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@ -117,6 +117,22 @@ static int rtp_hangup(struct ast_channel *ast)
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return 0;
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}
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static struct ast_format *derive_format_from_cap(struct ast_format_cap *cap)
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{
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struct ast_format *fmt = ast_format_cap_get_format(cap, 0);
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if (ast_format_cap_count(cap) == 1 && fmt == ast_format_slin) {
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/*
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* Because we have no SDP, we must use one of the static RTP payload
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* assignments. Signed linear @ 8kHz does not map, so if that is our
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* only capability, we force μ-law instead.
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*/
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fmt = ast_format_ulaw;
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}
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return fmt;
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}
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/*! \brief Function called when we should prepare to call the multicast destination */
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static struct ast_channel *multicast_rtp_request(const char *type, struct ast_format_cap *cap, const struct ast_assigned_ids *assignedids, const struct ast_channel *requestor, const char *data, int *cause)
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{
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@ -171,7 +187,7 @@ static struct ast_channel *multicast_rtp_request(const char *type, struct ast_fo
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fmt = ast_multicast_rtp_options_get_format(mcast_options);
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if (!fmt) {
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fmt = ast_format_cap_get_format(cap, 0);
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fmt = derive_format_from_cap(cap);
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}
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if (!fmt) {
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ast_log(LOG_ERROR, "No codec available for sending RTP to '%s'\n",
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@ -298,7 +314,7 @@ static struct ast_channel *unicast_rtp_request(const char *type, struct ast_form
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goto failure;
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}
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} else {
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fmt = ast_format_cap_get_format(cap, 0);
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fmt = derive_format_from_cap(cap);
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if (!fmt) {
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ast_log(LOG_ERROR, "No codec available for sending RTP to '%s'\n",
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args.destination);
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