[Asterisk-Dev] SIP CANCEL/BYE authorization patch

Tue Apr 8 06:04:51 MST 2003

Mark,

got the latest CVS. Still seeing extremely long delay between hangup and
tear-down of the RTP connection with Iconnect.

sip debug attached.

Gregg

On Sun, 2003-04-06 at 23:27, Mark Spencer wrote:
> > Mark: This fixes the second problem i've reports earlier today, BYE now
> > works without SIP481 error and hangup-delay. I'm still getting SIP487
> > on CANCEL, is this really an error or just a confirmation for the cancel?
> 
> It is, according to 3261, just the confirmation on the cancel.
> 
> Mark
> 
> _______________________________________________
> Asterisk-Dev mailing list
> Asterisk-Dev at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-dev
-------------- next part --------------
-48421825
CSeq: 102 INVITE
Proxy-Authenticate:DIGEST realm="deltathree.com", nonce="3e92c78c", algorithm=MD5
Content-Length: 0

8 headers, 0 lines
Transmitting:
ACK sip:777716176210060 at 213.137.73.140 SIP/2.0
Via: SIP/2.0/UDP 192.168.4.3:5060;branch=76e44f1a
From: "asterisk" <sip:asterisk at 192.168.4.3>;tag=as5c5597e8
To: <sip:777716176210060 at 213.137.73.140>;tag=3e819e9c-48421825
Contact: <sip:asterisk at 192.168.4.3>
Call-ID: 09b3c0ed31fbdb6a5f1a6c8d116d626c at 192.168.4.3
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

 (NAT) to 213.137.73.140:5060
We're at 192.168.4.3 port 47934
Answering with capability 2
Answering with capability 4
Answering with capability 8
Answering with non-codec capability 1
Reliably Transmitting:
INVITE sip:777716176210060 at 213.137.73.140 SIP/2.0
Via: SIP/2.0/UDP 192.168.4.3:5060;branch=76e44f1a
From: "asterisk" <sip:asterisk at 192.168.4.3>;tag=as5c5597e8
Contact: <sip:asterisk at 192.168.4.3>
To: <sip:777716176210060 at 213.137.73.140>
Call-ID: 09b3c0ed31fbdb6a5f1a6c8d116d626c at 192.168.4.3
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Proxy-Authorization: Digest username="85904362", realm="deltathree.com", algorithm="MD5", uri="sip:777716176210060 at 213.137.73.140", nonce="3e92c78c", response="1151999b797d8b7dd7ac5f3d3f1c0724"
Content-Type: application/sdp
Content-Length: 234

v=0
o=root 11851 11851 IN IP4 192.168.4.3
s=session
c=IN IP4 192.168.4.3
t=0 0
m=audio 47934 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
 (NAT) to 213.137.73.140:5060
Sip read: >
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.4.3:5060;branch=76e44f1a
Call-ID: 09b3c0ed31fbdb6a5f1a6c8d116d626c at 192.168.4.3
From: "asterisk" <sip:asterisk at 192.168.4.3>;tag=as5c5597e8
To: sip:777716176210060 at 192.168.4.3
CSeq: 103 INVITE
Content-Length: 0

7 headers, 0 lines
Sip read: >
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.4.3:5060;branch=76e44f1a
From: "asterisk" <sip:asterisk at 192.168.4.3>;tag=as5c5597e8
To: sip:777716176210060 at 192.168.4.3;tag=492470B4-A83
Date:Tue, 08 Apr 2003 12:58:55 GMT
Call-ID: 09b3c0ed31fbdb6a5f1a6c8d116d626c at 192.168.4.3
Server:Cisco-SIPGateway/IOS-12.x
CSeq: 103 INVITE
Allow-Events: telephone-event
Content-Type:application/sdp
Content-Disposition:session;handling=required
Content-Length: 184

v=0
o=CiscoSystemsSIP-GW-UserAgent 7645 9940 IN IP4 213.137.65.239
s=SIP Call
c=IN IP4 213.137.65.239
t=0 0
m=audio 18548 RTP/AVP 3
c=IN IP4 213.137.65.239
a=rtpmap:3 GSM/8000

