thr3ads.net - asterisk users - [Asterisk-Users] Problems in autnenticating with SER / PortaSIP [Nov 2004]

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Roberto Piola
2004-Nov-11 05:45 UTC
[Asterisk-Users] Problems in autnenticating with SER / PortaSIP

We have a problem in authenticating with a SIP server running PortaSIP.

first, my exten.conf says:

exten => _396262X.,1,Dial(SIP/${EXTEN:2}@to-uni)
exten => _39064040.,1,Dial(SIP/${EXTEN:2}@to-uni)

and sip.conf:

register=390645416983:XXXXXX@sip.uni.it/390645416983

[to-uni]
type=peer
secret=XXXXXX ; i tried also using md5secret=.... instead of secret=... but
it's the same
username=390645416983
fromuser=390645416983
host=sip.uni.it
nat=yes


our asterisk pbx correctly registers on sip.uni.it (it is displayed as
"registered" in sip show registry, and if I issue a sip debug I see
the
answer to the registration, correctly reporting the name of the remote
server and our balance:

SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.18.104.75:5060;branch=z9hG4bK1bd57ff9
From: <sip:390645416983@sip.uni.it>;tag=as4fb9a73e
To:
<sip:390645416983@sip.uni.it>;tag=a4a48d8b20978897d8e0f5c399e6cc29.fbc4
Call-ID: 7acae8557efdc1c01d7fed7e70ddbad5@10.196.1.18
CSeq: 103 REGISTER
PortaBilling: available-funds:5.00 currency:EUR
Contact: <sip:390645416983@217.18.104.75>;q=0.00;expires=115
Server: Sip EXpress router (0.8.14 (i386/freebsd))
Content-Length: 0

The problem is when I try to call a number on the othere side
(390640409999): the call is correctly routed, the remote server asks us for
the proper credentials, and it seems to me that asterisk answers their
challenge:

Authorization: Digest username="390645416983",
realm="sip.uni.it",
algorithm=MD5, uri="sip:0640409999@sip.uni.it",
nonce="419358e858969bef4a5c77326f2b205b97c672bf",
response="188824ee848f9ed095990999fb2e3893", opaque=""

but for some reason it seems that the remote server does not like the
answer. the helpdesk of uni.it says that this is an old bug of asterisk
(actually, the account works with an X-Lite softphone ).

I'm using CVS-v1-0-11/08/04-10:57:05. I hoped that the latest version
corrected this problem as well, but it appears that it is not the case

I enclose the sip debug trace of the call


    -- Executing Dial("IAX2/rpiola@rpiola/3",
"SIP/0640409999@touni") in new
stack
We're at 217.18.104.75 port 10880
Answering with preferred capability 0x4(ULAW)
Answering with non-codec capability 0x1(G723)
12 headers, 10 lines
Reliably Transmitting:
INVITE sip:0640409999@sip.uni.it SIP/2.0
Via: SIP/2.0/UDP 217.18.104.75:5060;branch=z9hG4bK2f983a20;rport
From: "Roberto Piola"
<sip:390645416983@sip.uni.it>;tag=as2fb0ecc6
To: <sip:0640409999@sip.uni.it>
Contact: <sip:390645416983@217.18.104.75>
Call-ID: 2aefee4e736a4aea18f8499e3ce6d47c@sip.uni.it
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Thu, 11 Nov 2004 12:14:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 218

v=0
o=root 12361 12361 IN IP4 217.18.104.75
s=session
c=IN IP4 217.18.104.75
t=0 0
m=audio 10880 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
 (NAT) to 217.72.100.4:5060
    -- Called 0640409999@touni
janis*CLI>

Sip read:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 217.18.104.75:5060;branch=z9hG4bK2f983a20;rport=5060
From: "Roberto Piola"
<sip:390645416983@sip.uni.it>;tag=as2fb0ecc6
To: <sip:0640409999@sip.uni.it>;tag=a4a48d8b20978897d8e0f5c399e6cc29.b98a
Call-ID: 2aefee4e736a4aea18f8499e3ce6d47c@sip.uni.it
CSeq: 102 INVITE
WWW-Authenticate: Digest realm="sip.uni.it",
nonce="419358e858969bef4a5c77326f2b205b97c672bf"
Server: Sip EXpress router (0.8.14 (i386/freebsd))
Content-Length: 0


9 headers, 0 lines
Transmitting:
ACK sip:0640409999@sip.uni.it SIP/2.0
Via: SIP/2.0/UDP 217.18.104.75:5060;branch=z9hG4bK2f983a20;rport
From: "Roberto Piola"
<sip:390645416983@sip.uni.it>;tag=as2fb0ecc6
To: <sip:0640409999@sip.uni.it>;tag=a4a48d8b20978897d8e0f5c399e6cc29.b98a
Contact: <sip:390645416983@217.18.104.75>
Call-ID: 2aefee4e736a4aea18f8499e3ce6d47c@sip.uni.it
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

