Hello,
Actually I have changed asterisk in such a way that any call that comes onto asterisk server will go into the voicemail() application for that user. I am sending the media through SIPp by putting the following action in scenario file: <!-- Play a pre-recorded PCAP file (RTP stream) --> <nop> <action> <exec play_pcap_audio="pcap/g711a.pcap"/> </action> </nop> Regards, Shalu Date: Wed, 11 Jan 2012 10:59:33 +0530 From: virendra bhati <[email protected]> Subject: Re: [asterisk-users] No audio available on SIP/172.16.129.13:5060-00000001?? To: Asterisk Users Mailing List - Non-Commercial Discussion <[email protected]> Message-ID: <cannhuhdoqvvovvyib7s0pnj+or_xy94d0yl8fe71eka+f4d...@mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" 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 < [email protected] > 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 >
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