Clif Jones
2004-Feb-10 06:03 UTC
[Asterisk-Users] Having problems with RTP packets and Hold
I'm having some problems with a SIP FXO gateway working with Asterisk when a call that involves the gateway is put on hold. This gateway was working up to a firmware upgrade but I believe it may have been working for the "wrong reasons". Here is what happens: 1. User calls in from PSTN to SIP FXO gateway. 2. FXO gateway calls extension on Asterisk. 3. SIP extension is rung and answered. (Canreinvite=no so voicepath goes from SIP extension - to - Asterisk - to - SIP FXO gateway.) 4. SIP extension puts incoming caller on hold (SIP extension re-invites Asterisk with SDP c=IN IP4 0.0.0.0) 5. Asterisk does not re-INVITE SIP gateway and apparently stops sending RTP packets to the SIP gateway. (I'm not using music on hold) 6. Gateway thinks something is wrong because the RTP packets stop arriving and sends a BYE to Asterisk. 7. PSTN caller is mad because they just got dropped. :) I think that the pre-release firmware for my SIP FXO gateway was not doing the RTP check so I was getting away with the above scenario without problems. The question is: How can I get Asterisk to either re-invite the FXO gateway with "held media" or send "silence" RTP packets? Any ideas would be appreciated.