A little correction on the ip addresses in the first step :
1) In sip.conf [sippuac] type=friend username=sippuac host=127.0.0.1 port=5061 context=test dtmfmode=rfc2833 insecure=very canreinvite=no nat=yes [sippuas] type=friend username=sippuas host=127.0.0.1 port=5062 context=test dtmfmode=rfc2833 insecure=very canreinvite=no nat=yes On Tue, Jun 14, 2016 at 11:54 AM, Hssan Driss <hssan.dr...@gmail.com> wrote: > 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
