I have a cisco 7960 phone. Worked fine in the office. I took it home. At home I have a linksys router that the phone is plugged into. The linksys router has DHCP enabled. I am getting the following error on the console from the 7960. I have tried it with nat=yes and nat=no in the sip.conf file. ----------------------- Transmitting (NAT) to 192.168.1.69:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.69:5060;branch=z9hG4bK1415360297;received=192.168.1.69;rport=5060 From: "Display Name" <sip:570 at xx>;tag=1683635072 To: "Display Name" <sip:570 at xx>;tag=as4c59a734 Call-ID: 1929465491 at 192.168.1.69 CSeq: 3091 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4a1c350c" Content-Length: 0 -------------------------- The username and secret are the same as they were in the office when it worked. I figure it has to be something easy but I have not found it yet. the sip.conf entry for this phone is: [570] type=friend dtmfmode=rfc2833 username=570 secret=XXXXXXXXXXX disallow=all allow=ulaw allow=alaw host=dynamic context=local-sip callerid="Home 570" <570> nat=no What might I try to get the phone working from home? Thanks, Jerry
Enable NAT on the phone itself and leave it enabled in *. Jerry Geis wrote:> I have a cisco 7960 phone. Worked fine in the office. > I took it home. At home I have a linksys router that the phone is > plugged into. > The linksys router has DHCP enabled. I am getting the following error on > the console from the 7960. > I have tried it with nat=yes and nat=no in the sip.conf file. > ----------------------- > > Transmitting (NAT) to 192.168.1.69:5060: > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP > 192.168.1.69:5060;branch=z9hG4bK1415360297;received=192.168.1.69;rport=5060 > From: "Display Name" <sip:570 at xx>;tag=1683635072 > To: "Display Name" <sip:570 at xx>;tag=as4c59a734 > Call-ID: 1929465491 at 192.168.1.69 > CSeq: 3091 REGISTER > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4a1c350c" > Content-Length: 0 > > -------------------------- > The username and secret are the same as they were in the office when it > worked. > > I figure it has to be something easy but I have not found it yet. the > sip.conf entry for this phone is: > [570] > type=friend > dtmfmode=rfc2833 > username=570 > secret=XXXXXXXXXXX > disallow=all > allow=ulaw > allow=alaw > host=dynamic > context=local-sip > callerid="Home 570" <570> > nat=no > > What might I try to get the phone working from home? > > Thanks, > > Jerry > > _______________________________________________ > -- 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 >