Looking at some old Polycom reference configs I find the following from the phonexxxxxxxx.cfg
"The users 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 <[EMAIL PROTECTED]> wrote: > >> From: hin lee <[EMAIL PROTECTED]> >> Subject: Re: [asterisk-users] Polycom low volume >> To: "Doug" <[EMAIL PROTECTED]>, "Asterisk Users" >> <asterisk-users@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 <[EMAIL PROTECTED]> wrote: >> >> > From: Doug <[EMAIL PROTECTED]> >> > Subject: Re: [asterisk-users] Polycom low volume >> > To: [EMAIL PROTECTED], asterisk-users@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 >> > <[EMAIL PROTECTED]> wrote: >> > > >> > >> From: Darrick Hartman >> > <[EMAIL PROTECTED]> >> > >> Subject: Re: [asterisk-users] Polycom low >> volume >> > >> To: "Asterisk Users Mailing List - >> > Non-Commercial Discussion" >> > ><asterisk-users@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:[EMAIL PROTECTED] >> > >> > 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:[EMAIL PROTECTED] 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