I have a question about chan_jingle in combination with sip.
In the same network I have a googletalk client and a sip-client. These are both behind NAT.
My asterisk server with jabber/jingle and sip is somewhere on the internet. When I connect asterisk to the XMPP server, I can see the user is coming up in my googletalk client.
When I call this user from googletalk, my sip client is ringing and i can answer the call. But i only have a audiostream from googletalk to the sip client and no stream from the sip client to googletalk.
In sip.conf I have specified 'nat=yes'.
When iIplace the asterisk server in my local lan, then everything work fine. So i think this is a NAT issue in combination with Asterisk.
Is there something what I might doing wrong?
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Theo
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