[Serusers] Can SER 2.0.0-rc1 pstn.cfg file actually work with BYE from PSTN?

Mon Sep 28 20:35:01 CEST 2009

Greger Viken Teigre wrote:
> Ok, Frank. You have told me that you don't want me to respond,

Actually I was hoping to nudge people who knew the answer to actually
provide the answer.  The last dozen or so questions asked here over
the past several months have pretty-much received no useful answers,
although the one where I provided a code fix saw the change quietly
introduced into a new release of that piece of software with no comment.

It seems that when I gave lots of detail, someone was bound to say I
needed to simplify the question because they didn't have time to
read it all, but if I reduced it to an A, B, C case, someone would
say they needed every SIP message and the .cfg file.
Can't seem to win, so I elected to provide what was requested.  :-)

Anyway, here are the pieces you asked about.  If you need more,
just ask. I realize how much you are getting paid for your assistance
(nearly some).   

If the PSTN switch really is making a mistake in this scenario
(the "BYE" seems okay and because of the presence of the Record-Route
it is only sent to the Record-Route address), please indicate
where it violates RFC so I can start the long process of getting
them to investigate and with luck I would have a fix in a year...  :-(

The original INVITE, as sent from a test Asterisk box.
IP addresses and phone numbers have been globally changed to
protect the innocent.

Our players in this sample:
66.77.88.99 Asterisk box originating the call
206.33.44.55 SER box in the middle
10.100.20.30 PSTN gateway (SIP is really handled at a
10.x.x.x address, all others public)
The SIP Phone placing the call is 3796661000
Someone out on the PSTN is the phone 9615551212 that
answers the call and goes on hook first, so the
PSTN switch receives a SS7 REL message and causes
it to emit the BYE back to SER.

U 2009/09/28 03:47:10.060467 66.77.88.99:5060 -> 206.33.44.55:5060
INVITE sip:9615551212 at 206.33.44.55 SIP/2.0
Via: SIP/2.0/UDP 66.77.88.99:5060;branch=z9hG4bK2a748d20;rport
From: "TEST PHONE 1000" <sip:3796661000 at 66.77.88.99>;tag=as6e0e68d0
To: <sip:9615551212 at 206.33.44.55>
Contact: <sip:3796661000 at 66.77.88.99>
Call-ID: 106cc02230e60e47223b9e643eee9f37 at 66.77.88.99
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 28 Sep 2009 03:47:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 415

v=0
o=root 45142 45142 IN IP4 66.77.88.99
s=session
c=IN IP4 66.77.88.99
t=0 0
m=audio 10918 RTP/AVP 0 3 8 112 5 10 7 97 111
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:111 G726-32/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

Here is the INVITE that SER sent on to the PSTN gateway:

U 2009/09/28 03:47:10.061905 206.33.44.55:5060 -> 10.100.20.30:5060
INVITE sip:9615551212 at 10.100.20.30 SIP/2.0
Record-Route: <sip:206.33.44.55;ftag=as6e0e68d0;lr=on>
Via: SIP/2.0/UDP 206.33.44.55;branch=z9hG4bKd151.5aa15121b87cf2215ad100fd89896056.0
Via: SIP/2.0/UDP 66.77.88.99:5060;branch=z9hG4bK2a748d20;rport=5060
From: "TEST PHONE 1000" <sip:3796661000 at 66.77.88.99>;tag=as6e0e68d0
To: <sip:9615551212 at 206.33.44.55>
Contact: <sip:3796661000 at 206.33.44.55:5060>
Call-ID: 106cc02230e60e47223b9e643eee9f37 at 66.77.88.99
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 16
Date: Mon, 28 Sep 2009 03:47:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 435

v=0
o=root 45142 45142 IN IP4 66.77.88.99
s=session
c=IN IP4 66.77.88.99
t=0 0
m=audio 31768 RTP/AVP 0 3 8 112 5 10 7 97 111
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:111 G726-32/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
a=nortpproxy:yes

Now, here is the BYE coming from the PSTN switch to SER,
which it would be really nice if SER sent on to the calling
asterisk box:

