[Asterisk-Users] Debug for SIP and reINVITES (ATA-186)

[John Todd]

I must be doing something incorrectly, or something is wrong with 
ATA-186 reINVITEs in SIP.  Perhaps someone more enlightened than me 
can lend me a hand.

I have been attempting to get two SIP phones to reINVITE to each 
other, and I've been unable to think of or uncover the correct 
method.  The calls always go through the Asterisk server, no matter 
what I try.  I've simplified things down to almost zero, but the 
audio insists on going through Asterisk instead of directly between 
the phones, which I thought was what the reinvite=yes setting 
allowed.  I seem to recall getting this to work this in the past, but 
cannot seem to make things work now.

Dialing from 2206 to 2208 doesn't make the RTP session end up in the 
"right" place.   The setup is two ATA-186 boxes, on the same 
ethernet, with the Asterisk server also on the same ethernet.  Both 
ATA-186 boxes are pretty "stock" except for the settings to make them 
work via NAT (the v.2.15 box has ConnectMode set to 0x00460400, while 
the v2.16 box doesn't need, and in fact will malfunction with that 
setting.  Go, Go, Cisco Standardization!)  Despite the setting of the 
ConnectMode on one of the boxes, neither box is behind NAT - all 
three addresses are "real" and on the same subnet with each other.

Clue: look at the very end of the packet dump, after I hang up 2208. 
There is an INVITE that happens after I hang up the phones!  That 
doesn't look right.  It's almost as if the reINVITE is happening 
after the BYE is sent.

; 2206 is an ATA-186 v.2.15 with no password set
[2206]
type=friend
username=2206
host=dynamic
context=foo
canreinvite=yes


; 2208 is an ATA-186 with v.2.16
[2208]
type=friend
username=2208
secret=nopasswordhere
host=dynamic
context=foo
canreinvite=yes




  -- extensions.conf ---

[foo]
exten => 2208,1,AbsoluteTimeout(9995)
exten => 2208,2,Dial(SIP/2208)
exten => 2208,3,Hangup



  -- debug output from tethereal --

202.22.13.7  = extension 2206
202.22.13.3  = extension 2208
202.22.13.10 = asterisk server



