Am 11.06.2019 um 21:10 schrieb Antony Stone: Hi,
> So, you have a SIP phone, connected to an Asterisk server on your local > network, which then connects to D Telekom's SIP server over the DSL line? Correct! >> The other party use VoIP, too, since they are in Germany (and Italy) and >> here there are just VoIP... Sigh! > > Are they also using a SIP phone? My mother yes, my father in law uses an ISDN phone connected to a FritzBox that convert the signal in VoIP. > Do they also have an Asterisk server on their local network? > >> Now I disabled the jitter (jbenable = no), and I called my father in >> law. He sayd me, the quality is really better, but I hear sometimes >> little noises... >> >> Any other suggestion? > > Have you considered trying some tool such as http://sipcapture.org/#about to > see if you can identify where the latency comes in? I must say, that I'm not an expert in VoIP, so I really don't know this tool and don't have any idea how to analyze the problem... Thanks Luca Bertoncello ([email protected]) -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
