[asterisk-dev] Anyone got "SIP/RTP" Working reliably using svn trunk?

[Arnd Vehling]

Hi,

i am having severe problems with asterisk svn trunk. SIP/RTP is pretty 
unreliable. Calls between 2 phones connected directly (sip) to the box always 
fail to establish a correct rtp stream. Looks like an NAT issue because the 
rtp stream failing/not getting setup is the one to the phone behind a NAT box. 
NAT is setup correct though. Works with older asterisk version.

Is this to be expected from svn trunk? I need a version with
imap<>voicemail support. Can i take any other svn release?

best regards,

   Arnd