[Asterisk-Users] Having problems with RTP packets and Hold

[Clif Jones]

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.