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 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 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" 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 _______________________________________________ -- 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