Re: [asterisk-users] Polycom Spectralink 8002 WiFi Phones
Michael Graves wrote: On Tue, 14 Jul 2009 22:58:14 + (UTC), Jeff LaCoursiere wrote: 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? Hi Cesar, I did actually get it to work as well, and was using Linksys WRT54G with dd-wrt. I *intended* for the phone to be useful at random wifi hotspots, however, and was a bit disappointed to find that that was not going to work. So it sits on a shelf gathering dust... I had one of these for evalution last spring. The resulting review is here: http://www.smallnetbuilder.com/content/view/30498/80/ They work well enough when paired with suitable APs. According to Polycom you must support WMM or all bets are off. In my case I had a Netgear WRT-2000. I had no issues at all with integration with Asterisk. However, I think that there truly isn't a dedicated Wifi handset that will satisfy if you want to be able to roam the world and make calls from public hotspots. Too many hotspots require a web login before you get access. Thus the best devices for this sort of thing seem to be Nokia dual mode phones with built-in web browsers. Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:mgra...@mstvp.onsip.com skype mjgraves fwd 54245 ___ -- 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, actually the intended use for this phones is at a customers warehouse and its trailer/truck yard, not roaming around public hotspots, so its now going to come down to selecting the proper AP's as you suggest, we were looking at Naonstation2 for their WDS implemetation, so maybe a set of linksys AP's with dd-wrt will do the job. I've been testing it with this Linksys WRT54G2 since yesterday and works great with asterisk. Great review btw. -Cesar ___ -- 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
[asterisk-users] Polycom Spectralink 8002 WiFi Phones
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:2...@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:2...@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 ___ -- 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
Re: [asterisk-users] Polycom Spectralink 8002 WiFi Phones
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:2...@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:2...@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 ___ -- 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
