[asterisk-users] Polycom Spectralink 8002 WiFi Phones

[Jeff LaCoursiere]

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 : 245 at 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:245 at 192.168.0.239:5060
>
> But after a few seconds the Status goes to UNKNOWN :
>
> Auto-Framing: No
> Status : UNKNOWN <<------
> Useragent : Slnk/12
> Reg. Contact : sip:245 at 192.168.0.239:5060
>
> 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
>
>
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