diff options
Diffstat (limited to 'channels/chan_sip.c')
| -rw-r--r-- | channels/chan_sip.c | 1055 |
1 files changed, 420 insertions, 635 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c index bd4f57e47..42894d3bc 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -1696,16 +1696,14 @@ static void unref_user(struct sip_user *user) static void registry_unref(struct sip_registry *reg) { - if (option_debug > 2) - ast_log(LOG_DEBUG, "SIP Registry %s: refcount now %d\n", reg->hostname, reg->refcount - 1); + ast_debug(3, "SIP Registry %s: refcount now %d\n", reg->hostname, reg->refcount - 1); ASTOBJ_UNREF(reg, sip_registry_destroy); } /*! \brief Add object reference to SIP registry */ static struct sip_registry *registry_addref(struct sip_registry *reg) { - if (option_debug > 2) - ast_log(LOG_DEBUG, "SIP Registry %s: refcount now %d\n", reg->hostname, reg->refcount + 1); + ast_debug(3, "SIP Registry %s: refcount now %d\n", reg->hostname, reg->refcount + 1); return ASTOBJ_REF(reg); /* Add pointer to registry in packet */ } @@ -1742,12 +1740,10 @@ static const char *referstatus2str(enum referstatus rstatus) a dialog */ static void initialize_initreq(struct sip_pvt *p, struct sip_request *req) { - if (option_debug) { - if (p->initreq.headers) - ast_log(LOG_DEBUG, "Initializing already initialized SIP dialog %s (presumably reinvite)\n", p->callid); - else - ast_log(LOG_DEBUG, "Initializing initreq for method %s - callid %s\n", sip_methods[req->method].text, p->callid); - } + if (p->initreq.headers) + ast_debug(1, "Initializing already initialized SIP dialog %s (presumably reinvite)\n", p->callid); + else + ast_debug(1, "Initializing initreq for method %s - callid %s\n", sip_methods[req->method].text, p->callid); /* Use this as the basis */ copy_request(&p->initreq, req); parse_request(&p->initreq); @@ -1758,8 +1754,7 @@ static void initialize_initreq(struct sip_pvt *p, struct sip_request *req) /*! \brief Encapsulate setting of SIP_ALREADYGONE to be able to trace it with debugging */ static void sip_alreadygone(struct sip_pvt *dialog) { - if (option_debug > 2) - ast_log(LOG_DEBUG, "Setting SIP_ALREADYGONE on dialog %s\n", dialog->callid); + ast_debug(3, "Setting SIP_ALREADYGONE on dialog %s\n", dialog->callid); ast_set_flag(&dialog->flags[0], SIP_ALREADYGONE); } @@ -1798,19 +1793,19 @@ static struct sip_proxy *proxy_allocate(char *name, char *port, int force) static struct sip_proxy *obproxy_get(struct sip_pvt *dialog, struct sip_peer *peer) { if (peer && peer->outboundproxy) { - if (option_debug && sipdebug) - ast_log(LOG_DEBUG, "OBPROXY: Applying peer OBproxy to this call\n"); + if (sipdebug) + ast_debug(1, "OBPROXY: Applying peer OBproxy to this call\n"); append_history(dialog, "OBproxy", "Using peer obproxy %s", peer->outboundproxy->name); return peer->outboundproxy; } if (global_outboundproxy.name[0]) { - if (option_debug && sipdebug) - ast_log(LOG_DEBUG, "OBPROXY: Applying global OBproxy to this call\n"); + if (sipdebug) + ast_debug(1, "OBPROXY: Applying global OBproxy to this call\n"); append_history(dialog, "OBproxy", "Using global obproxy %s", global_outboundproxy.name); return &global_outboundproxy; } - if (option_debug && sipdebug) - ast_log(LOG_DEBUG, "OBPROXY: Not applying OBproxy to this call\n"); + if (sipdebug) + ast_debug(1, "OBPROXY: Not applying OBproxy to this call\n"); return NULL; } @@ -1855,30 +1850,30 @@ static unsigned int parse_sip_options(struct sip_pvt *pvt, const char *supported return 0; temp = ast_strdupa(supported); - if (option_debug > 2 && sipdebug) - ast_log(LOG_DEBUG, "Begin: parsing SIP \"Supported: %s\"\n", supported); + if (sipdebug) + ast_debug(3, "Begin: parsing SIP \"Supported: %s\"\n", supported); for (next = temp; next; next = sep) { found = FALSE; if ( (sep = strchr(next, ',')) != NULL) *sep++ = '\0'; next = ast_skip_blanks(next); - if (option_debug > 2 && sipdebug) - ast_log(LOG_DEBUG, "Found SIP option: -%s-\n", next); + if (sipdebug) + ast_debug(3, "Found SIP option: -%s-\n", next); for (i=0; i < (sizeof(sip_options) / sizeof(sip_options[0])); i++) { if (!strcasecmp(next, sip_options[i].text)) { profile |= sip_options[i].id; found = TRUE; - if (option_debug > 2 && sipdebug) - ast_log(LOG_DEBUG, "Matched SIP option: %s\n", next); + if (sipdebug) + ast_debug(3, "Matched SIP option: %s\n", next); break; } } - if (!found && option_debug > 2 && sipdebug) { + if (!found && sipdebug) { if (!strncasecmp(next, "x-", 2)) - ast_log(LOG_DEBUG, "Found private SIP option, not supported: %s\n", next); + ast_debug(3, "Found private SIP option, not supported: %s\n", next); else - ast_log(LOG_DEBUG, "Found no match for SIP option: %s (Please file bug report!)\n", next); + ast_debug(3, "Found no match for SIP option: %s (Please file bug report!)\n", next); } } @@ -1986,10 +1981,8 @@ static enum sip_result ast_sip_ouraddrfor(struct in_addr *them, struct in_addr * ast_log(LOG_NOTICE, "Warning: Re-lookup of '%s' failed!\n", externhost); } *us = externip.sin_addr; - if (option_debug) { - ast_log(LOG_DEBUG, "Target address %s is not local, substituting externip\n", - ast_inet_ntoa(*(struct in_addr *)&them->s_addr)); - } + ast_debug(1, "Target address %s is not local, substituting externip\n", + ast_inet_ntoa(*(struct in_addr *)&them->s_addr)); } else if (bindaddr.sin_addr.s_addr) *us = bindaddr.sin_addr; return AST_SUCCESS; @@ -2042,13 +2035,13 @@ static int retrans_pkt(void *data) if (pkt->retrans < MAX_RETRANS) { pkt->retrans++; if (!pkt->timer_t1) { /* Re-schedule using timer_a and timer_t1 */ - if (sipdebug && option_debug > 3) - ast_log(LOG_DEBUG, "SIP TIMER: Not rescheduling id #%d:%s (Method %d) (No timer T1)\n", pkt->retransid, sip_methods[pkt->method].text, pkt->method); + if (sipdebug) + ast_debug(4, "SIP TIMER: Not rescheduling id #%d:%s (Method %d) (No timer T1)\n", pkt->retransid, sip_methods[pkt->method].text, pkt->method); } else { int siptimer_a; - if (sipdebug && option_debug > 3) - ast_log(LOG_DEBUG, "SIP TIMER: Rescheduling retransmission #%d (%d) %s - %d\n", pkt->retransid, pkt->retrans, sip_methods[pkt->method].text, pkt->method); + if (sipdebug) + ast_debug(4, "SIP TIMER: Rescheduling retransmission #%d (%d) %s - %d\n", pkt->retransid, pkt->retrans, sip_methods[pkt->method].text, pkt->method); if (!pkt->timer_a) pkt->timer_a = 2 ; else @@ -2061,8 +2054,7 @@ static int retrans_pkt(void *data) /* Reschedule re-transmit */ reschedule = siptimer_a; - if (option_debug > 3) - ast_log(LOG_DEBUG, "** SIP timers: Rescheduling retransmission %d to %d ms (t1 %d ms (Retrans id #%d)) \n", pkt->retrans +1, siptimer_a, pkt->timer_t1, pkt->retransid); + ast_debug(4, "** SIP timers: Rescheduling retransmission %d to %d ms (t1 %d ms (Retrans id #%d)) \n", pkt->retrans +1, siptimer_a, pkt->timer_t1, pkt->retransid); } if (sip_debug_test_pvt(pkt->owner)) { @@ -2119,8 +2111,7 @@ static int retrans_pkt(void *data) if (pkt->method != SIP_OPTIONS && pkt->method != SIP_REGISTER) { ast_set_flag(&pkt->owner->flags[0], SIP_NEEDDESTROY); sip_alreadygone(pkt->owner); - if (option_debug) - append_history(pkt->owner, "DialogKill", "Killing this failed dialog immediately"); + append_history(pkt->owner, "DialogKill", "Killing this failed dialog immediately"); } } } @@ -2176,8 +2167,8 @@ static enum sip_result __sip_reliable_xmit(struct sip_pvt *p, int seqno, int res /* Schedule retransmission */ pkt->retransid = ast_sched_add_variable(sched, siptimer_a, retrans_pkt, pkt, 1); - if (option_debug > 3 && sipdebug) - ast_log(LOG_DEBUG, "*** SIP TIMER: Initalizing retransmit timer on packet: Id #%d\n", pkt->retransid); + if (sipdebug) + ast_debug(4, "*** SIP TIMER: Initalizing retransmit timer on packet: Id #%d\n", pkt->retransid); pkt->next = p->packets; p->packets = pkt; if (sipmethod == SIP_INVITE) { @@ -2206,8 +2197,7 @@ static int __sip_autodestruct(void *data) transmit_state_notify(p, AST_EXTENSION_DEACTIVATED, 1, TRUE); /* Send last notification */ p->subscribed = NONE; append_history(p, "Subscribestatus", "timeout"); - if (option_debug > 2) - ast_log(LOG_DEBUG, "Re-scheduled destruction of SIP subsription %s\n", p->callid ? p->callid : "<unknown>"); + ast_debug(3, "Re-scheduled destruction of SIP subsription %s\n", p->callid ? p->callid : "<unknown>"); return 10000; /* Reschedule this destruction so that we know that it's gone */ } @@ -2222,15 +2212,13 @@ static int __sip_autodestruct(void *data) ast_log(LOG_WARNING, "Autodestruct on dialog '%s' with owner in place (Method: %s)\n", p->callid, sip_methods[p->method].text); ast_queue_hangup(p->owner); } else if (p->refer) { - if (option_debug > 2) - ast_log(LOG_DEBUG, "Finally hanging up channel after transfer: %s\n", p->callid); + ast_debug(3, "Finally hanging up channel after transfer: %s\n", p->callid); transmit_request_with_auth(p, SIP_BYE, 0, XMIT_RELIABLE, 1); append_history(p, "ReferBYE", "Sending BYE on transferer call leg %s", p->callid); sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT); } else { append_history(p, "AutoDestroy", "%s", p->callid); - if (option_debug) - ast_log(LOG_DEBUG, "Auto destroying SIP dialog '%s'\n", p->callid); + ast_debug(3, "Auto destroying SIP dialog '%s'\n", p->callid); sip_destroy(p); /* Go ahead and destroy dialog. All attempts to recover is done */ } return 0; @@ -2286,13 +2274,12 @@ static void __sip_ack(struct sip_pvt *p, int seqno, int resp, int sipmethod) if (ast_test_flag(cur, FLAG_RESPONSE) || cur->method == sipmethod) { msg = "Found"; if (!resp && (seqno == p->pendinginvite)) { - if (option_debug) - ast_log(LOG_DEBUG, "Acked pending invite %d\n", p->pendinginvite); + ast_debug(1, "Acked pending invite %d\n", p->pendinginvite); p->pendinginvite = 0; } if (cur->retransid > -1) { - if (sipdebug && option_debug > 3) - ast_log(LOG_DEBUG, "** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #%d\n", cur->retransid); + if (sipdebug) + ast_debug(4, "** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #%d\n", cur->retransid); ast_sched_del(sched, cur->retransid); cur->retransid = -1; } @@ -2302,9 +2289,8 @@ static void __sip_ack(struct sip_pvt *p, int seqno, int resp, int sipmethod) } } sip_pvt_unlock(p); - if (option_debug) - ast_log(LOG_DEBUG, "Stopping retransmission on '%s' of %s %d: Match %s\n", - p->callid, resp ? "Response" : "Request", seqno, msg); + ast_debug(1, "Stopping retransmission on '%s' of %s %d: Match %s\n", + p->callid, resp ? "Response" : "Request", seqno, msg); } /*! \brief Pretend to ack all packets @@ -2336,8 +2322,8 @@ static int __sip_semi_ack(struct sip_pvt *p, int seqno, int resp, int sipmethod) (ast_test_flag(cur, FLAG_RESPONSE) || method_match(sipmethod, cur->data))) { /* this is our baby */ if (cur->retransid > -1) { - if (option_debug > 3 && sipdebug) - ast_log(LOG_DEBUG, "*** SIP TIMER: Cancelling retransmission #%d - %s (got response)\n", cur->retransid, sip_methods[sipmethod].text); + if (sipdebug) + ast_debug(4, "*** SIP TIMER: Cancelling retransmission #%d - %s (got response)\n", cur->retransid, sip_methods[sipmethod].text); ast_sched_del(sched, cur->retransid); cur->retransid = -1; } @@ -2345,8 +2331,7 @@ static int __sip_semi_ack(struct sip_pvt *p, int seqno, int resp, int sipmethod) break; } } - if (option_debug) - ast_log(LOG_DEBUG, "(Provisional) Stopping retransmission (but retaining packet) on '%s' %s %d: %s\n", p->callid, resp ? "Response" : "Request", seqno, res ? "Not Found" : "Found"); + ast_debug(1, "(Provisional) Stopping retransmission (but retaining packet) on '%s' %s %d: %s\n", p->callid, resp ? "Response" : "Request", seqno, res ? "Not Found" : "Found"); return res; } @@ -2578,8 +2563,8 @@ static int sip_sendhtml(struct ast_channel *chan, int subclass, const char *data ast_string_field_build(p, url, "<%s>;mode=active", data); - if (sip_debug_test_pvt(p) && option_debug) - ast_log(LOG_DEBUG, "Send URL %s, state = %d!