[Asterisk-Users] Re: Asterisk as SIP Redirect Server -- Implemented - Not Working - Plz Help

[Kannaiyan Natesan]

/O

   Attached is the Debug information with the 300 Redirect implementation
with asterisk,

   You can get the source code from

    http://www.speak2world.com/asterisk/chan_sip.php


and when you compile and run it, you get the following info in the debug
o/p.


pbx*CLI> sip debug
SIP Debugging Enabled
pbx*CLI>

Sip read:
INVITE sip:rama at 69.15.152.35 SIP/2.0
Max-Forwards: 10
Record-Route: <sip:8138799 at 195.37.77.101;ftag=3758033922;lr=on>
Via: SIP/2.0/UDP 195.37.77.101;branch=z9hG4bKe65c.9f9e1c54.1
Via: SIP/2.0/UDP 81.86.234.10:5060;branch=129db1b09b88fc97f8aedf0767db7b61.0
Session-Expires: 3600
From: <sip:88138730 at iptel.org;user=phone>;tag=3758033922
To: <sip:8138799 at iptel.org;user=phone>
Call-ID: 995233702 at 192.168.0.2
CSeq: 2 INVITE
Contact:
<sip:sip%3a88138730%40192.168.0.2%3a5060%3buser=phone%3btransport=udp at 81.86.
234.10>
User-Agent: Cisco ATA 186  v3.0.0 atasip (031210A)
Proxy-Authorization: Digest
username="88138730",realm="iptel.org",nonce="400b262cb074761ee4458911df8b554
c31618ffc",uri="sip:8138799 at iptel.org",response="751a3ae3f06961159af4178d6d9
43402"
Expires: 300
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER
Content-Type: application/sdp
Content-Length: 254
P-hint: USRLOC

v=0
o=8813730 1818418 1818418 IN IP4 81.86.234.10
s=ATA186 Call
c=IN IP4 81.86.234.10
t=0 0
m=audio 35566 RTP/AVP 0 4 8 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:4 G723/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

18 headers, 11 lines
Transmitting (NAT):
SIP/2.0 300 Redirecting
Via: SIP/2.0/UDP
195.37.77.101;branch=z9hG4bKe65c.9f9e1c54.1;received=195.37.77.101
Via: SIP/2.0/UDP 81.86.234.10:5060;branch=129db1b09b88fc97f8aedf0767db7b61.0
From: <sip:88138730 at iptel.org;user=phone>;tag=3758033922
To: <sip:8138799 at iptel.org;user=phone>;tag=as6ad15254
Call-ID: 995233702 at 192.168.0.2
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:14551 at fwd.pulver.com>
Contact: <137830>
Content-Length: 0


 to 195.37.77.101:5060
pbx*CLI>

Sip read:
ACK sip:rama at 69.15.152.35 SIP/2.0
Via: SIP/2.0/UDP 195.37.77.101;branch=z9hG4bKe65c.9f9e1c54.1
From: <sip:88138730 at iptel.org;user=phone>;tag=3758033922
Call-ID: 995233702 at 192.168.0.2
To: <sip:8138799 at iptel.org;user=phone>;tag=as6ad15254
CSeq: 2 ACK
User-Agent: Sip EXpress router(0.8.12-tcp_nonb-tls (i386/linux))
Content-Length: 0

Kindly let me know if you need more information.
I have the confidence that I can make asterisk even as a very good SIP Proxy
might be like SER.

Kannaiyan