[asterisk-users] Asterisk refuses INVITE (401) and I don't know why

Thu Nov 24 15:26:43 CST 2011

just a quick observation, but not sure that it is critical

in this case, the first invite comes without Authorization header, then gets challenged then resends the invite (with increased cseq) with calculated response based on the challenge from the server.

In your AAstra case, the first invite already contained Authorization header (which is really impossible because you don't have all the pieces to calculate the response). Normally not an issue, as UAS should challenge it, but I wonder why it does it anyway. I would compare Authorize elements between 2 cases particularly response, uri and authorization user name. if response is the same between the two, I am lost.
On Nov 24, 2011, at 2:11 PM, Jonas Kellens wrote:

> On 11/22/2011 06:13 PM, Alex Vishnev wrote:
>> 
>> it is strange that Aastra acks 401, sends another invite but does not increase CSeq. Is that the same behavior with others?
>> On Nov 22, 2011, at 11:51 AM, Jonas Kellens wrote:
> This is a trace taken when an Alcatel-Lucent PBX sends an INVITE (no refusal by Asterisk). Do you see any difference ?
> 
> A1.A1.A1.A1 = IP-address Asterisk PBX
> AL.AL.AL.AL = IP-address Alcatel-Lucent PBX
> 
> 
> <--- SIP read from UDP:AL.AL.AL.AL:5060 --->
> INVITE sip:311083335533 at A1.A1.A1.A1;user=phone SIP/2.0
> Allow: INVITE, ACK, CANCEL, BYE, PRACK, NOTIFY, SUBSCRIBE, OPTIONS, UPDATE
> Supported: replaces, timer, 100rel
> User-Agent: OmniPCX Enterprise R9.1 i1.605.21
> Session-Expires: 1800;refresher=uac
> Min-SE: 900
> P-Asserted-Identity: "Dan Luc" <sip:328883300 at 192.168.8.10;user=phone>
> To: <sip:311083335533 at A1.A1.A1.A1;user=phone>
> From: "Dan Luc" <sip:328883300 at AL.AL.AL.AL:5060;user=phone>;tag=37a49f0486bab42b240be214b2d13153
> Contact: <sip:328883300 at AL.AL.AL.AL:5060;transport=UDP>
> Call-ID: 2fae0b0266919172cac1e23dc2567cd2 at 192.168.8.10
> CSeq: 443337258 INVITE
> Via: SIP/2.0/UDP AL.AL.AL.AL:5060;branch=z9hG4bK5dee58e3294e4b7f9fe34c65af7b4cae
> Max-Forwards: 70
> Content-Type: application/sdp
> Content-Length: 292
> 
> v=0
> o=OXE 1322045354 1322045354 IN IP4 AL.AL.AL.AL
> s=abs
> c=IN IP4 AL.AL.AL.AL
> t=0 0
> m=audio 34422 RTP/AVP 8 18 97
> a=sendrecv
> a=rtpmap:8 PCMA/8000
> a=ptime:20
> a=maxptime:30
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=ptime:20
> a=maxptime:40
> a=rtpmap:97 telephone-event/8000
> 
> 
> <--- Reliably Transmitting (NAT) to AL.AL.AL.AL:5060 --->
> SIP/2.0 401 Unauthorized
> Via: SIP/2.0/UDP AL.AL.AL.AL:5060;branch=z9hG4bK5dee58e3294e4b7f9fe34c65af7b4cae;received=AL.AL.AL.AL
> From: "Dan Luc" <sip:328883300 at AL.AL.AL.AL:5060;user=phone>;tag=37a49f0486bab42b240be214b2d13153
> To: <sip:311083335533 at A1.A1.A1.A1;user=phone>;tag=as1b6f387a
> Call-ID: 2fae0b0266919172cac1e23dc2567cd2 at 192.168.8.10
> CSeq: 443337258 INVITE
> Server: Asterisk PBX 1.6.2.20
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
> Supported: replaces, timer
> WWW-Authenticate: Digest algorithm=MD5, realm="domain.tld", nonce="7684ab1d"
> Content-Length: 0
> 
> 
> <--- SIP read from UDP:AL.AL.AL.AL:5060 --->
> INVITE sip:311083335533 at A1.A1.A1.A1;user=phone SIP/2.0
> Allow: INVITE, ACK, CANCEL, BYE, PRACK, NOTIFY, SUBSCRIBE, OPTIONS, UPDATE
> Supported: replaces, timer, 100rel
> User-Agent: OmniPCX Enterprise R9.1 i1.605.21
> Session-Expires: 1800;refresher=uac
> Min-SE: 900
> P-Asserted-Identity: "Dan Luc" <sip:328883300 at 192.168.8.10;user=phone>
> To: <sip:311083335533 at A1.A1.A1.A1;user=phone>
> From: "Dan Luc" <sip:328883300 at AL.AL.AL.AL:5060;user=phone>;tag=37a49f0486bab42b240be214b2d13153
> Contact: <sip:328883300 at AL.AL.AL.AL:5060;transport=UDP>
> Call-ID: 2fae0b0266919172cac1e23dc2567cd2 at 192.168.8.10
> CSeq: 443337259 INVITE
> Max-Forwards: 70
> Authorization: Digest username="SIPPEERusername",realm="domain.tld",nonce="7684ab1d",algorithm=MD5,uri="sip:311083335533 at A1.A1.A1.A1;user=phone",response="38bb824b9081bf2eefe9f9677d3eb005"
> Via: SIP/2.0/UDP AL.AL.AL.AL:5060;branch=z9hG4bK52dae2e7816406e20a9c02aa9cb86726
> Content-Type: application/sdp
> Content-Length: 292
> 
> v=0
> o=OXE 1322045354 1322045354 IN IP4 AL.AL.AL.AL
> s=abs
> c=IN IP4 AL.AL.AL.AL
> t=0 0
> m=audio 34422 RTP/AVP 8 18 97
> a=sendrecv
> a=rtpmap:8 PCMA/8000
> a=ptime:20
> a=maxptime:30
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=ptime:20
> a=maxptime:40
> a=rtpmap:97 telephone-event/8000
> 
> 
> <--- Transmitting (NAT) to AL.AL.AL.AL:5060 --->
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP AL.AL.AL.AL:5060;branch=z9hG4bK52dae2e7816406e20a9c02aa9cb86726;received=AL.AL.AL.AL
> From: "Dan Luc" <sip:328883300 at AL.AL.AL.AL:5060;user=phone>;tag=37a49f0486bab42b240be214b2d13153
> To: <sip:311083335533 at A1.A1.A1.A1;user=phone>
> Call-ID: 2fae0b0266919172cac1e23dc2567cd2 at 192.168.8.10
> CSeq: 443337259 INVITE
> Server: Asterisk PBX 1.6.2.20
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
> Supported: replaces, timer
> Session-Expires: 1800;refresher=uac
> Contact: <sip:311083335533 at A1.A1.A1.A1>
> Content-Length: 0
> 
> 
> Thanks !
> 
> Jonas.
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