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/*
* Asterisk config_site.h
*/
#include <sys/select.h>
/*
* Defining PJMEDIA_HAS_SRTP to 0 does NOT disable Asterisk's ability to use srtp.
* It only disables the pjmedia srtp transport which Asterisk doesn't use.
* The reason for the disable is that while Asterisk works fine with older libsrtp
* versions, newer versions of pjproject won't compile with them.
*/
#define PJMEDIA_HAS_SRTP 0
#define PJ_HAS_IPV6 1
#define NDEBUG 1
#define PJ_MAX_HOSTNAME (256)
#define PJSIP_MAX_URL_SIZE (512)
#ifdef PJ_HAS_LINUX_EPOLL
#define PJ_IOQUEUE_MAX_HANDLES (5000)
#else
#define PJ_IOQUEUE_MAX_HANDLES (FD_SETSIZE)
#endif
#define PJ_IOQUEUE_HAS_SAFE_UNREG 1
#define PJ_IOQUEUE_MAX_EVENTS_IN_SINGLE_POLL (16)
#define PJ_SCANNER_USE_BITWISE 0
#define PJ_OS_HAS_CHECK_STACK 0
#define PJ_LOG_MAX_LEVEL 3
#define PJ_ENABLE_EXTRA_CHECK 1
#define PJSIP_MAX_TSX_COUNT ((64*1024)-1)
#define PJSIP_MAX_DIALOG_COUNT ((64*1024)-1)
#define PJSIP_UDP_SO_SNDBUF_SIZE (512*1024)
#define PJSIP_UDP_SO_RCVBUF_SIZE (512*1024)
#define PJ_DEBUG 0
#define PJSIP_SAFE_MODULE 0
#define PJ_HAS_STRICMP_ALNUM 0
#define PJ_HASH_USE_OWN_TOLOWER 1
/*
It is imperative that PJSIP_UNESCAPE_IN_PLACE remain 0 or undefined.
Enabling it will result in SEGFAULTS when URIs containing escape sequences are encountered.
*/
#undef PJSIP_UNESCAPE_IN_PLACE
#define PJSIP_MAX_PKT_LEN 6000
#undef PJ_TODO
#define PJ_TODO(x)
/* Defaults too low for WebRTC */
#define PJ_ICE_MAX_CAND 32
#define PJ_ICE_MAX_CHECKS (PJ_ICE_MAX_CAND * 2)
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