Hi everyone, I'm trying to get sipp communicate with Asterisk in order to perform performance tests:
I've been through these steps: 1) In sip.conf [sippuac] type=friend username=sippuac host=192.168.101.9 port=5060 context=test dtmfmode=rfc2833 insecure=very canreinvite=no nat=yes [sippuas] type=friend username=sippuas host=192.168.101.9 port=5061 context=test dtmfmode=rfc2833 insecure=very canreinvite=no nat=yes 2) In extensions.conf [test] exten=>s,1,Dial(SIP/sippuas,20) 3) Running SIPp sipp -sn uas -rsa 127.0.0.1:5060 -p 5062 -i 127.0.0.1 -mp 6001 sipp -sn uac 127.0.0.1:5060 -s s -p 5061 -i 127.0.0.1 Finally I get on Asterisk : [Jun 14 07:36:56] WARNING[2600][C-00000120]: app_dial.c:2437 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent) How can I solve this and make the UAS receive the calls ? Thanks for your help ! Regards, Hssan ------------------------------------------------------------------------------ What NetFlow Analyzer can do for you? Monitors network bandwidth and traffic patterns at an interface-level. Reveals which users, apps, and protocols are consuming the most bandwidth. Provides multi-vendor support for NetFlow, J-Flow, sFlow and other flows. Make informed decisions using capacity planning reports. https://ad.doubleclick.net/ddm/clk/305295220;132659582;e _______________________________________________ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