\n", data, chan->_state); + if (sip_debug_test_pvt(p)) + ast_debug(1, "Send URL %s, state = %d!\n", data, chan->_state); switch (chan->_state) { case AST_STATE_RING: @@ -2698,8 +2683,7 @@ static void register_peer_exten(struct sip_peer *peer, int onoff) /*! \brief Destroy peer object from memory */ static void sip_destroy_peer(struct sip_peer *peer) { - if (option_debug > 2) - ast_log(LOG_DEBUG, "Destroying SIP peer %s\n", peer->name); + ast_debug(3, "Destroying SIP peer %s\n", peer->name); if (peer->outboundproxy) ast_free(peer->outboundproxy); @@ -2731,8 +2715,7 @@ static void sip_destroy_peer(struct sip_peer *peer) apeerobjs--; else if (ast_test_flag(&peer->flags[0], SIP_REALTIME)) { rpeerobjs--; - if (option_debug > 2) - ast_log(LOG_DEBUG,"-REALTIME- peer Destroyed. Name: %s. Realtime Peer objects: %d\n", peer->name, rpeerobjs); + ast_debug(3,"-REALTIME- peer Destroyed. Name: %s. Realtime Peer objects: %d\n", peer->name, rpeerobjs); } else speerobjs--; clear_realm_authentication(peer->auth); @@ -2833,8 +2816,7 @@ static struct sip_peer *realtime_peer(const char *newpeername, struct sockaddr_i return NULL; } - if (option_debug > 2) - ast_log(LOG_DEBUG,"-REALTIME- loading peer from database to memory. Name: %s. Peer objects: %d\n", peer->name, rpeerobjs); + ast_debug(3,"-REALTIME- loading peer from database to memory. Name: %s. Peer objects: %d\n", peer->name, rpeerobjs); if (ast_test_flag(&global_flags[1], SIP_PAGE2_RTCACHEFRIENDS)) { /* Cache peer */ @@ -2886,8 +2868,7 @@ static struct sip_peer *find_peer(const char *peer, struct sockaddr_in *sin, int /*! \brief Remove user object from in-memory storage */ static void sip_destroy_user(struct sip_user *user) { - if (option_debug > 2) - ast_log(LOG_DEBUG, "Destroying user object from memory: %s\n", user->name); + ast_debug(3, "Destroying user object from memory: %s\n", user->name); ast_free_ha(user->ha); if (user->chanvars) { ast_variables_destroy(user->chanvars); @@ -2961,23 +2942,19 @@ static void do_setnat(struct sip_pvt *p, int natflags) const char *mode = natflags ? "On" : "Off"; if (p->rtp) { - if (option_debug) - ast_log(LOG_DEBUG, "Setting NAT on RTP to %s\n", mode); + ast_debug(1, "Setting NAT on RTP to %s\n", mode); ast_rtp_setnat(p->rtp, natflags); } if (p->vrtp) { - if (option_debug) - ast_log(LOG_DEBUG, "Setting NAT on VRTP to %s\n", mode); + ast_debug(1, "Setting NAT on VRTP to %s\n", mode); ast_rtp_setnat(p->vrtp, natflags); } if (p->udptl) { - if (option_debug) - ast_log(LOG_DEBUG, "Setting NAT on UDPTL to %s\n", mode); + ast_debug(1, "Setting NAT on UDPTL to %s\n", mode); ast_udptl_setnat(p->udptl, natflags); } if (p->trtp) { - if (option_debug) - ast_log(LOG_DEBUG, "Setting NAT on TRTP to %s\n", mode); + ast_debug(1, "Setting NAT on TRTP to %s\n", mode); ast_rtp_setnat(p->trtp, natflags); } } @@ -3016,8 +2993,7 @@ static int create_addr_from_peer(struct sip_pvt *dialog, struct sip_peer *peer) else if (ast_udptl_get_error_correction_scheme(dialog->udptl) == UDPTL_ERROR_CORRECTION_NONE ) dialog->t38.capability |= T38FAX_UDP_EC_NONE; dialog->t38.capability |= T38FAX_RATE_MANAGEMENT_TRANSFERED_TCF; - if (option_debug > 1) - ast_log(LOG_DEBUG,"Our T38 capability (%d)\n", dialog->t38.capability); + ast_debug(2,"Our T38 capability (%d)\n", dialog->t38.capability); } dialog->t38.jointcapability = dialog->t38.capability; } else if (dialog->udptl) { @@ -3222,8 +3198,7 @@ static int sip_call(struct ast_channel *ast, char *dest, int timeout) p->options->replaces = ast_var_value(current); } else if (!strcasecmp(ast_var_name(current), "T38CALL")) { p->t38.state = T38_LOCAL_DIRECT; - if (option_debug) - ast_log(LOG_DEBUG,"T38State change to %d on channel %s\n", p->t38.state, ast->name); + ast_debug(1,"T38State change to %d on channel %s\n", p->t38.state, ast->name); } } @@ -3235,15 +3210,14 @@ static int sip_call(struct ast_channel *ast, char *dest, int timeout) char buf[BUFSIZ/2]; if (referer) { - if (sipdebug && option_debug > 2) - ast_log(LOG_DEBUG, "Call for %s transfered by %s\n", p->username, referer); + if (sipdebug) + ast_debug(3, "Call for %s transfered by %s\n", p->username, referer); snprintf(buf, sizeof(buf)-1, "-> %s (via %s)", p->cid_name, referer); } else snprintf(buf, sizeof(buf)-1, "-> %s", p->cid_name); ast_string_field_set(p, cid_name, buf); } - if (option_debug) - ast_log(LOG_DEBUG, "Outgoing Call for %s\n", p->username); + ast_debug(1, "Outgoing Call for %s\n", p->username); res = update_call_counter(p, INC_CALL_RINGING); @@ -3262,8 +3236,7 @@ static int sip_call(struct ast_channel *ast, char *dest, int timeout) int xmitres; p->t38.jointcapability = p->t38.capability; - if (option_debug > 1) - ast_log(LOG_DEBUG,"Our T38 capability (%d), joint T38 capability (%d)\n", p->t38.capability, p->t38.jointcapability); + ast_debug(2,"Our T38 capability (%d), joint T38 capability (%d)\n", p->t38.capability, p->t38.jointcapability); xmitres = transmit_invite(p, SIP_INVITE, 1, 2); if (xmitres == XMIT_ERROR) return -1; @@ -3281,15 +3254,13 @@ static int sip_call(struct ast_channel *ast, char *dest, int timeout) static void sip_registry_destroy(struct sip_registry *reg) { /* Really delete */ - if (option_debug > 2) - ast_log(LOG_DEBUG, "Destroying registry entry for %s@%s\n", reg->username, reg->hostname); + ast_debug(3, "Destroying registry entry for %s@%s\n", reg->username, reg->hostname); if (reg->call) { /* Clear registry before destroying to ensure we don't get reentered trying to grab the registry lock */ reg->call->registry = NULL; - if (option_debug > 2) - ast_log(LOG_DEBUG, "Destroying active SIP dialog for registry %s@%s\n", reg->username, reg->hostname); + ast_debug(3, "Destroying active SIP dialog for registry %s@%s\n", reg->username, reg->hostname); sip_destroy(reg->call); } if (reg->expire > -1) @@ -3308,13 +3279,12 @@ static void __sip_destroy(struct sip_pvt *p, int lockowner, int lockdialoglist) struct sip_pvt *cur, *prev = NULL; struct sip_pkt *cp; - if (sip_debug_test_pvt(p) || option_debug > 2) + if (sip_debug_test_pvt(p)) ast_verbose("Really destroying SIP dialog '%s' Method: %s\n", p->callid, sip_methods[p->method].text); if (ast_test_flag(&p->flags[0], SIP_INC_COUNT)) { update_call_counter(p, DEC_CALL_LIMIT); - if (option_debug > 1) - ast_log(LOG_DEBUG, "This call did not properly clean up call limits. Call ID %s\n", p->callid); + ast_debug(2, "This call did not properly clean up call limits. Call ID %s\n", p->callid); } /* Remove link from peer to subscription of MWI */ @@ -3358,8 +3328,7 @@ static void __sip_destroy(struct sip_pvt *p, int lockowner, int lockdialoglist) if (p->owner) { if (lockowner) ast_channel_lock(p->owner); - if (option_debug) - ast_log(LOG_DEBUG, "Detaching from %s\n", p->owner->name); + ast_debug(1, "Detaching from %s\n", p->owner->name); p->owner->tech_pvt = NULL; if (lockowner) ast_channel_unlock(p->owner); @@ -3429,8 +3398,7 @@ static int update_call_counter(struct sip_pvt *fup, int event) struct sip_user *u = NULL; struct sip_peer *p = NULL; - if (option_debug > 2) - ast_log(LOG_DEBUG, "Updating call counter for %s call\n", outgoing ? "outgoing" : "incoming"); + ast_debug(3, "Updating call counter for %s call\n", outgoing ? "outgoing" : "incoming"); /* Test if we need to check call limits, in order to avoid realtime lookups if we do not need it */ @@ -3451,8 +3419,7 @@ static int update_call_counter(struct sip_pvt *fup, int event) ast_copy_string(name, fup->peername, sizeof(name)); } if (!p && !u) { - if (option_debug > 1) - ast_log(LOG_DEBUG, "%s is not a local device, no call limit\n", name); + ast_debug(2, "%s is not a local device, no call limit\n", name); return 0; } @@ -3473,8 +3440,8 @@ static int update_call_counter(struct sip_pvt *fup, int event) /* Decrement onhold count if applicable */ if (ast_test_flag(&fup->flags[1], SIP_PAGE2_CALL_ONHOLD) && global_notifyhold) sip_peer_hold(fup, FALSE); - if (option_debug > 1 || sipdebug) - ast_log(LOG_DEBUG, "Call %s %s '%s' removed from call limit %d\n", outgoing ? "to" : "from", u ? "user":"peer", name, *call_limit); + if (sipdebug) + ast_debug(2, "Call %s %s '%s' removed from call limit %d\n", outgoing ? "to" : "from", u ? "user":"peer", name, *call_limit); break; case INC_CALL_RINGING: @@ -3499,8 +3466,8 @@ static int update_call_counter(struct sip_pvt *fup, int event) /* Continue */ ast_atomic_fetchadd_int(inuse, +1); ast_set_flag(&fup->flags[0], SIP_INC_COUNT); - if (option_debug > 1 || sipdebug) { - ast_log(LOG_DEBUG, "Call %s %s '%s' is %d out of %d\n", outgoing ? "to" : "from", u ? "user":"peer", name, *inuse, *call_limit); + if (sipdebug) { + ast_debug(2, "Call %s %s '%s' is %d out of %d\n", outgoing ? "to" : "from", u ? "user":"peer", name, *inuse, *call_limit); } break; @@ -3525,8 +3492,7 @@ static int update_call_counter(struct sip_pvt *fup, int event) /*! \brief Destroy SIP call structure */ static void sip_destroy(struct sip_pvt *p) { - if (option_debug > 2) - ast_log(LOG_DEBUG, "Destroying SIP dialog %s\n", p->callid); + ast_debug(3, "Destroying SIP dialog %s\n", p->callid); __sip_destroy(p, TRUE, TRUE); } @@ -3681,8 +3647,7 @@ static const char *hangup_cause2sip(int cause) case AST_CAUSE_NOTDEFINED: default: - if (option_debug) - ast_log(LOG_DEBUG, "AST hangup cause %d (no match found in SIP)\n", cause); + ast_debug(1, "AST hangup cause %d (no match found in SIP)\n", cause); return NULL; } @@ -3701,19 +3666,17 @@ static int sip_hangup(struct ast_channel *ast) struct ast_channel *oldowner = ast; if (!p) { - if (option_debug) - ast_log(LOG_DEBUG, "Asked to hangup channel that was not connected\n"); + ast_debug(1, "Asked to hangup channel that was not connected\n"); return 0; } if (ast_test_flag(&p->flags[0], SIP_DEFER_BYE_ON_TRANSFER)) { if (ast_test_flag(&p->flags[0], SIP_INC_COUNT)) { - if (option_debug && sipdebug) - ast_log(LOG_DEBUG, "update_call_counter(%s) - decrement call limit counter on hangup\n", p->username); + if (sipdebug) + ast_debug(1, "update_call_counter(%s) - decrement call limit counter on hangup\n", p->username); update_call_counter(p, DEC_CALL_LIMIT); } - if (option_debug >3) - ast_log(LOG_DEBUG, "SIP Transfer: Not hanging up right now... Rescheduling hangup for %s.\n", p->callid); + ast_debug(4, "SIP Transfer: Not hanging up right now... Rescheduling hangup for %s.\n", p->callid); if (p->autokillid > -1) sip_cancel_destroy(p); sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT); @@ -3723,21 +3686,18 @@ static int sip_hangup(struct ast_channel *ast) p->owner = NULL; /* Owner will be gone after we return, so take it away */ return 0; } - if (option_debug) { - if (ast_test_flag(ast, AST_FLAG_ZOMBIE) && p->refer && option_debug) - ast_log(LOG_DEBUG, "SIP Transfer: Hanging up Zombie channel %s after transfer ... Call-ID: %s\n", ast->name, p->callid); - else { - if (option_debug) - ast_log(LOG_DEBUG, "Hangup call %s, SIP callid %s)\n", ast->name, p->callid); - } - } - if (option_debug && ast_test_flag(ast, AST_FLAG_ZOMBIE)) - ast_log(LOG_DEBUG, "Hanging up zombie call. Be scared.\n"); + if (ast_test_flag(ast, AST_FLAG_ZOMBIE) && p->refer) + ast_debug(1, "SIP Transfer: Hanging up Zombie channel %s after transfer ... Call-ID: %s\n", ast->name, p->callid); + else + ast_debug(1, "Hangup call %s, SIP callid %s)\n", ast->name, p->callid); + + if (ast_test_flag(ast, AST_FLAG_ZOMBIE)) + ast_debug(1, "Hanging up zombie call. Be scared.