12 headers, 8 lines
Capabilities: us - 14, them - 2, combined - 2
Non-codec capabilities: us - 1, them - 0, combined - 0
Sip read: >
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.4.3:5060;branch=76e44f1a
From: "asterisk" <sip:asterisk at 192.168.4.3>;tag=as5c5597e8
To: sip:777716176210060 at 192.168.4.3;tag=492470B4-A83
Date:Tue, 08 Apr 2003 12:58:55 GMT
Call-ID: 09b3c0ed31fbdb6a5f1a6c8d116d626c at 192.168.4.3
Server:Cisco-SIPGateway/IOS-12.x
CSeq: 103 INVITE
Allow:INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO
Allow-Events: telephone-event
Contact:sip:777716176210060 at 213.137.65.239:5060
Record-Route: <sip:213.137.79.80>
Record-Route: <sip:213.137.79.78>
Record-Route: <sip:777716176210060 at 213.137.73.178:5060;maddr=213.137.73.176>
Record-Route: <sip:213.137.73.141:5060;maddr=213.137.73.140>
Content-Type:application/sdp
Content-Length: 184

v=0
o=CiscoSystemsSIP-GW-UserAgent 7645 9940 IN IP4 213.137.65.239
s=SIP Call
c=IN IP4 213.137.65.239
t=0 0
m=audio 18548 RTP/AVP 3
c=IN IP4 213.137.65.239
a=rtpmap:3 GSM/8000

17 headers, 8 lines
Capabilities: us - 14, them - 2, combined - 2
Non-codec capabilities: us - 1, them - 0, combined - 0
list_route: hop: <sip:213.137.73.141:5060;maddr=213.137.73.140>
list_route: hop: <sip:777716176210060 at 213.137.73.178:5060;maddr=213.137.73.176>
list_route: hop: <sip:213.137.79.78>
list_route: hop: <sip:213.137.79.80>
list_route: hop: <sip:777716176210060 at 213.137.65.239:5060>
set_destination: Parsing <sip:213.137.73.141:5060;maddr=213.137.73.140> for address/port to send to
set_destination: set destination to 213.137.73.140, port 5060
Transmitting:
ACK sip:777716176210060 at 213.137.73.140 SIP/2.0
Via: SIP/2.0/UDP 192.168.4.3:5060;branch=76e44f1a
Route: <sip:777716176210060 at 213.137.73.178:5060;maddr=213.137.73.176>,<sip:213.137.79.78>,<sip:213.137.79.80>,<sip:777716176210060 at 213.137.65.239:5060>
From: "asterisk" <sip:asterisk at 192.168.4.3>;tag=as5c5597e8
To: <sip:777716176210060 at 213.137.73.140>;tag=3e819e9c-48421825
Contact: <sip:asterisk at 192.168.4.3>
Call-ID: 09b3c0ed31fbdb6a5f1a6c8d116d626c at 192.168.4.3
CSeq: 103 ACK
User-Agent: Asterisk PBX
Content-Length: 0

 (NAT) to 213.137.73.140:5060
    -- SIP/iconnecthere-6eef answered Phone/phone0
set_destination: Parsing <sip:213.137.73.141:5060;maddr=213.137.73.140> for address/port to send to
set_destination: set destination to 213.137.73.140, port 5060
Reliably Transmitting:
BYE sip:777716176210060 at 213.137.73.140 SIP/2.0
Via: SIP/2.0/UDP 192.168.4.3:5060;branch=76e44f1a
Route: <sip:777716176210060 at 213.137.73.178:5060;maddr=213.137.73.176>,<sip:213.137.79.78>,<sip:213.137.79.80>,<sip:777716176210060 at 213.137.65.239:5060>
From: "asterisk" <sip:asterisk at 192.168.4.3>;tag=as5c5597e8
To: <sip:777716176210060 at 213.137.73.140>;tag=3e819e9c-48421825
Contact: <sip:asterisk at 192.168.4.3>
Call-ID: 09b3c0ed31fbdb6a5f1a6c8d116d626c at 192.168.4.3
CSeq: 104 BYE
User-Agent: Asterisk PBX
Proxy-Authorization: Digest username="85904362", realm="deltathree.com", algorithm="MD5", uri="sip:777716176210060 at 213.137.73.140", nonce="3e92c78c", response="562a00d88f626445e73da6fbce7b2eb9"
Content-Length: 0

