[asterisk-users] Sip incoming call issue with Asterisk 1.6

Sun Nov 15 15:44:39 CST 2009

Leif,

Le dimanche 15 novembre 2009 à 16:18 -0500, Leif Madsen a écrit :

> I'm not sure you've provided enough of the trace here. It finds the peer, but 
> rejects it with a 401 Unauthorized, which is not uncommon. And I don't see any 
> authentication information in the first INVITE. This is why the 401 is sent 
> back, as the WWW-Authenticate line contains the realm and nonce which should be 
> used by the other end to generate the authentication, and then send another 
> INVITE back with authentication.
> 
> Since you've only shown the two packets in the trace, it is impossible to tell 
> what is going on beyond the 401 response from Asterisk.
> 
> Leif.

I attach more packets (from a new test run).

Thanks,

Eric

<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '074e14720543d49c21e4999c4786b686 at 192.168.4.2' Method: OPTIONS
asterisk*CLI> 
<--- SIP read from UDP:212.27.52.5:5060 --->
INVITE sip:s at 192.168.4.2:5060;transport=udp SIP/2.0
Call-ID: 01749-LO-0eb4458a-552a137e7 at freephonie.net
Contact: <sip:172.17.20.241:5062>
Content-Type: application/sdp
CSeq: 240171363 INVITE
From: "Caller" <sip:095245XXXX at freephonie.net;user=phone>;tag=01749-RV-0eb4458b-093bd4160
Max-Forwards: 28
Record-Route: <sip:C=on-88.165.134.117.5060;t=XNJDG at 212.27.52.5:5060;lr>
To: <sip:095199YYYY at 172.17.20.241;user=phone>
Via: SIP/2.0/UDP 212.27.52.5:5060;branch=z9hG4bK-XNJD-0079618e-344c12fe
Allow: UPDATE,REFER,INFO
User-Agent: Cirpack/v4.41c (gw_sip)
Content-Length: 173

v=0
o=cp10 125832072268 125832072268 IN IP4 212.27.52.130
s=SIP Call
c=IN IP4 212.27.52.130
t=0 0
m=audio 33092 RTP/AVP 8
b=AS:64
a=rtpmap:8 PCMA/8000/1
a=ptime:30

<------------->
--- (13 headers 9 lines) ---
  == Using SIP RTP CoS mark 5
Sending to 212.27.52.5 : 5060 (no NAT)
Using INVITE request as basis request - 01749-LO-0eb4458a-552a137e7 at freephonie.net
Found peer 'freephonie_appelsortant' for '095245XXXX' from 212.27.52.5:5060
asterisk*CLI> 
<--- Reliably Transmitting (no NAT) to 212.27.52.5:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 212.27.52.5:5060;branch=z9hG4bK-XNJD-0079618e-344c12fe;received=212.27.52.5
From: "Caller" <sip:095245XXXX at freephonie.net;user=phone>;tag=01749-RV-0eb4458b-093bd4160
To: <sip:095199YYYY at 172.17.20.241;user=phone>;tag=as7249497d
Call-ID: 01749-LO-0eb4458a-552a137e7 at freephonie.net
CSeq: 240171363 INVITE
Server: Asterisk PBX 1.6.2.0~rc2-0ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4a3516d7"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog '01749-LO-0eb4458a-552a137e7 at freephonie.net' in 6400 ms (Method: INVITE)
asterisk*CLI> 
<--- SIP read from UDP:212.27.52.5:5060 --->
ACK sip:s at 192.168.4.2:5060;transport=udp SIP/2.0
Call-ID: 01749-LO-0eb4458a-552a137e7 at freephonie.net
CSeq: 240171363 ACK
From: "Caller" <sip:095245XXXX at freephonie.net;user=phone>;tag=01749-RV-0eb4458b-093bd4160
Max-Forwards: 28
To: <sip:095199YYYY at 172.17.20.241;user=phone>;tag=as7249497d
Via: SIP/2.0/UDP 212.27.52.5:5060;branch=z9hG4bK-XNJD-0079618e-344c12fe
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '01749-LO-0eb4458a-552a137e7 at freephonie.net' Method: ACK
Reliably Transmitting (no NAT) to 212.27.52.5:5060:
OPTIONS sip:freephonie.net SIP/2.0
Via: SIP/2.0/UDP 192.168.4.2:5060;branch=z9hG4bK41486e10;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk at 192.168.4.2>;tag=as096de98b
To: <sip:freephonie.net>
Contact: <sip:asterisk at 192.168.4.2>
Call-ID: 3334481068f263823a05fdef0480f2af at 192.168.4.2
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.0~rc2-0ubuntu1
Date: Sun, 15 Nov 2009 21:32:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

