[Asterisk-Users] SIPP with asterisk

[Tulika Pradhan]

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" <tim at timsnet.com>
>Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion 
><asterisk-users at lists.digium.com>
>To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" 
><asterisk-users at lists.digium.com>
>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: asterisk-users-bounces at lists.digium.com
>[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Tulika 
>Pradhan
>Sent: Wednesday, May 04, 2005 11:29 PM
>To: asterisk-users at lists.digium.com
>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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