res_rtp_asterisk.c: Fix bridged_payload matching with sample rate for DTMF

Fixes #1004

(cherry picked from commit 9060a267e0adf901f0e94dc31c03adf2e0795600)

diff --git a/res/res_rtp_asterisk.c b/res/res_rtp_asterisk.c
index 028150f9503e6eef17600ca9ca0e3ebf73c36b9c..1953e15e2942259fe439ffe1b0ea71cdfe19ec23 100644 (file)
--- a/res/res_rtp_asterisk.c
+++ b/res/res_rtp_asterisk.c
@@ -7240,8 +7240,8 @@ static int bridge_p2p_rtp_write(struct ast_rtp_instance *instance,
        }
 
        /* Otherwise adjust bridged payload to match */
-       bridged_payload = ast_rtp_codecs_payload_code_tx(ast_rtp_instance_get_codecs(instance1),
-               payload_type->asterisk_format, payload_type->format, payload_type->rtp_code);
+       bridged_payload = ast_rtp_codecs_payload_code_tx_sample_rate(ast_rtp_instance_get_codecs(instance1),
+               payload_type->asterisk_format, payload_type->format, payload_type->rtp_code, payload_type->sample_rate);
 
        /* If no codec could be matched between instance and instance1, then somehow things were made incompatible while we were still bridged.  Bail. */
        if (bridged_payload < 0) {