[Asterisk-Dev] SIP CANCEL/BYE authorization patch

Gregg Lebovitz gregg at lebovitz.net
Tue Apr 8 10:26:30 MST 2003 

Heres a trace with the line open. I don't see any activity between the
call answer and the hangup.

Gregg



On Tue, 2003-04-08 at 10:11, Stephen Davies wrote:
> On 8 Apr 2003, Gregg Lebovitz wrote:
> 
> > 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
> 
> Hi Gregg,
> 
> I'd like to see a call where you hold the line open for a while - in order
> to see whether they keep sending their 200 OK - ie don't like (or don't
> get) our ACK.
> 
> Thanks,
> Steve
> 
> 
> _______________________________________________
> Asterisk-Dev mailing list
> Asterisk-Dev at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-dev
-------------- next part --------------

Asterisk CVS-04/08/03-08:46:35, Copyright (C) 1999-2001 Linux Support Services, Inc.

Written by Mark Spencer <markster at linux-support.net>

=========================================================================

Connected to Asterisk CVS-04 currently running on bigcat (pid = 11845)
bigcat*CLI> 
    -- Executing Dial("Phone/phone0", "SIP/777716176210060 at iconnecthere") in new stack
 ^Dbigcat*CLI> 
Interface is eth0
 ^Dbigcat*CLI> 
IP Address is 192.168.4.3
 ^Dbigcat*CLI> 
We're at 192.168.4.3 port 37450
 ^Dbigcat*CLI> 
Answering with capability 2
 ^Dbigcat*CLI> 
Answering with capability 4
 ^Dbigcat*CLI> 
Answering with capability 8
 ^Dbigcat*CLI> 
Answering with non-codec capability 1
 ^Dbigcat*CLI> 
10 headers, 11 lines
 ^Dbigcat*CLI> 
Reliably Transmitting:
INVITE sip:777716176210060 at 213.137.73.140 SIP/2.0
Via: SIP/2.0/UDP 192.168.4.3:5060;branch=1f10f58a
From: "asterisk" <sip:asterisk at 192.168.4.3>;tag=as2cebae5c
Contact: <sip:asterisk at 192.168.4.3>
To: <sip:777716176210060 at 213.137.73.140>
Call-ID: 4969610c7df4b9ab321725a83bad1a53 at 192.168.4.3
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Content-Type: application/sdp
Content-Length: 234

v=0
o=root 11897 11897 IN IP4 192.168.4.3
s=session
c=IN IP4 192.168.4.3
t=0 0
m=audio 37450 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
 ^Dbigcat*CLI> 
    -- Called 777716176210060 at iconnecthere
 ^Dbigcat*CLI> 
Sip read: 
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.4.3:5060;branch=1f10f58a
Call-ID: 4969610c7df4b9ab321725a83bad1a53 at 192.168.4.3
From: "asterisk" <sip:asterisk at 192.168.4.3>;tag=as2cebae5c
To: sip:777716176210060 at 192.168.4.3
CSeq: 102 INVITE
Content-Length: 0


 7 headers, 0 lines
 ^Dbigcat*CLI> 
Sip read: 
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.4.3:5060;branch=1f10f58a
Call-ID: 4969610c7df4b9ab321725a83bad1a53 at 192.168.4.3
From: "asterisk" <sip:asterisk at 192.168.4.3>;tag=as2cebae5c
To: sip:777716176210060 at 192.168.4.3;tag=1d67756-378a83d6
CSeq: 102 INVITE
Proxy-Authenticate:DIGEST realm="deltathree.com", nonce="3e930388", 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=1f10f58a
From: "asterisk" <sip:asterisk at 192.168.4.3>;tag=as2cebae5c
To: <sip:777716176210060 at 213.137.73.140>;tag=1d67756-378a83d6
Contact: <sip:asterisk at 192.168.4.3>
Call-ID: 4969610c7df4b9ab321725a83bad1a53 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 37450
 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=1f10f58a
From: "asterisk" <sip:asterisk at 192.168.4.3>;tag=as2cebae5c
Contact: <sip:asterisk at 192.168.4.3>
To: <sip:777716176210060 at 213.137.73.140>
Call-ID: 4969610c7df4b9ab321725a83bad1a53 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="3e930388", response="a4260204d0528096d90f272d9f2e0307"
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 37450 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
 ^Dbigcat*CLI> 
Sip read: 
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.4.3:5060;branch=1f10f58a
Call-ID: 4969610c7df4b9ab321725a83bad1a53 at 192.168.4.3
From: "asterisk" <sip:asterisk at 192.168.4.3>;tag=as2cebae5c
To: sip:777716176210060 at 192.168.4.3
CSeq: 103 INVITE
Content-Length: 0


