[Asterisk-Users] SIP Problem (previous post) .. information might be relevant

Tue Jul 8 14:28:15 MST 2003

regarding my previous post about SIP outgoing calls
dropping with an error 481 ..

this is my output from  a SIP debug.
the call dropped occurs at the end.
Asterisk is mine, Cisco-SIPGateway is the other end (remote) and not in my
control.

help :) please!!

Dave

Signal=0
Duration=250
 (no NAT) to 216.52.153.207:5060
Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.168.168.5:5060;branch=z9hG4bK5a5cde5e
From: "21382890" <sip:21382890 at 217.168.168.5>;tag=as6556b0d9
To: <sip:723 at 216.52.153.207>;tag=26845C24-FDA
Date: Tue, 08 Jul 2003 22:22:57 GMT
Call-ID: 14bce0f47fb42b734f7904ca351a4220 at 217.168.168.5
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
CSeq: 117 INFO
Contact: <sip:723 at 216.52.153.207:5060>

10 headers, 0 lines
set_destination: Parsing <sip:723 at 216.52.153.207:5060> for address/port to
send to
set_destination: set destination to 216.52.153.207, port 5060
Reliably Transmitting:
INFO sip:723 at 216.52.153.207 SIP/2.0
Via: SIP/2.0/UDP 217.168.168.5:5060;branch=z9hG4bK5a5cde5e
From: "21382890" <sip:21382890 at 217.168.168.5>;tag=as6556b0d9
To: <sip:723 at 216.52.153.207>;tag=26845C24-FDA
Contact: <sip:21382890 at 217.168.168.5>
Call-ID: 14bce0f47fb42b734f7904ca351a4220 at 217.168.168.5
CSeq: 118 INFO
User-Agent: Asterisk PBX
Content-Type: application/dtmf-relay
Content-Length: 24

Signal=1
Duration=250
 (no NAT) to 216.52.153.207:5060
Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.168.168.5:5060;branch=z9hG4bK5a5cde5e
From: "21382890" <sip:21382890 at 217.168.168.5>;tag=as6556b0d9
To: <sip:723 at 216.52.153.207>;tag=26845C24-FDA
Date: Tue, 08 Jul 2003 22:22:58 GMT
Call-ID: 14bce0f47fb42b734f7904ca351a4220 at 217.168.168.5
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
CSeq: 118 INFO
Contact: <sip:723 at 216.52.153.207:5060>

10 headers, 0 lines
set_destination: Parsing <sip:723 at 216.52.153.207:5060> for address/port to
send to
set_destination: set destination to 216.52.153.207, port 5060
Reliably Transmitting:
INFO sip:723 at 216.52.153.207 SIP/2.0
Via: SIP/2.0/UDP 217.168.168.5:5060;branch=z9hG4bK5a5cde5e
From: "21382890" <sip:21382890 at 217.168.168.5>;tag=as6556b0d9
To: <sip:723 at 216.52.153.207>;tag=26845C24-FDA
Contact: <sip:21382890 at 217.168.168.5>
Call-ID: 14bce0f47fb42b734f7904ca351a4220 at 217.168.168.5
CSeq: 119 INFO
User-Agent: Asterisk PBX
Content-Type: application/dtmf-relay
Content-Length: 24

Signal=2
Duration=250
 (no NAT) to 216.52.153.207:5060
Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.168.168.5:5060;branch=z9hG4bK5a5cde5e
From: "21382890" <sip:21382890 at 217.168.168.5>;tag=as6556b0d9
To: <sip:723 at 216.52.153.207>;tag=26845C24-FDA
Date: Tue, 08 Jul 2003 22:22:58 GMT
Call-ID: 14bce0f47fb42b734f7904ca351a4220 at 217.168.168.5
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
CSeq: 119 INFO
Contact: <sip:723 at 216.52.153.207:5060>

10 headers, 0 lines
set_destination: Parsing <sip:723 at 216.52.153.207:5060> for address/port to
send to
set_destination: set destination to 216.52.153.207, port 5060
Reliably Transmitting:
INFO sip:723 at 216.52.153.207 SIP/2.0
Via: SIP/2.0/UDP 217.168.168.5:5060;branch=z9hG4bK5a5cde5e
From: "21382890" <sip:21382890 at 217.168.168.5>;tag=as6556b0d9
To: <sip:723 at 216.52.153.207>;tag=26845C24-FDA
Contact: <sip:21382890 at 217.168.168.5>
Call-ID: 14bce0f47fb42b734f7904ca351a4220 at 217.168.168.5
CSeq: 120 INFO
User-Agent: Asterisk PBX
Content-Type: application/dtmf-relay
Content-Length: 24

Signal=5
Duration=250
 (no NAT) to 216.52.153.207:5060
set_destination: Parsing <sip:723 at 216.52.153.207:5060> for address/port to
send to
set_destination: set destination to 216.52.153.207, port 5060
Reliably Transmitting:
INFO sip:723 at 216.52.153.207 SIP/2.0
Via: SIP/2.0/UDP 217.168.168.5:5060;branch=z9hG4bK5a5cde5e
From: "21382890" <sip:21382890 at 217.168.168.5>;tag=as6556b0d9
To: <sip:723 at 216.52.153.207>;tag=26845C24-FDA
Contact: <sip:21382890 at 217.168.168.5>
Call-ID: 14bce0f47fb42b734f7904ca351a4220 at 217.168.168.5
CSeq: 121 INFO
User-Agent: Asterisk PBX
Content-Type: application/dtmf-relay
Content-Length: 24

