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

[Joshua Colp]

Arnd Vehling wrote:
> 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
> 
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Can you do an rtp debug while it is in this state and see what the 
output is like? It would help immensely.

Joshua Colp
Digium