Hi Mahesh,
actually my guess was correct - something used for restoring the old
FROM uri was change. OK, it wasn't the Route parameter, but it is worst
- it is the FROM new URI.
the UAS receives :
From: "Membersevice
Queue"<sip:6462174165@flareon.aptela.com>;tag=10.10.35.100+1+8c020003+dd5437f3
but when generating the BYE, it sends:
To: "Membersevice
Queue"<sip:6462174165@69.25.47.136>;tag=10.10.35.100+1+8c020003+dd5437f3
changing the FROM/TO uris and tags is against RFC3261.
regards,
bogdan
Mahesh Paolini-Subramanya wrote:
Bogdan
I'm sorry - I commited the cardinal cut/paste sin (copying stuff
incorrectly :-( )
Anyhow, I've *correctly* cut/paste the INVITE and BYE headers below,
and it looks to me like the route headers *are* being rewritten correctly.
So, it doesn't look like that was the issue - unless you are referring
to something else?
cheers
--- Original Message -----
From: Bogdan-Andrei Iancu <bogdan(a)voice-system.ro>
To: mahesh(a)aptela.com
Cc: users(a)openser.org
Sent: Thursday, February 8, 2007 3:29:26 AM GMT-0600
Subject: Re: [Users] Corrupted 'To' after uac_restore_from
Hi Mahesh,
I suspect you have a client that does not preserve the
Route/Record-Route parameters. According to RFC326, the parameters must
not be altered at al, but I saw UAC doing lowercase on the param's
value. Most probably this is your case also - the uac module use B64
encoding of the old uri in RR param, so this is case sensitive.
Get the full trace of the call and check the original RR param (inserted
by openser in INVITE) against the one in put in the BYE.
regards,
bogdan
---- CORRECTED MESSAGE-----
I'm seeing this *very* weird error where uac_restore_from is ending up
with a corrupted 'From' header.
The weird thing is that it happens with only one end point (a
Televantage SIP Trunk).
With virtually *everything* else - and I am talking about tens of
thousands of end points, spanning all sorts of sip devices, we are
having *no* problems.
The call is set up as follows -> the endpoint sends the INVITE, which
is forwarded to a PSTN gateway
The ACKs and the 200 OK all have their From/To headers appropriately
rewritten
The problem arises when the PSTN gateway initiates the BYE As you
can see below, the 'To' header gets corrupted
The *only* thing I can think of is that it has something to do with
the structure of the Televantage tags (the '+' signs?).
Any ideas?
FYI, Openser 1.1.0 (latest CVS checkout), and
modparam("uac","from_restore_mode","auto")
modparam("uac","rr_store_param","aptelavsf")
--INVITE-- UAC->proxy
INVITE sip:17038677006@69.25.47.136 SIP/2.0.
Via: SIP/2.0/UDP
10.10.35.100;rport;branch=z9hG4bK+ec6a2a20436621337016546fa4dc9e87+10.10.35.100+1.
Allow-Events: message-summary.
Allow-Events: presence.
Allow-Events: refer.
Max-Forwards: 70.
Call-ID: 56796C1E(a)10.10.35.100.
From: "Member
service"<sip:user100.crunchhq@69.25.47.136>;tag=10.10.35.100+1+8c020003+dd5437f3.
To: <sip:17038677006@69.25.47.136>.
CSeq: 132610708 INVITE.
Expires: 180.
Supported: replaces.
Supported: 100rel.
Contact: <sip:6462174165@10.10.35.100>.
Content-Type: application/sdp.
Content-Length: 242.
.
User-Agent: TeleVantage-UA/7.50.4817.
.
v=0.
o=TeleVantage 10671 0 IN IP4 10.10.35.100.
s=phone-call.
c=IN IP4 10.10.35.100.
t=0 0.
m=audio 6036 RTP/AVP 0 18 101.
a=rtpmap:0 pcmu/8000/1.
a=rtpmap:18 g729/8000/1.
a=fmtp:18 annexb=no.
a=rtpmap:101 telephone-event/8000/1.
a=ptime:20.
