Cant register sip phone - wan - wrong password

Asterisk Asterisk Support
1

hello everyone

i am desperate for help on a problem ive had for months - but still havent found the answer.

i use plain asterisk - its a recent installtion running on ubuntu

i have created an extension called “intphone”

when i enter the details in my nokia or softfone locally - adding intphone@192.168.1.2 with the password and all - everything works fine .

when i try entering intphone@MYSTATICpublicIPADRESS , with same password
i always get “WRONG PASSWORD” in debug…
the phone - be it nokia/ or ekigo will not register - its nothing to do with wrong password in phone- alll ive done is entered the public user address of my asterisk box- instead of local.

5060-5082
10000-20000 ports are all forwarded

the particular extension , has nat=yes in sip.conf

localnet/externip is set correctly in sip.conf ( i think) localnet= my ubuntos static address/ and i think 255.255.255.0 is right

ive tried stun on the phones
and ive correctly entered realm in phones…

bottom line ,
since all these parameters work locally (intphone@192.168.1.2) - i must have entered them right i guess.

if i delete the secret in sip.conf for intphone , then i can register externally, but obviously this is unsatisfactory…

what do you guys think im doing wrong ?

can someone work on this for some paypal magic ?

id really like to get to the bottom of this …

one final thing - im using an ACER REVO 3600 , which uses a smart/ arm cpu ? somewhere ive heard this chipset is known to give just such problems and has been noted by digium ,

any thoughts anyone ???

2

Can you show to us the CLI output with verbose 4 when you try to register with your local ip and then with your external ip?

Then the debug log, and the sip.conf file.

3

hi navismo
the below is with the phone lan side.

bottom is wan side

all ive done is deleted the 192.168.1.2 in the nokia and substituted for my home ip
leaving passswords etc

Transmitting (NAT) to 192.168.1.64:5060 —>
SIP/2.0 401 Unauthorized

Via: SIP/2.0/UDP 192.168.1.64:5060;branch=z9hG4bK58a7p4kpflhc6fmv0mhjpc9;received=192.168.1.64;rport=5060

From: sip:intphone@192.168.1.2;tag=5fonp4hd1dhc7iv50mhj

To: sip:intphone@192.168.1.2;tag=as4b22c203

Call-ID: QytYAHgIoIfQhMmzf7KEAg-ly7VXpX

CSeq: 2079 REGISTER

Server: Asterisk PBX 1.6.2.5-0ubuntu1.3

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO

Supported: replaces, timer

WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“615ae1e1”

Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘QytYAHgIoIfQhMmzf7KEAg-ly7VXpX’ in 32000 ms (Method: REGISTER)

e[Kripon-desktop*CLI>

<— SIP read from UDP:192.168.1.64:5060 —>
REGISTER sip:192.168.1.2;transport=UDP SIP/2.0

Route: sip:192.168.1.2;lr;transport=UDP

Via: SIP/2.0/UDP 192.168.1.64:5060;branch=z9hG4bKerga52d2h7k6dqn5fi9f343;rport

From: sip:intphone@192.168.1.2;tag=5fonp4hd1dhc7iv50mhj

To: sip:intphone@192.168.1.2

Contact: sip:intphone@192.168.1.64;transport=UDP;expires=3600

CSeq: 2080 REGISTER

Call-ID: QytYAHgIoIfQhMmzf7KEAg-ly7VXpX

Supported: sec-agree

User-Agent: E71-1 RM-346 400.21.013

Max-Forwards: 70

Authorization: Digest realm=“asterisk”,nonce=“615ae1e1”,algorithm=MD5,username=“intphone”,uri=“sip:192.168.1.2;transport=UDP”,response=“6f1301a558f43ce850463c6624a6a582”

Content-Length: 0

<------------->
— (13 headers 0 lines) —
Sending to 192.168.1.64 : 5060 (NAT)

e[Kripon-desktop*CLI>
– Registered SIP ‘intphone’ at 192.168.1.64 port 5060
> Saved useragent “E71-1 RM-346 400.21.013” for peer intphone

<— Transmitting (NAT) to 192.168.1.64:5060 —>
SIP/2.0 200 OK

Via: SIP/2.0/UDP 192.168.1.64:5060;branch=z9hG4bKerga52d2h7k6dqn5fi9f343;received=192.168.1.64;rport=5060

From: sip:intphone@192.168.1.2;tag=5fonp4hd1dhc7iv50mhj

To: sip:intphone@192.168.1.2;tag=as4b22c203

Call-ID: QytYAHgIoIfQhMmzf7KEAg-ly7VXpX

CSeq: 2080 REGISTER

Server: Asterisk PBX 1.6.2.5-0ubuntu1.3

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO

Supported: replaces, timer

Expires: 3600

Contact: sip:intphone@192.168.1.64;transport=UDP;expires=3600

Date: Sat, 19 Feb 2011 18:28:46 GMT

Content-Length: 0

wan side…

om UDP:82.132.242.57:17571 —>
REGISTER sip:192.168.1.2;transport=UDP SIP/2.0

Route: sip:192.168.1.2;lr;transport=UDP

Via: SIP/2.0/UDP 82.132.139.252:31657;branch=z9hG4bKejtnp4haudhc6el20q1f9ju;rport

