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:[email protected];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:[email protected];user=phone>
To: <sip:[email protected];user=phone>
From: "Dan Luc"
<sip:[email protected]:5060;user=phone>;tag=37a49f0486bab42b240be214b2d13153
Contact: <sip:[email protected]:5060;transport=UDP>
Call-ID: [email protected]
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:[email protected]:5060;user=phone>;tag=37a49f0486bab42b240be214b2d13153
To: <sip:[email protected];user=phone>;tag=as1b6f387a
Call-ID: [email protected]
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:[email protected];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:[email protected];user=phone>
To: <sip:[email protected];user=phone>
From: "Dan Luc"
<sip:[email protected]:5060;user=phone>;tag=37a49f0486bab42b240be214b2d13153
Contact: <sip:[email protected]:5060;transport=UDP>
Call-ID: [email protected]
CSeq: 443337259 INVITE
Max-Forwards: 70
Authorization: Digest
username="SIPPEERusername",realm="domain.tld",nonce="7684ab1d",algorithm=MD5,uri="sip:[email protected];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:[email protected]:5060;user=phone>;tag=37a49f0486bab42b240be214b2d13153
To: <sip:[email protected];user=phone>
Call-ID: [email protected]
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:[email protected]>
Content-Length: 0
Thanks !
Jonas.
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