I have found with a number of clients to who we have installed the LinkSys phones, that when you get the input gains to 6, that the phones have a tendency to pick up too much background noise. Have you experienced this at all?
Cheers, Daniel Cole -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kev S Sent: Wednesday, 9 January 2008 12:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Linksys SPA-9xx Audio Issues The issues i have been having are probably similar to the original message, I use the Linksys 9XX Series phones and we used to always receive complaints from the person we were calling that they could hardly hear us. I fixed this by: Going into the Phone section of the config and setting the Handset, Speakerphone and Headset input gain to 6. And i also went into SIP and changed the RTP Packet Size to 0.020 This resolved the low volume issue, Sorry if you have a no sound issue, but thats how i resolved very low volume. Phones sound great now! Regards, Kevin Sandalin Daniel Cole wrote: > Can you describe the issue more please? Can the remote person not hear you at > all? Or is there distorted/broken voice? > > > Cheers, > > Daniel Cole > > > -----Original Message----- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Andrew > Joakimsen > Sent: Wednesday, 9 January 2008 9:26 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [asterisk-users] Linksys SPA-9xx Audio Issues > > Anyone else have problems with phones like SPA-922, SPA-921, etc? > Inbound audio is perfect but the remote end reports audio quality issues on > the audio the handset is sending out. It's not the network.... I've tried > asterisk 1.2, 1.4. I've used ulaw, G726, G793 & G729. Ulaw seems to be the > least problematic but its still an issue. > Doesn't matter if we send the calls to the PSTN via IAX or SIP or PRI. > I don't know it if happens all the time but about 40% of the time the remote > caller reports they cannot hear me. > > _______________________________________________ > -- 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 > -- This message has been scanned for viruses and dangerous content by Mail Call antivirus software, and is believed to be clean. _______________________________________________ -- 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
