Jeff LaCoursiere wrote: > 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 > > Jeff, yeah i saw the posts, i followed Bob Pierce config and had no luck, BUT it just started to work, i changed AP's, seems like theres something wrong with Ubiquiti NanoStation2 WMM implementation, i used a Linksys WRT54G2 and viola! it started to work, i guess i should've done that to begin with... :(
I'll play around whit the Nanostations QoS settings and see if i can get it to work on those AP's. What AP's were you using? -Cesar > 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 > _______________________________________________ -- 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