---INVITE--- proxy->gateway
INVITE sip:17038677006@65.91.91.16:5060;transport=udp SIP/2.0.
Record-Route:
<sip:69.25.47.136;lr;ftag=10.10.35.100+1+8c020003+dd5437f3;nat=yes;aptelavsf=dGVzdDciIDR0YndrISJGfHtsaWBbPWQ7NiQ4LCJlISgv>.
Via: SIP/2.0/UDP 69.25.47.136;branch=z9hG4bKf67d.2d31dbb.0.
Via: SIP/2.0/UDP
10.10.35.100;rport=5060;branch=z9hG4bK+ec6a2a20436621337016546fa4dc9e87+10.10.35.100+1.
Allow-Events: message-summary.
Allow-Events: presence.
Allow-Events: refer.
Max-Forwards: 69.
Remote-Party-ID: "Membersevice Queue"
<sip:6462174165@flareon.aptela.com>;party=calling;id-type=subscriber;screen=yes;privacy=off.
Supported: replaces.
Call-ID: 56796C1E(a)10.10.35.100.
From: "Membersevice
Queue"<sip:6462174165@flareon.aptela.com>;tag=10.10.35.100+1+8c020003+dd5437f3.
To: <sip:17038677006@69.25.47.136>.
CSeq: 132610708 INVITE.
Expires: 180.
Contact: <sip:6462174165@10.10.35.100:5060>.
Content-Type: application/sdp.
Content-Length: 262.
User-Agent: TeleVantage-UA/7.50.4817.
.
v=0.
o=TeleVantage 10671 0 IN IP4 10.10.35.100.
s=phone-call.
c=IN IP4 66.150.122.23.
t=0 0.
m=audio 59520 RTP/AVP 0 18 101.
a=rtpmap:0 pcmu/8000/1.
a=rtpmap:18 g729/8000/1.
a=fmtp:18 annexb=no.
a=rtpmap:101 telephone-event/8000/1.
a=ptime:20.
a=nortpproxy:yes.
---BYE---gateway->proxy
BYE sip:6462174165@10.10.35.100:5060 SIP/2.0.
Via: SIP/2.0/UDP
66.52.236.25:5060;branch=z9hG4bKd8orpb20eg2hais2c2g1sd0000g00.1.
To: "Membersevice
Queue"<sip:6462174165@69.25.47.136>;tag=10.10.35.100+1+8c020003+dd5437f3.
From: <sip:17038677006@66.52.236.25>;tag=1386496.
Call-ID: 56796C1E(a)10.10.35.100.
CSeq: 1 BYE.
Content-Length: 0.
Route:
<sip:69.25.47.136;lr;ftag=10.10.35.100+1+8c020003+dd5437f3;nat=yes;aptelavsf=dGVzdDciIDR0YndrISJGfHtsaWBbPWQ7NiQ4LCJlISgv>.
Max-Forwards: 70.
---BYE--- proxy->UAC
BYE sip:6462174165@10.10.35.100:5060 SIP/2.0.
Record-Route: <sip:69.25.47.136;lr;ftag=1386496>.
Via: SIP/2.0/UDP 69.25.47.136;branch=z9hG4bK7791.48e0e2f5.0.
Via: SIP/2.0/UDP
66.52.236.25:5060;branch=z9hG4bKd8orpb20eg2hais2c2g1sd0000g00.1.
To: "Membersevice Queue"<sip:user100.cru>6'(!.l
asr}XV.R\[>;tag=10.10.35.100+1+8c020003+dd5437f3.
From: <sip:17038677006@66.52.236.25>;tag=1386496.
Call-ID: 56796C1E(a)10.10.35.100.
CSeq: 1 BYE.
Content-Length: 0.
Max-Forwards: 69.
--
*******************************************
Mahesh Paolini-Subramanya (703) 386-1500 x9100
CTO mahesh(a)aptela.com
Aptela, Inc.
http://www.aptela.com
"Aptela: How Business Answers The Call"
*******************************************