Hi to all, i’m just testing my asterisk box with the command : ./sipp -sn uac -d 100000 -s 1001 127.0.0.1 -l 2000 -trace_err
and i’m getting a lot of errors on my asterisk box about lack of RTP packets, look here: [Mar 6 12:26:16] NOTICE[18311] chan_sip.c: Disconnecting call 'SIP/sipp-00005f7a' for lack of RTP activity in 6 seconds we use this timeout because many providers doesn’t hangup properly on PSTN lines, so we are detecting the HangUp using the RTP timeout. My question is, how to test with sipp without having got these RTP timeout issue? There is no way to simulate RTP activity? -- Roberto Innaimi roberto.inna...@gmail.com
------------------------------------------------------------------------------
_______________________________________________ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users