[Asterisk-Dev] Record-route Issues

[Glenn Dalgliesh]

Could some please confirm that this behavior is incorrect. I am seeing
issues where it appears that asterisk is not following the Record-route on
it's reply messages. Please let me know if you require any other
information.

Thanks

Example:

xxx.yyy.154.243(PSTN-GW) <--sip--> xxx.yyy.77.23(Asterisk) <--sip-->
xxx.yyy.91.74(SNOM or SER proxy) <--sip----> xxx.yyy.165.201(ATA186)

1) Call place from PSTN to xxx9931211
2) Asterisk via rules in extension.conf sends call to xxx.yyy.91.74
3) xxx.yyy.91.74 sends call to xxx.yyy.165.201 which is registered as
xxx9931211
4) Call completes fine and audio works
5) PSTN Hangs up and the sends a BYE to Asterisk
6) Asterisk recieves the bye and sends BYE directly to the UA skipping the
proxy
7) Results is that the proxy never recieves a BYE

Complete SIP Trace
http://www.routerboy.com/sip2004041902.html

Example of actual Record-Route issue:
-Message 9 the Proxy sends a Record-Route to Asterisk and Message 10 seems
to be built are return to the Proxy but not with the infomation from the
Record-Route Statement.)

     SIP MESSAGE 9        xxx.yyy.91.74:5060(4) -> xxx.yyy.77.23:5060(2)
     UDP Frame 9        19/Apr/04 18:17:49.9304
TimeFromPreviousSipFrame=0.3592 TimeFromStart=2.1462
SIP/2.0 200 OK
Via: SIP/2.0/UDP xxx.yyy.77.23:5060;branch=z9hG4bK019f067c
Record-Route: <sip:proxy.abccorp.net:5060;maddr=xxx.yyy.91.74>
From: "xxx3427216" <sip:xxx3427216 at xxx.yyy.77.23>;tag=as5304af8c
To: <sip:xxx9931211 at proxy.abccorp.net>;tag=1841513983
Call-ID: 58d4bd4e5e29ff254db520665915ac83 at xxx.yyy.77.23
CSeq: 102 INVITE
Contact: <sip:xxx9931211 at xxx.yyy.165.201:5060;user=phone;transport=udp>
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER
Server: Cisco ATA 186  v3.0.0 atasip (031210A)
Content-Type: application/sdp
Content-Length: 205

v=0
o=xxx9931211 18172 18172 IN IP4 xxx.yyy.165.201
s=ATA186 Call
c=IN IP4 xxx.yyy.165.201
t=0 0
m=audio 16384 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15


----------------------------------------------------------------------------
----

     SIP MESSAGE 10       xxx.yyy.77.23:5060(2) -> xxx.yyy.91.74:5060(4)
UDP Frame 10       19/Apr/04 18:17:49.9310 TimeFromPreviousSipFrame=0.0006
TimeFromStart=2.1468 ACK sip:xxx9931211 at xxx.yyy.165.201:5060 SIP/2.0 Via:
SIP/2.0/UDP xxx.yyy.77.23:5060;branch=z9hG4bK50c814eb Route:
<sip:xxx9931211 at xxx.yyy.165.201:5060;user=phone;transport=udp> From:
"xxx3427216" <sip:xxx3427216 at xxx.yyy.77.23>;tag=as5304af8c To:
<sip:xxx9931211 at proxy.abccorp.net>;tag=1841513983 Contact:
<sip:xxx3427216 at xxx.yyy.77.23> Call-ID:
58d4bd4e5e29ff254db520665915ac83 at xxx.yyy.77.23 CSeq: 102 ACK User-Agent:
Asterisk PBX Content-Length: 0