 (NAT) to 213.137.73.140:5060
  == Spawn extension (default, 16176210060, 1) exited non-zero on 'Phone/phone0'    -- Hungup 'Phone/phone0'
Sip read: >
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.4.3:5060;branch=76e44f1a
Call-ID: 09b3c0ed31fbdb6a5f1a6c8d116d626c at 192.168.4.3
From: "asterisk" <sip:asterisk at 192.168.4.3>;tag=as5c5597e8
To: sip:777716176210060 at 192.168.4.3;tag=3e819e9c-48421825
CSeq: 104 BYE
Content-Length: 0

7 headers, 0 lines
Sip read: >
SIP/2.0 481 Call Leg/Transaction Does Not Exist
Via: SIP/2.0/UDP 192.168.4.3:5060;branch=76e44f1a
From: "asterisk" <sip:asterisk at 192.168.4.3>;tag=as5c5597e8
To: sip:777716176210060 at 192.168.4.3;tag=3e819e9c-48421825
Call-ID: 09b3c0ed31fbdb6a5f1a6c8d116d626c at 192.168.4.3
CSeq: 104 BYE
Content-Length: 0

7 headers, 0 lines
Interface is eth0
IP Address is 192.168.4.3
Sip read: >
BYE sip:66.30.28.60:5060;maddr=213.137.73.140 SIP/2.0
Record-Route:<sip:777716176210060 at 213.137.73.178:5060;maddr=213.137.73.176>
Via: SIP/2.0/UDP 213.137.73.141:5060;maddr=213.137.73.140
Via: SIP/2.0/UDP 213.137.73.176:5060;branch=e8fba87e-3f981ad9-febefb34-a08e1a11-1
Via: SIP/2.0/UDP 213.137.79.78:5060
Via: SIP/2.0/UDP 213.137.79.80:5060
Via: SIP/2.0/UDP 213.137.65.239:5060
From: sip:777716176210060 at 213.137.73.140;tag=492470B4-A83
To: "asterisk" <sip:asterisk at 192.168.4.3>;tag=as5c5597e8
Date:Tue, 08 Apr 2003 12:59:02 GMT
Call-ID: 09b3c0ed31fbdb6a5f1a6c8d116d626c at 192.168.4.3
User-Agent:Cisco-SIPGateway/IOS-12.x
Max-Forwards:9
Route: <sip:asterisk at 192.168.4.3:5060>
Timestamp:1049806811
CSeq: 101 BYE
Content-Length: 0

17 headers, 0 lines
Interface is eth0
IP Address is 192.168.4.3
Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 213.137.73.141:5060;maddr=213.137.73.140
Via: SIP/2.0/UDP 213.137.73.176:5060;branch=e8fba87e-3f981ad9-febefb34-a08e1a11-1
Via: SIP/2.0/UDP 213.137.79.78:5060
Via: SIP/2.0/UDP 213.137.79.80:5060
Via: SIP/2.0/UDP 213.137.65.239:5060
Record-Route: <sip:777716176210060 at 213.137.73.178:5060;maddr=213.137.73.176>
From: sip:777716176210060 at 213.137.73.140;tag=492470B4-A83
To: "asterisk" <sip:asterisk at 192.168.4.3>;tag=as5c5597e8
Call-ID: 09b3c0ed31fbdb6a5f1a6c8d116d626c at 192.168.4.3
CSeq: 101 BYE
User-Agent: Asterisk PBX
Contact:
Content-Length: 0

 to 213.137.73.140:5060
bigcat*CLI>