Since I have a partial answer for completeness: When I moved (duplicated) the PROXYn_ADDRESS definition to the user specific config it then registers with the proxy. It seems like a defect to me.
I'd still like to see how people have set up different users/lines using LINEn and LINEn_AUTH. I'm struggling a bit with the relationship between the user registrations in the phone admin and the lines/users in the sip_xxxx.cfg and sip.conf. 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...@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
