Hi Shalu, How you are invoking call in dialplan. it's completely depends on that. And error show that no voice is there for store in voicemail .
On Wed, Jan 11, 2012 at 10:05 AM, shalu dhamija < shalu.dham...@rancoretech.com> wrote: > Hello, > > > > I am trying to run load on asterisk server(version 1.8.7.1) for the > voicemail() application using SIPp tool. I am just running sipp at call > rate of 1 cps with the following command: > > > > ./sipp -m 9000 -r 1 -rp 1000 -d 45 -max_socket 65536 -sf > uac_msg_deposit.xml -i 172.16.129.13 -s 1 172.16.129.14 --trace_err > > > > I am trying to deposit 9000 messages in the mailbox of user 1 (given by > the -s option) but the following warning is coming on the asterisk server > due to which the message does not get deposited into the users mailbox: > > > > No audio available on SIP/172.16.129.13:5060-00000001?? > > > > I have set rtpstart=6000 and rtpend=20000 in rtp.conf. > > > > > > Can someone please let me know how to avoid these kind of warnings. > > > > Thanks. > > > > Shalu > > > > > > > > Thanks and Regards, > Shalu Dhamija > Rancore Technologies(P) Ltd. > Gurgaon > Ph : 0124-4200691 > +91-9910995356(M) > > -- > _____________________________________________________________________ > -- 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 > -- 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