thufir
June 29, 2016, 1:25pm
1
Why does sipsak fail with:
error: didn't received '200 OK' on register
?
I have thufir
registered on a Cisco IP phone. From the same computer where Asterisk is running:
thufir@mordor:~$
thufir@mordor:~$ sudo sipsak -vvv --usrloc-mode --from sip:demo_alice@localhost --password 123 --sip-uri sip:6003@localhost
No SRV record: _sip._tcp.localhost
No SRV record: _sip._udp.localhost
using A record: localhost
warning: ignoring -i option when in usrloc mode
fqdnhostname: 127.0.1.1
our Via-Line: Via: SIP/2.0/UDP 127.0.1.1:57946;branch=z9hG4bK.0238e203;rport;alias
New message with Via-Line:
REGISTER sip:localhost SIP/2.0
Via: SIP/2.0/UDP 127.0.1.1:57946;branch=z9hG4bK.0238e203;rport;alias
From: sip:6003@localhost;tag=1c556565
To: sip:6003@localhost
Call-ID: 475358565@127.0.1.1
CSeq: 1 REGISTER
Content-Length: 0
Max-Forwards: 70
User-Agent: sipsak 0.9.6
Expires: 15
Contact: sip:6003@127.0.1.1:57946
registering user 6003...
request:
REGISTER sip:localhost SIP/2.0
Via: SIP/2.0/UDP 127.0.1.1:57946;branch=z9hG4bK.0238e203;rport;alias
From: sip:6003@localhost;tag=1c556565
To: sip:6003@localhost
Call-ID: 475358565@127.0.1.1
CSeq: 1 REGISTER
Content-Length: 0
Max-Forwards: 70
User-Agent: sipsak 0.9.6
Expires: 15
Contact: sip:6003@127.0.1.1:57946
send to: UDP:127.0.0.1:5060
authorizing
received:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 127.0.1.1:57946;branch=z9hG4bK.0238e203;alias;received=127.0.0.1;rport=57946
From: sip:6003@localhost;tag=1c556565
To: sip:6003@localhost;tag=as390fd2fe
Call-ID: 475358565@127.0.1.1
CSeq: 1 REGISTER
Server: Asterisk PBX 13.1.0~dfsg-1.1ubuntu3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="38ab0be7"
Content-Length: 0
registering user 6003...
request:
REGISTER sip:localhost SIP/2.0
Authorization: Digest username="6003", uri="sip:localhost", algorithm=MD5, realm="asterisk", nonce="38ab0be7", response="fa5cac509e948b3d2965586917e2a5fd"
Via: SIP/2.0/UDP 127.0.1.1:57946;branch=z9hG4bK.4ad883b2;rport;alias
From: sip:6003@localhost;tag=1c556565
To: sip:6003@localhost
Call-ID: 475358565@127.0.1.1
CSeq: 2 REGISTER
Content-Length: 0
Max-Forwards: 70
User-Agent: sipsak 0.9.6
Expires: 15
Contact: sip:6003@127.0.1.1:57946
send to: UDP:127.0.0.1:5060
received:
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 127.0.1.1:57946;branch=z9hG4bK.4ad883b2;alias;received=127.0.0.1;rport=57946
From: sip:6003@localhost;tag=1c556565
To: sip:6003@localhost;tag=as390fd2fe
Call-ID: 475358565@127.0.1.1
CSeq: 2 REGISTER
Server: Asterisk PBX 13.1.0~dfsg-1.1ubuntu3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Length: 0
error: didn't received '200 OK' on register (see above). aborting
thufir@mordor:~$
the 200 OK is return after The SIP server validates the user’scredentials. It registers the user in its contact database and returns a response (200 OK)
thufir:
SIP/2.0 403 Forbidden
server is replyinn back with a 403 Forbidden error
thufir
June 30, 2016, 7:05am
3
what is being allowed here? The user credentials are accepted with the 200 OK?
