asterisk users - Apr 2003 - Bug? * not correctly honouring tag on To?

Hi Mark,

Current CVS, * isn't correctly remembering the tag added to the To header
by a server.

For instance:

Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 81.96.69.210:5062;branch=7b858c09
From: steve-ata186 <sip:asterisk@81.96.69.210:5062>;tag=14925711
To: <sip:18478974611@4.42.235.170>;tag=t2907cab0911c8g
Call-ID: 232752ec518d398f25ce03f45a8940f3@81.96.69.210
CSeq: 102 INVITE
Allow: INVITE, CANCEL, REFER, BYE, ACK
Contact: <sip:0403242876@195.217.255.36:5061>
Content-Type: application/sdp
Record-Route: <sip:4.42.235.170:5060;lr>
Server: DTA SIP/0.11.7 NNOS/VR30
Content-Length: 144

v=0
o=0403242876 0 2 IN IP4 195.217.255.36
s=-
c=IN IP4 4.42.235.170
t=0 0
m=audio 16082 RTP/AVP 8 101
a=rtpmap:101 telephone-event/8000


Transmitting:
ACK sip:18478974611@4.42.235.170 SIP/2.0
Via: SIP/2.0/UDP 81.96.69.210:5062;branch=7b858c09
Route: <sip:0403242876@195.217.255.36:5061>
From: "steve-ata186"
<sip:asterisk@81.96.69.210:5062>;tag=14925711
To: <sip:18478974611@4.42.235.170>;tag=14925711
Contact: <sip:asterisk@81.96.69.210:5062>
Call-ID: 232752ec518d398f25ce03f45a8940f3@81.96.69.210
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0


Notice that the tag in the ACK's To doesn't match that set by the server
in the 200 OK.

Steve

Can you find me on IRC?  I think I just committed a fix but I'm not sure
I've done it right.  Also, I am not convinced of my own logic in respprep
for handlign the tag.

irc.freenode.net in #asterisk, i'm "kram"

Mark

On Sun, 6 Apr 2003, Stephen Davies wrote:
> Hi Mark,
>
> Current CVS, * isn't correctly remembering the tag added to the To
header
> by a server.
>
> For instance:
>
> Sip read:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 81.96.69.210:5062;branch=7b858c09
> From: steve-ata186 <sip:asterisk@81.96.69.210:5062>;tag=14925711
> To: <sip:18478974611@4.42.235.170>;tag=t2907cab0911c8g
> Call-ID: 232752ec518d398f25ce03f45a8940f3@81.96.69.210
> CSeq: 102 INVITE
> Allow: INVITE, CANCEL, REFER, BYE, ACK
> Contact: <sip:0403242876@195.217.255.36:5061>
> Content-Type: application/sdp
> Record-Route: <sip:4.42.235.170:5060;lr>
> Server: DTA SIP/0.11.7 NNOS/VR30
> Content-Length: 144
>
> v=0
> o=0403242876 0 2 IN IP4 195.217.255.36
> s=-
> c=IN IP4 4.42.235.170
> t=0 0
> m=audio 16082 RTP/AVP 8 101
> a=rtpmap:101 telephone-event/8000
>
>
> Transmitting:
> ACK sip:18478974611@4.42.235.170 SIP/2.0
> Via: SIP/2.0/UDP 81.96.69.210:5062;branch=7b858c09
> Route: <sip:0403242876@195.217.255.36:5061>
> From: "steve-ata186"
<sip:asterisk@81.96.69.210:5062>;tag=14925711
> To: <sip:18478974611@4.42.235.170>;tag=14925711
> Contact: <sip:asterisk@81.96.69.210:5062>
> Call-ID: 232752ec518d398f25ce03f45a8940f3@81.96.69.210
> CSeq: 102 ACK
> User-Agent: Asterisk PBX
> Content-Length: 0
>
>
> Notice that the tag in the ACK's To doesn't match that set by the
server
> in the 200 OK.
>
> Steve
>
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