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? -- Howard. LANNet Computing Associates; Your Linux people <http://www.lannetlinux.com> ------------------------------------------ "When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft." ------------------------------------------ "Flatter government, not fatter government; Get rid of the Australian states."
Eric Wieling aka ManxPower
2004-Dec-11 09:46 UTC
[Asterisk-Users] What might be blocking RTP
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 -- 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.