 7 headers, 0 lines
 ^Dbigcat*CLI> 
Sip read: 
SIP/2.0 183 Session Progress  
Via: SIP/2.0/UDP 192.168.4.3:5060;branch=1f10f58a
From: "asterisk" <sip:asterisk at 192.168.4.3>;tag=as2cebae5c
To: sip:777716176210060 at 192.168.4.3;tag=126C5F40-22E8
Date:Tue, 08 Apr 2003 17:14:49 GMT
Call-ID: 4969610c7df4b9ab321725a83bad1a53 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: 183

v=0
o=CiscoSystemsSIP-GW-UserAgent 9665 309 IN IP4 213.137.65.235
s=SIP Call
c=IN IP4 213.137.65.235
t=0 0
m=audio 19310 RTP/AVP 3
c=IN IP4 213.137.65.235
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
 ^Dbigcat*CLI> 
Sip read: 
SIP/2.0 200 OK  
Via: SIP/2.0/UDP 192.168.4.3:5060;branch=1f10f58a
From: "asterisk" <sip:asterisk at 192.168.4.3>;tag=as2cebae5c
To: sip:777716176210060 at 192.168.4.3;tag=126C5F40-22E8
Date:Tue, 08 Apr 2003 17:14:49 GMT
Call-ID: 4969610c7df4b9ab321725a83bad1a53 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 213.137.65.235:5060
Record-Route: <sip:213.137.79.83>
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: 183

v=0
o=CiscoSystemsSIP-GW-UserAgent 9665 309 IN IP4 213.137.65.235
s=SIP Call
c=IN IP4 213.137.65.235
t=0 0
m=audio 19310 RTP/AVP 3
c=IN IP4 213.137.65.235
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:213.137.79.83>
 list_route: hop: <sip:777716176210060 at 213.137.65.235:5060>
 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=1f10f58a
Route: <sip:777716176210060 at 213.137.73.178:5060;maddr=213.137.73.176>,<sip:213.137.79.78>,<sip:213.137.79.83>,<sip:777716176210060 at 213.137.65.235:5060>
From: "asterisk" <sip:asterisk at 192.168.4.3>;tag=as2cebae5c
To: <sip:777716176210060 at 213.137.73.140>;tag=1d67756-378a83d6
Contact: <sip:asterisk at 192.168.4.3>
Call-ID: 4969610c7df4b9ab321725a83bad1a53 at 192.168.4.3
CSeq: 103 ACK
User-Agent: Asterisk PBX
Content-Length: 0

 (NAT) to 213.137.73.140:5060
     -- SIP/iconnecthere-93ce answered Phone/phone0
 ^Dbigcat*CLI> 
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=1f10f58a
Route: <sip:777716176210060 at 213.137.73.178:5060;maddr=213.137.73.176>,<sip:213.137.79.78>,<sip:213.137.79.83>,<sip:777716176210060 at 213.137.65.235:5060>
From: "asterisk" <sip:asterisk at 192.168.4.3>;tag=as2cebae5c
To: <sip:777716176210060 at 213.137.73.140>;tag=1d67756-378a83d6
Contact: <sip:asterisk at 192.168.4.3>
Call-ID: 4969610c7df4b9ab321725a83bad1a53 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="3e930388", response="5ac4e5c545a0b4ae5d3ec65bf15635b7"
Content-Length: 0

 (NAT) to 213.137.73.140:5060
   == Spawn extension (default, 16176210060, 1) exited non-zero on 'Phone/phone0'
     -- Hungup 'Phone/phone0'
 ^Dbigcat*CLI> 
Sip read: 
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.4.3:5060;branch=1f10f58a
Call-ID: 4969610c7df4b9ab321725a83bad1a53 at 192.168.4.3
From: "asterisk" <sip:asterisk at 192.168.4.3>;tag=as2cebae5c
To: sip:777716176210060 at 192.168.4.3;tag=1d67756-378a83d6
CSeq: 104 BYE
Content-Length: 0


 7 headers, 0 lines
 ^Dbigcat*CLI> 
Sip read: 
SIP/2.0 481 Call Leg/Transaction Does Not Exist  
Via: SIP/2.0/UDP 192.168.4.3:5060;branch=1f10f58a
From: "asterisk" <sip:asterisk at 192.168.4.3>;tag=as2cebae5c
To: sip:777716176210060 at 192.168.4.3;tag=1d67756-378a83d6
Call-ID: 4969610c7df4b9ab321725a83bad1a53 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
 ^Dbigcat*CLI> 
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=31dab3a-f7f09c4f-b694ee20-84045f26-1
Via: SIP/2.0/UDP 213.137.79.78:5060
Via: SIP/2.0/UDP 213.137.79.83:5060
Via: SIP/2.0/UDP 213.137.65.235:5060
From: sip:777716176210060 at 213.137.73.140;tag=126C5F40-22E8
To: "asterisk" <sip:asterisk at 192.168.4.3>;tag=as2cebae5c
Date:Tue, 08 Apr 2003 17:14:54 GMT
Call-ID: 4969610c7df4b9ab321725a83bad1a53 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:1049822240
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=31dab3a-f7f09c4f-b694ee20-84045f26-1
Via: SIP/2.0/UDP 213.137.79.78:5060
Via: SIP/2.0/UDP 213.137.79.83:5060
Via: SIP/2.0/UDP 213.137.65.235:5060
Record-Route: <sip:777716176210060 at 213.137.73.178:5060;maddr=213.137.73.176>
From: sip:777716176210060 at 213.137.73.140;tag=126C5F40-22E8
To: "asterisk" <sip:asterisk at 192.168.4.3>;tag=as2cebae5c
Call-ID: 4969610c7df4b9ab321725a83bad1a53 at 192.168.4.3
CSeq: 101 BYE
User-Agent: Asterisk PBX
Contact: 
Content-Length: 0


 to 213.137.73.140:5060
 ^Dbigcat*CLI>

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