Search the archives - we had a big discussion about this phone about six months ago. If you make it work and want another one "I will give you special price!".
j On Tue, 14 Jul 2009, Cesar Gonzalez wrote: > Has anyone played with this phone? i cant seem to get it to work > properly, i manged to get it registered and can make calls from it, but > i havent been able to make it receive calls. Weird thing its that if you > make a call from it and while you are on that call you dial its number > does calls go thru in second line, but as soon as you terminate both > calls it wont recieve any calls again. > > Heres a look from the asterisk CLI : > > -- Registered SIP '245' at 192.168.0.239 port 5060 expires 60 > trixbox1*CLI> sip show peer 245 > trixbox1*CLI> > > Name : 245 > Secret : Set > MD5Secret : Not set > Context : from-internal > Subscr.Cont. : Not set > Language : > AMA flags : Unknown > Transfer mode: open > CallingPres : Presentation Allowed, Not Screened > Callgroup : > Pickupgroup : > Mailbox : 2...@device > VM Extension : *97 > LastMsgsSent : 32767/65535 > Call limit : 50 > Dynamic : Yes > Callerid : "device" <245> > MaxCallBR : 384 kbps > Expire : 67 > Insecure : no > Nat : RFC3581 > ACL : No > T38 pt UDPTL : No > CanReinvite : No > PromiscRedir : No > User=Phone : No > Video Support: Yes > Trust RPID : No > Send RPID : No > Subscriptions: Yes > Overlap dial : Yes > DTMFmode : rfc2833 > LastMsg : 0 > ToHost : > Addr->IP : 192.168.0.239 Port 5060 > Defaddr->IP : 0.0.0.0 Port 5060 > Def. Username: 245 > SIP Options : (none) > Codecs : 0x4 (ulaw) > Codec Order : (ulaw:20) > Auto-Framing: No > Status : OK (124 ms) > Useragent : Slnk/12 > Reg. Contact : sip:[email protected]:5060 > > But after a few seconds the Status goes to UNKNOWN : > > Auto-Framing: No > Status : UNKNOWN <<------ > Useragent : Slnk/12 > Reg. Contact : sip:[email protected]:5060 > > This are the config files : > > sip_245.cfg > AUTH = 245; 123456 > LINE1 = 245 > LINE1_PROXY = 1 > LINE1_CALLID = Wireless > LINE1_AUTH = 245; 123456 > LINE2 = 245 > LINE2_PROXY = 1 > LINE2_CALLID = Wireless > LINE2_AUTH = 245; 123456 > > sip_allusers.cfg > CODECS = g711u, g711a > PROXY1_TYPE = Asterisk > PROXY1_ADDR = 192.168.0.253:5060 > #PROXY1_KEYPRESS_2833 = enable > PROXY1_KEYPRESS_INFO = disable > PROXY1_HOLD_IP0 = disable > #PROXY1_PRACK = enable > PROXY1_REREG_SECS=3600 > PROXY1_KEEPALIVE_SECS=14 > #PROXY1_DOMAIN = 192.168.0.253 > PROXY1_CALLID_PER_LINE = disable > PROXY1_MAIL_ACCESS = *97 > > Access Points are Ubiquitki NanoStation 2 in WDS Mode with QoS enabled. > > One last thing is that while you're on a call you can ping the phone and > soon as the call ends phone stops pinging. > > Any Ideas? > Thanks > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