\n"); sip_pvt_lock(p); if (ast_test_flag(&p->flags[0], SIP_INC_COUNT)) { - if (option_debug && sipdebug) - ast_log(LOG_DEBUG, "update_call_counter(%s) - decrement call limit counter on hangup\n", p->username); + if (sipdebug) + ast_debug(1, "update_call_counter(%s) - decrement call limit counter on hangup\n", p->username); update_call_counter(p, DEC_CALL_LIMIT); } @@ -3751,8 +3711,7 @@ static int sip_hangup(struct ast_channel *ast) /* In case of re-invites, the call might be UP even though we have an incomplete invite transaction */ if (p->invitestate < INV_COMPLETED && p->owner->_state != AST_STATE_UP) { needcancel = TRUE; - if (option_debug > 3) - ast_log(LOG_DEBUG, "Hanging up channel in state %s (not UP)\n", ast_state2str(ast->_state)); + ast_debug(4, "Hanging up channel in state %s (not UP)\n", ast_state2str(ast->_state)); } stop_media_flows(p); /* Immediately stop RTP, VRTP and UDPTL as applicable */ @@ -3889,12 +3848,10 @@ static int sip_answer(struct ast_channel *ast) try_suggested_sip_codec(p); ast_setstate(ast, AST_STATE_UP); - if (option_debug) - ast_log(LOG_DEBUG, "SIP answering channel: %s\n", ast->name); + ast_debug(1, "SIP answering channel: %s\n", ast->name); if (p->t38.state == T38_PEER_DIRECT) { p->t38.state = T38_ENABLED; - if (option_debug > 1) - ast_log(LOG_DEBUG,"T38State change to %d on channel %s\n", p->t38.state, ast->name); + ast_debug(2,"T38State change to %d on channel %s\n", p->t38.state, ast->name); res = transmit_response_with_t38_sdp(p, "200 OK", &p->initreq, XMIT_CRITICAL); } else res = transmit_response_with_sdp(p, "200 OK", &p->initreq, XMIT_CRITICAL); @@ -4003,10 +3960,10 @@ static int sip_fixup(struct ast_channel *oldchan, struct ast_channel *newchan) int ret = -1; struct sip_pvt *p; - if (newchan && ast_test_flag(newchan, AST_FLAG_ZOMBIE) && option_debug) - ast_log(LOG_DEBUG, "New channel is zombie\n"); - if (oldchan && ast_test_flag(oldchan, AST_FLAG_ZOMBIE) && option_debug) - ast_log(LOG_DEBUG, "Old channel is zombie\n"); + if (newchan && ast_test_flag(newchan, AST_FLAG_ZOMBIE)) + ast_debug(1, "New channel is zombie\n"); + if (oldchan && ast_test_flag(oldchan, AST_FLAG_ZOMBIE)) + ast_debug(1, "Old channel is zombie\n"); if (!newchan || !newchan->tech_pvt) { if (!newchan) @@ -4026,8 +3983,7 @@ static int sip_fixup(struct ast_channel *oldchan, struct ast_channel *newchan) p->owner = newchan; ret = 0; } - if (option_debug > 2) - ast_log(LOG_DEBUG, "SIP Fixup: New owner for dialogue %s: %s (Old parent: %s)\n", p->callid, p->owner->name, oldchan->name); + ast_debug(3, "SIP Fixup: New owner for dialogue %s: %s (Old parent: %s)\n", p->callid, p->owner->name, oldchan->name); sip_pvt_unlock(p); return ret; @@ -4252,15 +4208,13 @@ static struct ast_channel *sip_new(struct sip_pvt *i, int state, const char *tit /* Set the native formats for audio and merge in video */ tmp->nativeformats = ast_codec_choose(&i->prefs, what, 1) | video | text; - if (option_debug > 2) { - char buf[BUFSIZ]; - ast_log(LOG_DEBUG, "*** Our native formats are %s \n", ast_getformatname_multiple(buf, BUFSIZ, tmp->nativeformats)); - ast_log(LOG_DEBUG, "*** Joint capabilities are %s \n", ast_getformatname_multiple(buf, BUFSIZ, i->jointcapability)); - ast_log(LOG_DEBUG, "*** Our capabilities are %s \n", ast_getformatname_multiple(buf, BUFSIZ, i->capability)); - ast_log(LOG_DEBUG, "*** AST_CODEC_CHOOSE formats are %s \n", ast_getformatname_multiple(buf, BUFSIZ, ast_codec_choose(&i->prefs, what, 1))); - if (i->prefcodec) - ast_log(LOG_DEBUG, "*** Our preferred formats from the incoming channel are %s \n", ast_getformatname_multiple(buf, BUFSIZ, i->prefcodec)); - } + char buf[BUFSIZ]; + ast_debug(3, "*** Our native formats are %s \n", ast_getformatname_multiple(buf, BUFSIZ, tmp->nativeformats)); + ast_debug(3, "*** Joint capabilities are %s \n", ast_getformatname_multiple(buf, BUFSIZ, i->jointcapability)); + ast_debug(3, "*** Our capabilities are %s \n", ast_getformatname_multiple(buf, BUFSIZ, i->capability)); + ast_debug(3, "*** AST_CODEC_CHOOSE formats are %s \n", ast_getformatname_multiple(buf, BUFSIZ, ast_codec_choose(&i->prefs, what, 1))); + if (i->prefcodec) + ast_debug(3, "*** Our preferred formats from the incoming channel are %s \n", ast_getformatname_multiple(buf, BUFSIZ, i->prefcodec)); /* XXX Why are we choosing a codec from the native formats?? */ fmt = ast_best_codec(tmp->nativeformats); @@ -4283,12 +4237,10 @@ static struct ast_channel *sip_new(struct sip_pvt *i, int state, const char *tit needtext = i->jointcapability & AST_FORMAT_TEXT_MASK; /* Inbound call */ } - if (option_debug > 2) { - if (needvideo) - ast_log(LOG_DEBUG, "This channel can handle video! HOLLYWOOD next!\n"); - else - ast_log(LOG_DEBUG, "This channel will not be able to handle video.\n"); - } + if (needvideo) + ast_debug(3, "This channel can handle video! HOLLYWOOD next!\n"); + else + ast_debug(3, "This channel will not be able to handle video.\n"); @@ -4566,14 +4518,11 @@ static struct ast_frame *sip_rtp_read(struct ast_channel *ast, struct sip_pvt *p if (f && f->subclass != (p->owner->nativeformats & AST_FORMAT_AUDIO_MASK)) { if (!(f->subclass & p->jointcapability)) { - if (option_debug) { - ast_log(LOG_DEBUG, "Bogus frame of format '%s' received from '%s'!\n", - ast_getformatname(f->subclass), p->owner->name); - } + ast_debug(1, "Bogus frame of format '%s' received from '%s'!\n", + ast_getformatname(f->subclass), p->owner->name); return &ast_null_frame; } - if (option_debug) - ast_log(LOG_DEBUG, "Oooh, format changed to %d\n", f->subclass); + ast_debug(1, "Oooh, format changed to %d\n", f->subclass); p->owner->nativeformats = (p->owner->nativeformats & (AST_FORMAT_VIDEO_MASK | AST_FORMAT_TEXT_MASK)) | f->subclass; ast_set_read_format(p->owner, p->owner->readformat); ast_set_write_format(p->owner, p->owner->writeformat); @@ -4583,11 +4532,10 @@ static struct ast_frame *sip_rtp_read(struct ast_channel *ast, struct sip_pvt *p f = ast_dsp_process(p->owner, p->vad, f); if (f && f->frametype == AST_FRAME_DTMF) { if (ast_test_flag(&p->t38.t38support, SIP_PAGE2_T38SUPPORT_UDPTL) && f->subclass == 'f') { - if (option_debug) - ast_log(LOG_DEBUG, "Fax CNG detected on %s\n", ast->name); + ast_debug(1, "Fax CNG detected on %s\n", ast->name); *faxdetect = 1; - } else if (option_debug) { - ast_log(LOG_DEBUG, "* Detected inband DTMF '%c'\n", f->subclass); + } else { + ast_debug(1, "* Detected inband DTMF '%c'\n", f->subclass); } } } @@ -4611,16 +4559,13 @@ static struct ast_frame *sip_read(struct ast_channel *ast) if (faxdetected && ast_test_flag(&p->t38.t38support, SIP_PAGE2_T38SUPPORT_UDPTL) && (p->t38.state == T38_DISABLED) && !(ast_bridged_channel(ast))) { if (!ast_test_flag(&p->flags[0], SIP_GOTREFER)) { if (!p->pendinginvite) { - if (option_debug > 2) - ast_log(LOG_DEBUG, "Sending reinvite on SIP (%s) for T.38 negotiation.\n",ast->name); + ast_debug(3, "Sending reinvite on SIP (%s) for T.38 negotiation.\n",ast->name); p->t38.state = T38_LOCAL_REINVITE; transmit_reinvite_with_sdp(p, TRUE); - if (option_debug > 1) - ast_log(LOG_DEBUG, "T38 state changed to %d on channel %s\n", p->t38.state, ast->name); + ast_debug(2, "T38 state changed to %d on channel %s\n", p->t38.state, ast->name); } } else if (!ast_test_flag(&p->flags[0], SIP_PENDINGBYE)) { - if (option_debug > 2) - ast_log(LOG_DEBUG, "Deferring reinvite on SIP (%s) - it will be re-negotiated for T.38\n", ast->name); + ast_debug(3, "Deferring reinvite on SIP (%s) - it will be re-negotiated for T.38\n", ast->name); ast_set_flag(&p->flags[0], SIP_NEEDREINVITE); } } @@ -4799,8 +4744,7 @@ static struct sip_pvt *sip_alloc(ast_string_field callid, struct sockaddr_in *si p->next = dialoglist; dialoglist = p; dialoglist_unlock(); - if (option_debug) - ast_log(LOG_DEBUG, "Allocating new SIP dialog for %s - %s (%s)\n", callid ? callid : "(No Call-ID)", sip_methods[intended_method].text, p->rtp ? "With RTP" : "No RTP"); + ast_debug(1, "Allocating new SIP dialog for %s - %s (%s)\n", callid ? callid : "(No Call-ID)", sip_methods[intended_method].text, p->rtp ? "With RTP" : "No RTP"); return p; } @@ -4836,19 +4780,16 @@ static struct sip_pvt *find_call(struct sip_request *req, struct sockaddr_in *si tag = (req->method == SIP_RESPONSE) ? totag : fromtag; - if (option_debug > 4 ) - ast_log(LOG_DEBUG, "= Looking for Call ID: %s (Checking %s) --From tag %s --To-tag %s \n", callid, req->method==SIP_RESPONSE ? "To" : "From", fromtag, totag); + ast_debug(5, "= Looking for Call ID: %s (Checking %s) --From tag %s --To-tag %s \n", callid, req->method==SIP_RESPONSE ? "To" : "From", fromtag, totag); /* All messages must always have From: tag */ if (ast_strlen_zero(fromtag)) { - if (option_debug > 4 ) - ast_log(LOG_DEBUG, "%s request has no from tag, dropping callid: %s from: %s\n", sip_methods[req->method].text , callid, from ); + ast_debug(5, "%s request has no from tag, dropping callid: %s from: %s\n", sip_methods[req->method].text , callid, from ); return NULL; } /* reject requests that must always have a To: tag */ if (ast_strlen_zero(totag) && (req->method == SIP_ACK || req->method == SIP_BYE || req->method == SIP_INFO )) { - if (option_debug > 4) - ast_log(LOG_DEBUG, "%s must have a to tag. dropping callid: %s from: %s\n", sip_methods[req->method].text , callid, from ); + ast_debug(5, "%s must have a to tag. dropping callid: %s from: %s\n", sip_methods[req->method].text , callid, from ); return NULL; } } @@ -4865,8 +4806,7 @@ static struct sip_pvt *find_call(struct sip_request *req, struct sockaddr_in *si found = (!strcmp(p->callid, callid) && (!pedanticsipchecking || !tag || ast_strlen_zero(p->theirtag) || !strcmp(p->theirtag, tag))) ; - if (option_debug > 4) - ast_log(LOG_DEBUG, "= %s Their Call ID: %s Their Tag %s Our tag: %s\n", found ? "Found" : "No match", p->callid, p->theirtag, p->tag); + ast_debug(5, "= %s Their Call ID: %s Their Tag %s Our tag: %s\n", found ? "Found" : "No match", p->callid, p->theirtag, p->tag); /* If we get a new request within an existing to-tag - check the to tag as well */ if (pedanticsipchecking && found && req->method != SIP_RESPONSE) { /* SIP Request */ @@ -4878,8 +4818,8 @@ static struct sip_pvt *find_call(struct sip_request *req, struct sockaddr_in *si found = FALSE; /* This is not our packet */ } } - if (!found && option_debug > 4) - ast_log(LOG_DEBUG, "= Being pedantic: This is not our match on request: Call ID: %s Ourtag <null> Totag %s Method %s\n", p->callid, totag, sip_methods[req->method].text); + if (!found) + ast_debug(5, "= Being pedantic: This is not our match on request: Call ID: %s Ourtag <null> Totag %s Method %s\n", p->callid, totag, sip_methods[req->method].text); } @@ -4916,26 +4856,23 @@ static struct sip_pvt *find_call(struct sip_request *req, struct sockaddr_in *si Sorry, we apologize for the inconvienience */ transmit_response_using_temp(callid, sin, 1, intended_method, req, "500 Server internal error"); - if (option_debug > 3) - ast_log(LOG_DEBUG, "Failed allocating SIP dialog, sending 500 Server internal error and giving up\n"); + ast_debug(4, "Failed allocating SIP dialog, sending 500 Server internal error and giving up\n"); } } return p; } else if( sip_methods[intended_method].can_create == CAN_CREATE_DIALOG_UNSUPPORTED_METHOD) { /* A method we do not support, let's take it on the volley */ transmit_response_using_temp(callid, sin, 1, intended_method, req, "501 Method Not Implemented"); - if (option_debug > 1 ) - ast_log(LOG_DEBUG, "Got a request with unsupported SIP method.\n"); + ast_debug(2, "Got a request with unsupported SIP method.