[asterisk-users] Polycom Spectralink 8002 WiFi Phones

[Cesar Gonzalez]

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 : 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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>
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