Ok, no worries :) Most of our clients have a relatively open common work area, where the phones are located. I would be interested to know what your sales manager has experienced.
Cheers, Daniel Cole -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kev S Sent: Wednesday, 9 January 2008 2:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Linksys SPA-9xx Audio Issues No, I haven't experienced this. I think were lucky because most voip phones are in there own offices, I will check with our sales manager this afternoon who sits in the call center and see what the background noise is like on her phone. I guess i'm just lucky that its a quiet environment, But there are a few people who *may* be affected and i will check this out and let you know. Regards, Kevin Daniel Cole wrote: > 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 > -- 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
