On Tue, Dec 27, 2011 at 6:33 AM, virendra bhati <[email protected]> wrote:
> Hi Sammy, > > I did the same and start calling. And it's working find but Now I want to > the server max capacity with this script then what is the correct process..? > There is a nice tutorial on how you can do this in the asterisk source code: ./doc/chan_sip-perf-testing.txt murf > > > On Tue, Dec 27, 2011 at 6:36 PM, Sammy Govind <[email protected]> wrote: > >> Hi, >> as the Logs say clearly you need to create an extension in default >> context named service >> >> [default] >> ..... >> exten => service,1,NOOP(Incoming call from SIPp) >> ..... >> >> Regards, >> Sammy >> >> >> On Tue, Dec 27, 2011 at 5:48 PM, virendra bhati <[email protected]>wrote: >> >>> Hi list, >>> >>> I have installed SIPp into my server. But not able to used it properly. >>> how to configure with my server ? how to see logs on webpage ? >>> how to start call testing .... >>> >>> when i start SIPp then found verious hits on myserver. >>> >>> *CLI:- * >>> [Dec 27 17:37:54] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: >>> Call from '' to extension 'service' rejected because extension not found in >>> context 'default'. >>> == Using SIP RTP CoS mark 5 >>> [Dec 27 17:37:54] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: >>> Call from '' to extension 'service' rejected because extension not found in >>> context 'default'. >>> == Using SIP RTP CoS mark 5 >>> [Dec 27 17:37:54] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: >>> Call from '' to extension 'service' rejected because extension not found in >>> context 'default'. >>> == Using SIP RTP CoS mark 5 >>> [Dec 27 17:37:54] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: >>> Call from '' to extension 'service' rejected because extension not found in >>> context 'default'. >>> == Using SIP RTP CoS mark 5 >>> [Dec 27 17:37:55] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: >>> Call from '' to extension 'service' rejected because extension not found in >>> context 'default'. >>> == Using SIP RTP CoS mark 5 >>> [Dec 27 17:37:55] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: >>> Call from '' to extension 'service' rejected because extension not found in >>> context 'default'. >>> == Using SIP RTP CoS mark 5 >>> [Dec 27 17:37:55] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: >>> Call from '' to extension 'service' rejected because extension not found in >>> context 'default'. >>> == Using SIP RTP CoS mark 5 >>> [Dec 27 17:37:55] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: >>> Call from '' to extension 'service' rejected because extension not found in >>> context 'default'. >>> == Using SIP RTP CoS mark 5 >>> [Dec 27 17:37:55] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: >>> Call from '' to extension 'service' rejected because extension not found in >>> context 'default'. >>> haddock8-astrx*CLI> >>> >>> >>> >>> -- >>> >>> Thanks and regards >>> >>> Virendra Bhati >>> +91-8885268942 >>> Software Engineer >>> >>> >> > > > -- > > Thanks and regards > > Virendra Bhati > +91-8885268942 > Software Engineer > > > -- > _____________________________________________________________________ > -- 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 > -- Steve Murphy ParseTree Corporation 57 Lane 17 Cody, WY 82414 ✉ [email protected] ☎ 307-899-5535
-- _____________________________________________________________________ -- 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
