[Asterisk-Dev] Missing max-forwards header in INVITE message

Chintan Thakker cthakker at ipnetfusion.com
Wed Jul 9 14:02:14 MST 2003 

Whenver asterisx forwards an INVITE, it does not put the mandatory 
max-forwards header in it.
The trace for a run is attached below.

Thanks,

-- start asterix sip debug trace --
*CLI>
Sip read:
INVITE sip:9727610001 at 192.1.2.17 SIP/2.0
Via: SIP/2.0/UDP 192.1.2.88:5060;branch=z9hG4bK222444590
Max-Forwards: 70
From: 9727619271 <sip:9727619271 at 192.1.2.88>;tag=222444590
To: 9727610001 <sip:9727610001 at 192.1.2.17>
Call-ID: 222444590 at 192.1.2.88
CSeq: 1 INVITE
Contact: <sip:9727619271 at 192.1.2.88>
Content-Type: application/sdp
Content-Length: 138

v=0
o=username 222444590 222444590 IN IP4 192.1.2.88
s=Session SDP
c=IN IP4 192.1.2.88
t=0 0
m=audio 54454 RTP/AVP 0
a=rtpmap:0 PCMU/8000

10 headers, 7 lines
Interface is eth0
IP Address is 192.1.2.17
Using latest request as basis request
Sending to 192.1.2.88 : 5060 (non-NAT)
Capabilities: us - 14, them - 4, combined - 4
Non-codec capabilities: us - 1, them - 0, combined - 0
Looking for 9727610001 in sip
list_route: hop: <sip:9727619271 at 192.1.2.88>
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.1.2.88:5060;branch=z9hG4bK222444590
From: 9727619271 <sip:9727619271 at 192.1.2.88>;tag=222444590
To: 9727610001 <sip:9727610001 at 192.1.2.17>;tag=as491fadd2
Call-ID: 222444590 at 192.1.2.88
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Contact: <sip:9727610001 at 192.1.2.17>
Content-Length: 0


 to 192.1.2.88:5060
    -- Executing Dial("SIP/9727619271-4585", "SIP/9727610001") in new stack
Interface is eth0
IP Address is 192.1.2.17
We're at 192.1.2.17 port 46372
Answering with capability 2
Answering with capability 4
Answering with capability 8
Answering with non-codec capability 1
10 headers, 11 lines
Reliably Transmitting:
INVITE sip:9727610001 at 192.1.2.223 SIP/2.0
Via: SIP/2.0/UDP 192.1.2.17:5060;branch=z9hG4bK5312283c
From: "9727619271" <sip:9727610001 at 192.1.2.17>;tag=as27e35563
To: <sip:9727610001 at 192.1.2.223>
Contact: <sip:9727610001 at 192.1.2.17>
Call-ID: 281e94e27d505ca8376ecf7007cef1d3 at 192.1.2.17
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Content-Type: application/sdp
Content-Length: 232

v=0
o=root 10628 10628 IN IP4 192.1.2.17
s=session
c=IN IP4 192.1.2.17
t=0 0
m=audio 46372 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
 (no NAT) to 192.1.2.223:5060
    -- Called 9727610001
Retransmitting #1 (no NAT):
INVITE sip:9727610001 at 192.1.2.223 SIP/2.0
Via: SIP/2.0/UDP 192.1.2.17:5060;branch=z9hG4bK5312283c
From: "9727619271" <sip:9727610001 at 192.1.2.17>;tag=as27e35563
To: <sip:9727610001 at 192.1.2.223>
Contact: <sip:9727610001 at 192.1.2.17>
Call-ID: 281e94e27d505ca8376ecf7007cef1d3 at 192.1.2.17
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Content-Type: application/sdp
Content-Length: 232

v=0
o=root 10628 10628 IN IP4 192.1.2.17
s=session
c=IN IP4 192.1.2.17
t=0 0
m=audio 46372 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

 to 192.1.2.223:5060
Retransmitting #2 (no NAT):
INVITE sip:9727610001 at 192.1.2.223 SIP/2.0
Via: SIP/2.0/UDP 192.1.2.17:5060;branch=z9hG4bK5312283c
From: "9727619271" <sip:9727610001 at 192.1.2.17>;tag=as27e35563
To: <sip:9727610001 at 192.1.2.223>
Contact: <sip:9727610001 at 192.1.2.17>
Call-ID: 281e94e27d505ca8376ecf7007cef1d3 at 192.1.2.17
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Content-Type: application/sdp
Content-Length: 232

