Am 13.06.2020 um 08:28 schrieb Luca Bertoncello: > Hi! > > I have a Asterisk installation to manage my phones at home (provider is > Deutsche Telekom). > It works, but very often the voice is "broken"... > Yesterday during a call it was very difficult to understand what my > partner sayd... > > It can NOT be a problem of other downloads/uploads, since in that moment > there were no ones...
Hi again! Just a detail: I tried an internal call (from my phone, to my wife's phone) and it works wonderful, no broken, no delay, top quality. So the problem _MUST_ be in the settings of the communication with Deutsche Telekom and MessageNet (the providers I used). The settings for Deutsche Telekom are: [pbxluca] type=peer defaultuser=<mylogin>-0001 secret= <myverysecretpassword> dtmfmode=rfc2833 host=tel.t-online.de context=luca_incoming outboundproxy=tel.t-online.de port=5060 fromuser=0351xxxxxxx fromdomain=tel.t-online.de usereqphone=yes canreinvite=yes insecure=port,invite nat=force_rport,comedia qualify=yes qualifyfreq=600 disallow=all allow=alaw allow=ulaw and the settings for MessageNet are: [messagenet] type=peer defaultuser=<mylogin> secret=<myveryverysecretpassword> dtmfmode=rfc2833 host=sip.messagenet.it context=messagenet_incoming outboundproxy=sip.messagenet.it port=5060 fromuser=<mylogin> fromdomain=sip.messagenet.it usereqphone=yes canreinvite=yes insecure=invite qualify=yes qualifyfreq=60 disallow=all allow=alaw allow=ulaw allow=gsm Any idea? Thanks a lot 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
