[asterisk-users] stress-test realtime voicemail with sipp

2007-01-23 Thread Julian Lyndon-Smith
We are in the process of implementing realtime voicemail. I was wanting to stress-test the system to see if or when it would fall over. Is it possible to use sipp to create say 250 calls, each of which leaves a message in the voicemail ? My dialplan is currently [default] exten =

Re: [asterisk-users] stress-test realtime voicemail with sipp

2007-01-23 Thread Victor Toofic
El mar, ene 23 de 2007 a las 14:44 +, Julian Lyndon-Smith comentaba: however, if I use sipp to test this, I get [Jan 23 14:43:51] WARNING[22782]: app.c:599 __ast_play_and_record: No audio available on SIP/sipp-b7c274b0?? I suspect that's because sipp itself is not sending audio.

Re: [asterisk-users] stress-test realtime voicemail with sipp (Solved)

2007-01-23 Thread Julian Lyndon-Smith
Thanks Victor for the heads up. I've got it to work with the following: [default] exten = stress,1,Answer() exten = stress,2(vm),Voicemail() exten = stress,3,Hangup() and a sipp command line of ./sipp -d 4 -r 5 -t un -sn uac_pcap -l 50 -m 250 -s stress 127.0.0.1 this created 250

Re: [asterisk-users] stress-test realtime voicemail with sipp

2007-01-23 Thread Marco Mouta
As far as I know: You need to compile sipp with media streaming and authentication or if you just want first to test you may provide an extension named service in the context defined in general section of your sip conf for external calls coming to your asterisk server without authentication:

Re: [asterisk-users] stress-test realtime voicemail with sipp

2007-01-23 Thread Olle E Johansson
23 jan 2007 kl. 16.07 skrev Victor Toofic: El mar, ene 23 de 2007 a las 14:44 +, Julian Lyndon-Smith comentaba: however, if I use sipp to test this, I get [Jan 23 14:43:51] WARNING[22782]: app.c:599 __ast_play_and_record: No audio available on SIP/sipp-b7c274b0?? I suspect that's