v=0
o=root 10628 10628 IN IP4 192.1.2.17
s=session
c=IN IP4 192.1.2.17
t=0 0
m=audio 46372 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

 to 192.1.2.223:5060
Retransmitting #3 (no NAT):
INVITE sip:9727610001 at 192.1.2.223 SIP/2.0
Via: SIP/2.0/UDP 192.1.2.17:5060;branch=z9hG4bK5312283c
From: "9727619271" <sip:9727610001 at 192.1.2.17>;tag=as27e35563
To: <sip:9727610001 at 192.1.2.223>
Contact: <sip:9727610001 at 192.1.2.17>
Call-ID: 281e94e27d505ca8376ecf7007cef1d3 at 192.1.2.17
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Content-Type: application/sdp
Content-Length: 232

v=0
o=root 10628 10628 IN IP4 192.1.2.17
s=session
c=IN IP4 192.1.2.17
t=0 0
m=audio 46372 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

 to 192.1.2.223:5060
Retransmitting #4 (no NAT):
INVITE sip:9727610001 at 192.1.2.223 SIP/2.0
Via: SIP/2.0/UDP 192.1.2.17:5060;branch=z9hG4bK5312283c
From: "9727619271" <sip:9727610001 at 192.1.2.17>;tag=as27e35563
To: <sip:9727610001 at 192.1.2.223>
Contact: <sip:9727610001 at 192.1.2.17>
Call-ID: 281e94e27d505ca8376ecf7007cef1d3 at 192.1.2.17
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Content-Type: application/sdp
Content-Length: 232

v=0
o=root 10628 10628 IN IP4 192.1.2.17
s=session
c=IN IP4 192.1.2.17
t=0 0
m=audio 46372 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

 to 192.1.2.223:5060
Retransmitting #5 (no NAT):
INVITE sip:9727610001 at 192.1.2.223 SIP/2.0
Via: SIP/2.0/UDP 192.1.2.17:5060;branch=z9hG4bK5312283c
From: "9727619271" <sip:9727610001 at 192.1.2.17>;tag=as27e35563
To: <sip:9727610001 at 192.1.2.223>
Contact: <sip:9727610001 at 192.1.2.17>
Call-ID: 281e94e27d505ca8376ecf7007cef1d3 at 192.1.2.17
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Content-Type: application/sdp
Content-Length: 232

v=0
o=root 10628 10628 IN IP4 192.1.2.17
s=session
c=IN IP4 192.1.2.17
t=0 0
m=audio 46372 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

 to 192.1.2.223:5060
WARNING[40966]: File chan_sip.c, Line 388 (retrans_pkt): Maximum retries 
exceede
d on call 281e94e27d505ca8376ecf7007cef1d3 at 192.1.2.17 for seqno 102 
(Request)
  == No one is available to answer at this time
Reliably Transmitting:
CANCEL sip:9727610001 at 192.1.2.223 SIP/2.0
Via: SIP/2.0/UDP 192.1.2.17:5060;branch=z9hG4bK5312283c
From: "9727619271" <sip:9727610001 at 192.1.2.17>;tag=as27e35563
To: <sip:9727610001 at 192.1.2.223>
Contact: <sip:9727610001 at 192.1.2.17>
Call-ID: 281e94e27d505ca8376ecf7007cef1d3 at 192.1.2.17
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Content-Length: 0

 (no NAT) to 192.1.2.223:5060
Retransmitting #1 (no NAT):
CANCEL sip:9727610001 at 192.1.2.223 SIP/2.0
Via: SIP/2.0/UDP 192.1.2.17:5060;branch=z9hG4bK5312283c
From: "9727619271" <sip:9727610001 at 192.1.2.17>;tag=as27e35563
To: <sip:9727610001 at 192.1.2.223>
Contact: <sip:9727610001 at 192.1.2.17>
Call-ID: 281e94e27d505ca8376ecf7007cef1d3 at 192.1.2.17
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Content-Length: 0


 to 192.1.2.223:5060
Retransmitting #2 (no NAT):
CANCEL sip:9727610001 at 192.1.2.223 SIP/2.0
Via: SIP/2.0/UDP 192.1.2.17:5060;branch=z9hG4bK5312283c
From: "9727619271" <sip:9727610001 at 192.1.2.17>;tag=as27e35563
To: <sip:9727610001 at 192.1.2.223>
Contact: <sip:9727610001 at 192.1.2.17>
Call-ID: 281e94e27d505ca8376ecf7007cef1d3 at 192.1.2.17
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Content-Length: 0


