Hi all,
         i am doing my master thesis on server perfromance evaluation i am
using asterisk as sip proxy server and sipp tool as traffic generator...

i have run basic testing of asterisk like as shown in website
http://sipp.sourceforge.net/wiki/index.php/Howto_test_an_Asterisk_server_using_SIPp


when i have copied sip.conf and extensions.conf from the site and run the
client and server i am getting error like this

NOTICE[2715]: chan_sip.c:20314 handle_request_invite: Call from '' to
extension 'service' rejected because extension not found in context
'default'

i dont know y this is coming its really troubling me a
lot...................................





please any one send me some xml, dial plan and sip.conf files for
registering and for inviting. I have been trying for this a lot if any one
help me i would be more thankful .....



BR
viswavardhanreddy
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
               http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Reply via email to