[asterisk-users] Polycom Spectralink 8002 Configuration

Tue Feb 24 20:34:21 CST 2009

It seems to me that based upon your comments you miss the point of the
product. It's design targets large commercial concerns, school
campuses, corporate parks, etc...not making free calls from Starbucks.

I had one under test for several months and it behaved really well on
my WLAN using a Netgear comsumer N type rouiter/AP with WMM. WMM is
essentially a wireless QoS mechanism. Without it you cannot assure
voice quality if there's anything else using the WLAN.

Granted, the phone is a bit fiddly to provision. In it's intended
target markets that's not a problem. If you want to make free calls
from hotspots you're far better of with trashy consumer oriented stuff
that has a built-in web browser. In many cases you need it to
authenticate against the hotspot.

The best option seems to be a SIP client on a dual mode cell phone. But
then, why use the wifi when you have a cell phone in your hand? Minutes
are cheap in either case.

Michael

On Tue, 24 Feb 2009 22:50:53 +0000 (UTC), Jeff LaCoursiere wrote:

>I have one of these seemingly useless devices too.  Please let me know if 
>you get anywhere with it.  I bought it thinking it would be a good phone 
>to take around to various hotspots and keep my extension.  Turns out it 
>really wants to be only in its home "hotspot" and has some stringent 
>restrictions on wifi options (WMM ONLY?!?!) that will probably not be 
>present at Starbucks.  I'm pretty disgusted with it.  Bummer, too, because 
>otherwise Polycom has fantastic VoIP products.
>
>j
>
>On Mon, 23 Feb 2009, M Hulber wrote:
>
>> I have a new Polycom Spectralink 8002 and am having trouble with the
>> configuration or the unit but I can't see what's wrong.  The unit does
>> not seem to even attempt to register with the Asterisk proxy but I can
>> make calls to it.  I have viewed the syslog from the device which it
>> will actually write to the asterisk server so I know it can be reached.
>> I have also run a sip debug and see no registration traffic from the
>> unit.  It also pulls the configs from the tftp server on the asterisk
>> box ok.
>>
>> Does anyone have a sample set of configs that work?  I have samples for
>> the Polycom side but haven't seen the match on the asterisk side.  Since
>> I don't even see traffic, I can't think that it's even an authentication
>> issue.
>>
>> When I dial from the device it just sits there, basically.
>>
>> MARK.
>>
>> ----------
>>
>> sip_allusers.cfg:  (I've tried most variations on theses settings)
>>
>> ## FOR PROXY1_TYPE = ASTERISK
>>
>> #PROXY1_ADDR = 192.168.2.80:5060        # replace the ip address with
>> the Asterisk Server's Address
>> PROXY1_ADDR = 192.168.2.80      # replace the ip address with the
>> Asterisk Server's Address
>> PROXY1_KEYPRESS_2833 = enable
>> PROXY1_KEYPRESS_INFO = enable
>> PROXY1_HOLD_IP0 = disable
>> PROXY1_PRACK = enable
>> #PROXY1_REREG_SECS=3600
>> PROXY1_REREG_SECS=35
>> PROXY1_KEEPALIVE_SECS=14
>> #PROXY1_DOMAIN = asterisk        # Replace this with your SIP Domain's name
>> PROXY1_CALLID_PER_LINE = disable
>> PROXY1_MAIL_ACCESS = 864                 # Put Your Voice Mail Sytem's
>> Pilot Number here
>>
>> sip_2000.cfg:
>>
>> LINE1         = 2000
>> LINE1_PROXY   = 1
>> LINE1_CALLID  = 2000
>> #LINE1_AUTH    = 2000; 2000
>>
>> sip.conf:
>>
>> ; Polycom Spectralink 8002
>> [2000]
>>   type=friend
>>   host=192.168.3.123
>>   ;port=5060
>>   secret=2000
>>   username=2000
>>   ;fromuser=2000
>>   ;authuser=2000
>>   qualify=no   ; turned this off to stop asterisk side initiated traffic
>>   context=spectra_default
>>   dtmfmode=rfc2833
>>   disallow=all
>>   allow=ulaw
>>   mailbox=99 at default
>>   canreinvite=yes
>>   callgroup=1
>>   pickupgroup=1
>>   accountcode=Home
>>   nat=no
>>
>>
>> Syslog:
>>
>> Feb 23 20:25:06 192.168.3.123 Jan  1 00:18:24.57 0090.7a0a.13f3
>> (192.168.003.123) [0007] Call start, AP 0014.d1c2.70fe (-32 dBm)
>> Feb 23 20:25:09 192.168.3.123 Jan  1 00:18:26.87 0090.7a0a.13f3
>> (192.168.003.123) [0008] Number Abufs: 26
>> Feb 23 20:25:09 192.168.3.123 Jan  1 00:18:26.87 0090.7a0a.13f3
>> (192.168.003.123) [0009] Number Fbufs: 2
>> Feb 23 20:25:09 192.168.3.123 Jan  1 00:18:26.88 0090.7a0a.13f3
>> (192.168.003.123) [000a] Max Number Abufs: 359
>> Feb 23 20:25:09 192.168.3.123 Jan  1 00:18:26.88 0090.7a0a.13f3
>> (192.168.003.123) [000b] Max Number Fbufs: 33
>> Feb 23 20:25:11 192.168.3.123 Jan  1 00:18:29.57 0090.7a0a.13f3
>> (192.168.003.123) [000c] NStat: 0014.d1c2.70fe (-30 dBm), Tx 3704, Rx
>> 43841, BTx 2, BRx 2766, MTx 0, MRx 0, Tx Drop 3 (0.1%), Tx Retry 96
>> (2.7%), Rx Retry 19 (0.0%)
>> Feb 23 20:25:16 192.168.3.123 Jan  1 00:18:33.87 0090.7a0a.13f3
>> (192.168.003.123) [000d] Number Abufs: 46
>> Feb 23 20:25:16 192.168.3.123 Jan  1 00:18:33.87 0090.7a0a.13f3
>> (192.168.003.123) [000e] Number Fbufs: 3
>> Feb 23 20:25:16 192.168.3.123 Jan  1 00:18:34.57 0090.7a0a.13f3
>> (192.168.003.123) [000f] NStat: 0014.d1c2.70fe (-36 dBm), Tx 3707, Rx
>> 43996, BTx 2, BRx 2773, MTx 0, MRx 0, Tx Drop 3 (0.0%), Tx Retry 96
>> (0.0%), Rx Retry 19 (0.0%)
>> Feb 23 20:25:21 192.168.3.123 Jan  1 00:18:39.57 0090.7a0a.13f3
>> (192.168.003.123) [0010] NStat: 0014.d1c2.70fe (-36 dBm), Tx 3708, Rx
>> 44284, BTx 2, BRx 2792, MTx 0, MRx 0, Tx Drop 3 (0.0%), Tx Retry 96
>> (0.0%), Rx Retry 19 (0.0%)
>> Feb 23 20:25:26 192.168.3.123 Jan  1 00:18:44.36 0090.7a0a.13f3
>> (192.168.003.123) [0011] Call end, AP 0014.d1c2.70fe (-36 dBm)
>>
>>
>>
>>
>> _______________________________________________
>> -- 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
>

--
Michael Graves
mgraves<at>mstvp.com
http://blog.mgraves.org
o713-861-4005
c713-201-1262
sip:mgraves at mstvp.onsip.com
skype mjgraves
fwd 54245