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I am getting one way audio when sending calls to OpenSIPS. The OpenSIPS in return route calls to asterisk server that plays a file. The problem I am having is Asterisk is not getting any DTMF back from the originating SIP call.
I looked in the Contact-URI and it is using local ip of Asterisk. So the INVITE would come through just fine and gets routed to Asterisk and I see that call in the console. The audio's working fine too but somehow Asterisk's not able to detect any DTMF entry.
I've tried the subst method to rewrite the contact info but to no avail.
Can someone point me to the right direction? Would rewriting the contact-uri to external ip help?