[asterisk-dev] SIP OAC

[info at magnussolution.com]

Hello everyone

I’m learn how to implement AOC using chan_sip. On the doc I see is necessary active option snom_aoc_enabled=yes 

but it not work.

analyzing the code I see app_dial.c that the channel is answered is checked the option aoc_s_rate_list to set the flag AST_CONTROL_AOC

if (o->aoc_s_rate_list) {
							size_t encoded_size;
							struct ast_aoc_encoded *encoded;
							if ((encoded = ast_aoc_encode(o->aoc_s_rate_list, &encoded_size, o->chan))) {
								ast_indicate_data(in, AST_CONTROL_AOC, encoded, encoded_size);
								ast_aoc_destroy_encoded(encoded);
							}
						}


So, what is aoc_s_rate_list? where I set it?

I find in all asterisk code aoc_s_rate_list reference but I not found.


Best regards
Adilson Magnus from MagnusBilling