308.484146 202.22.13.7 -> 202.22.13.10 SIP/SDP Request: INVITE 
sip:2208 at 202.22.13.10;user=phone, with session description
308.485628 202.22.13.10 -> 202.22.13.7 SIP Status: 100 Trying
308.487933 202.22.13.10 -> 202.22.13.3 SIP/SDP Request: INVITE 
sip:2208 at 202.22.13.3, with session description
308.489363 202.22.13.1 -> 202.22.13.10 ICMP Redirect
308.500403 202.22.13.3 -> 202.22.13.10 SIP Status: 100 Trying
308.502071 202.22.13.3 -> 202.22.13.1 NTP NTP
308.506036 202.22.13.1 -> 202.22.13.3 NTP NTP
308.526069 202.22.13.3 -> 202.22.13.10 SIP Status: 180 Ringing
308.527027 202.22.13.10 -> 202.22.13.7 SIP Status: 180 Ringing
[I pick up extension 2208]
309.456148 202.22.13.3 -> 202.22.13.10 SIP/SDP Status: 200 OK, with 
session description
309.456985 202.22.13.10 -> 202.22.13.3 SIP Request: ACK sip:2208 at 202.22.13.3
309.457906 202.22.13.10 -> 202.22.13.7 SIP/SDP Status: 200 OK, with 
session description
309.460650 202.22.13.1 -> 202.22.13.10 ICMP Redirect
309.465647 202.22.13.7 -> 202.22.13.10 SIP Request: ACK sip:2208 at 202.22.13.10
309.479795 202.22.13.7 -> 202.22.13.10 UDP Source port: 16384 
Destination port: 30128
309.485033 202.22.13.3 -> 202.22.13.10 UDP Source port: 16384 
Destination port: 12886
[lots more of the same line, over and over again]
309.985277 202.22.13.3 -> 202.22.13.10 UDP Source port: 16384 
Destination port: 12886
310.000075 202.22.13.7 -> 202.22.13.10 UDP Source port: 16384 
Destination port: 30128
310.005275 202.22.13.3 -> 202.22.13.10 UDP Source port: 16384 
Destination port: 12886
310.005824 202.22.13.10 -> 202.22.13.7 SIP/SDP Request: INVITE 
sip:2206 at 202.22.13.10, with session description
310.006086 202.22.13.10 -> 202.22.13.3 SIP/SDP Request: INVITE 
sip:2208 at 202.22.13.3, with session description
310.006629 202.22.13.10 -> 202.22.13.3 UDP Source port: 12886 
Destination port: 16384
310.007177 202.22.13.10 -> 202.22.13.7 UDP Source port: 30128 
Destination port: 16384
310.007264 202.22.13.10 -> 202.22.13.3 UDP Source port: 12886 
Destination port: 16384
310.007337 202.22.13.10 -> 202.22.13.7 UDP Source port: 30128 
Destination port: 16384
310.009548 202.22.13.1 -> 202.22.13.10 ICMP Redirect
310.020540 202.22.13.7 -> 202.22.13.10 SIP/SDP Status: 200 OK, with 
session description
310.021024 202.22.13.7 -> 202.22.13.10 UDP Source port: 16384 
Destination port: 30128
310.021632 202.22.13.10 -> 202.22.13.7 SIP Request: ACK sip:2206 at 202.22.13.10
310.022079 202.22.13.10 -> 202.22.13.3 UDP Source port: 12886 
Destination port: 16384
310.025775 202.22.13.3 -> 202.22.13.10 UDP Source port: 16384 
Destination port: 12886
310.025916 202.22.13.10 -> 202.22.13.7 UDP Source port: 30128 
Destination port: 16384
310.028582 202.22.13.3 -> 202.22.13.10 SIP/SDP Status: 200 OK, with 
session description
310.029514 202.22.13.10 -> 202.22.13.3 SIP Request: ACK sip:2208 at 202.22.13.3
310.040127 202.22.13.7 -> 202.22.13.10 UDP Source port: 16384 
Destination port: 30128
310.040257 202.22.13.10 -> 202.22.13.3 UDP Source port: 12886 
Destination port: 16384
310.046028 202.22.13.3 -> 202.22.13.10 UDP Source port: 16384 
Destination port: 12886
[lots more of the same lines, over and over again - this is RTP]
311.726517 202.22.13.3 -> 202.22.13.10 UDP Source port: 16384 
Destination port: 12886
311.726636 202.22.13.10 -> 202.22.13.7 UDP Source port: 30128 
Destination port: 16384
311.741810 202.22.13.7 -> 202.22.13.10 UDP Source port: 16384 
Destination port: 30128
311.741932 202.22.13.10 -> 202.22.13.3 UDP Source port: 12886 
Destination port: 16384
[and now, I hang up extension 2208...]
311.747604 202.22.13.3 -> 202.22.13.10 SIP Request: BYE sip:2206 at 202.22.13.10
311.747920 202.22.13.10 -> 202.22.13.3 SIP Status: 200 OK
[HEY!  What's this INVITE doing here?!?]
311.748662 202.22.13.10 -> 202.22.13.7 SIP/SDP Request: INVITE 
sip:2206 at 202.22.13.10, with session description
311.751322 202.22.13.1 -> 202.22.13.10 ICMP Redirect
311.759873 202.22.13.7 -> 202.22.13.10 SIP/SDP Status: 200 OK, with 
session description
311.760910 202.22.13.10 -> 202.22.13.7 SIP Request: ACK sip:2206 at 202.22.13.10
311.761034 202.22.13.10 -> 202.22.13.7 SIP Request: BYE sip:2206 at 202.22.13.10
311.762659 202.22.13.7 -> 202.22.13.10 UDP Source port: 16384 
Destination port: 30128
311.776319 202.22.13.7 -> 202.22.13.10 UDP Source port: 16384 
Destination port: 30128
311.784068 202.22.13.7 -> 202.22.13.10 SIP Status: 200 OK