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hi, I have the following configuration: Grandstream --> NAT (Netgear RP614)-->Internet-->Asterisk(public IP) i can register fine and call ringing is working as good. The problem is = i cant hear audio both ways and i get this error: WARNING[22544]: File rtp.c, Line 375 (ast_rtp_read): RTP Read error: Resource temporarily unavailable my sip.conf file is as follows:

Hi, I'm reasonably new to Asterisk and have a question which I hoping somebody can help me with. I've looked through the archives and config guides, etc but unfortunately I can find no mention of the information that I'm looking for. I have an Asterisk server running version 1.07. On the user side there are several Grandstream phones and two 24 port fxs gateways. Asterisk is