---
Reliably Transmitting (no NAT) to 212.27.52.5:5060:
OPTIONS sip:freephonie.net SIP/2.0
Via: SIP/2.0/UDP 192.168.4.2:5060;branch=z9hG4bK49b10bb3;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk at 192.168.4.2>;tag=as0c4787d3
To: <sip:freephonie.net>
Contact: <sip:asterisk at 192.168.4.2>
Call-ID: 00cb37381237588a64cbe39a43ca088b at 192.168.4.2
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.0~rc2-0ubuntu1
Date: Sun, 15 Nov 2009 21:32:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

---
asterisk*CLI> 
<--- SIP read from UDP:212.27.52.5:5060 --->
SIP/2.0 501 Not Implemented Yet
Call-ID: 3334481068f263823a05fdef0480f2af at 192.168.4.2
CSeq: 102 OPTIONS
From: "asterisk" <sip:asterisk at 192.168.4.2>;tag=as096de98b
To: <sip:freephonie.net>;tag=00-32733-00efde5e-4f3d2a1b6
Via: SIP/2.0/UDP 192.168.4.2:5060;received=88.165.134.117;rport=5060;branch=z9hG4bK41486e10
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '3334481068f263823a05fdef0480f2af at 192.168.4.2' Method: OPTIONS
asterisk*CLI> 
<--- SIP read from UDP:212.27.52.5:5060 --->
SIP/2.0 501 Not Implemented Yet
Call-ID: 00cb37381237588a64cbe39a43ca088b at 192.168.4.2
CSeq: 102 OPTIONS
From: "asterisk" <sip:asterisk at 192.168.4.2>;tag=as0c4787d3
To: <sip:freephonie.net>;tag=00-31015-00efde62-60b13b660
Via: SIP/2.0/UDP 192.168.4.2:5060;received=88.165.134.117;rport=5060;branch=z9hG4bK49b10bb3
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '00cb37381237588a64cbe39a43ca088b at 192.168.4.2' Method: OPTIONS
Reliably Transmitting (no NAT) to 212.27.52.5:5060:
OPTIONS sip:freephonie.net SIP/2.0
Via: SIP/2.0/UDP 192.168.4.2:5060;branch=z9hG4bK2470d659;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk at 192.168.4.2>;tag=as4b104230
To: <sip:freephonie.net>
Contact: <sip:asterisk at 192.168.4.2>
Call-ID: 7477f6c44c972eca31b9ea3d5baa4648 at 192.168.4.2
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.0~rc2-0ubuntu1
Date: Sun, 15 Nov 2009 21:33:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

---
Reliably Transmitting (no NAT) to 212.27.52.5:5060:
OPTIONS sip:freephonie.net SIP/2.0
Via: SIP/2.0/UDP 192.168.4.2:5060;branch=z9hG4bK00867a4e;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk at 192.168.4.2>;tag=as223230cb
To: <sip:freephonie.net>
Contact: <sip:asterisk at 192.168.4.2>
Call-ID: 0dc16b0031fe1e952ebb71437ef55b77 at 192.168.4.2
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.0~rc2-0ubuntu1
Date: Sun, 15 Nov 2009 21:33:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

---
asterisk*CLI> 
<--- SIP read from UDP:212.27.52.5:5060 --->
SIP/2.0 501 Not Implemented Yet
Call-ID: 7477f6c44c972eca31b9ea3d5baa4648 at 192.168.4.2
CSeq: 102 OPTIONS
From: "asterisk" <sip:asterisk at 192.168.4.2>;tag=as4b104230
To: <sip:freephonie.net>;tag=00-30943-00efebab-2818f75c0
Via: SIP/2.0/UDP 192.168.4.2:5060;received=88.165.134.117;rport=5060;branch=z9hG4bK2470d659
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '7477f6c44c972eca31b9ea3d5baa4648 at 192.168.4.2' Method: OPTIONS
asterisk*CLI> 
<--- SIP read from UDP:212.27.52.5:5060 --->
SIP/2.0 501 Not Implemented Yet
Call-ID: 0dc16b0031fe1e952ebb71437ef55b77 at 192.168.4.2
CSeq: 102 OPTIONS
From: "asterisk" <sip:asterisk at 192.168.4.2>;tag=as223230cb
To: <sip:freephonie.net>;tag=00-30939-00efebae-7a2b576c6
Via: SIP/2.0/UDP 192.168.4.2:5060;received=88.165.134.117;rport=5060;branch=z9hG4bK00867a4e
Content-Length: 0