[asterisk-users] stress-test realtime voicemail with sipp (Solved)
Julian Lyndon-Smith
asterisk at dotr.com
Tue Jan 23 08:29:32 MST 2007
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(7777)
exten => stress,3,Hangup()
and a sipp command line of
./sipp -d 40000 -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 +0000, 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
>
>
More information about the asterisk-users
mailing list