[asterisk-users] Asterisk 1.8: Warning messages in CLI while putting a SIP-Call on hold

[Karsten Wemheuer]

Am Mittwoch, den 06.10.2010, 15:11 +0200 schrieb Karsten Wemheuer:
> Hi,
> 
> while testing current release candidate 1.8.0-rc2 I stumbled on a weird
> behavior. I did not find any hints in the archives or at the bug
> tracker.
> 
> Two SIP-Clients are connected (both on the local net, no NAT). The RTP
> stream flows directly between the phones. If I set phone A on hold, the
> music on hold is played. On the CLI I see the following message running:
> WARNING[2470]: res_rtp_asterisk.c:1939 bridge_p2p_rtp_write: RTP
> Transmission error of packet to (null): Invalid argument
> 
> The message is running until the phones are connected again. In the
> meantime the CLI is nearly unusable. This does not happen, if I
> configure asterisk to stay in the media path.

for the archives: This behavior seems to be fixed in 1.8.0-rc3.

Karsten