[Asterisk-Dev] SIP 407 error
jerk face
jerkface2098 at yahoo.com
Thu Dec 4 11:57:26 MST 2003
I have been having a problem with calling my SIP
provider.
First of all, I’m running Asterisk
CVS-12/04/03-13:16:25, so I should have the recent SIP
updates.
Anyways, I can register with my provider, but I cannot
make an outgoing call.
I have included the output when I make a call below
(this stuff goes way beyond my qualifications as a
cafeteria worker)
- Starting simple switch on 'Zap/1-1'
-- Executing Dial("Zap/1-1",
"SIP/13239381067 at sipPROVIDER") in new stack
We're at MY.IP.ADD.RESS port 13182
Answering with preferred capability 8
Answering with preferred capability 4
Answering with non-codec capability 1
11 headers, 10 lines
Reliably Transmitting:
INVITE sip:13239381067 at SIP.PROXY.ADDRESS SIP/2.0
Via: SIP/2.0/UDP
MY.IP.ADD.RESS:5060;branch=z9hG4bK6aa623e5
From: "asterisk"
<sip:asterisk at MY.IP.ADD.RESS>;tag=as7108ed74
To: <sip:13239381067 at SIP.PROXY.ADDRESS>
Contact: <sip:asterisk at MY.IP.ADD.RESS>
Call-ID:
05b58f7e76eae6bb53a87dc142a1a7a0 at MY.IP.ADD.RESS
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 215
v=0
o=root 15930 15930 IN IP4 MY.IP.ADD.RESS
s=session
c=IN IP4 MY.IP.ADD.RESS
t=0 0
m=audio 13182 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
(no NAT) to 216.187.111.6:5060
-- Called 13239381067 at sipPROVIDER
Sip read:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
MY.IP.ADD.RESS:5060;branch=z9hG4bK6aa623e5
From: "asterisk"
<sip:asterisk at MY.IP.ADD.RESS>;tag=as7108ed74
To: <sip:13239381067 at SIP.PROXY.ADDRESS>
Call-Id:
05b58f7e76eae6bb53a87dc142a1a7a0 at MY.IP.ADD.RESS
Cseq: 102 INVITE
Date: Thu, 04 Dec 2003 19:30:35 GMT
Proxy-Authenticate: Digest realm="SIP.PROXY.ADDRESS",
nonce="308aefae000c2523", algorithm=MD5
Content-Length: 0
9 headers, 0 lines
Transmitting:
ACK sip:13239381067 at SIP.PROXY.ADDRESS SIP/2.0
Via: SIP/2.0/UDP
MY.IP.ADD.RESS:5060;branch=z9hG4bK6aa623e5
From: "asterisk"
<sip:asterisk at MY.IP.ADD.RESS>;tag=as7108ed74
To: <sip:13239381067 at SIP.PROXY.ADDRESS>
Contact: <sip:asterisk at MY.IP.ADD.RESS>
Call-ID:
05b58f7e76eae6bb53a87dc142a1a7a0 at MY.IP.ADD.RESS
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0
(no NAT) to 216.187.111.6:5060
We're at MY.IP.ADD.RESS port 13182
Answering with preferred capability 8
Answering with preferred capability 4
Answering with non-codec capability 1
Reliably Transmitting:
INVITE sip:13239381067 at SIP.PROXY.ADDRESS SIP/2.0
Via: SIP/2.0/UDP
MY.IP.ADD.RESS:5060;branch=z9hG4bK6aa623e5
From: "asterisk"
<sip:asterisk at MY.IP.ADD.RESS>;tag=as7108ed74
To: <sip:13239381067 at SIP.PROXY.ADDRESS>
Contact: <sip:asterisk at MY.IP.ADD.RESS>
Call-ID:
05b58f7e76eae6bb53a87dc142a1a7a0 at MY.IP.ADD.RESS
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Proxy-Authorization: Digest username="6045551634",
realm="SIP.PROXY.ADDRESS", algorithm="MD5",
uri="sip:13239381067 at SIP.PROXY.ADDRESS",
nonce="308aefae000c2523",
response="36f377a5a405adc3a3d2b15e84b83109"
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 215
v=0
o=root 15930 15931 IN IP4 MY.IP.ADD.RESS
s=session
c=IN IP4 MY.IP.ADD.RESS
t=0 0
m=audio 13182 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