 to 192.1.2.223:5060
Retransmitting #3 (no NAT):
CANCEL sip:9727610001 at 192.1.2.223 SIP/2.0
Via: SIP/2.0/UDP 192.1.2.17:5060;branch=z9hG4bK5312283c
From: "9727619271" <sip:9727610001 at 192.1.2.17>;tag=as27e35563
To: <sip:9727610001 at 192.1.2.223>
Contact: <sip:9727610001 at 192.1.2.17>
Call-ID: 281e94e27d505ca8376ecf7007cef1d3 at 192.1.2.17
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Content-Length: 0


 to 192.1.2.223:5060
Retransmitting #4 (no NAT):
CANCEL sip:9727610001 at 192.1.2.223 SIP/2.0
Via: SIP/2.0/UDP 192.1.2.17:5060;branch=z9hG4bK5312283c
From: "9727619271" <sip:9727610001 at 192.1.2.17>;tag=as27e35563
To: <sip:9727610001 at 192.1.2.223>
Contact: <sip:9727610001 at 192.1.2.17>
Call-ID: 281e94e27d505ca8376ecf7007cef1d3 at 192.1.2.17
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Content-Length: 0


 to 192.1.2.223:5060
Retransmitting #5 (no NAT):
CANCEL sip:9727610001 at 192.1.2.223 SIP/2.0
Via: SIP/2.0/UDP 192.1.2.17:5060;branch=z9hG4bK5312283c
From: "9727619271" <sip:9727610001 at 192.1.2.17>;tag=as27e35563
To: <sip:9727610001 at 192.1.2.223>
Contact: <sip:9727610001 at 192.1.2.17>
Call-ID: 281e94e27d505ca8376ecf7007cef1d3 at 192.1.2.17
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Content-Length: 0


 to 192.1.2.223:5060
WARNING[40966]: File chan_sip.c, Line 388 (retrans_pkt): Maximum retries 
exceede
d on call 281e94e27d505ca8376ecf7007cef1d3 at 192.1.2.17 for seqno 102 
(Request)
    -- Timeout on SIP/9727619271-4585
    -- Executing Goto("SIP/9727619271-4585", "#|1") in new stack
    -- Goto (sip,#,1)
    -- Executing Playback("SIP/9727619271-4585", "demo-thanks") in new stack
We're at 192.1.2.17 port 14292
Answering with capability 4
Reliably Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.1.2.88:5060;branch=z9hG4bK222444590
From: 9727619271 <sip:9727619271 at 192.1.2.88>;tag=222444590
To: 9727610001 <sip:9727610001 at 192.1.2.17>;tag=as491fadd2
Call-ID: 222444590 at 192.1.2.88
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Contact: <sip:9727610001 at 192.1.2.17>
Content-Type: application/sdp
Content-Length: 129

v=0
o=root 10628 10628 IN IP4 192.1.2.17
s=session
c=IN IP4 192.1.2.17
t=0 0
m=audio 14292 RTP/AVP 0
a=rtpmap:0 PCMU/8000

 to 192.1.2.88:5060
    -- Playing 'demo-thanks'
Sip read:
ACK sip:9727610001 at 192.1.2.17 SIP/2.0
Via: SIP/2.0/UDP 192.1.2.88:5060;branch=z9hG4bK222444590
Max-Forwards: 70
From: 9727619271 <sip:9727619271 at 192.1.2.88>;tag=222444590
To: 9727610001 <sip:9727610001 at 192.1.2.17>;tag=as491fadd2
Call-ID: 222444590 at 192.1.2.88
CSeq: 1 ACK
Content-Length: 0


8 headers, 0 lines
    -- Executing Hangup("SIP/9727619271-4585", "") in new stack
  == Spawn extension (sip, #, 2) exited non-zero on 'SIP/9727619271-4585'
set_destination: Parsing <sip:9727619271 at 192.1.2.88> for address/port to 
send to
set_destination: set destination to 192.1.2.88, port 5060
Reliably Transmitting:
BYE sip:9727619271 at 192.1.2.88 SIP/2.0
Via: SIP/2.0/UDP 192.1.2.17:5060;branch=z9hG4bK77fb44b4
From: 9727610001 <sip:9727610001 at 192.1.2.17>;tag=as491fadd2
To: 9727619271 <sip:9727619271 at 192.1.2.88>;tag=222444590
Contact: <sip:9727610001 at 192.1.2.17>
Call-ID: 222444590 at 192.1.2.88
CSeq: 102 BYE
User-Agent: Asterisk PBX
Content-Length: 0

