asterisk users - Apr 2007 - Play "blank" sound while VM recording?

Greetings,

(Apologies if this is an FAQ, but I've Googled for hours and haven't 
come up with anything yet.)

I have an Asterisk system deployed at a customer's site. It is connected 
to the outside world by a local SIP provider. When someone calls in 
through the trunk to leave a voicemail, Asterisk is not sending any RTP 
packets back through the trunk after the beep is played. This is fine 
and probably should be the expected behavior, except that after 30 
seconds to a minute of not seeing any RTP traffic coming from the PBX, 
the trunk appears to make the faulty assumption that the PBX is gone 
and hangs up the call.

I've called the trunk provider and they said two things. 1) This is 
indeed what their trunk was programmed to do. 2) No, they won't change 
it.

We're working on switching the customer to a trunk provider with a bit 
more clue, but in the meantime, how can I have Asterisk play an empty 
sound file while the caller is leaving a voicemail message just to keep 
the RTP traffic flowing? This installation of Asterisk was designed by 
someone else and I have limited personal experience with Asterisk 
configuration files, so an example would be appreciated if possible.

Thanks!

-- 
Charles Ulrich
Ideal Solution, LLC -- http://www.idealso.com

Charles Ulrich wrote:
> I have an Asterisk system deployed at a customer's site. It is
connected
> to the outside world by a local SIP provider. When someone calls in 
> through the trunk to leave a voicemail, Asterisk is not sending any RTP 
> packets back through the trunk after the beep is played. This is fine 
> and probably should be the expected behavior, except that after 30 
> seconds to a minute of not seeing any RTP traffic coming from the PBX, 
> the trunk appears to make the faulty assumption that the PBX is gone 
> and hangs up the call.
Maybe this is what you need?:

;rtpkeepalive=<secs>		; Send keepalives in the RTP stream to keep NAT open
				; (default is off - zero)
(in sip.conf, [general] section)

Regards,
  Philipp

-- 
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
     Let's use IT to solve problems and not to create new ones.
           Asterisk? -> http://www.das-asterisk-buch.de

Gesch?ftsf?hrer: Stefan Wintermeyer
Handelsregister: Neuwied B 14998

On Tuesday 03 April 2007 20:09, asterisk-users-request@lists.digium.com 
wrote:
> Charles Ulrich wrote:
> > I have an Asterisk system deployed at a customer's site. It is
> > connected to the outside world by a local SIP provider. When
> > someone calls in through the trunk to leave a voicemail, Asterisk
> > is not sending any RTP packets back through the trunk after the
> > beep is played. This is fine and probably should be the expected
> > behavior, except that after 30 seconds to a minute of not seeing
> > any RTP traffic coming from the PBX, the trunk appears to make the
> > faulty assumption that the PBX is gone and hangs up the call.
>
> Maybe this is what you need?:
>
> ;rtpkeepalive=<secs>????????????; Send keepalives in the RTP stream
> to keep NAT open ; (default is off - zero)
> (in sip.conf, [general] section)
>
> Regards,
> ? Philipp
That was exactly what I needed, thanks!

-- 
Charles Ulrich
Ideal Solution, LLC -- http://www.idealso.com