On Sat, 13 Jun 2009, Jim Gottlieb wrote: > I'm evaluating using Polycom phones for our call center and I've set > up my first phone (a SoundPoint 560) to give it a try. > > The phone is working and can successfully place and receive calls. > But every minute, there's an error in the log file: > > chan_sip.c: Registration from '<sip:6193644...@jtsd05>' failed for > '192.168.200.99' - Username/auth name mismatch > > Turning on SIP debug, it appears it's asterisk trying to register with > the phone: > > Using latest REGISTER request as basis request > Sending to 192.168.200.99 : 5060 (non-NAT) > Transmitting (no NAT) to 192.168.200.99:5060: > SIP/2.0 404 Not found > Via: SIP/2.0/UDP > 192.168.200.99;branch=z9hG4bKd732a3486F528693;received=192.168.200.99 > From: "6193644850" <sip:6193644...@jtsd05>;tag=A1BB38FF-7161AAEA > To: <sip:6193644...@jtsd05>;tag=as3d68239c > Call-ID: 20f907fe-db323389-f4569...@192.168.200.99 > CSeq: 1 REGISTER > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Content-Length: 0
This isn't asterisk registering with the phone - it is asterisk sending back the "404 Not Found". > > But then, the From: and To: lines seem to both show it from hostname > jtsd05, though there's also the line saying it's going to > 192.168.200.99 (the phone). > > I've played with all sorts of settings in sip.conf, but the messages > persist. Here's what I've got: > > [hft0] > type=friend > username=hft0 > secret=mysecret > context=outtrunk-office > host=192.168.200.99 Change the above to host=dynamic > disallow=all > allow=ulaw > dtmfmode=rfc2833 > progressinband=no ;Polycom phones have trouble with the > progressinband=never > callerid="HFT Booth 0 <(619) 364-4850>" > allowsubscribe=yes > > And some of the Polycom phone config: > reg reg.1.displayName="HFT0" > reg.1.address="6193644850" > reg.1.label="4850" > reg.1.type="private" > reg.1.lcs="" > reg.1.csta="" > reg.1.thirdPartyName="" > reg.1.auth.userId="hft0" > reg.1.auth.password="mysecret" > reg.1.auth.optimizedInFailover="" > reg.1.musicOnHold.uri="" > reg.1.server.1.address="jtsd05" Can the phone resolve this unqualified name? > reg.1.server.1.port="" > reg.1.server.1.transport="DNSnaptr" > reg.1.server.2.transport="DNSnaptr" > reg.1.server.1.expires="" > reg.1.server.1.expires.overlap="" > reg.1.server.1.register="" > reg.1.server.1.retryTimeOut="" > reg.1.server.1.retryMaxCount="" > reg.1.server.1.expires.lineSeize="" > reg.1.server.1.lcs="" > reg.1.outboundProxy.address="" > > Any ideas would be welcomed. Thanks... > > ...Jim Gottlieb, San Diego, California > I think host=dynamic will fix you up. j _______________________________________________ -- 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