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Merge "pjsip: Fix a few media bugs with reinvites and asymmetric payloads." into 13
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@@ -919,6 +919,14 @@
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On outbound requests, force the user portion of the Contact header to this value.
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</para></description>
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</configOption>
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<configOption name="asymmetric_rtp_codec" default="no">
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<synopsis>Allow the sending and receiving RTP codec to differ</synopsis>
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<description><para>
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When set to "yes" the codec in use for sending will be allowed to differ from
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that of the received one. PJSIP will not automatically switch the sending one
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to the receiving one.
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</para></description>
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</configOption>
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</configObject>
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<configObject name="auth">
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<synopsis>Authentication type</synopsis>
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@@ -1939,6 +1939,7 @@ int ast_res_pjsip_initialize_configuration(const struct ast_module_info *ast_mod
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ast_sorcery_object_field_register_custom(sip_sorcery, "endpoint", "contact_acl", "", endpoint_acl_handler, contact_acl_to_str, NULL, 0, 0);
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ast_sorcery_object_field_register(sip_sorcery, "endpoint", "subscribe_context", "", OPT_CHAR_ARRAY_T, 0, CHARFLDSET(struct ast_sip_endpoint, subscription.context));
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ast_sorcery_object_field_register_custom(sip_sorcery, "endpoint", "contact_user", "", contact_user_handler, contact_user_to_str, NULL, 0, 0);
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ast_sorcery_object_field_register(sip_sorcery, "endpoint", "asymmetric_rtp_codec", "no", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, asymmetric_rtp_codec));
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if (ast_sip_initialize_sorcery_transport()) {
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ast_log(LOG_ERROR, "Failed to register SIP transport support with sorcery\n");
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@@ -370,6 +370,11 @@ static int set_caps(struct ast_sip_session *session, struct ast_sip_session_medi
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session->dsp = NULL;
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}
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}
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if (ast_channel_is_bridged(session->channel)) {
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ast_channel_set_unbridged_nolock(session->channel, 1);
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}
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ast_channel_unlock(session->channel);
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}
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