il ping verso messagenet e' a posto:
Messagenet2/535xxxx 212.97.59.76 N 5061 OK (67 ms)
messagenet/539xxx 212.97.59.76 N 5061 OK (67 ms)
idem lo speedtest:
[email protected]:~# ./speedtest
Retrieving speedtest.net configuration...
Retrieving speedtest.net server list...
Testing from NGI SpA (81.xxx.xxx.xxx)...
Selecting best server based on ping...
Hosted by Comeser Srl (Fidenza) [65.01 km]: 24.319 ms
Testing download speed........................................
Download: 9.88 Mbit/s
Testing upload speed..................................................
Upload: 1.00 Mbit/s
per cui non e' un mio problema!
Global Settings:
----------------
UDP SIP Port: 5060
UDP Bindaddress: 0.0.0.0
TCP SIP Port: Disabled
TLS SIP Port: Disabled
Videosupport: No
Textsupport: No
Ignore SDP sess. ver.: No
AutoCreate Peer: No
Match Auth Username: No
Allow unknown access: No
Allow subscriptions: Yes
Allow overlap dialing: Yes
Allow promsic. redir: No
Enable call counters: No
SIP domain support: No
Realm. auth: No
Our auth realm asterisk
Call to non-local dom.: Yes
URI user is phone no: No
Always auth rejects: Yes
Direct RTP setup: No
User Agent: FPBX-2.8.1(1.6.2.24)
SDP Session Name: Asterisk PBX 1.6.2.24
SDP Owner Name: root
Reg. context: (not set)
Regexten on Qualify: No
Caller ID: Unknown
From: Domain:
Record SIP history: Off
Call Events: Off
Auth. Failure Events: Off
T.38 support: No
T.38 EC mode: Unknown
T.38 MaxDtgrm: -1
SIP realtime: Disabled
Qualify Freq : 60000 ms
Network QoS Settings:
---------------------------
IP ToS SIP: CS3
IP ToS RTP audio: EF
IP ToS RTP video: AF41
IP ToS RTP text: CS0
802.1p CoS SIP: 4
802.1p CoS RTP audio: 5
802.1p CoS RTP video: 6
802.1p CoS RTP text: 5
Jitterbuffer enabled: Yes
Jitterbuffer forced: No
Jitterbuffer max size: 200
Jitterbuffer resync: 1000
Jitterbuffer impl: adaptive
Jitterbuffer log: No
Network Settings:
---------------------------
SIP address remapping: Enabled using externip
Externhost: <none>
Externip: 81.xxx.xxx.xxx.:5060
Externrefresh: 10
Localnet: 192.168.1.0/255.255.255.0
127.0.0.0/255.255.255.0
STUN server: 0.0.0.0:0
Global Signalling Settings:
---------------------------
Codecs: 0x1d1e (gsm|ulaw|alaw|g726|g729|ilbc|g726aal2|g722)
Codec Order: ulaw:20,alaw:20,gsm:20,g722:20,g729:20,g726:20,g726aal2:20,ilbc:30
Relax DTMF: No
RFC2833 Compensation: No
Compact SIP headers: No
RTP Keepalive: 0 (Disabled)
RTP Timeout: 30
RTP Hold Timeout: 300
MWI NOTIFY mime type: application/simple-message-summary
DNS SRV lookup: No
Pedantic SIP support: No
Reg. min duration 60 secs
Reg. max duration: 3600 secs
Reg. default duration: 120 secs
Outbound reg. timeout: 120 secs
Outbound reg. attempts: 0
Notify ringing state: Yes
Include CID: No
Notify hold state: Yes
SIP Transfer mode: open
Max Call Bitrate: 256 kbps
Auto-Framing: No
Outb. proxy: <not set>
Session Timers: Accept
Session Refresher: uas
Session Expires: 1800 secs
Session Min-SE: 90 secs
Timer T1: 500
Timer T1 minimum: 100
Timer B: 32000
No premature media: Yes
Default Settings:
-----------------
Allowed transports: UDP
Outbound transport: UDP
Context: from-sip-external
Nat: Always
DTMF: rfc2833
Qualify: 0
Use ClientCode: No
Progress inband: Never
Language: it
MOH Interpret: default
MOH Suggest:
Voice Mail Extension: *97
Forward Detected Loops: Yes