[asterisk-dev] No audio available on SIP/172.16.129.13:5060-00000001??

[shalu dhamija]

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) 
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