\n"); } else if (intended_method != SIP_RESPONSE && intended_method != SIP_ACK) { /* This is a request outside of a dialog that we don't know about */ transmit_response_using_temp(callid, sin, 1, intended_method, req, "481 Call leg/transaction does not exist"); - if (option_debug > 1) - ast_log(LOG_DEBUG, "That's odd... Got a request in unknown dialog. Callid %s\n", callid ? callid : "<unknown>"); + ast_debug(2, "That's odd... Got a request in unknown dialog. Callid %s\n", callid ? callid : "<unknown>"); } /* We do not respond to responses for dialogs that we don't know about, we just drop the session quickly */ - if (option_debug > 1 && intended_method == SIP_RESPONSE) - ast_log(LOG_DEBUG, "That's odd... Got a response on a call we dont know about. Callid %s\n", callid ? callid : "<unknown>"); + if (intended_method == SIP_RESPONSE) + ast_debug(2, "That's odd... Got a response on a call we dont know about. Callid %s\n", callid ? callid : "<unknown>"); return p; } @@ -5079,8 +5016,8 @@ static void parse_request(struct sip_request *req) *c = '\0'; else if (*c == '\n') { /* end of this line */ *c = '\0'; - if (sipdebug && option_debug > 3) - ast_log(LOG_DEBUG, "%7s %2d [%3d]: %s\n", + if (sipdebug) + ast_debug(4, "%7s %2d [%3d]: %s\n", req->headers < 0 ? "Header" : "Body", i, (int)strlen(dst[i]), dst[i]); if (ast_strlen_zero(dst[i]) && req->headers < 0) { @@ -5099,8 +5036,8 @@ static void parse_request(struct sip_request *req) but since some devices send without, we'll be generous in what we accept. */ if (!ast_strlen_zero(dst[i])) { - if (sipdebug && option_debug > 3) - ast_log(LOG_DEBUG, "%7s %2d [%3d]: %s\n", + if (sipdebug) + ast_debug(4, "%7s %2d [%3d]: %s\n", req->headers < 0 ? "Header" : "Body", i, (int)strlen(dst[i]), dst[i]); i++; @@ -5368,12 +5305,10 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req) if (p->owner && p->lastinvite) { p->t38.state = T38_PEER_REINVITE; /* T38 Offered in re-invite from remote party */ - if (option_debug > 1) - ast_log(LOG_DEBUG, "T38 state changed to %d on channel %s\n", p->t38.state, p->owner ? p->owner->name : "<none>" ); + ast_debug(2, "T38 state changed to %d on channel %s\n", p->t38.state, p->owner ? p->owner->name : "<none>" ); } else { p->t38.state = T38_PEER_DIRECT; /* T38 Offered directly from peer in first invite */ - if (option_debug > 1) - ast_log(LOG_DEBUG, "T38 state changed to %d on channel %s\n", p->t38.state, p->owner ? p->owner->name : "<none>"); + ast_debug(2, "T38 state changed to %d on channel %s\n", p->t38.state, p->owner ? p->owner->name : "<none>"); } } else ast_log(LOG_WARNING, "Unsupported SDP media type in offer: %s\n", m); @@ -5433,11 +5368,11 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req) sin.sin_port = htons(udptlportno); ast_udptl_set_peer(p->udptl, &sin); if (debug) - ast_log(LOG_DEBUG,"Peer T.38 UDPTL is at port %s:%d\n",ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port)); + ast_debug(1,"Peer T.38 UDPTL is at port %s:%d\n",ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port)); } else { ast_udptl_stop(p->udptl); if (debug) - ast_log(LOG_DEBUG, "Peer doesn't provide T.38 UDPTL\n"); + ast_debug(1, "Peer doesn't provide T.38 UDPTL\n"); } } @@ -5529,8 +5464,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req) framing = strtol(tmp, NULL, 10); if (framing == LONG_MIN || framing == LONG_MAX) { framing = 0; - if (option_debug) - ast_log(LOG_DEBUG, "Can't read framing from SDP: %s\n", a); + ast_debug(1, "Can't read framing from SDP: %s\n", a); } } if (framing && last_rtpmap_codec) { @@ -5542,8 +5476,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req) format = ast_rtp_codec_getformat(found_rtpmap_codecs[codec_n]); if (!format) /* non-codec or not found */ continue; - if (option_debug) - ast_log(LOG_DEBUG, "Setting framing for %d to %ld\n", format, framing); + ast_debug(1, "Setting framing for %d to %ld\n", format, framing); ast_codec_pref_setsize(pref, format, framing); } ast_rtp_codec_setpref(p->rtp, pref); @@ -5585,12 +5518,10 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req) while ((a = get_sdp_iterate(&iterator, req, "a"))[0] != '\0') { if ((sscanf(a, "T38FaxMaxBuffer:%d", &x) == 1)) { found = 1; - if (option_debug > 2) - ast_log(LOG_DEBUG, "MaxBufferSize:%d\n",x); + ast_debug(3, "MaxBufferSize:%d\n",x); } else if ((sscanf(a, "T38MaxBitRate:%d", &x) == 1)) { found = 1; - if (option_debug > 2) - ast_log(LOG_DEBUG,"T38MaxBitRate: %d\n",x); + ast_debug(3,"T38MaxBitRate: %d\n",x); switch (x) { case 14400: peert38capability |= T38FAX_RATE_14400 | T38FAX_RATE_12000 | T38FAX_RATE_9600 | T38FAX_RATE_7200 | T38FAX_RATE_4800 | T38FAX_RATE_2400; @@ -5613,49 +5544,42 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req) } } else if ((sscanf(a, "T38FaxVersion:%d", &x) == 1)) { found = 1; - if (option_debug > 2) - ast_log(LOG_DEBUG, "FaxVersion: %d\n",x); + ast_debug(3, "FaxVersion: %d\n",x); if (x == 0) peert38capability |= T38FAX_VERSION_0; else if (x == 1) peert38capability |= T38FAX_VERSION_1; } else if ((sscanf(a, "T38FaxMaxDatagram:%d", &x) == 1)) { found = 1; - if (option_debug > 2) - ast_log(LOG_DEBUG, "FaxMaxDatagram: %d\n",x); + ast_debug(3, "FaxMaxDatagram: %d\n",x); ast_udptl_set_far_max_datagram(p->udptl, x); ast_udptl_set_local_max_datagram(p->udptl, x); } else if ((sscanf(a, "T38FaxFillBitRemoval:%d", &x) == 1)) { found = 1; - if (option_debug > 2) - ast_log(LOG_DEBUG, "FillBitRemoval: %d\n",x); + ast_debug(3, "FillBitRemoval: %d\n",x); if (x == 1) peert38capability |= T38FAX_FILL_BIT_REMOVAL; } else if ((sscanf(a, "T38FaxTranscodingMMR:%d", &x) == 1)) { found = 1; - if (option_debug > 2) - ast_log(LOG_DEBUG, "Transcoding MMR: %d\n",x); + ast_debug(3, "Transcoding MMR: %d\n",x); if (x == 1) peert38capability |= T38FAX_TRANSCODING_MMR; } if ((sscanf(a, "T38FaxTranscodingJBIG:%d", &x) == 1)) { found = 1; - if (option_debug > 2) - ast_log(LOG_DEBUG, "Transcoding JBIG: %d\n",x); + ast_debug(3, "Transcoding JBIG: %d\n",x); if (x == 1) peert38capability |= T38FAX_TRANSCODING_JBIG; } else if ((sscanf(a, "T38FaxRateManagement:%255s", s) == 1)) { found = 1; - if (option_debug > 2) - ast_log(LOG_DEBUG, "RateManagement: %s\n", s); + ast_debug(3, "RateManagement: %s\n", s); if (!strcasecmp(s, "localTCF")) peert38capability |= T38FAX_RATE_MANAGEMENT_LOCAL_TCF; else if (!strcasecmp(s, "transferredTCF")) peert38capability |= T38FAX_RATE_MANAGEMENT_TRANSFERED_TCF; } else if ((sscanf(a, "T38FaxUdpEC:%255s", s) == 1)) { found = 1; - if (option_debug > 2) - ast_log(LOG_DEBUG, "UDP EC: %s\n", s); + ast_debug(3, "UDP EC: %s\n", s); if (!strcasecmp(s, "t38UDPRedundancy")) { peert38capability |= T38FAX_UDP_EC_REDUNDANCY; ast_udptl_set_error_correction_scheme(p->udptl, UDPTL_ERROR_CORRECTION_REDUNDANCY); @@ -5675,14 +5599,13 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req) p->t38.jointcapability |= (peert38capability & p->t38.capability); /* Put the lower of our's and peer's speed */ } if (debug) - ast_log(LOG_DEBUG, "Our T38 capability = (%d), peer T38 capability (%d), joint T38 capability (%d)\n", + ast_debug(1, "Our T38 capability = (%d), peer T38 capability (%d), joint T38 capability (%d)\n", p->t38.capability, p->t38.peercapability, p->t38.jointcapability); } else { p->t38.state = T38_DISABLED; - if (option_debug > 2) - ast_log(LOG_DEBUG, "T38 state changed to %d on channel %s\n", p->t38.state, p->owner ? p->owner->name : "<none>"); + ast_debug(3, "T38 state changed to %d on channel %s\n", p->t38.state, p->owner ? p->owner->name : "<none>"); } /* Now gather all of the codecs that we are asked for: */ @@ -5718,8 +5641,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req) /* Do NOT Change current setting */ return -1; } else { - if (option_debug > 2) - ast_log(LOG_DEBUG, "Have T.38 but no audio codecs, accepting offer anyway\n"); + ast_debug(3, "Have T.38 but no audio codecs, accepting offer anyway\n"); return 0; } } @@ -5771,21 +5693,18 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req) } /* Ok, we're going with this offer */ - if (option_debug > 1) { - char buf[BUFSIZ]; - ast_log(LOG_DEBUG, "We're settling with these formats: %s\n", ast_getformatname_multiple(buf, BUFSIZ, p->jointcapability)); - } + char buf[BUFSIZ]; + ast_debug(2, "We're settling with these formats: %s\n", ast_getformatname_multiple(buf, BUFSIZ, p->jointcapability)); if (!p->owner) /* There's no open channel owning us so we can return here. For a re-invite or so, we proceed */ return 0; - if (option_debug > 3) - ast_log(LOG_DEBUG, "We have an owner, now see if we need to change this call\n"); + ast_debug(4, "We have an owner, now see if we need to change this call\n"); if (!(p->owner->nativeformats & p->jointcapability) && (p->jointcapability & AST_FORMAT_AUDIO_MASK)) { if (debug) { char s1[BUFSIZ], s2[BUFSIZ]; - ast_log(LOG_DEBUG, "Oooh, we need to change our audio formats since our peer supports only %s and not %s\n", + ast_debug(1, "Oooh, we need to change our audio formats since our peer supports only %s and not %s\n", ast_getformatname_multiple(s1, BUFSIZ, p->jointcapability), ast_getformatname_multiple(s2, BUFSIZ, p->owner->nativeformats)); } @@ -6212,7 +6131,7 @@ static int reqprep(struct sip_request *req, struct sip_pvt *p, int sipmethod, in if (p->route && !ast_strlen_zero(p->route->hop) && strstr(p->route->hop,";lr") == NULL) { is_strict = TRUE; if (sipdebug) - ast_log(LOG_DEBUG, "Strict routing enforced for session %s\n", p->callid); + ast_debug(1, "Strict routing enforced for session %s\n", p->callid); } if (sipmethod == SIP_CANCEL) @@ -6589,32 +6508,25 @@ static int t38_get_rate(int t38cap) int maxrate = (t38cap & (T38FAX_RATE_14400 | T38FAX_RATE_12000 | T38FAX_RATE_9600 | T38FAX_RATE_7200 | T38FAX_RATE_4800 | T38FAX_RATE_2400)); if (maxrate & T38FAX_RATE_14400) { - if (option_debug > 1) - ast_log(LOG_DEBUG, "T38MaxFaxRate 14400 found\n"); + ast_debug(2, "T38MaxFaxRate 14400 found\n"); return 14400; } else if (maxrate & T38FAX_RATE_12000) { - if (option_debug > 1) - ast_log(LOG_DEBUG, "T38MaxFaxRate 12000 found\n"); + ast_debug(2, "T38MaxFaxRate 12000 found\n"); return 12000; } else if (maxrate & T38FAX_RATE_9600) { - if (option_debug > 1) - ast_log(LOG_DEBUG, "T38MaxFaxRate 9600 found\n"); + ast_debug(2, "T38MaxFaxRate 9600 found\n"); return 9600; } else if (maxrate & T38FAX_RATE_7200) { - if (option_debug > 1) - ast_log(LOG_DEBUG, "T38MaxFaxRate 7200 found\n"); + ast_debug(2, "T38MaxFaxRate 7200 found\n"); return 7200; } else if (maxrate & T38FAX_RATE_4800) { - if (option_debug > 1) - ast_log(LOG_DEBUG, "T38MaxFaxRate 4800 found\n"); + ast_debug(2, "T38MaxFaxRate 4800 found\n"); return 4800; } else if (maxrate & T38FAX_RATE_2400) { - if (option_debug > 1) - ast_log(LOG_DEBUG, "T38MaxFaxRate 2400 found\n"); + ast_debug(2, "T38MaxFaxRate 2400 found\n"); return 2400; } else { - if (option_debug > 1) - ast_log(LOG_DEBUG, "Strange, T38MaxFaxRate NOT found in peers T38 SDP.\n"); + ast_debug(2, "Strange, T38MaxFaxRate NOT found in peers T38 SDP.