My understanding is that the sip.conf definitions use qualify=yes, which means Asterisk should send a SIP NOTIFY message to each handset every 60 seconds or so, retransmitting 7 times if a response isn't received and the peers should go UNKNOWN if they fail to respond after the 7 attempts. Using SIP debug I can see quite a number of these messages passing along with SIP OK in return.
However, after a varying period of time it appears that Asterisk no longer sends the OPTIONS messages, and the next time I run sip show peers a handset or two will show UNKNOWN status and cannot be called. If I search back through the SIP debug logs, I cannot see any OPTIONS messages being sent to this handset since the last time I ran the show peers and the previous OK 200 message prior to this.
Features: The IP Handset, TT-403670B, includes the following main units: TT-3672A IPHandset TT-3674A IP cradle This IP Handset connects wirelessly to the below deck unit. The cradle connects with a fixed LAN cable to a LAN port with PoE, for example in a BGAN terminal (Broadband Global Area Network). Type TT-3672A IP Handset Display 2.2 , 240 x 320 pixel TFT color LCD Operating temperature -25°C to +45°C Storage temperature -25°C to +55°C Humidity Up to 95% without condensation Power Power over Ethernet (PoE) class 2 Alternative B of IEEE802.3af is not supported Power consumption Max. 7 Watt Protection category IP55, dust proof and splash proof LAN interface 10/100 Mbps Network Protocol Internet Protocol (IP) VoIP Protocol SIP v2 Session Initiation Protocol (RFC3261), SDP (RFC2327) Voice Codecs G.711 and G.729 A/B Physical interfaces RJ-45 male connector on fixed cable Headset 2.5 mm jack Mini USB Certifications CE, EN60950
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