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List:       asterisk-dev
Subject:    [asterisk-dev] Anyone got "SIP/RTP" Working reliably using svn
From:       Arnd Vehling <av () nethead ! de>
Date:       2006-08-31 17:01:56
Message-ID: 44F71604.5070606 () nethead ! de
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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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