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" > <[email protected]> > 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], [email protected] > > 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" > > ><[email protected]> > > >> 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
