[asterisk-users] stress-test realtime voicemail with sipp
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 = stress,1,Answer() exten = stress,2(vm),Voicemail(|su) exten = stress,3,Hangup() 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. Is there any tricks I can do in the dialplan to get an extension to answer sipp and then send it to voicemail, but play some audio for the voicemail ? Thanks. Julian. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] stress-test realtime voicemail with sipp
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. Why don't you use sipp with pcap support enabled? http://sipp.sourceforge.net/doc/reference.html You can modify a little bit some of the integrated scenarios to allow sipp to interoperate with your voicemail extension. http://sipp.sourceforge.net/doc/reference.html#UAC+with+media ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] stress-test realtime voicemail with sipp (Solved)
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 voicemail messages (with 50 simultaneous calls) leaving a 6-7 second voicemail (using .wav, .WAV and .gsm) I'm *really* going to try and hurt it now ;) Julian Victor Toofic wrote: 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. Why don't you use sipp with pcap support enabled? http://sipp.sourceforge.net/doc/reference.html You can modify a little bit some of the integrated scenarios to allow sipp to interoperate with your voicemail extension. http://sipp.sourceforge.net/doc/reference.html#UAC+with+media ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] stress-test realtime voicemail with sipp
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: http://sipp.sourceforge.net/doc/reference.html#Installing+SIPp - *With PCAP playhttp://sipp.sourceforge.net/doc/reference.html#pcapplayand without authenticationhttp://sipp.sourceforge.net/doc/reference.html#authenticationsupport *: # gunzip sipp-xxx.tar.gz # tar -xvf sipp-xxx.tar # cd sipp # make pcapplay - *With PCAP playhttp://sipp.sourceforge.net/doc/reference.html#pcapplayand authenticationhttp://sipp.sourceforge.net/doc/reference.html#authenticationsupport *: # gunzip sipp-xxx.tar.gz # tar -xvf sipp-xxx.tar # cd sipp # make pcapplay_ossl Example: - Sipp being used as a SIP user agent Client: - Call Duration 1ms - Dialing Calls with RTP using ulaw ./sipp -sf uac_pcap.xml -d 1 192.168.34.6 -trace_err Where this IP is my * . Hope this helps, Plse provid some feedback. I would like also to learn from community how to understand Load average results with Top command while incrementing calls dial from sipp to asterisk, and how to determine max calls on Asterisk. This max calls is defined when Sipp calls to * starts being discarded? Best regards, Marco Mouta On 1/23/07, Julian Lyndon-Smith [EMAIL PROTECTED] wrote: 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 = stress,1,Answer() exten = stress,2(vm),Voicemail(|su) exten = stress,3,Hangup() 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. Is there any tricks I can do in the dialplan to get an extension to answer sipp and then send it to voicemail, but play some audio for the voicemail ? Thanks. Julian. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] stress-test realtime voicemail with sipp
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 because sipp itself is not sending audio. Why don't you use sipp with pcap support enabled? http://sipp.sourceforge.net/doc/reference.html You can modify a little bit some of the integrated scenarios to allow sipp to interoperate with your voicemail extension. http://sipp.sourceforge.net/doc/reference.html#UAC+with+media Easier is to use another ASterisk server, or two... /O ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
