Thanks for the hint. I had indeed set the rtp-ip to the fqdn of the server in internal.xml, as this way I got rid of one way audio for internal calls. Calls through external gateways didn't work though due to the error we are talking about here. Now I added a <param name="ext-rtp-ip" value="stun:stun.freeswitch.org"/> entry to internal.xml and external.xml in the SIP profiles and now the RTP stream via internal and external profile is o.k. I added a new chapter "Freeswitch behind NAT" to the wiki, where I describe my settings: http://wiki.freeswitch.org/wiki/NAT_Traversal Best regards Peter Brian West schrieb:
Ok this is why.. DO NOT put hostnames in the external RTP ip or the sip ip's those are ONLY IP addresses.
/b
On Nov 21, 2008, at 2:00 PM, Peter P GMX wrote:
o=FreeSWITCH 1227284490 1227284492 IN IP4 my.domain.com s=FreeSWITCH c=IN IP4 my.domain.com
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