calls get setup with the config below but in the stats all calls are failed. BYE are sent from uac to asterisk but after no more than 1 response is receieved. All calls show to be failed in stats printed by sipp (uac).
isn't the uas part of sipp supposed to received 200 messages after BYE has been sent. That is not coming into picture.


From: "Tim Connolly" <[EMAIL PROTECTED]>
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion <[email protected]>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" <[email protected]>
Subject: RE: [Asterisk-Users] SIPP with asterisk
Date: Thu, 5 May 2005 00:41:25 -0500


How about:
exten => 9111222,1,answer
exten => 9111222,2,wait(10)
exten => 9111222,3,Hangup



-----Original Message-----
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tulika Pradhan
Sent: Wednesday, May 04, 2005 11:29 PM
To: [email protected]
Subject: [Asterisk-Users] SIPP with asterisk


i am trying to do load testing on asterisk using sipp testing tool. i am
able to send invite requests to asterisk by using
     sipp -sn uac <ip address> -s 9111222 -d 10000 -r 10
i am also running
     sipp -sn uas
on the same box
but no message arrives on uas part. and asterisk returns error while
dialing.
     "Unable to create channel of type SIP"
the extensions.conf has
exten => 9111222,1,Dial(SIP/9111222)
exten => 9111222,2,Hangup

what config changes should i make to answer the calls landing on asterisk ?

tulika

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