I have two machines on the internet. Box A and Box B. Box A has a SIP trunk to the world, Making calls box A works fine I have audio to my cell and all works.
I defined a SIP trunk between box B and A. tried to make a call originating from box B - to box A and then over the SIP trunk to my cell. My cell rings but then no audio. I have defined SIP trunks before between boxes pretty straight forward. I have checked and my firewalls are open for SIP/RTP -A INPUT -m state --state NEW -m udp -p udp --dport 5060 -j ACCEPT -A INPUT -m state --state NEW -m tcp -p tcp --dport 8000:60000 -j ACCEPT -A INPUT -m state --state NEW -m udp -p udp --dport 8000:60000 -j ACCEPT I am using asterisk 11.16 box A is [boxab_sip] type=friend username=boxa_sip secret=*** disallow=all allow=ulaw allow=alaw allow=gsm dtmfmode=rfc2833 host=DNS Name here context=sip_trunk insecure=port,invite box B is [boxab_sip] type=friend username=boxab_sip secret=*** disallow=all allow=ulaw allow=alaw allow=gsm dtmfmode=rfc2833 host=DNS Name here context=sip_turnk insecure=port,invite Is there something I am missing? The one piece I have not done before is SIP trunk - to - SIP trunk. But the phone rings - so its routed - just no audio. Thoughts? Thanks, Jerry
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