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

[Olle E Johansson]

23 jan 2007 kl. 16.07 skrev Victor Toofic:

> 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

Easier is to use another ASterisk server, or two...

/O