[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:[EMAIL PROTECTED]/redirectconfig

[redirectconfig]
redirect=yes
redirecturi=sip:[EMAIL PROTECTED]
redirecturi=sip:[EMAIL PROTECTED]
redirecturi=sip:[EMAIL PROTECTED] 

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

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

[Olle E. Johansson]

Could you please explain what you want to do, why you want asterisk to register but 
not take
the calls?
You could take the calls into the dialplan (extensions.conf) and dial out from there 
with an agi
script that performed the same thing. If you have canreinvite=yes, asterisk will leave 
the media
back to the clients. It's not a redirect, but it's more Asterisk.
Asterisk SIP channel is not designed as a SIP proxy so I believe this is hard to do. 
From the
code it looks like a good attempt, but since you didn't include any debug output I 
don't know
what did not work for you. Please send SIP DEBUG output.
/O

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