\n"); return 0; } } @@ -6665,13 +6577,13 @@ static int add_t38_sdp(struct sip_request *resp, struct sip_pvt *p) } if (debug) - ast_log(LOG_DEBUG, "T.38 UDPTL is at %s port %d\n", ast_inet_ntoa(p->ourip), ntohs(udptlsin.sin_port)); + ast_debug(1, "T.38 UDPTL is at %s port %d\n", ast_inet_ntoa(p->ourip), ntohs(udptlsin.sin_port)); /* We break with the "recommendation" and send our IP, in order that our peer doesn't have to ast_gethostbyname() us */ if (debug) { - ast_log(LOG_DEBUG, "Our T38 capability (%d), peer T38 capability (%d), joint capability (%d)\n", + ast_debug(1, "Our T38 capability (%d), peer T38 capability (%d), joint capability (%d)\n", p->t38.capability, p->t38.peercapability, p->t38.jointcapability); @@ -6841,12 +6753,10 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p) capability = p->jointcapability; - if (option_debug > 1) { - char codecbuf[BUFSIZ]; - ast_log(LOG_DEBUG, "** Our capability: %s Video flag: %s Text flag: %s\n", ast_getformatname_multiple(codecbuf, sizeof(codecbuf), capability), - ast_test_flag(&p->flags[0], SIP_NOVIDEO) ? "True" : "False", ast_test_flag(&p->flags[1], SIP_PAGE2_NOTEXT) ? "True" : "False"); - ast_log(LOG_DEBUG, "** Our prefcodec: %s \n", ast_getformatname_multiple(codecbuf, sizeof(codecbuf), p->prefcodec)); - } + char codecbuf[BUFSIZ]; + ast_debug(1, "** Our capability: %s Video flag: %s Text flag: %s\n", ast_getformatname_multiple(codecbuf, sizeof(codecbuf), capability), + ast_test_flag(&p->flags[0], SIP_NOVIDEO) ? "True" : "False", ast_test_flag(&p->flags[1], SIP_PAGE2_NOTEXT) ? "True" : "False"); + ast_debug(1, "** Our prefcodec: %s \n", ast_getformatname_multiple(codecbuf, sizeof(codecbuf), p->prefcodec)); #ifdef WHEN_WE_HAVE_T38_FOR_OTHER_TRANSPORTS if (ast_test_flag(&p->t38.t38support, SIP_PAGE2_T38SUPPORT_RTP)) { @@ -6859,10 +6769,9 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p) if ((capability & AST_FORMAT_VIDEO_MASK) && !ast_test_flag(&p->flags[0], SIP_NOVIDEO)) { if (p->vrtp) { needvideo = TRUE; - if (option_debug > 1) - ast_log(LOG_DEBUG, "This call needs video offers!\n"); - } else if (option_debug > 1) - ast_log(LOG_DEBUG, "This call needs video offers, but there's no video support enabled!\n"); + ast_debug(2, "This call needs video offers!\n"); + } else + ast_debug(2, "This call needs video offers, but there's no video support enabled!\n"); } /* Get our media addresses */ @@ -6889,10 +6798,9 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p) if (p->trtp) { if (sipdebug_text) ast_verbose("And we have a text rtp object\n"); needtext = TRUE; - if (option_debug > 1) - ast_log(LOG_DEBUG, "This call needs text offers! \n"); - } else if (option_debug > 1) - ast_log(LOG_DEBUG, "This call needs text offers, but there's no text support enabled ! \n"); + ast_debug(2, "This call needs text offers! \n"); + } else + ast_debug(2, "This call needs text offers, but there's no text support enabled ! \n"); } /* Ok, we need text. Let's add what we need for text and set codecs. @@ -7005,8 +6913,7 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p) debug); } - if (option_debug > 2) - ast_log(LOG_DEBUG, "-- Done with adding codecs to SDP\n"); + ast_debug(3, "-- Done with adding codecs to SDP\n"); if (!p->owner || !ast_internal_timing_enabled(p->owner)) ast_build_string(&a_audio_next, &a_audio_left, "a=silenceSupp:off - - - -\r\n"); @@ -7064,10 +6971,8 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p) /* Update lastrtprx when we send our SDP */ p->lastrtprx = p->lastrtptx = time(NULL); /* XXX why both ? */ - if (option_debug > 2) { - char buf[BUFSIZ]; - ast_log(LOG_DEBUG, "Done building SDP. Settling with this capability: %s\n", ast_getformatname_multiple(buf, BUFSIZ, capability)); - } + char buf[BUFSIZ]; + ast_debug(3, "Done building SDP. Settling with this capability: %s\n", ast_getformatname_multiple(buf, BUFSIZ, capability)); return AST_SUCCESS; } @@ -7124,8 +7029,7 @@ static int transmit_response_with_sdp(struct sip_pvt *p, const char *msg, const respprep(&resp, p, msg, req); if (p->rtp) { if (!p->autoframing && !ast_test_flag(&p->flags[0], SIP_OUTGOING)) { - if (option_debug) - ast_log(LOG_DEBUG, "Setting framing from config on incoming call\n"); + ast_debug(1, "Setting framing from config on incoming call\n"); ast_rtp_codec_setpref(p->rtp, &p->prefs); } try_suggested_sip_codec(p); @@ -7533,7 +7437,7 @@ static int transmit_invite(struct sip_pvt *p, int sipmethod, int sdp, int init) add_header(&req, headdup, content); if (sipdebug) - ast_log(LOG_DEBUG, "Adding SIP Header \"%s\" with content :%s: \n", headdup, content); + ast_debug(1, "Adding SIP Header \"%s\" with content :%s: \n", headdup, content); } } } @@ -7542,8 +7446,7 @@ static int transmit_invite(struct sip_pvt *p, int sipmethod, int sdp, int init) if (sdp) { if (p->udptl && p->t38.state == T38_LOCAL_DIRECT) { ast_udptl_offered_from_local(p->udptl, 1); - if (option_debug) - ast_log(LOG_DEBUG, "T38 is in state %d on channel %s\n", p->t38.state, p->owner ? p->owner->name : "<none>"); + ast_debug(1, "T38 is in state %d on channel %s\n", p->t38.state, p->owner ? p->owner->name : "<none>"); add_t38_sdp(&req, p); } else if (p->rtp) add_sdp(&req, p); @@ -8010,8 +7913,7 @@ static int transmit_register(struct sip_registry *r, int sipmethod, const char * ast_sched_del(sched, r->timeout); } r->timeout = ast_sched_add(sched, global_reg_timeout * 1000, sip_reg_timeout, r); - if (option_debug) - ast_log(LOG_DEBUG, "Scheduled a registration timeout for %s id #%d \n", r->hostname, r->timeout); + ast_debug(1, "Scheduled a registration timeout for %s id #%d \n", r->hostname, r->timeout); } if (strchr(r->username, '@')) { @@ -8069,7 +7971,7 @@ static int transmit_register(struct sip_registry *r, int sipmethod, const char * /* We have auth data to reuse, build a digest header! */ if (sipdebug) - ast_log(LOG_DEBUG, " >>> Re-using Auth data for %s@%s\n", r->username, r->hostname); + ast_debug(1, " >>> Re-using Auth data for %s@%s\n", r->username, r->hostname); ast_string_field_set(p, realm, r->realm); ast_string_field_set(p, nonce, r->nonce); ast_string_field_set(p, domain, r->domain); @@ -8137,8 +8039,8 @@ static int transmit_refer(struct sip_pvt *p, const char *dest) char *ttag, *ftag; char *theirtag = ast_strdupa(p->theirtag); - if (option_debug || sipdebug) - ast_log(LOG_DEBUG, "SIP transfer of %s to %s\n", p->callid, dest); + if (sipdebug) + ast_debug(1, "SIP transfer of %s to %s\n", p->callid, dest); /* Are we transfering an inbound or outbound call ? */ if (ast_test_flag(&p->flags[0], SIP_OUTGOING)) { @@ -8315,8 +8217,8 @@ static int expire_register(void *data) /* Do we need to release this peer from memory? Only for realtime peers and autocreated peers */ - if (option_debug > 2 && ast_test_flag(&peer->flags[0], SIP_REALTIME)) - ast_log(LOG_DEBUG,"-REALTIME- peer expired registration. Name: %s. Realtime peer objects now %d\n", peer->name, rpeerobjs); + if (ast_test_flag(&peer->flags[0], SIP_REALTIME)) + ast_debug(3,"-REALTIME- peer expired registration. Name: %s. Realtime peer objects now %d\n", peer->name, rpeerobjs); if (ast_test_flag(&peer->flags[1], SIP_PAGE2_SELFDESTRUCT) || ast_test_flag(&peer->flags[1], SIP_PAGE2_RTAUTOCLEAR)) { @@ -8376,9 +8278,8 @@ static void reg_source_db(struct sip_peer *peer) if (contact) ast_copy_string(peer->fullcontact, contact, sizeof(peer->fullcontact)); - if (option_debug > 1) - ast_log(LOG_DEBUG, "SIP Seeding peer from astdb: '%s' at %s@%s:%d for %d\n", - peer->name, peer->username, ast_inet_ntoa(in), port, expiry); + ast_debug(2, "SIP Seeding peer from astdb: '%s' at %s@%s:%d for %d\n", + peer->name, peer->username, ast_inet_ntoa(in), port, expiry); memset(&peer->addr, 0, sizeof(peer->addr)); peer->addr.sin_family = AF_INET; @@ -8626,8 +8527,7 @@ static void build_route(struct sip_pvt *p, struct sip_request *req, int backward /* Once a persistant route is set, don't fool with it */ if (p->route && p->route_persistant) { - if (option_debug) - ast_log(LOG_DEBUG, "build_route: Retaining previous route: <%s>\n", p->route->hop); + ast_debug(1, "build_route: Retaining previous route: <%s>\n", p->route->hop); return; } @@ -8658,8 +8558,7 @@ static void build_route(struct sip_pvt *p, struct sip_request *req, int backward if ((thishop = ast_malloc(sizeof(*thishop) + len))) { /* ast_calloc is not needed because all fields are initialized in this block */ ast_copy_string(thishop->hop, rr, len); - if (option_debug > 1) - ast_log(LOG_DEBUG, "build_route: Record-Route hop: <%s>\n", thishop->hop); + ast_debug(2, "build_route: Record-Route hop: <%s>\n", thishop->hop); /* Link in */ if (backwards) { /* Link in at head so they end up in reverse order */ @@ -8687,8 +8586,7 @@ static void build_route(struct sip_pvt *p, struct sip_request *req, int backward /* Can be multiple Contact headers, comma separated values - we just take the first */ contact = get_header(req, "Contact"); if (!ast_strlen_zero(contact)) { - if (option_debug > 1) - ast_log(LOG_DEBUG, "build_route: Contact hop: %s\n", contact); + ast_debug(2, "build_route: Contact hop: %s\n", contact); /* Look for <: delimited address */ c = strchr(contact, '<'); if (c) { @@ -9308,8 +9206,7 @@ static int get_destination(struct sip_pvt *p, struct sip_request *oreq) domain_context[0] = '\0'; if (!check_sip_domain(p->domain, domain_context, sizeof(domain_context))) { if (!allow_external_domains && (req->method == SIP_INVITE || req->method == SIP_REFER)) { - if (option_debug) - ast_log(LOG_DEBUG, "Got SIP %s to non-local domain '%s'; refusing request.\n", sip_methods[req->method].text, p->domain); + ast_debug(1, "Got SIP %s to non-local domain '%s'; refusing request.\n", sip_methods[req->method].text, p->domain); return -2; } } @@ -9351,8 +9248,8 @@ static struct sip_pvt *get_sip_pvt_byid_locked(const char *callid, const char *t struct sip_pvt *sip_pvt_ptr; - if (option_debug > 3 && totag) - ast_log(LOG_DEBUG, "Looking for callid %s (fromtag %s totag %s)\n", callid, fromtag ? fromtag : "<no fromtag>", totag ? totag : "<no totag>"); + if (totag) + ast_debug(4, "Looking for callid %s (fromtag %s totag %s)\n", callid, fromtag ? fromtag : "<no fromtag>", totag ? totag : "<no totag>"); /* Search dialogs and find the match */ dialoglist_lock(); @@ -9376,8 +9273,8 @@ static struct sip_pvt *get_sip_pvt_byid_locked(const char *callid, const char *t continue; } - if (option_debug > 3 && totag) - ast_log(LOG_DEBUG, "Matched %s call - their tag is %s Our tag is %s\n", + if (totag) + ast_debug(4, "Matched %s call - their tag is %s Our tag is %s\n", ast_test_flag(&sip_pvt_ptr->flags[0], SIP_OUTGOING) ? "OUTGOING": "INCOMING", sip_pvt_ptr->theirtag, sip_pvt_ptr->tag); @@ -9391,8 +9288,8 @@ static struct sip_pvt *get_sip_pvt_byid_locked(const char *callid, const char *t } } dialoglist_unlock(); - if (option_debug > 3 && !sip_pvt_ptr) - ast_log(LOG_DEBUG, "Found no match for callid %s to-tag %s from-tag %s\n", callid, totag, fromtag); + if (!sip_pvt_ptr) + ast_debug(4, "Found no match for callid %s to-tag %s from-tag %s\n", callid, totag, fromtag); return sip_pvt_ptr; } @@ -9497,12 +9394,10 @@ static int get_refer_info(struct sip_pvt *transferer, struct sip_request *outgoi ast_copy_string(referdata->replaces_callid_fromtag, ptr, sizeof(referdata->replaces_callid_fromtag)); } - if (option_debug > 1) { - if (!pedanticsipchecking) - ast_log(LOG_DEBUG,"Attended transfer: Will use Replace-Call-ID : %s (No check of from/to tags)\n", referdata->replaces_callid ); - else - ast_log(LOG_DEBUG,"Attended transfer: Will use Replace-Call-ID : %s F-tag: %s T-tag: %s\n", referdata->replaces_callid, referdata->replaces_callid_fromtag ? referdata->replaces_callid_fromtag : "<none>", referdata->replaces_callid_totag ? referdata->replaces_callid_totag : "<none>" ); - } + if (!pedanticsipchecking) + ast_debug(2,"Attended transfer: Will use Replace-Call-ID : %s (No check of from/to tags)\n", referdata->replaces_callid ); + else + ast_debug(2,"Attended transfer: Will use Replace-Call-ID : %s F-tag: %s T-tag: %s\n", referdata->replaces_callid, referdata->replaces_callid_fromtag ? referdata->replaces_callid_fromtag : "<none>", referdata->replaces_callid_totag ? referdata->replaces_callid_totag : "<none>" ); } } @@ -9598,8 +9493,7 @@ static int get_also_info(struct sip_pvt *p, struct sip_request *oreq) } if (ast_exists_extension(NULL, transfer_context, c, 