asterisk users - Oct 2010 - Asterisk 1.8: Warning messages in CLI while putting a SIP-Call on hold

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.

Is this a new bug or do I something wrong?

File sip.conf looks like this:

[general]
bindaddr = 0.0.0.0
disallow = all
allow = alaw
allow = ulaw
language = de
allowguest = no
fromdomain = 192.168.10.70
tos_sip = 96
tos_audio = 184

[katrin]
type = friend
host = dynamic
callerid = Katrin Wemheuer <200>
context = Standard
mailbox = 200

[max]
type = friend
host = dynamic
callerid = Max M?ller <245>
context = Standard
mailbox = 245

Thanks,

Karsten

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

I have a Quintum AX Tenor gateway sending calls to Asterisk from BT 
analogue lines connected to FXO.
The agents hear an echo on their side but incoming callers hear the 
conversation fine. I can't seem to find the problem. Anyone seen this 
issue before?

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