Signal=6
Duration=250
 (no NAT) to 216.52.153.207:5060
Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.168.168.5:5060;branch=z9hG4bK5a5cde5e
From: "21382890" <sip:21382890 at 217.168.168.5>;tag=as6556b0d9
To: <sip:723 at 216.52.153.207>;tag=26845C24-FDA
Date: Tue, 08 Jul 2003 22:22:59 GMT
Call-ID: 14bce0f47fb42b734f7904ca351a4220 at 217.168.168.5
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
CSeq: 121 INFO
Contact: <sip:723 at 216.52.153.207:5060>

10 headers, 0 lines
Retransmitting #1 (no NAT):
INFO sip:723 at 216.52.153.207 SIP/2.0
Via: SIP/2.0/UDP 217.168.168.5:5060;branch=z9hG4bK5a5cde5e
From: "21382890" <sip:21382890 at 217.168.168.5>;tag=as6556b0d9
To: <sip:723 at 216.52.153.207>;tag=26845C24-FDA
Contact: <sip:21382890 at 217.168.168.5>
Call-ID: 14bce0f47fb42b734f7904ca351a4220 at 217.168.168.5
CSeq: 120 INFO
User-Agent: Asterisk PBX
Content-Type: application/dtmf-relay
Content-Length: 24

Signal=5
Duration=250

 to 216.52.153.207:5060
set_destination: Parsing <sip:723 at 216.52.153.207:5060> for address/port to
send to
set_destination: set destination to 216.52.153.207, port 5060
Reliably Transmitting:
INFO sip:723 at 216.52.153.207 SIP/2.0
Via: SIP/2.0/UDP 217.168.168.5:5060;branch=z9hG4bK5a5cde5e
From: "21382890" <sip:21382890 at 217.168.168.5>;tag=as6556b0d9
To: <sip:723 at 216.52.153.207>;tag=26845C24-FDA
Contact: <sip:21382890 at 217.168.168.5>
Call-ID: 14bce0f47fb42b734f7904ca351a4220 at 217.168.168.5
CSeq: 122 INFO
User-Agent: Asterisk PBX
Content-Type: application/dtmf-relay
Content-Length: 24

Signal=4
Duration=250
 (no NAT) to 216.52.153.207:5060
Sip read:
SIP/2.0 481 Invalid CSeq Number
Via: SIP/2.0/UDP 217.168.168.5:5060;branch=z9hG4bK5a5cde5e
From: "21382890" <sip:21382890 at 217.168.168.5>;tag=as6556b0d9
To: <sip:723 at 216.52.153.207>;tag=26845C24-FDA
Call-ID: 14bce0f47fb42b734f7904ca351a4220 at 217.168.168.5
CSeq: 120 INFO
Content-Length: 0

7 headers, 0 lines
    -- Got SIP response 481 "Invalid CSeq Number" back from 216.52.153.207
  == Spawn extension (incoming, s, 2) exited non-zero on 'Zap/1-1'
    -- Hungup 'Zap/1-1'
Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.168.168.5:5060;branch=z9hG4bK5a5cde5e
From: "21382890" <sip:21382890 at 217.168.168.5>;tag=as6556b0d9
To: <sip:723 at 216.52.153.207>;tag=26845C24-FDA
Date: Tue, 08 Jul 2003 22:22:59 GMT
Call-ID: 14bce0f47fb42b734f7904ca351a4220 at 217.168.168.5
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
CSeq: 122 INFO
Contact: <sip:723 at 216.52.153.207:5060>

10 headers, 0 lines
Sip read:
BYE sip:21382890 at 217.168.168.5:5060 SIP/2.0
Via: SIP/2.0/UDP  216.52.153.207:5060
From: <sip:723 at 216.52.153.207>;tag=26845C24-FDA
To: "21382890" <sip:21382890 at 217.168.168.5>;tag=as6556b0d9
Date: Tue, 08 Jul 2003 22:22:59 GMT
Call-ID: 14bce0f47fb42b734f7904ca351a4220 at 217.168.168.5
User-Agent: Cisco-SIPGateway/IOS-12.x
Max-Forwards: 6
Timestamp: 1057703051
CSeq: 101 BYE
Content-Length: 0

11 headers, 0 lines
Sending to 216.52.153.207 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP  216.52.153.207:5060
From: <sip:723 at 216.52.153.207>;tag=26845C24-FDA
To: "21382890" <sip:21382890 at 217.168.168.5>;tag=as6556b0d9
Call-ID: 14bce0f47fb42b734f7904ca351a4220 at 217.168.168.5
CSeq: 101 BYE
User-Agent: Asterisk PBX
Contact:
Content-Length: 0

 to 216.52.153.207:5060