asterisk users - Sep 2013 - No remote address on RTP instance - On Ringing

Hello Everyone,

I have a new problem where when placing the call, asterisk will
automatically go into music on hold until the call is connected (ie,
no ringing). It was kind of confusing because sometimes `SESSION
PROGRESS` takes longer than others, during this time we are in MOH.
The call does eventually connect and the MOH stops. When debugging I
saw the following debug message:

[Sep 10 10:56:12] DEBUG[7930]: res_rtp_asterisk.c:1228 ast_rtp_write:
No remote address on RTP instance '0xb6e00b20' so dropping frame
[Sep 10 10:56:12] DEBUG[7930]: res_rtp_asterisk.c:1228 ast_rtp_write:
No remote address on RTP instance '0xb6e00b20' so dropping frame
[Sep 10 10:56:12] DEBUG[7930]: res_rtp_asterisk.c:1228 ast_rtp_write:
No remote address on RTP instance '0xb6e00b20' so dropping frame
[Sep 10 10:56:12] DEBUG[7930]: res_rtp_asterisk.c:1228 ast_rtp_write:
No remote address on RTP instance '0xb6e00b20' so dropping frame



This is a straight SIP channel. No DAHDI.

Your Help is Greatly Appreciated. This is a new one for me :)

Nick.

Does your Dial() command include the "m" option?

jg