[asterisk-users] Polycom low volume

Michael Graves mgraves at mstvp.com
Sun Nov 16 08:12:19 CST 2008


Looking at some old Polycom reference configs I find the following from
the phonexxxxxxxx.cfg

"The user’s selection of the receive volume during a call can be
remembered between calls. This can be configured per termination
(handset, headset and hands-free/chassis). In some countries
regulations exist which dictate that receive volume should be reset to
nominal at the start of each call on handset and headset."

voice.volume.persist.handset
voice.volume.persist.headset
voice.volume.persist.handsfree

If set to 1 any of these will cause call valume to be remembered once
set by the user. My users like this a lot.

OTOH, more general line level issues can be addressed in the Zaptel
config. Beware of related echo.

Michael

On Sat, 15 Nov 2008 22:55:43 -0800 (PST), hin lee wrote:

>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
>
>
>
>--- 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
>> > >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
>

--
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






More information about the asterisk-users mailing list