From: sip:intphone@192.168.1.2;tag=74nnp4kuglhc78lr0q1f

To: sip:intphone@192.168.1.2

Contact: sip:intphone@82.132.242.57:31657;transport=UDP;expires=3600

CSeq: 2081 REGISTER

Call-ID: fsJYALpEoIdjrwq2Z86BZAbNrpkHLE

Supported: sec-agree

User-Agent: E71-1 RM-346 400.21.013

Max-Forwards: 70

Content-Length: 0

<------------->
— (12 headers 0 lines) —
[Feb 19 18:31:13] DEBUG[1299]: chan_sip.c:7289 sip_alloc: Allocating new SIP dialog for fsJYALpEoIdjrwq2Z86BZAbNrpkHLE - REGISTER (No RTP)
Sending to 82.132.139.252 : 31657 (no NAT)

<— Transmitting (NAT) to 82.132.242.57:17571 —>
SIP/2.0 401 Unauthorized

Via: SIP/2.0/UDP 82.132.139.252:31657;branch=z9hG4bKejtnp4haudhc6el20q1f9ju;received=82.132.242.57;rport=17571

From: sip:intphone@192.168.1.2;tag=74nnp4kuglhc78lr0q1f

To: sip:intphone@192.168.1.2;tag=as6dbf5af2

Call-ID: fsJYALpEoIdjrwq2Z86BZAbNrpkHLE

CSeq: 2081 REGISTER

Server: Asterisk PBX 1.6.2.5-0ubuntu1.3

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO

Supported: replaces, timer

WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“7777f0e3”

Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘fsJYALpEoIdjrwq2Z86BZAbNrpkHLE’ in 32000 ms (Method: REGISTER)

e[Kripon-desktop*CLI>

<— SIP read from UDP:82.132.242.57:17571 —>
REGISTER sip:192.168.1.2;transport=UDP SIP/2.0

Route: sip:192.168.1.2;lr;transport=UDP

Via: SIP/2.0/UDP 82.132.139.252:31657;branch=z9hG4bKb5cirk1dq1duhs82fi9b7l3;rport

From: sip:intphone@192.168.1.2;tag=74nnp4kuglhc78lr0q1f

To: sip:intphone@192.168.1.2

Contact: sip:intphone@82.132.242.57:31657;transport=UDP;expires=3600

CSeq: 2082 REGISTER

Call-ID: fsJYALpEoIdjrwq2Z86BZAbNrpkHLE

Supported: sec-agree

User-Agent: E71-1 RM-346 400.21.013

Max-Forwards: 70

Authorization: Digest realm=“asterisk”,nonce=“7777f0e3”,algorithm=MD5,username=“intphone”,uri=“sip:192.168.1.2;transport=UDP”,response=“c68956acf5c7900ac55a16a8548b4249”

Content-Length: 0

<------------->
— (13 headers 0 lines) —
Sending to 82.132.242.57 : 17571 (NAT)

<— Transmitting (NAT) to 82.132.242.57:17571 —>
SIP/2.0 403 Forbidden (Bad auth)

Via: SIP/2.0/UDP 82.132.139.252:31657;branch=z9hG4bKb5cirk1dq1duhs82fi9b7l3;received=82.132.242.57;rport=17571

From: sip:intphone@192.168.1.2;tag=74nnp4kuglhc78lr0q1f

To: sip:intphone@192.168.1.2;tag=as6dbf5af2

Call-ID: fsJYALpEoIdjrwq2Z86BZAbNrpkHLE

CSeq: 2082 REGISTER

Server: Asterisk PBX 1.6.2.5-0ubuntu1.3

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO

Supported: replaces, timer

Content-Length: 0

<------------>
[Feb 19 18:31:14] NOTICE[1299]: chan_sip.c:21500 handle_request_register: Registration from ‘sip:intphone@192.168.1.2’ failed for ‘82.132.242.57’ - Wrong password
Scheduling destruction of SIP dialog ‘fsJYALpEoIdjrwq2Z86BZAbNrpkHLE’ in 32000 ms (Method: REGISTER)
[Feb 19 18:31:14] DEBUG[1299]: chan_sip.c:21978 handle_request_do: SIP message could not be handled, bad request: fsJYALpEoIdjrwq2Z86BZAbNrpkHLE.

sip.conf

[general]
bindaddr=0.0.0.0
bindport=5060
localnet=192.168.1.2/255.255.255.0
externip= MY STATIC IP ADDRESS
realm=asterisk
nat=yes

[intphone]
context=default
type=friend
host=dynamic
nat=yes
canreinvite=no
secret=lololo
qualify=yes

apart from the above…
all defaults in sip.conf, are left as defaults
no stun setup for the server, just nat=yes

what do you think ?