[asterisk-users] Why Asterisk doesn't accept sip302 redirect?

[Vitaly]

Thanks for your answer, see details below:

U 10.10.10.10.67:5060 -> 10.10.10.107:5060
  INVITE sip:12345678 at 10.10.10.107 SIP/2.0..v:
SIP/2.0/UDP
10.10.10.67:5060;branch=z9hG4bK0264a8da;rport..f:
"2519494"
   <sip:2519494 at 10.10.10.67>;tag=as1d5e5664..t:
<sip:12124441782 at 10.10.10.107>..m:
<sip:2519494 at 10.10.10.67>..i: 503f1f3a
  320241cd2af48cba4147ae73 at 10.10.10.67..CSeq: 102
INVITE..User-Agent: Asterisk PBX..Max-Forwards:
70..Date: Wed, 10 Oct 2
  007 10:01:31 GMT..Allow: INVITE, ACK, CANCEL,
OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..c:
application/sdp..l: 259....v=0
  ..o=root 2423 2423 IN IP4
10.10.10.67..s=session..c=IN IP4 10.10.10.67..t=0
0..m=audio 17250 RTP/AVP 18 4 101..a=rtpmap
  :18 G729/8000..a=fmtp:18 annexb=no..a=rtpmap:4
G723/8000..a=rtpmap:101
telephone-event/8000..a=fmtp:101 0-16..a=silence
  Supp:off - - - -..
#
U 10.10.10.107:5060 -> 10.10.10.67:5060
  SIP/2.0 302 Redirect..Contact:
<sip:12345678 at 10.10.10.100:11060>..v: SIP/2.0/UDP
10.10.10.67:5060;branch=z9hG4bK0264
  a8da;rport..CSeq: 102 INVITE..Content-Length: 0....

Master.csv:

"","2519494","001112345678","to-sip","2519494","SIP/10.10.10.66-09e0a8b0","SIP/out4-09e15578","Dial","SIP/12345678
@out4","2007-10-10 15:01:31",,"2007-10-10
15:02:01",30,0,"NO ANSWER","DOCUMENTATION"


--- Alex Balashov <abalashov at evaristesys.com> wrote:

> 
> Vitaly,
> 
> Can you provide details of what is going on in the
> packet capture exactly?
> What is the Contact: URI that the peer provides in
> the 302 Moved response?
> What does Asterisk do subsequently?
> 
> Cheers,
> 
> -- Alex
> 
> On Wed, 10 Oct 2007, Vitaly wrote:
> 
> > My asterisk should follow 302 redirect which it
> > receives from other sip server(10.10.10.10). By
> > running network sniffer I see, that asterisk
> receives
> > 302 answer, but doesn't follow it.
> > My config is:
> >
> > sip.conf:
> > .......
> > [out4]
> > type=peer
> > host=10.10.10.10
> > canreinvite=no
> > promiscredir=yes
> > insecure=very
> > disallow=all
> > allow=g729
> > allow=g723
> > .......
> >
> > extensions.conf:
> > [to-sip]
> > exten => _0011X., 1, Dial(SIP/${EXTEN:4}@out4)
> > exten => _0011X., 2, Hangup()
> >
> >
> > Any ideas?
> > Vitaly
> >
> >
> >
> >
> >
> >
> >
>
____________________________________________________________________________________
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> >
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> 
> --
> Alex Balashov
> Evariste Systems
> Web    : http://www.evaristesys.com/
> Tel    : +1-678-954-0670
> Direct : +1-678-954-0671
> 
> _______________________________________________
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____________________________________________________________________________________
Be a better Heartthrob. Get better relationship answers from someone who knows. Yahoo! Answers - Check it out. 
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