 (NAT) to 217.72.100.4:5060
We're at 217.18.104.75 port 10880
Answering with preferred capability 0x4(ULAW)
Answering with non-codec capability 0x1(G723)
Reliably Transmitting:
INVITE sip:0640409999@sip.uni.it SIP/2.0
Via: SIP/2.0/UDP 217.18.104.75:5060;branch=z9hG4bK588b0624;rport
From: "Roberto Piola"
<sip:390645416983@sip.uni.it>;tag=as2fb0ecc6
To: <sip:0640409999@sip.uni.it>
Contact: <sip:390645416983@217.18.104.75>
Call-ID: 2aefee4e736a4aea18f8499e3ce6d47c@sip.uni.it
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Authorization: Digest username="390645416983",
realm="sip.uni.it",
algorithm=MD5, uri="sip:0640409999@sip.uni.it",
nonce="419358e858969bef4a5c77326f2b205b97c672bf",
response="188824ee848f9ed095990999fb2e3893", opaque=""
Date: Thu, 11 Nov 2004 12:14:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 218

v=0
o=root 12361 12362 IN IP4 217.18.104.75
s=session
c=IN IP4 217.18.104.75
t=0 0
m=audio 10880 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
 (NAT) to 217.72.100.4:5060
janis*CLI>

Sip read:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 217.18.104.75:5060;branch=z9hG4bK588b0624;rport=5060
From: "Roberto Piola"
<sip:390645416983@sip.uni.it>;tag=as2fb0ecc6
To: <sip:0640409999@sip.uni.it>;tag=a4a48d8b20978897d8e0f5c399e6cc29.5919
Call-ID: 2aefee4e736a4aea18f8499e3ce6d47c@sip.uni.it
CSeq: 103 INVITE
WWW-Authenticate: Digest realm="sip.uni.it",
nonce="419358e858969bef4a5c77326f2b205b97c672bf"
Server: Sip EXpress router (0.8.14 (i386/freebsd))
Content-Length: 0


9 headers, 0 lines
Transmitting:
ACK sip:0640409999@sip.uni.it SIP/2.0
Via: SIP/2.0/UDP 217.18.104.75:5060;branch=z9hG4bK588b0624;rport
From: "Roberto Piola"
<sip:390645416983@sip.uni.it>;tag=as2fb0ecc6
To: <sip:0640409999@sip.uni.it>;tag=a4a48d8b20978897d8e0f5c399e6cc29.5919
Contact: <sip:390645416983@217.18.104.75>
Call-ID: 2aefee4e736a4aea18f8499e3ce6d47c@sip.uni.it
CSeq: 103 ACK
User-Agent: Asterisk PBX
Content-Length: 0

 (NAT) to 217.72.100.4:5060
We're at 217.18.104.75 port 10880
Answering with preferred capability 0x4(ULAW)
Answering with non-codec capability 0x1(G723)
Reliably Transmitting:
INVITE sip:0640409999@sip.uni.it SIP/2.0
Via: SIP/2.0/UDP 217.18.104.75:5060;branch=z9hG4bK2dae1f5b;rport
From: "Roberto Piola"
<sip:390645416983@sip.uni.it>;tag=as2fb0ecc6
To: <sip:0640409999@sip.uni.it>
Contact: <sip:390645416983@217.18.104.75>
Call-ID: 2aefee4e736a4aea18f8499e3ce6d47c@sip.uni.it
CSeq: 104 INVITE
User-Agent: Asterisk PBX
Authorization: Digest username="390645416983",
realm="sip.uni.it",
algorithm=MD5, uri="sip:0640409999@sip.uni.it",
nonce="419358e858969bef4a5c77326f2b205b97c672bf",
response="188824ee848f9ed095990999fb2e3893", opaque=""
Date: Thu, 11 Nov 2004 12:14:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 218

v=0
o=root 12361 12363 IN IP4 217.18.104.75
s=session
c=IN IP4 217.18.104.75
t=0 0
m=audio 10880 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
 (NAT) to 217.72.100.4:5060
janis*CLI>

Sip read:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 217.18.104.75:5060;branch=z9hG4bK2dae1f5b;rport=5060
From: "Roberto Piola"
<sip:390645416983@sip.uni.it>;tag=as2fb0ecc6
To: <sip:0640409999@sip.uni.it>;tag=a4a48d8b20978897d8e0f5c399e6cc29.3ee7
Call-ID: 2aefee4e736a4aea18f8499e3ce6d47c@sip.uni.it
CSeq: 104 INVITE
WWW-Authenticate: Digest realm="sip.uni.it",
nonce="419358e858969bef4a5c77326f2b205b97c672bf"
Server: Sip EXpress router (0.8.14 (i386/freebsd))
Content-Length: 0


9 headers, 0 lines
Transmitting:
ACK sip:0640409999@sip.uni.it SIP/2.0
Via: SIP/2.0/UDP 217.18.104.75:5060;branch=z9hG4bK2dae1f5b;rport
From: "Roberto Piola"
<sip:390645416983@sip.uni.it>;tag=as2fb0ecc6
To: <sip:0640409999@sip.uni.it>;tag=a4a48d8b20978897d8e0f5c399e6cc29.3ee7
Contact: <sip:390645416983@217.18.104.75>
Call-ID: 2aefee4e736a4aea18f8499e3ce6d47c@sip.uni.it
CSeq: 104 ACK
User-Agent: Asterisk PBX
Content-Length: 0

 (NAT) to 217.72.100.4:5060
Nov 11 13:14:52 NOTICE[12366]: chan_sip.c:6774 handle_response: Failed to
authenticate on INVITE to '"Roberto Piola"
<sip:390645416983@sip.uni.it>;tag=as2fb0ecc6'
asterisk users - Nov 2004 - Problems in autnenticating with SER / PortaSIP

[Asterisk-Users] Problems in autnenticating with SER / PortaSIP
