM
 >> > Attached, my configuration files for the polycom
 >> phone.  As
 >> > you'll see, majority of my settings are default.
 >> Hope
 >> > it will help you to determine where my problem is at.
 >> >
 >> > Thanks!
 >> > Hin
 >> >
 >> > --- On Sat, 11/15/08, Doug <Doug at NaTel.net>
 >> wrote:
 >> >
 >> > > From: Doug <Doug at NaTel.net>
 >> > > Subject: Re: [asterisk-users] Polycom low volume
 >> > > To: hin87 at yahoo.com,
 >> asterisk-users at lists.digium.com
 >> > > Date: Saturday, November 15, 2008, 7:20 PM
 >> > > At 21:06 11/15/2008, hin lee wrote:
 >> > > >Here are more information as requested:
 >> > > >
 >> > > >Asterisk v. 1.4 (running PBX in a Flash)
 >> > > >Using Zaptel, TDM800P card
 >> > > >Polycom running:  3.03 SIP Firmware
 >> > > >Provisioning by: FTP
 >> > > >
 >> > > >I am calling from my Polycom to other land
 >> line
 >> > phones.
 >> > >  Hope I
 >> > > >provided enough information.
 >> > >
 >> > > Why don't you post a link to your sip.cfg?
 >> > >
 >> > > Typical PhoneXXXXXXXXXX.cfg?
 >> > >
 >> > >
 >> > > >
 >> > > >Thanks!
 >> > > >Hin
 >> > > >
 >> > > >
 >> > > >--- On Sat, 11/15/08, Darrick Hartman
 >> > > <dhartman at djhsolutions.com> wrote:
 >> > > >
 >> > > >> From: Darrick Hartman
 >> > > <dhartman at djhsolutions.com>
 >> > > >> Subject: Re: [asterisk-users] Polycom
 >> low
 >> > volume
 >> > > >> To: "Asterisk Users Mailing List -
 >> > > Non-Commercial Discussion"
 >> > > ><asterisk-users at lists.digium.com>
 >> > > >> Date: Saturday, November 15, 2008, 1:44
 >> PM
 >> > > >> Actually, it could be within Asterisk,
 >> but
 >> > only if
 >> > > you have
 >> > > >> Zaptel
 >> > > >> hardware.  If you are only using SIP
 >> devices,
 >> > then
 >> > > the
 >> > > >> problem is with
 >> > > >> the phone configuration.  You really
 >> > don't
 >> > > provide
 >> > > >> enough information to
 >> > > >> determine what is causing your problem.
 >> How
 >> > are
 >> > > you
 >> > > >> provisioning the
 >> > > >> phones?  What version of the SIP
 >> firmware is
 >> > used
 >> > > on the
 >> > > >> phones?  Are
 >> > > >> you calling from one phone to the other?
 >> > > >>
 >> > > >> Darrick
 >> > > >>
 >> > > >> Michael Graves wrote:
 >> > > >> > Probably has nothing to do with
 >> > Asterisk. You
 >> > > can set
 >> > > >> the volume and
 >> > > >> > persistence in the phones config
 >> files.
 >> > > >> >
 >> > > >> > Michael
 >> > > >> >
 >> > > >> > On Fri, 14 Nov 2008 22:43:45 -0800
 >> > (PST), hin
 >> > > lee
 >> > > >> wrote:
 >> > > >> >
 >> > > >> >> Using a Polycom 550 and 650
 >> phones
 >> > on my
 >> > > Asterisk
 >> > > >> server for testing.  I can't figure
 >> out
 >> > why
 >> > > the volume
 >> > > >> is so low.  How can I adjust the volume
 >> > control on
 >> > > Asterisk?
 >> > > >>  It's at max on the handset phones.
 >> > > >> >>
 >> > > >> >> Thanks!
 >> > > >> >> Hin
 >> > > >> >>
 >> > > >> >>
 >> > > >> >>
 >> > > >> >>
 >> > > >> >>
 >> > > _______________________________________________
 >> > > >> >> -- Bandwidth and Colocation
 >> Provided
 >> > by
 >> > > >> http://www.api-digital.com --
 >> > > >> >>
 >> > > >> >> asterisk-users mailing list
 >> > > >> >> To UNSUBSCRIBE or update
 >> options
 >> > visit:
 >> > > >> >>
 >> > > >>
 >> > >
 >> >
 >> http://lists.digium.com/mailman/listinfo/asterisk-users
 >> > > >> >>
 >> > > >> >
 >> > > >> > --
 >> > > >> > Michael Graves
 >> > > >> > mgraves<at>mstvp.com
 >> > > >> > http://blog.mgraves.org
 >> > > >> > o713-861-4005
 >> > > >> > c713-201-1262
 >> > > >> > sip:mjgraves at pixelpower.onsip.com
 >> > > >> > skype mjgraves
 >> > > >> > fwd 54245
 >> > > >> >
 >> > > >> >
 >> > > >> >
 >> > > >> >
 >> > > >> >
 >> > > _______________________________________________
 >> > > >> > -- Bandwidth and Colocation
 >> Provided by
 >> > > >> http://www.api-digital.com --
 >> > > >> >
 >> > > >> > 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 --
 >> > > >>
 >> > > >> 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 --
 >> > > >
 >> > > >asterisk-users mailing list
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 >> > > >
 >> > >
 >> >
 >> http://lists.digium.com/mailman/listinfo/asterisk-users
 >>
 >>
 >>
 >>
 >> _______________________________________________
 >> -- Bandwidth and Colocation Provided by
 >> http://www.api-digital.com --
 >>
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 >
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 >
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