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:6193644850 at 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:6193644850 at jtsd05>;tag=A1BB38FF-7161AAEA To: <sip:6193644850 at jtsd05>;tag=as3d68239c Call-ID: 20f907fe-db323389-f4569534 at 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
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:6193644850 at 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:6193644850 at jtsd05>;tag=A1BB38FF-7161AAEA > To: <sip:6193644850 at jtsd05>;tag=as3d68239c > Call-ID: 20f907fe-db323389-f4569534 at 192.168.200.99 > CSeq: 1 REGISTER > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Content-Length: 0This 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=192.168.200.99Change 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
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:6193644850 at 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:6193644850 at jtsd05>;tag=A1BB38FF-7161AAEA > To: <sip:6193644850 at jtsd05>;tag=as3d68239c > Call-ID: 20f907fe-db323389-f4569534 at 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="" >Try changing reg.1.address to "hft0". My hunch is asterisk is looking at the from of "6193644850 at jtsd05" and going "huh? I don't know a 6193644850 at jtsd05". -Dave