Just as an interesting follow-up/additional information, if I place a call to Site 2 on a POTS line, someone at Site 2 answers the call (using one of the Cisco phones) and then transfers it to me across the VPN the call sounds fine.
So I think Bob's question was on the right track with it being a CODEC issue, but I'm not sure how I need to deal with that for the ZAP channel type. Thanks again, Lincoln -- Lincoln King-Cliby, CTS Applications Engineer ControlWorks Consulting, LLC Crestron Authorized Independent Programmer -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lincoln King-Cliby Sent: Monday, November 03, 2008 1:17 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Call quality issue across VPN-> POTS vs SIP Bob, It's conceivable, but how would I verify this and how would I change it if that was the problem? The site that those calls terminate at is using an Asterisk Appliance so most of the config is "done for you" but it is possible to tweak the underlying configuration files (and I also have SSH access so I can do asterisk -vvvvv -r) -- If I know what I need to tweak. Thanks, Lincoln -- Lincoln King-Cliby, CTS Applications Engineer ControlWorks Consulting, LLC Crestron Authorized Independent Programmer -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bob Pierce Sent: Monday, November 03, 2008 12:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call quality issue across VPN-> POTS vs SIP On Mon, 2008-11-03 at 11:14 -0500, Lincoln King-Cliby wrote: > Any ideas why the audio quality would be so markedly different when > the only thing that seems to be different is where the call is > originating from (POTS line vs. SIP phone)? Is it possible that calls from your POTS line are going across the VPN as uLaw while the calls from the sip phones are using a compressed codec? Bob _______________________________________________ -- 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