However, the REGISTER for 6003 fails? there’s no 6003 per se, it’s thufir
at extension 6003:
mordor*CLI>
mordor*CLI> dialplan show internal
[ Context 'internal' created by 'pbx_config' ]
'6001' => 1. Dial(SIP/demo_alice,20) [pbx_config]
'6002' => 1. Dial(SIP/demo_bob,20) [pbx_config]
'6003' => 1. Dial(SIP/thufir,20) [pbx_config]
-= 3 extensions (3 priorities) in 1 context. =-
mordor*CLI>
mordor*CLI> sip show peers
Name/username Host Dyn Forcerport Comedia ACL Port Status Description
demo_alice/demo_alice 192.168.1.6 D Yes Yes 5060 Unmonitored
demo_bob/demo_bob (Unspecified) D Yes Yes 0 Unmonitored
thufir/thufir 192.168.1.5 D Yes Yes 5062 Unmonitored
3 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 1 offline]
mordor*CLI>
jcolp
June 30, 2016, 10:03am
4
I have not used sipsak so I can’t speak on the proper configuration to pass to it to make it work, but you are indeed registering as 6003 instead of thufir so the problem is there, not in Asterisk.
thufir
June 30, 2016, 10:10am
5
Thanks. I thought that I’d tried with thufir
but apparently not. Limited success, I think:
thufir @mordor :~$
thufir@mordor:~$ sudo sipsak -vvv --usrloc-mode --from sip:demo_alice@localhost --password 123 --sip-uri sip:thufir@localhost
No SRV record: _sip._tcp.localhost
No SRV record: _sip._udp.localhost
using A record: localhost
warning: ignoring -i option when in usrloc mode
fqdnhostname: 127.0.1.1
our Via-Line: Via: SIP/2.0/UDP 127.0.1.1:53492;branch=z9hG4bK.1c1bca40;rport;alias
New message with Via-Line:
REGISTER sip:localhost SIP/2.0
Via: SIP/2.0/UDP 127.0.1.1:53492;branch=z9hG4bK.1c1bca40;rport;alias
From: sip:thufir@localhost;tag=13534fc0
To: sip:thufir@localhost
Call-ID: 324227008@127.0.1.1
CSeq: 1 REGISTER
Content-Length: 0
Max-Forwards: 70
User-Agent: sipsak 0.9.6
Expires: 15
Contact: sip:thufir@127.0.1.1:53492
registering user thufir...
request:
REGISTER sip:localhost SIP/2.0
Via: SIP/2.0/UDP 127.0.1.1:53492;branch=z9hG4bK.1c1bca40;rport;alias
From: sip:thufir@localhost;tag=13534fc0
To: sip:thufir@localhost
Call-ID: 324227008@127.0.1.1
CSeq: 1 REGISTER
Content-Length: 0
Max-Forwards: 70
User-Agent: sipsak 0.9.6
Expires: 15
Contact: sip:thufir@127.0.1.1:53492
send to: UDP:127.0.0.1:5060
authorizing
received:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 127.0.1.1:53492;branch=z9hG4bK.1c1bca40;alias;received=127.0.0.1;rport=53492
From: sip:thufir@localhost;tag=13534fc0
To: sip:thufir@localhost;tag=as11a64156
Call-ID: 324227008@127.0.1.1
CSeq: 1 REGISTER
Server: Asterisk PBX 13.1.0~dfsg-1.1ubuntu3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="39db0bfd"
Content-Length: 0
registering user thufir...
request:
REGISTER sip:localhost SIP/2.0
Authorization: Digest username="thufir", uri="sip:localhost", algorithm=MD5, realm="asterisk", nonce="39db0bfd", response="e54fe210187a1255f5af4fd0a1a361c0"
Via: SIP/2.0/UDP 127.0.1.1:53492;branch=z9hG4bK.0b9d9eb6;rport;alias
From: sip:thufir@localhost;tag=13534fc0
To: sip:thufir@localhost
Call-ID: 324227008@127.0.1.1
CSeq: 2 REGISTER
Content-Length: 0
Max-Forwards: 70
User-Agent: sipsak 0.9.6
Expires: 15
Contact: sip:thufir@127.0.1.1:53492
send to: UDP:127.0.0.1:5060
OK
SIP/2.0 200 OK
Via: SIP/2.0/UDP 127.0.1.1:53492;branch=z9hG4bK.0b9d9eb6;alias;received=127.0.0.1;rport=53492
From: sip:thufir@localhost;tag=13534fc0
To: sip:thufir@localhost;tag=as11a64156
Call-ID: 324227008@127.0.1.1
CSeq: 2 REGISTER
Server: Asterisk PBX 13.1.0~dfsg-1.1ubuntu3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Expires: 60
Contact: <sip:thufir@127.0.1.1:53492>;expires=60
Date: Thu, 30 Jun 2016 10:08:40 GMT
Content-Length: 0
All usrloc tests completed successful.
received last message 11.159 ms after first request (test duration).
thufir@mordor:~$
Surely it’s possible to at least get the phone to ring if not play a default sound, a sort of “hello world”?
jcolp
June 30, 2016, 10:15am
6
It is, many people do it using a softphone. Like I said I can’t speak for sipsak.