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... I already had the problem in the past, solved it enabling the jitter, but the problem with jitter enabled was a long delay (1-2 seconds) in the communication, so I disabled it. Can someone suggest me what can I do? I can send extract of my configuration if needed. Thanks a lot Luca Bertoncello (lucabert at lucabert.de)
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 (lucabert at lucabert.de)
Am 13.06.2020 09:30, schrieb Luca Bertoncello: Hi again (again) I noticed right now another strange detail... I made a call using my mobile phone (connected to the Asterisk). The quality was top... Maybe is the problem in a codec used from our phones at homes? Could someone suggest me how to check the codec used by my mobile phone and the codec used by the phones at home? Thanks Luca Bertoncello (lucabert at lucabert.de)
Hi Luca, Am Samstag, den 13.06.2020, 08:28 +0200 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...The product is "All-IP" and not the SIP trunk, right? The call starts normally and after about 15 minutes the quality is disturbed? Try to set "session-timers = refuse" in the sip.conf in the global section. I have observed that when updating the session this error occurs. Best regards, Karsten
Am 17.06.2020 14:37, schrieb Karsten Wemheuer: Hi Karsten!> The product is "All-IP" and not the SIP trunk, right? > The call starts normally and after about 15 minutes the quality is > disturbed?No, current we have Magenta Zuhause. Tomorrow we'll change to DeutschlandLAN IP (business contract). The quality is disturbed from the first second... I had the problem, that the connection will be *dropped* after 15 minutes, and I solved it with "session-timers = refuse" Bye Luca Bertoncello (lucabert at lucabert.de)