1, NULL)) { /* This is a blind transfer */ - if (option_debug) - ast_log(LOG_DEBUG,"SIP Bye-also transfer to Extension %s@%s \n", c, transfer_context); + ast_debug(1,"SIP Bye-also transfer to Extension %s@%s \n", c, transfer_context); ast_copy_string(referdata->refer_to, c, sizeof(referdata->refer_to)); ast_copy_string(referdata->referred_by, "", sizeof(referdata->referred_by)); ast_copy_string(referdata->refer_contact, "", sizeof(referdata->refer_contact)); @@ -11669,17 +11563,17 @@ static void sip_dump_history(struct sip_pvt *dialog) return; } - ast_log(LOG_DEBUG, "\n---------- SIP HISTORY for '%s' \n", dialog->callid); + ast_debug(1, "\n---------- SIP HISTORY for '%s' \n", dialog->callid); if (dialog->subscribed) - ast_log(LOG_DEBUG, " * Subscription\n"); + ast_debug(1, " * Subscription\n"); else - ast_log(LOG_DEBUG, " * SIP Call\n"); + ast_debug(1, " * SIP Call\n"); if (dialog->history) AST_LIST_TRAVERSE(dialog->history, hist, list) - ast_log(LOG_DEBUG, " %-3.3d. %s\n", ++x, hist->event); + ast_debug(1, " %-3.3d. %s\n", ++x, hist->event); if (!x) - ast_log(LOG_DEBUG, "Call '%s' has no history\n", dialog->callid); - ast_log(LOG_DEBUG, "\n---------- END SIP HISTORY for '%s' \n", dialog->callid); + ast_debug(1, "Call '%s' has no history\n", dialog->callid); + ast_debug(1, "\n---------- END SIP HISTORY for '%s' \n", dialog->callid); } @@ -11803,8 +11697,7 @@ static void handle_request_info(struct sip_pvt *p, struct sip_request *req) } ast_unlock_call_features(); - if (option_debug) - ast_log(LOG_DEBUG, "Got a Request to Record the channel, state %s\n", c); + ast_debug(1, "Got a Request to Record the channel, state %s\n", c); transmit_response(p, "200 OK", req); return; } @@ -12015,8 +11908,7 @@ static int do_proxy_auth(struct sip_pvt *p, struct sip_request *req, enum sip_au p->authtries++; auth_headers(code, &header, &respheader); - if (option_debug > 1) - ast_log(LOG_DEBUG, "Auth attempt %d on %s\n", p->authtries, sip_methods[sipmethod].text); + ast_debug(2, "Auth attempt %d on %s\n", p->authtries, sip_methods[sipmethod].text); memset(digest, 0, sizeof(digest)); if (reply_digest(p, req, header, sipmethod, digest, sizeof(digest) )) { /* No way to authenticate */ @@ -12138,7 +12030,7 @@ static int build_reply_digest(struct sip_pvt *p, int method, char* digest, int d secret = auth->secret; md5secret = auth->md5secret; if (sipdebug) - ast_log(LOG_DEBUG,"Using realm %s authentication for call %s\n", p->realm, p->callid); + ast_debug(1,"Using realm %s authentication for call %s\n", p->realm, p->callid); } else { /* No authentication, use peer or register= config */ username = p->authname; @@ -12545,8 +12437,7 @@ static void parse_moved_contact(struct sip_pvt *p, struct sip_request *req) e = strchr(s, '/'); if (e) *e = '\0'; - if (option_debug) - ast_log(LOG_DEBUG, "Found promiscuous redirection to 'SIP/%s'\n", s); + ast_debug(2, "Found promiscuous redirection to 'SIP/%s'\n", s); if (p->owner) ast_string_field_build(p->owner, call_forward, "SIP/%s", s); } else { @@ -12567,8 +12458,7 @@ static void parse_moved_contact(struct sip_pvt *p, struct sip_request *req) e = strchr(s, ';'); /* And username ; parameters? */ if (e) *e = '\0'; - if (option_debug > 1) - ast_log(LOG_DEBUG, "Received 302 Redirect to extension '%s' (domain %s)\n", s, domain); + ast_debug(2, "Received 302 Redirect to extension '%s' (domain %s)\n", s, domain); if (p->owner) { pbx_builtin_setvar_helper(p->owner, "SIPDOMAIN", domain); ast_string_field_set(p->owner, call_forward, s); @@ -12590,8 +12480,7 @@ static void check_pendings(struct sip_pvt *p) ast_clear_flag(&p->flags[0], SIP_PENDINGBYE); sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT); } else if (ast_test_flag(&p->flags[0], SIP_NEEDREINVITE)) { - if (option_debug) - ast_log(LOG_DEBUG, "Sending pending reinvite on '%s'\n", p->callid); + ast_debug(2, "Sending pending reinvite on '%s'\n", p->callid); /* Didn't get to reinvite yet, so do it now */ transmit_reinvite_with_sdp(p, FALSE); ast_clear_flag(&p->flags[0], SIP_NEEDREINVITE); @@ -12607,16 +12496,13 @@ static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, stru int reinvite = (p->owner && p->owner->_state == AST_STATE_UP); struct ast_channel *bridgepeer = NULL; - if (option_debug > 3) { - if (reinvite) - ast_log(LOG_DEBUG, "SIP response %d to RE-invite on %s call %s\n", resp, outgoing ? "outgoing" : "incoming", p->callid); - else - ast_log(LOG_DEBUG, "SIP response %d to standard invite\n", resp); - } + if (reinvite) + ast_debug(4, "SIP response %d to RE-invite on %s call %s\n", resp, outgoing ? "outgoing" : "incoming", p->callid); + else + ast_debug(4, "SIP response %d to standard invite\n", resp); if (ast_test_flag(&p->flags[0], SIP_ALREADYGONE)) { /* This call is already gone */ - if (option_debug) - ast_log(LOG_DEBUG, "Got response on call that is already terminated: %s (ignoring)\n", p->callid); + ast_debug(1, "Got response on call that is already terminated: %s (ignoring)\n", p->callid); return; } @@ -12741,31 +12627,25 @@ static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, stru sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT); } } else { - if (option_debug > 1) - ast_log(LOG_DEBUG, "Strange... The other side of the bridge does not have a udptl struct\n"); + ast_debug(2, "Strange... The other side of the bridge does not have a udptl struct\n"); sip_pvt_lock(bridgepvt); bridgepvt->t38.state = T38_DISABLED; sip_pvt_unlock(bridgepvt); - if (option_debug) - ast_log(LOG_DEBUG,"T38 state changed to %d on channel %s\n", bridgepvt->t38.state, bridgepeer->tech->type); + ast_debug(1,"T38 state changed to %d on channel %s\n", bridgepvt->t38.state, bridgepeer->tech->type); p->t38.state = T38_DISABLED; - if (option_debug > 1) - ast_log(LOG_DEBUG,"T38 state changed to %d on channel %s\n", p->t38.state, p->owner ? p->owner->name : "<none>"); + ast_debug(2,"T38 state changed to %d on channel %s\n", p->t38.state, p->owner ? p->owner->name : "<none>"); } } else { /* Other side is not a SIP channel */ - if (option_debug > 1) - ast_log(LOG_DEBUG, "Strange... The other side of the bridge is not a SIP channel\n"); + ast_debug(2, "Strange... The other side of the bridge is not a SIP channel\n"); p->t38.state = T38_DISABLED; - if (option_debug > 1) - ast_log(LOG_DEBUG,"T38 state changed to %d on channel %s\n", p->t38.state, p->owner ? p->owner->name : "<none>"); + ast_debug(2,"T38 state changed to %d on channel %s\n", p->t38.state, p->owner ? p->owner->name : "<none>"); } } if ((p->t38.state == T38_LOCAL_REINVITE) || (p->t38.state == T38_LOCAL_DIRECT)) { /* If there was T38 reinvite and we are supposed to answer with 200 OK than this should set us to T38 negotiated mode */ p->t38.state = T38_ENABLED; - if (option_debug) - ast_log(LOG_DEBUG, "T38 changed state to %d on channel %s\n", p->t38.state, p->owner ? p->owner->name : "<none>"); + ast_debug(1, "T38 changed state to %d on channel %s\n", p->t38.state, p->owner ? p->owner->name : "<none>"); } if (!ast_test_flag(req, SIP_PKT_IGNORE) && p->owner) { @@ -12916,8 +12796,7 @@ static void handle_response_refer(struct sip_pvt *p, int resp, char *rest, struc /* The transferee is now sending INVITE to target */ p->refer->status = REFER_ACCEPTED; /* Now wait for next message */ - if (option_debug > 2) - ast_log(LOG_DEBUG, "Got 202 accepted on transfer\n"); + ast_debug(3, "Got 202 accepted on transfer\n"); /* We should hang along, waiting for NOTIFY's here */ break; @@ -13042,11 +12921,9 @@ static int handle_response_register(struct sip_pvt *p, int resp, char *rest, str r->regtime = time(NULL); /* Reset time of last succesful registration */ manager_event(EVENT_FLAG_SYSTEM, "Registry", "ChannelType: SIP\r\nDomain: %s\r\nStatus: %s\r\n", r->hostname, regstate2str(r->regstate)); r->regattempts = 0; - if (option_debug) - ast_log(LOG_DEBUG, "Registration successful\n"); + ast_debug(1, "Registration successful\n"); if (r->timeout > -1) { - if (option_debug) - ast_log(LOG_DEBUG, "Cancelling timeout %d\n", r->timeout); + ast_debug(1, "Cancelling timeout %d\n", r->timeout); ast_sched_del(sched, r->timeout); } r->timeout=-1; @@ -13240,8 +13117,8 @@ static void handle_response(struct sip_pvt *p, int resp, char *rest, struct sip_ if (!p->refer) { ast_log(LOG_WARNING, "Notify answer on an owned channel? - %s\n", p->owner->name); ast_queue_hangup(p->owner); - } else if (option_debug > 3) - ast_log(LOG_DEBUG, "Got OK on REFER Notify message\n"); + } else + ast_debug(4, "Got OK on REFER Notify message\n"); } else { if (p->subscribed == NONE) ast_set_flag(&p->flags[0], SIP_NEEDDESTROY); @@ -13308,8 +13185,7 @@ static void handle_response(struct sip_pvt *p, int resp, char *rest, struct sip_ res = handle_response_register(p, resp, rest, req, seqno); else if (sipmethod == SIP_BYE) { ast_set_flag(&p->flags[0], SIP_NEEDDESTROY); - if (option_debug) - ast_log(LOG_DEBUG, "Got timeout on bye. Thanks for the answer. Now, kill this call\n"); + ast_debug(4, "Got timeout on bye. Thanks for the answer. Now, kill this call\n"); } else { if (owner) ast_queue_control(p->owner, AST_CONTROL_CONGESTION); @@ -13346,8 +13222,7 @@ static void handle_response(struct sip_pvt *p, int resp, char *rest, struct sip_ if (sipmethod == SIP_INVITE) handle_response_invite(p, resp, rest, req, seqno); else { - if (option_debug) - ast_log(LOG_DEBUG, "Got 491 on %s, unspported. Call ID %s\n", sip_methods[sipmethod].text, p->callid); + ast_debug(1, "Got 491 on %s, unspported. Call ID %s\n", sip_methods[sipmethod].text, p->callid); ast_set_flag(&p->flags[0], SIP_NEEDDESTROY); } break; @@ -13393,8 +13268,7 @@ static void handle_response(struct sip_pvt *p, int resp, char *rest, struct sip_ is yet another failure of not having a layer 2 (again, YAY IETF for thinking ahead). So we treat this as a call forward and hope we end up at the right place... */ - if (option_debug) - ast_log(LOG_DEBUG, "Hairpin detected, setting up call forward for what it's worth\n"); + ast_debug(1, "Hairpin detected, setting up call forward for what it's worth\n"); if (p->owner) ast_string_field_build(p->owner, call_forward, "Local/%s@%s", p->username, p->context); @@ -13461,16 +13335,14 @@ static void handle_response(struct sip_pvt *p, int resp, char *rest, struct sip_ if (sipmethod == SIP_INVITE) { handle_response_invite(p, resp, rest, req, seqno); } else if (sipmethod == SIP_CANCEL) { - if (option_debug) - ast_log(LOG_DEBUG, "Got 200 OK on CANCEL\n"); + ast_debug(1, "Got 200 OK on CANCEL\n"); /* Wait for 487, then destroy */ } else if (sipmethod == SIP_NOTIFY) { /* They got the notify, this is the end */ if (p->owner) { if (p->refer) { - if (option_debug) - ast_log(LOG_DEBUG, "Got 200 OK on NOTIFY for transfer\n"); + ast_debug(1, "Got 200 OK on NOTIFY for transfer\n"); } else ast_log(LOG_WARNING, "Notify answer on an owned channel?