<HTML>
<FONT SIZE="2" POINTSIZE="11" DEFAULT="SIZE">I have several models running; IP335, IP430, IP450 & IP60. I'm running<BR>
v3.2.1.<BR>
<BR>
I don't have convenient access to the configs at the moment. But if you<BR>
download the newer firmware from Polycom their stock configs will give<BR>
you all that you need.<BR>
<BR>
http://downloads.polycom.com/voice/voip/sip_sw_releases_matrix.html<BR>
<BR>
Michael<BR>
<BR>
<BR>
--Original Message Text---<BR>
<B>From:</B> hin lee<BR>
<B>Date:</B> Sat, 30 Jan 2010 16:01:19 -0800 (PST)<BR>
<BR>
Which model do you have? what version of sip are you running on the Polycom? I am on sip 3.0.2. Can you email me your sip.cfg so I can compare the differences? <BR>
<BR>
<BR>
<B><FONT FACE="Tahoma"><FONT SIZE="2" POINTSIZE="10">From:</B> Michael Graves <mgraves@mstvp.com><BR>
<B>To:</B> Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com><BR>
<B>Sent:</B> Sat, January 30, 2010 9:02:44 AM<BR>
<B>Subject:</B> Re: [asterisk-users] microphone on Polycom 550/650<BR>
<BR>
There's no issue here. There are parameters in the Polycom config files for the various gain settings. The headset, handset and speakerphone volume and mic gains are all separate. They can be tweaked as you like via the config files.<BR>
<BR>
Michael<BR>
<BR>
--Original Message Text---<BR>
<B>From:</B> hin lee<BR>
<B>Date:</B> Sat, 30 Jan 2010 08:37:24 -0800 (PST)<BR>
<BR>
Yes, the external calls are going over DAHDI. The problem is on the Polycom phones b/c if I pick up the handset, the other end can hear me fine. The problem is when using the hands-free (speakerphone) instead the handset. <BR>
<BR>
Here are some similar posting of the same issue.<BR>
<BR>
http://www.trixbox.org/forums/vendor-forums-certified/polycom/increasing-speakerphone-tx-gain<FONT COLOR=000000 DEFAULT="COLOR"><BR>
http://www.trixbox.org/forums/vendor-moderated-forums/polycom/430-sound-volume-gain<BR>
<BR>
Most of our phones are IP 550. Where and what do I need to adjust the setting to fix this issue? Any Polycom experts in this mailing list?<BR>
<BR>
<BR>
<B>From:</B> Danny Nicholas <danny@debsinc.com><BR>
<B>To:</B> Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com><BR>
<B>Sent:</B> Fri, January 29, 2010 10:18:29 AM<BR>
<B>Subject:</B> Re: [asterisk-users] microphone on Polycom 550/650<BR>
<BR>
<BR>
<BR>
<FONT FACE="Arial" DEFAULT="FACE">You don’t state this, but the assumption would be that your external calls are DAHDI based, so you might need to tweak txgain in dahdi.conf. <BR>
<BR>
<FONT FACE="Times New Roman"><FONT SIZE="3" POINTSIZE="12"><BR>
<BR>
<BR>
<B><FONT FACE="Tahoma"><FONT SIZE="2" POINTSIZE="10">From:</B> asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] <B>On Behalf Of </B>hin lee<BR>
<B>Sent:</B> Friday, January 29, 2010 12:08 PM<BR>
<B>To:</B> Asterisk Users<BR>
<B>Subject:</B> [asterisk-users] microphone on Polycom 550/650 <BR>
<BR>
<BR>
<FONT FACE="Times New Roman"><FONT SIZE="3" POINTSIZE="12"><BR>
<BR>
<FONT SIZE="2" POINTSIZE="10">I have quite a number of users complaining that when they are using handsfree to talk over a landline, the other end can't hear them. It's like the person is speaking 5 feet away and can barely hear their voice. However between internal SIP calls, it's fine.<BR>
<BR>
What could be the problem? <BR>
<BR>
<BR>
<BR>
<FONT SIZE="3" POINTSIZE="12"><BR>
<BR>
<BR>
<BR>
<BR>
<BR>
<BR>
<FONT FACE="Arial" DEFAULT="FACE"><FONT SIZE="2" POINTSIZE="10">--<BR>
Michael Graves<BR>
mgraves<at>mstvp.com<BR>
http://www.mgraves.org<BR>
o713-861-4005<BR>
c713-201-1262<BR>
sip:mgraves@mstvp.onsip.com<BR>
skype mjgraves<BR>
Twitter mjgraves<BR>
<BR>
<BR>
<BR>
<BR>
<BR>
<BR>
<BR>
</HTML>
<HTML>
<LEFT>
<FONT FACE="Arial" COLOR="#000001" SIZE="2" POINTSIZE="11" DEFAULT="ALL">
--<br>
Michael Graves<br>
mgraves<at>mstvp.com<br>
http://www.mgraves.org<br>
o713-861-4005<br>
c713-201-1262<br>
sip:mgraves@mstvp.onsip.com<br>
skype mjgraves<br>
Twitter mjgraves<br>
<br>
</HTML>