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

[Kannaiyan Natesan]

I have coded chan_sip.c so that you can have

// sip.conf

register =>  username:password at somedomain.com/redirectconfig

[redirectconfig]
redirect=yes
redirecturi=sip:12345 at domain1.com
redirecturi=sip:34556 at domain2.com
redirecturi=sip:87877 at domain3.com ....

so when you receive a call it will redirect to the alternating uri's with a
SIP 300 Message.

It works with the following sequence,

INVITE  -- Receives INVITE
REDIRECT  -- Sends 300 Successfully
ACK -- Receives ACK

But the actuall call is not redirected.
Can anyone please help what is problem with the SIP redirection message and
anyhelp to test this functionality please.

You can download the source code from
http://www.speak2world.com/asterisk/chan_sip.php
Here is the procedure to compile and run it.

1. cd to /usr/src/asterisk/channels/
2. Backup your existing chan_sip.c
3. replace the chan_sip.c with the current one
4. Type, "make install"


when you receive a call, it should now pass the SIP 300 message to the
caller which you can see with sip debug.

Can anyone please help me, what could be the problem.

Thanks in advance.

Kannaiyan