\n"); /* ast_queue_hangup(p->owner); Disabled */ @@ -13512,7 +13384,7 @@ static void handle_response(struct sip_pvt *p, int resp, char *rest, struct sip_ } else if (sipmethod == SIP_BYE) { ast_set_flag(&p->flags[0], SIP_NEEDDESTROY); } else if (sipdebug) { - ast_log (LOG_DEBUG, "Remote host can't match request %s to call '%s'. Giving up\n", sip_methods[sipmethod].text, p->callid); + ast_debug(1, "Remote host can't match request %s to call '%s'. Giving up\n", sip_methods[sipmethod].text, p->callid); } break; case 501: /* Not Implemented */ @@ -13579,8 +13451,7 @@ static void *sip_park_thread(void *stuff) ast_log(LOG_ERROR, "Missing channels for parking! Transferer %s Transferee %s\n", transferer ? "<available>" : "<missing>", transferee ? "<available>" : "<missing>" ); return NULL; } - if (option_debug > 3) - ast_log(LOG_DEBUG, "SIP Park: Transferer channel %s, Transferee %s\n", transferer->name, transferee->name); + ast_debug(4, "SIP Park: Transferer channel %s, Transferee %s\n", transferer->name, transferee->name); ast_channel_lock(transferee); if (ast_do_masquerade(transferee)) { @@ -13613,13 +13484,11 @@ static void *sip_park_thread(void *stuff) transmit_notify_with_sipfrag(transferer->tech_pvt, d->seqno, "200 OK", TRUE); transferer->hangupcause = AST_CAUSE_NORMAL_CLEARING; ast_hangup(transferer); /* This will cause a BYE */ - if (option_debug) - ast_log(LOG_DEBUG, "SIP Call parked on extension '%d'\n", ext); + ast_debug(1, "SIP Call parked on extension '%d'\n", ext); } else { transmit_notify_with_sipfrag(transferer->tech_pvt, d->seqno, "503 Service Unavailable", TRUE); append_history(transferer->tech_pvt, "SIPpark","Parking failed\n"); - if (option_debug) - ast_log(LOG_DEBUG, "SIP Call parked failed \n"); + ast_debug(1, "SIP Call parked failed \n"); /* Do not hangup call */ } return NULL; @@ -13687,12 +13556,10 @@ static int sip_park(struct ast_channel *chan1, struct ast_channel *chan2, struct ast_channel_unlock(transferer); if (!transferer || !transferee) { if (!transferer) { - if (option_debug) - ast_log(LOG_DEBUG, "No transferer channel, giving up parking\n"); + ast_debug(1, "No transferer channel, giving up parking\n"); } if (!transferee) { - if (option_debug) - ast_log(LOG_DEBUG, "No transferee channel, giving up parking\n"); + ast_debug(1, "No transferee channel, giving up parking\n"); } return -1; } @@ -13737,40 +13604,36 @@ static int attempt_transfer(struct sip_dual *transferer, struct sip_dual *target /* We will try to connect the transferee with the target and hangup all channels to the transferer */ - if (option_debug > 3) { - ast_log(LOG_DEBUG, "Sip transfer:--------------------\n"); - if (transferer->chan1) - ast_log(LOG_DEBUG, "-- Transferer to PBX channel: %s State %s\n", transferer->chan1->name, ast_state2str(transferer->chan1->_state)); - else - ast_log(LOG_DEBUG, "-- No transferer first channel - odd??? \n"); - if (target->chan1) - ast_log(LOG_DEBUG, "-- Transferer to PBX second channel (target): %s State %s\n", target->chan1->name, ast_state2str(target->chan1->_state)); - else - ast_log(LOG_DEBUG, "-- No target first channel ---\n"); - if (transferer->chan2) - ast_log(LOG_DEBUG, "-- Bridged call to transferee: %s State %s\n", transferer->chan2->name, ast_state2str(transferer->chan2->_state)); - else - ast_log(LOG_DEBUG, "-- No bridged call to transferee\n"); - if (target->chan2) - ast_log(LOG_DEBUG, "-- Bridged call to transfer target: %s State %s\n", target->chan2 ? target->chan2->name : "<none>", target->chan2 ? ast_state2str(target->chan2->_state) : "(none)"); - else - ast_log(LOG_DEBUG, "-- No target second channel ---\n"); - ast_log(LOG_DEBUG, "-- END Sip transfer:--------------------\n"); - } + ast_debug(4, "Sip transfer:--------------------\n"); + if (transferer->chan1) + ast_debug(4, "-- Transferer to PBX channel: %s State %s\n", transferer->chan1->name, ast_state2str(transferer->chan1->_state)); + else + ast_debug(4, "-- No transferer first channel - odd??? \n"); + if (target->chan1) + ast_debug(4, "-- Transferer to PBX second channel (target): %s State %s\n", target->chan1->name, ast_state2str(target->chan1->_state)); + else + ast_debug(4, "-- No target first channel ---\n"); + if (transferer->chan2) + ast_debug(4, "-- Bridged call to transferee: %s State %s\n", transferer->chan2->name, ast_state2str(transferer->chan2->_state)); + else + ast_debug(4, "-- No bridged call to transferee\n"); + if (target->chan2) + ast_debug(4, "-- Bridged call to transfer target: %s State %s\n", target->chan2 ? target->chan2->name : "<none>", target->chan2 ? ast_state2str(target->chan2->_state) : "(none)"); + else + ast_debug(4, "-- No target second channel ---\n"); + ast_debug(4, "-- END Sip transfer:--------------------\n"); if (transferer->chan2) { /* We have a bridge on the transferer's channel */ peera = transferer->chan1; /* Transferer - PBX -> transferee channel * the one we hangup */ peerb = target->chan1; /* Transferer - PBX -> target channel - This will get lost in masq */ peerc = transferer->chan2; /* Asterisk to Transferee */ peerd = target->chan2; /* Asterisk to Target */ - if (option_debug > 2) - ast_log(LOG_DEBUG, "SIP transfer: Four channels to handle\n"); + ast_debug(3, "SIP transfer: Four channels to handle\n"); } else if (target->chan2) { /* Transferer has no bridge (IVR), but transferee */ peera = target->chan1; /* Transferer to PBX -> target channel */ peerb = transferer->chan1; /* Transferer to IVR*/ peerc = target->chan2; /* Asterisk to Target */ peerd = transferer->chan2; /* Nothing */ - if (option_debug > 2) - ast_log(LOG_DEBUG, "SIP transfer: Three channels to handle\n"); + ast_debug(3, "SIP transfer: Three channels to handle\n"); } if (peera && peerb && peerc && (peerb != peerc)) { @@ -13793,13 +13656,12 @@ static int attempt_transfer(struct sip_dual *transferer, struct sip_dual *target peerb->cdr = peerc->cdr; peerc->cdr = NULL; - if (option_debug > 3) - ast_log(LOG_DEBUG, "SIP transfer: trying to masquerade %s into %s\n", peerc->name, peerb->name); + ast_debug(4, "SIP transfer: trying to masquerade %s into %s\n", peerc->name, peerb->name); if (ast_channel_masquerade(peerb, peerc)) { ast_log(LOG_WARNING, "Failed to masquerade %s into %s\n", peerb->name, peerc->name); res = -1; } else - ast_log(LOG_DEBUG, "SIP transfer: Succeeded to masquerade channels.\n"); + ast_debug(4, "SIP transfer: Succeeded to masquerade channels.\n"); return res; } else { ast_log(LOG_NOTICE, "SIP Transfer attempted with no appropriate bridged calls to transfer\n"); @@ -13849,8 +13711,8 @@ static int handle_request_notify(struct sip_pvt *p, struct sip_request *req, str eventid = sep; } - if (option_debug > 1 && sipdebug) - ast_log(LOG_DEBUG, "Got NOTIFY Event: %s\n", event); + if (sipdebug) + ast_debug(2, "Got NOTIFY Event: %s\n", event); if (strcmp(event, "refer")) { /* We don't understand this event. */ @@ -13910,8 +13772,7 @@ static int handle_request_notify(struct sip_pvt *p, struct sip_request *req, str If there are several REFERs in the same dialog, we need to match the ID of the event header... */ - if (option_debug > 2) - ast_log(LOG_DEBUG, "* SIP Transfer NOTIFY Attachment: \n---%s\n---\n", buf); + ast_debug(3, "* SIP Transfer NOTIFY Attachment: \n---%s\n---\n", buf); cmd = ast_skip_blanks(buf); code = cmd; /* We are at SIP/2.0 */ @@ -14007,20 +13868,17 @@ static int handle_invite_replaces(struct sip_pvt *p, struct sip_request *req, in if (!(targetcall = ast_bridged_channel(replacecall))) { /* We have no bridge */ if (!earlyreplace) { - if (option_debug > 1) - ast_log(LOG_DEBUG, " Attended transfer attempted to replace call with no bridge (maybe ringing). Channel %s!\n", replacecall->name); + ast_debug(2, " Attended transfer attempted to replace call with no bridge (maybe ringing). Channel %s!\n", replacecall->name); oneleggedreplace = 1; } } - if (option_debug > 3 && targetcall && targetcall->_state == AST_STATE_RINGING) - ast_log(LOG_DEBUG, "SIP transfer: Target channel is in ringing state\n"); + if (targetcall && targetcall->_state == AST_STATE_RINGING) + ast_debug(4, "SIP transfer: Target channel is in ringing state\n"); - if (option_debug > 3) { - if (targetcall) - ast_log(LOG_DEBUG, "SIP transfer: Invite Replace incoming channel should bridge to channel %s while hanging up channel %s\n", targetcall->name, replacecall->name); - else - ast_log(LOG_DEBUG, "SIP transfer: Invite Replace incoming channel should replace and hang up channel %s (one call leg)\n", replacecall->name); - } + if (targetcall) + ast_debug(4, "SIP transfer: Invite Replace incoming channel should bridge to channel %s while hanging up channel %s\n", targetcall->name, replacecall->name); + else + ast_debug(4, "SIP transfer: Invite Replace incoming channel should replace and hang up channel %s (one call leg)\n", replacecall->name); if (ast_test_flag(req, SIP_PKT_IGNORE)) { ast_log(LOG_NOTICE, "Ignoring this INVITE with replaces in a stupid way.\n"); @@ -14068,8 +13926,7 @@ static int handle_invite_replaces(struct sip_pvt *p, struct sip_request *req, in /* Stop music on hold and other generators */ ast_quiet_chan(replacecall); ast_quiet_chan(targetcall); - if (option_debug > 3) - ast_log(LOG_DEBUG, "Invite/Replaces: preparing to masquerade %s into %s\n", c->name, replacecall->name); + ast_debug(4, "Invite/Replaces: preparing to masquerade %s into %s\n", c->name, replacecall->name); /* Unlock clone, but not original (replacecall) */ ast_channel_unlock(c); @@ -14082,8 +13939,8 @@ static int handle_invite_replaces(struct sip_pvt *p, struct sip_request *req, in /* Prepare the masquerade - if this does not happen, we will be gone */ if(ast_channel_masquerade(replacecall, c)) ast_log(LOG_ERROR, "Failed to masquerade C into Replacecall\n"); - else if (option_debug > 3) - ast_log(LOG_DEBUG, "Invite/Replaces: Going to masquerade %s into %s\n", c->name, replacecall->name); + else + ast_debug(4, "Invite/Replaces: Going to masquerade %s into %s\n", c->name, replacecall->name); /* The masquerade will happen as soon as someone reads a frame from the channel */ @@ -14096,8 +13953,7 @@ static int handle_invite_replaces(struct sip_pvt *p, struct sip_request *req, in if ((f = ast_read(replacecall))) { /* Force the masq to happen */ ast_frfree(f); f = NULL; - if (option_debug > 3) - ast_log(LOG_DEBUG, "Invite/Replace: Could successfully read frame from RING channel!\n"); + ast_debug(4, "Invite/Replace: Could successfully read frame from RING channel!\n"); } else { ast_log(LOG_WARNING, "Invite/Replace: Could not read frame from RING channel \n"); } @@ -14108,8 +13964,7 @@ static int handle_invite_replaces(struct sip_pvt *p, struct sip_request *req, in /* Masq ok */ ast_frfree(f); f = NULL; - if (option_debug > 2) - ast_log(LOG_DEBUG, "Invite/Replace: Could successfully read frame from channel! Masq done.\n"); + ast_debug(3, "Invite/Replace: Could successfully read frame from channel! Masq done.