[asterisk-dev] chan_jingle

Theo Belder tbelder at gmail.com
Wed Sep 13 23:54:11 MST 2006


Hi,

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?

--
Theo
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-dev/attachments/20060913/fb6e09f7/attachment.htm


More information about the asterisk-dev mailing list