[Asterisk-Users] What might be blocking RTP

[Eric Wieling aka ManxPower]

Howard Lowndes wrote:
> When I make a call from a SIP phone to a speaking extension on *, such
> as one that speaks digits or similar, when I monitor * in very verbose
> mode I can see it running through the routine associated with the
> extension, but I am getting no RTP data stream back to the phone.
> 
> Does the machine housing * need a sound card?
> Does it need OSS or ALSA modules installed?
> What actually generates the RTP data stream?
> 

You don't need a soundcard.

Is Asterisk behind NAT?  If so look at localnet= and externip= in 
sip.conf and look into portforwarding and rtp.conf.  Remember AUDIO on 
SIP/H323/MGCP/SCCP are sent using the RTP protocol.  SIP is just a 
signaling protocol.

--Eric

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