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
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