[asterisk-users] Trying to test my voicemail

[Martin]

Have you compiled sipp with

make pcapplay

? If so maybe the prompt you're playing is too short and after the
initial intoduction "please leave a message for
one-four-thee-five-blah blah"
the prompt simply ends ... I'm not sure if sipp is going to play it
over and over again ...

Although it should repeat it ... You might also check whether there's
a firewall somewhere and the RTP session can work properly ...
You'd have to inspect the UDP ports mentioned in the SDP of INVITE and
200 OK to INVITE (turn on sip debug on asteirsk CLI to catch the SIP
messages)

Martin

On Thu, Apr 2, 2009 at 2:30 AM, Pepo <pepo at ecualug.org> wrote:
> Hi friends...
>
> I am trying to test my voicemail with Asterisk using SIPP (SIPP is running in
> Debian Squezze and Asterisk is running in OpenSuSE-11.1), the command that I
> use is:
>
> sipp -sn uac_pcap -l 1 -m 1 -s 55 -trace_err 192.168.13.6
>
> But, If I use the file g711a.pcap included in the sources of sipp or if use
> some file captured for me the result is the same ---> error ... the message
> in Asterisk is:
>
> [Apr  2 02:16:14] WARNING[21197]: app.c:674 __ast_play_and_record: No audio
> available on SIP/sipp-082402b0??
>
> I've tested compiling the sources but still with the same error. I've changed
> the xml file but I keep failing.
>
> Please, How do I test my voicemail but recording audio? Is there other tool to
> help me?
>
> A lot of thanks.
>
> Pepo.
>
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>
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