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
--
I am seeking part or full time employment in the Greater Toronto Area, My preference is part time employment with some telecommuting, but all offers will be considered. Contact eric at fnords.org.
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