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

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Fixes #1004
This commit is contained in:
Alexey Vasilyev
2024-11-25 09:41:48 +01:00
committed by asterisk-org-access-app[bot]
parent e020ee5795
commit dc64d485d7
1 changed files with 2 additions and 2 deletions
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res/res_rtp_asterisk.c
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@@ -7240,8 +7240,8 @@ static int bridge_p2p_rtp_write(struct ast_rtp_instance *instance,
} }
/* Otherwise adjust bridged payload to match */ /* Otherwise adjust bridged payload to match */
bridged_payload = ast_rtp_codecs_payload_code_tx(ast_rtp_instance_get_codecs(instance1), 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->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 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) { if (bridged_payload < 0) {