(no NAT) to 216.187.111.6:5060
Sip read:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
MY.IP.ADD.RESS:5060;branch=z9hG4bK6aa623e5
From: "asterisk"
<sip:asterisk at MY.IP.ADD.RESS>;tag=as7108ed74
To: <sip:13239381067 at SIP.PROXY.ADDRESS>
Call-Id:
05b58f7e76eae6bb53a87dc142a1a7a0 at MY.IP.ADD.RESS
Cseq: 103 INVITE
Date: Thu, 04 Dec 2003 19:30:35 GMT
Proxy-Authenticate: Digest realm="SIP.PROXY.ADDRESS",
nonce="308aefae000c2523", algorithm=MD5
Content-Length: 0
9 headers, 0 lines
Transmitting:
ACK sip:13239381067 at SIP.PROXY.ADDRESS SIP/2.0
Via: SIP/2.0/UDP
MY.IP.ADD.RESS:5060;branch=z9hG4bK6aa623e5
From: "asterisk"
<sip:asterisk at MY.IP.ADD.RESS>;tag=as7108ed74
To: <sip:13239381067 at SIP.PROXY.ADDRESS>
Contact: <sip:asterisk at MY.IP.ADD.RESS>
Call-ID:
05b58f7e76eae6bb53a87dc142a1a7a0 at MY.IP.ADD.RESS
CSeq: 103 ACK
User-Agent: Asterisk PBX
Content-Length: 0
(no NAT) to 216.187.111.6:5060
We're at MY.IP.ADD.RESS port 13182
Answering with preferred capability 8
Answering with preferred capability 4
Answering with non-codec capability 1
Reliably Transmitting:
INVITE sip:13239381067 at SIP.PROXY.ADDRESS SIP/2.0
Via: SIP/2.0/UDP
MY.IP.ADD.RESS:5060;branch=z9hG4bK6aa623e5
From: "asterisk"
<sip:asterisk at MY.IP.ADD.RESS>;tag=as7108ed74
To: <sip:13239381067 at SIP.PROXY.ADDRESS>
Contact: <sip:asterisk at MY.IP.ADD.RESS>
Call-ID:
05b58f7e76eae6bb53a87dc142a1a7a0 at MY.IP.ADD.RESS
CSeq: 104 INVITE
User-Agent: Asterisk PBX
Proxy-Authorization: Digest username="6045551634",
realm="SIP.PROXY.ADDRESS", algorithm="MD5",
uri="sip:13239381067 at SIP.PROXY.ADDRESS",
nonce="308aefae000c2523",
response="36f377a5a405adc3a3d2b15e84b83109"
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 215
v=0
o=root 15930 15932 IN IP4 MY.IP.ADD.RESS
s=session
c=IN IP4 MY.IP.ADD.RESS
t=0 0
m=audio 13182 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
(no NAT) to 216.187.111.6:5060
Sip read:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
MY.IP.ADD.RESS:5060;branch=z9hG4bK6aa623e5
From: "asterisk"
<sip:asterisk at MY.IP.ADD.RESS>;tag=as7108ed74
To: <sip:13239381067 at SIP.PROXY.ADDRESS>
Call-Id:
05b58f7e76eae6bb53a87dc142a1a7a0 at MY.IP.ADD.RESS
Cseq: 104 INVITE
Date: Thu, 04 Dec 2003 19:30:35 GMT
Proxy-Authenticate: Digest realm="SIP.PROXY.ADDRESS",
nonce="308aefae000c2523", algorithm=MD5
Content-Length: 0
9 headers, 0 lines
Transmitting:
ACK sip:13239381067 at SIP.PROXY.ADDRESS SIP/2.0
Via: SIP/2.0/UDP
MY.IP.ADD.RESS:5060;branch=z9hG4bK6aa623e5
From: "asterisk"
<sip:asterisk at MY.IP.ADD.RESS>;tag=as7108ed74
To: <sip:13239381067 at SIP.PROXY.ADDRESS>
Contact: <sip:asterisk at MY.IP.ADD.RESS>
Call-ID:
05b58f7e76eae6bb53a87dc142a1a7a0 at MY.IP.ADD.RESS
CSeq: 104 ACK
User-Agent: Asterisk PBX
Content-Length: 0
(no NAT) to 216.187.111.6:5060
NOTICE[1125329600]: File chan_sip.c, Line 4802
(handle_response): Failed to authenticate on INVITE to
'"asterisk"
<sip:asterisk at MY.IP.ADD.RESS>;tag=as7108ed74'
WARNING[1200825920]: File app_dial.c, Line 317
(wait_for_answer): Unable to forward voice
Any help is always appreciated.
Thank you for your time.
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