U 2009/09/28 03:47:27.619185 10.100.20.30:5060 -> 206.33.44.55:5060
BYE sip:3796661000 at 206.33.44.55:5060 SIP/2.0
Via: SIP/2.0/UDP 10.100.20.30;branch=z9hG4bK+2403b90cbb5388c3a1fbf676690bb105+000a0283+1
Max-Forwards: 70
Call-ID: 106cc02230e60e47223b9e643eee9f37 at 66.77.88.99
From: <sip:9615551212 at 206.33.44.55>;tag=000a0283+1+4906002d+2fd2af7a
To: "TEST PHONE 1000" <sip:3796661000 at 66.77.88.99>;tag=as6e0e68d0
CSeq: 969167137 BYE
Route: <sip:206.33.44.55;ftag=as6e0e68d0;lr=on>
Supported: timer
Content-Length: 0

and here is the BYE that SER sends right back to the PSTN
switch, which wonders why it was sent this junk:

U 2009/09/28 03:47:27.627511 206.33.44.55:5060 -> 10.100.20.30:5060
BYE sip:3796661000 at 10.100.20.30:5060 SIP/2.0
Record-Route: <sip:206.33.44.55;ftag=000a0283+1+4906002d+2fd2af7a;lr=on>
Record-Route: <sip:206.33.44.55;ftag=000a0283+1+4906002d+2fd2af7a;lr=on>
Via: SIP/2.0/UDP 206.33.44.55;branch=z9hG4bK9f22.8367f454f9911340deec0901b216f326.0
Via: SIP/2.0/UDP 206.33.44.55;branch=z9hG4bK9f22.55530ed60f502428f904e8008519abec.0
Via: SIP/2.0/UDP 10.100.20.30;branch=z9hG4bK+2403b90cbb5388c3a1fbf676690bb105+000a0283+1
Max-Forwards: 15
Call-ID: 106cc02230e60e47223b9e643eee9f37 at 66.77.88.99
From: <sip:9615551212 at 206.33.44.55>;tag=000a0283+1+4906002d+2fd2af7a
To: "TEST PHONE 1000" <sip:3796661000 at 66.77.88.99>;tag=as6e0e68d0
CSeq: 969167137 BYE
Supported: timer
Content-Length: 0

and of course, the PSTN switch replies "Huh?":

U 2009/09/28 03:47:27.629429 10.100.20.30:5060 -> 206.33.44.55:5060
SIP/2.0 481 Call/Transaction Does Not Exist
Record-Route: <sip:206.33.44.55;ftag=000a0283+1+4906002d+2fd2af7a;lr=on>
Record-Route: <sip:206.33.44.55;ftag=000a0283+1+4906002d+2fd2af7a;lr=on>
Via: SIP/2.0/UDP 206.33.44.55;branch=z9hG4bK9f22.8367f454f9911340deec0901b216f326.0
Via: SIP/2.0/UDP 206.33.44.55;branch=z9hG4bK9f22.55530ed60f502428f904e8008519abec.0
Via: SIP/2.0/UDP 10.100.20.30;branch=z9hG4bK+2403b90cbb5388c3a1fbf676690bb105+000a0283+1
Max-Forwards: 15
Call-ID: 106cc02230e60e47223b9e643eee9f37 at 66.77.88.99
From: <sip:9615551212 at 206.33.44.55>;tag=000a0283+1+4906002d+2fd2af7a
To: "TEST PHONE 1000" <sip:3796661000 at 66.77.88.99>;tag=as6e0e68d0
CSeq: 969167137 BYE
Supported: timer
Content-Length: 0

and SER responds right back with "Double-Huh?"

U 2009/09/28 03:47:27.629558 206.33.44.55:5060 -> 10.100.20.30:5060
SIP/2.0 481 Call/Transaction Does Not Exist
Record-Route: <sip:206.33.44.55;ftag=000a0283+1+4906002d+2fd2af7a;lr=on>
Record-Route: <sip:206.33.44.55;ftag=000a0283+1+4906002d+2fd2af7a;lr=on>
Via: SIP/2.0/UDP 10.100.20.30;branch=z9hG4bK+2403b90cbb5388c3a1fbf676690bb105+000a0283+1
Max-Forwards: 15
Call-ID: 106cc02230e60e47223b9e643eee9f37 at 66.77.88.99
From: <sip:9615551212 at 206.33.44.55>;tag=000a0283+1+4906002d+2fd2af7a
To: "TEST PHONE 1000" <sip:3796661000 at 66.77.88.99>;tag=as6e0e68d0
CSeq: 969167137 BYE
Supported: timer
Content-Length: 0

and so on...

Thanks for looking!