 (no NAT) to 192.1.2.88:5060
Retransmitting #1 (no NAT):
BYE sip:9727619271 at 192.1.2.88 SIP/2.0
Via: SIP/2.0/UDP 192.1.2.17:5060;branch=z9hG4bK77fb44b4
From: 9727610001 <sip:9727610001 at 192.1.2.17>;tag=as491fadd2
To: 9727619271 <sip:9727619271 at 192.1.2.88>;tag=222444590
Contact: <sip:9727610001 at 192.1.2.17>
Call-ID: 222444590 at 192.1.2.88
CSeq: 102 BYE
User-Agent: Asterisk PBX
Content-Length: 0

gth: 1©
 to 192.1.2.88:5060
Retransmitting #2 (no NAT):
BYE sip:9727619271 at 192.1.2.88 SIP/2.0
Via: SIP/2.0/UDP 192.1.2.17:5060;branch=z9hG4bK77fb44b4
From: 9727610001 <sip:9727610001 at 192.1.2.17>;tag=as491fadd2
To: 9727619271 <sip:9727619271 at 192.1.2.88>;tag=222444590
Contact: <sip:9727610001 at 192.1.2.17>
Call-ID: 222444590 at 192.1.2.88
CSeq: 102 BYE
User-Agent: Asterisk PBX
Content-Length: 0

gth: 1
 to 192.1.2.88:5060
Retransmitting #3 (no NAT):
BYE sip:9727619271 at 192.1.2.88 SIP/2.0
Via: SIP/2.0/UDP 192.1.2.17:5060;branch=z9hG4bK77fb44b4
From: 9727610001 <sip:9727610001 at 192.1.2.17>;tag=as491fadd2
To: 9727619271 <sip:9727619271 at 192.1.2.88>;tag=222444590
Contact: <sip:9727610001 at 192.1.2.17>
Call-ID: 222444590 at 192.1.2.88
CSeq: 102 BYE
User-Agent: Asterisk PBX
Content-Length: 0

gth: 1
 to 192.1.2.88:5060
Retransmitting #4 (no NAT):
BYE sip:9727619271 at 192.1.2.88 SIP/2.0
Via: SIP/2.0/UDP 192.1.2.17:5060;branch=z9hG4bK77fb44b4
From: 9727610001 <sip:9727610001 at 192.1.2.17>;tag=as491fadd2
To: 9727619271 <sip:9727619271 at 192.1.2.88>;tag=222444590
Contact: <sip:9727610001 at 192.1.2.17>
Call-ID: 222444590 at 192.1.2.88
CSeq: 102 BYE
User-Agent: Asterisk PBX
Content-Length: 0

gth: 1
 to 192.1.2.88:5060
Retransmitting #5 (no NAT):
BYE sip:9727619271 at 192.1.2.88 SIP/2.0
Via: SIP/2.0/UDP 192.1.2.17:5060;branch=z9hG4bK77fb44b4
From: 9727610001 <sip:9727610001 at 192.1.2.17>;tag=as491fadd2
To: 9727619271 <sip:9727619271 at 192.1.2.88>;tag=222444590
Contact: <sip:9727610001 at 192.1.2.17>
Call-ID: 222444590 at 192.1.2.88
CSeq: 102 BYE
User-Agent: Asterisk PBX
Content-Length: 0

gth: 1
 to 192.1.2.88:5060
WARNING[40966]: File chan_sip.c, Line 388 (retrans_pkt): Maximum retries 
exceede
d on call 222444590 at 192.1.2.88 for seqno 102 (Request)
Sip read:
BYE sip:9727610001 at 192.1.2.17 SIP/2.0
Via: SIP/2.0/UDP 192.1.2.88:5060;branch=z9hG4bK222444590
Max-Forwards: 70
From: 9727619271 <sip:9727619271 at 192.1.2.88>;tag=222444590
To: 9727610001 <sip:9727610001 at 192.1.2.17>;tag=as491fadd2
Call-ID: 222444590 at 192.1.2.88
CSeq: 1 BYE
Contact: <sip:9727619271 at 192.1.2.88>
Content-Length: 0


9 headers, 0 lines
Interface is eth0
IP Address is 192.1.2.17
Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.1.2.88:5060;branch=z9hG4bK222444590
From: 9727619271 <sip:9727619271 at 192.1.2.88>;tag=222444590
To: 9727610001 <sip:9727610001 at 192.1.2.17>;tag=as491fadd2
Call-ID: 222444590 at 192.1.2.88
CSeq: 1 BYE
User-Agent: Asterisk PBX
Contact:
Content-Length: 0


 to 192.1.2.88:5060

-- end asterix sip debug trace

More information about the asterisk-dev mailing list