[SR-Users] no audio

[Rizwan Qureshi]

Hi List,
I am using dispatcher module to route everything from registerations to
calls toa asterisk pbx. thats all what openser is doing at the moment.
outgoing calls are fine but I get CHANUNAVAILABLE status for incoming calls
on Asterisk. I guess the reverse sip dialogue are not working properly
through openser. Following is my open ser config

route{

        if (!mf_process_maxfwd_header("10")) {
                sl_send_reply("483","Too Many Hops");
                exit;
        }

        if(!sanity_check("1511", "7"))
        {
                xlog("Malformed SIP message from $si:$sp\n");
                exit;
        }

        #record_route();
        ds_select_dst("1","2");
        forward();

}

I have tried with and without record_route() but nothing working for
incoming calls.

Any ideas?
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