[Asterisk-Users] Problem with SIP authentication

Tue Oct 14 21:52:35 MST 2003

Yes Sean 
It looks fine with 200 Responses, but actually it doesnt ring while dialing from other phone. Might this be the problem of SIP UA? if i try X-Lite that continously goes in loop to register(One bad thing with X-Lite is that it uses transparent proxy's IP as its own., so never gets reply from asterisk, going in loop of register process)

Secondly i tried to live with that "Acquired" message, and using extension.conf, then the dialing UA gets message "Established" and Terminates media channel in 2,3 seconds. While there is no actual activity on dialed UA 
strange..:)

JF

"Sean P. Robertson" <spr at netxusa.com> wrote:
It looks like you are registering fine.  If you dial 12321 from another phone, does it not ring?

This is the transaction as I see it in the log that you attached:

Phone: REGISTER
Asterisk: Proxy Authentication Required (Send me your credentials)
Phone: REGISTER with CREDENTIALS
Asterisk: 200 OK (You are now registered)
Asterisk: NOTIFY (You have 0/0 messages in your voicemail.)
Phone: 200 OK (Thanks for letting me know)

Sean

_______________________________________________

Sean Robertson

NETXUSA
p. 800-289-6389
f.  864-233-4344                  "Ask me about Voice over IP."
http://www.netxusa.com/

----- Original Message ----- 
From: John Foster 
To: asterisk-users at lists.digium.com 
Sent: Tuesday, October 14, 2003 12:49 AM
Subject: [Asterisk-Users] Problem with SIP authentication

Hi List,

After going through mailing list and manual of asterisk, I still could not properly get authenticated with my SIP UA to asterisk. i m using a username for UA "12321" and following are SIP.conf file user params

[12321]
type=friend
username=12321
host=dynamic
secret=ccarta
context=default
mailbox=1234,2345               ; Mailbox for message waiting indicator

[77777]
type=friend
username=77777
host=dynamic
secret=atracc
context=default
mailbox=1234,2345

m trying with SIPPS UA, that gets status "Acquired", not "Registered", Can anyone give any idea about it? I tried same with X-Lite, didnt work.
Sip debug messages are pasted below.

Best Regards,
JF

Sip read:
REGISTER sip:192.168.100.71 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.66:5062;branch=z9hG4bKnp992800961-40de3e5f192.168.100.66
From: <sip:12321 at 192.168.100.71>;tag=3b2cf0ba
To: <sip:12321 at 192.168.100.71>
Call-ID: 990209125-415b5d61 at 990209122-415b5d5e
Contact: ccarta <sip:12321 at 192.168.100.66:5062>;expires=600;q=0.500
Expires: 600
CSeq: 1 REGISTER
Content-Length: 0
User-Agent: Ahead SIPPS IP Phone Version 2.0.42.13

10 headers, 0 lines
Using latest request as basis request
Sending to 192.168.100.66 : 5062 (non-NAT)
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.100.66:5062;branch=z9hG4bKnp992800961-40de3e5f192.168.100.66
From: <sip:12321 at 192.168.100.71>;tag=3b2cf0ba
To: <sip:12321 at 192.168.100.71>;tag=as648287fa
Call-ID: 990209125-415b5d61 at 990209122-415b5d5e
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:12321 at 192.168.100.71>
Content-Length: 0

 to 192.168.100.66:5062
Transmitting (no NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.100.66:5062;branch=z9hG4bKnp992800961-40de3e5f192.168.100.66
From: <sip:12321 at 192.168.100.71>;tag=3b2cf0ba
To: <sip:12321 at 192.168.100.71>;tag=as648287fa
Call-ID: 990209125-415b5d61 at 990209122-415b5d5e
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:12321 at 192.168.100.71>
Proxy-Authenticate: Digest realm="asterisk", nonce="7c7fba4d"
Content-Length: 0

 to 192.168.100.66:5062
Sip read:
REGISTER sip:192.168.100.71 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.66:5062;branch=z9hG4bKnp992804895-411b471d192.168.100.66
From: <sip:12321 at 192.168.100.71>;tag=3b2d0018
To: <sip:12321 at 192.168.100.71>
Call-ID: 990209125-415b5d61 at 990209122-415b5d5e
Contact: ccarta <sip:12321 at 192.168.100.66:5062>;expires=600;q=0.500
Expires: 600
CSeq: 2 REGISTER
Content-Length: 0
Proxy-Authorization: Digest username="12321",realm="asterisk",uri="sip:192.168.100.66",nonce="7c7fba4d",nc="00000001",response="b8b1d7fc53eff354dfc31dfa3f800749"
User-Agent: Ahead SIPPS IP Phone Version 2.0.42.13

11 headers, 0 lines
Using latest request as basis request
Sending to 192.168.100.66 : 5062 (non-NAT)
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.100.66:5062;branch=z9hG4bKnp992804895-411b471d192.168.100.66
From: <sip:12321 at 192.168.100.71>;tag=3b2d0018
To: <sip:12321 at 192.168.100.71>;tag=as648287fa
Call-ID: 990209125-415b5d61 at 990209122-415b5d5e
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:12321 at 192.168.100.71>
Content-Length: 0

 to 192.168.100.66:5062
Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.66:5062;branch=z9hG4bKnp992804895-411b471d192.168.100.66
From: <sip:12321 at 192.168.100.71>;tag=3b2d0018
To: <sip:12321 at 192.168.100.71>;tag=as648287fa
Call-ID: 990209125-415b5d61 at 990209122-415b5d5e
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Expires: 600
Contact: <sip:12321 at 192.168.100.71>;expires=600
Date: Tue, 14 Oct 2003 13:46:14 GMT
Content-Length: 0

 to 192.168.100.66:5062
11 headers, 2 lines
Reliably Transmitting:
NOTIFY sip:12321 at 192.168.100.66:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.71:5060;branch=z9hG4bK3ecb7a3b
From: "asterisk" <sip:asterisk at 192.168.100.71>;tag=as3f6e8c0e
To: <sip:12321 at 192.168.100.66:5062>
Contact: <sip:asterisk at 192.168.100.71>
Call-ID: 074dfadf24b95dd75e17b56c67ffcaf1 at 192.168.100.71
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 36
Messages-Waiting: no
Voicemail: 0/0
 (no NAT) to 192.168.100.66:5062
Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.71;branch=z9hG4bK3ecb7a3b
From: "asterisk" <sip:asterisk at 192.168.100.71>;tag=as3f6e8c0e
To: <sip:12321 at 192.168.100.66:5062>;tag=3b302259
Call-ID: 074dfadf24b95dd75e17b56c67ffcaf1 at 192.168.100.71
CSeq: 102 NOTIFY
User-Agent: Ahead SIPPS IP Phone Version 2.0.42.13
Content-Length: 0

8 headers, 0 lines

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