\n"); } else { ast_log(LOG_WARNING, "Invite/Replace: Could not read frame from channel. Transfer failed\n"); } @@ -14118,23 +13973,21 @@ static int handle_invite_replaces(struct sip_pvt *p, struct sip_request *req, in sip_pvt_unlock(p->refer->refer_call); ast_setstate(c, AST_STATE_DOWN); - if (option_debug > 3) { - struct ast_channel *test; - ast_log(LOG_DEBUG, "After transfer:----------------------------\n"); - ast_log(LOG_DEBUG, " -- C: %s State %s\n", c->name, ast_state2str(c->_state)); - if (replacecall) - ast_log(LOG_DEBUG, " -- replacecall: %s State %s\n", replacecall->name, ast_state2str(replacecall->_state)); - if (p->owner) { - ast_log(LOG_DEBUG, " -- P->owner: %s State %s\n", p->owner->name, ast_state2str(p->owner->_state)); - test = ast_bridged_channel(p->owner); - if (test) - ast_log(LOG_DEBUG, " -- Call bridged to P->owner: %s State %s\n", test->name, ast_state2str(test->_state)); - else - ast_log(LOG_DEBUG, " -- No call bridged to C->owner \n"); - } else - ast_log(LOG_DEBUG, " -- No channel yet \n"); - ast_log(LOG_DEBUG, "End After transfer:----------------------------\n"); - } + struct ast_channel *test; + ast_debug(4, "After transfer:----------------------------\n"); + ast_debug(4, " -- C: %s State %s\n", c->name, ast_state2str(c->_state)); + if (replacecall) + ast_debug(4, " -- replacecall: %s State %s\n", replacecall->name, ast_state2str(replacecall->_state)); + if (p->owner) { + ast_debug(4, " -- P->owner: %s State %s\n", p->owner->name, ast_state2str(p->owner->_state)); + test = ast_bridged_channel(p->owner); + if (test) + ast_debug(4, " -- Call bridged to P->owner: %s State %s\n", test->name, ast_state2str(test->_state)); + else + ast_debug(4, " -- No call bridged to C->owner \n"); + } else + ast_debug(4, " -- No channel yet \n"); + ast_debug(4, "End After transfer:----------------------------\n"); ast_channel_unlock(p->owner); /* Unlock new owner */ sip_pvt_unlock(p); /* Unlock SIP structure */ @@ -14201,8 +14054,7 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int if (!ast_test_flag(req, SIP_PKT_IGNORE) && p->pendinginvite) { /* We already have a pending invite. Sorry. You are on hold. */ transmit_response(p, "491 Request Pending", req); - if (option_debug) - ast_log(LOG_DEBUG, "Got INVITE on call where we already have pending INVITE, deferring that - %s\n", p->callid); + ast_debug(1, "Got INVITE on call where we already have pending INVITE, deferring that - %s\n", p->callid); /* Don't destroy dialog here */ return 0; } @@ -14217,15 +14069,14 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int int error = 0; if (p->owner) { - if (option_debug > 2) - ast_log(LOG_DEBUG, "INVITE w Replaces on existing call? Refusing action. [%s]\n", p->callid); + ast_debug(3, "INVITE w Replaces on existing call? Refusing action. [%s]\n", p->callid); transmit_response(p, "400 Bad request", req); /* The best way to not not accept the transfer */ /* Do not destroy existing call */ return -1; } - if (sipdebug && option_debug > 2) - ast_log(LOG_DEBUG, "INVITE part of call transfer. Replaces [%s]\n", p_replaces); + if (sipdebug) + ast_debug(3, "INVITE part of call transfer. Replaces [%s]\n", p_replaces); /* Create a buffer we can manipulate */ replace_id = ast_strdupa(p_replaces); ast_uri_decode(replace_id); @@ -14260,8 +14111,8 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int } } - if (sipdebug && option_debug > 3) - ast_log(LOG_DEBUG,"Invite/replaces: Will use Replace-Call-ID : %s Fromtag: %s Totag: %s\n", replace_id, fromtag ? fromtag : "<no from tag>", totag ? totag : "<no to tag>"); + if (sipdebug) + ast_debug(4,"Invite/replaces: Will use Replace-Call-ID : %s Fromtag: %s Totag: %s\n", replace_id, fromtag ? fromtag : "<no from tag>", totag ? totag : "<no to tag>"); /* Try to find call that we are replacing @@ -14326,8 +14177,8 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int check_via(p, req); copy_request(&p->initreq, req); /* Save this INVITE as the transaction basis */ - if (sipdebug && option_debug) - ast_log(LOG_DEBUG, "Initializing initreq for method %s - callid %s\n", sip_methods[req->method].text, p->callid); + if (sipdebug) + ast_debug(1, "Initializing initreq for method %s - callid %s\n", sip_methods[req->method].text, p->callid); if (!p->owner) { /* Not a re-invite */ if (debug) ast_verbose("Using INVITE request as basis request - %s\n", p->callid); @@ -14346,8 +14197,7 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int } } else { p->jointcapability = p->capability; - if (option_debug) - ast_log(LOG_DEBUG, "Hm.... No sdp for the moment\n"); + ast_debug(1, "Hm.... No sdp for the moment\n"); } if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY)) /* This is a response, note what it was for */ append_history(p, "ReInv", "Re-invite received"); @@ -14385,14 +14235,12 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int transmit_response_reliable(p, "488 Not acceptable here", req); p->invitestate = INV_COMPLETED; sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT); - if (option_debug) - ast_log(LOG_DEBUG, "No compatible codecs for this SIP call.\n"); + ast_debug(1, "No compatible codecs for this SIP call.\n"); return -1; } } else { /* No SDP in invite, call control session */ p->jointcapability = p->capability; - if (option_debug > 1) - ast_log(LOG_DEBUG, "No SDP in Invite, third party call control\n"); + ast_debug(2, "No SDP in Invite, third party call control\n"); } /* Queue NULL frame to prod ast_rtp_bridge if appropriate */ @@ -14407,8 +14255,7 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int /* Check number of concurrent calls -vs- incoming limit HERE */ - if (option_debug) - ast_log(LOG_DEBUG, "Checking SIP call limits for device %s\n", p->username); + ast_debug(1, "Checking SIP call limits for device %s\n", p->username); if ((res = update_call_counter(p, INC_CALL_LIMIT))) { if (res < 0) { ast_log(LOG_NOTICE, "Failed to place call for user %s, too many calls\n", p->username); @@ -14458,11 +14305,11 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int } } } else { - if (option_debug > 1 && sipdebug) { + if (sipdebug) { if (!ast_test_flag(req, SIP_PKT_IGNORE)) - ast_log(LOG_DEBUG, "Got a SIP re-invite for call %s\n", p->callid); + ast_debug(2, "Got a SIP re-invite for call %s\n", p->callid); else - ast_log(LOG_DEBUG, "Got a SIP re-transmit of INVITE for call %s\n", p->callid); + ast_debug(2, "Got a SIP re-transmit of INVITE for call %s\n", p->callid); } reinvite = 1; c = p->owner; @@ -14473,8 +14320,8 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int if (replace_id) { /* Attended transfer or call pickup - we're the target */ /* Go and take over the target call */ - if (sipdebug && option_debug > 3) - ast_log(LOG_DEBUG, "Sending this call to the invite/replcaes handler %s\n", p->callid); + if (sipdebug) + ast_debug(4, "Sending this call to the invite/replcaes handler %s\n", p->callid); return handle_invite_replaces(p, req, debug, ast_test_flag(req, SIP_PKT_IGNORE), seqno, sin); } @@ -14482,8 +14329,7 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int if (c) { /* We have a call -either a new call or an old one (RE-INVITE) */ switch(c->_state) { case AST_STATE_DOWN: - if (option_debug > 1) - ast_log(LOG_DEBUG, "%s: New call is still down.... Trying... \n", c->name); + ast_debug(2, "%s: New call is still down.... Trying... \n", c->name); transmit_response(p, "100 Trying", req); p->invitestate = INV_PROCEEDING; ast_setstate(c, AST_STATE_RING); @@ -14556,8 +14402,7 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int p->invitestate = INV_PROCEEDING; break; case AST_STATE_UP: - if (option_debug > 1) - ast_log(LOG_DEBUG, "%s: This call is UP.... \n", c->name); + ast_debug(2, "%s: This call is UP.... \n", c->name); if (p->t38.state == T38_PEER_REINVITE) { struct ast_channel *bridgepeer = NULL; @@ -14577,8 +14422,7 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int sip_pvt_lock(bridgepvt); bridgepvt->t38.state = T38_DISABLED; sip_pvt_unlock(bridgepvt); - if (option_debug > 1) - ast_log(LOG_DEBUG,"T38 state changed to %d on channel %s\n", bridgepvt->t38.state, bridgepeer->name); + ast_debug(2,"T38 state changed to %d on channel %s\n", bridgepvt->t38.state, bridgepeer->name); if (ast_test_flag(req, SIP_PKT_IGNORE)) transmit_response(p, "488 Not acceptable here", req); else @@ -14589,8 +14433,7 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int /* The other side is already setup for T.38 most likely so we need to acknowledge this too */ transmit_response_with_t38_sdp(p, "200 OK", req, XMIT_CRITICAL); p->t38.state = T38_ENABLED; - if (option_debug) - ast_log(LOG_DEBUG, "T38 state changed to %d on channel %s\n", p->t38.state, p->owner ? p->owner->name : "<none>"); + ast_debug(1, "T38 state changed to %d on channel %s\n", p->t38.state, p->owner ? p->owner->name : "<none>"); } } else { /* Other side is not a SIP channel */ @@ -14599,8 +14442,7 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int else transmit_response_reliable(p, "488 Not acceptable here", req); p->t38.state = T38_DISABLED; - if (option_debug > 1) - ast_log(LOG_DEBUG,"T38 state changed to %d on channel %s\n", p->t38.state, p->owner ? p->owner->name : "<none>"); + ast_debug(2,"T38 state changed to %d on channel %s\n", p->t38.state, p->owner ? p->owner->name : "<none>"); if (!p->lastinvite) /* Only destroy if this is *not* a re-invite */ sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT); @@ -14609,8 +14451,7 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int /* we are not bridged in a call */ transmit_response_with_t38_sdp(p, "200 OK", req, XMIT_CRITICAL); p->t38.state = T38_ENABLED; - if (option_debug) - ast_log(LOG_DEBUG,"T38 state changed to %d on channel %s\n", p->t38.state, p->owner ? p->owner->name : "<none>"); + ast_debug(1,"T38 state changed to %d on channel %s\n", p->t38.state, p->owner ? p->owner->name : "<none>"); } } else if (p->t38.state == T38_DISABLED) { /* Channel doesn't have T38 offered or enabled */ int sendok = TRUE; @@ -14692,8 +14533,7 @@ static int local_attended_transfer(struct sip_pvt *transferer, struct sip_dual * return -1; } /* Fall through for remote transfers that we did not find locally */ - if (option_debug > 2) - ast_log(LOG_DEBUG, "SIP attended transfer: Not our call - generating INVITE with replaces\n"); + ast_debug(3, "SIP attended transfer: Not our call - generating INVITE with replaces\n"); return 0; } @@ -14701,8 +14541,7 @@ static int local_attended_transfer(struct sip_pvt *transferer, struct sip_dual * transmit_response(transferer, "202 Accepted", req); append_history(transferer, "Xfer", "Refer accepted"); if (!targetcall_pvt->owner) { /* No active channel */ - if (option_debug > 3) - ast_log(LOG_DEBUG, "SIP attended transfer: Error: No owner of target call\n"); + ast_debug(4, "SIP attended transfer: Error: No owner of target call\n"); /* Cancel transfer */ transmit_notify_with_sipfrag(transferer, seqno, "503 Service Unavailable", TRUE); append_history(transferer, "Xfer", "Refer failed"); @@ -14719,22 +14558,20 @@ static int local_attended_transfer(struct sip_pvt *transferer, struct sip_dual * if (!target.chan2 || !(target.chan2->_state == AST_STATE_UP || target.chan2->_state == AST_STATE_RINGING) ) { /* Wrong state of new channel */ - if (option_debug > 3) { - if (target.chan2) - ast_log(LOG_DEBUG, "SIP attended transfer: Error: Wrong state of target call: %s\n", ast_state2str(target.chan2->_state)); - else if (target.chan1->_state != AST_STATE_RING) - ast_log(LOG_DEBUG, "SIP attended transfer: Error: No target channel\n"); - else - ast_log(LOG_DEBUG, "SIP attended transfer: Attempting transfer in ringing state\n"); - } + if (target.chan2) + ast_debug(4, "SIP attended transfer: Error: Wrong state of target call: %s\n", ast_state2str(target.chan2->_state)); + else if (target.chan1->_state != AST_STATE_RING) + ast_debug(4, "SIP attended transfer: Error: No target channel\n"); + else + ast_debug(4, "SIP attended transfer: Attempting transfer in ringing state\n"); } /* Transfer */ - if (option_debug > 3 && sipdebug) { + if (sipdebug) { if (current->chan2) /* We have two bridges */ - ast_log(LOG_DEBUG, "SIP attended transfer: trying to bridge %s and %s\n", target.chan1->name, current->chan2->name); + ast_debug(4, "SIP attended transfer: trying to bridge %s and %s\n", target.chan1->name, current->chan2->name); else /* One bridge, propably transfer of IVR/voicemail etc */ - ast_log(LOG_DEBUG, "SIP attended transfer: trying to make %s take over (masq) %s\n", target.chan1->name, current->chan1->name); + ast_debug(4, "SIP attended transfer: trying to make %s take over (masq) %s\n", target.chan1->name, current->chan1->name); } ast_set_flag(&transferer->flags[0], SIP_DEFER_BYE_ON_TRANSFER); /* Delay hangup */ @@ -14759,8 +14596,7 @@ static int local_attended_transfer(struct sip_pvt *transferer, struct sip_dual * append_history(transferer, "Xfer", "Refer succeeded"); transferer->refer->status = REFER_200OK; if (targetcall_pvt->owner) { - if (option_debug) - ast_log(LOG_DEBUG, "SIP attended transfer: Unlocking channel %s\n", targetcall_pvt->owner->name); + ast_debug(1, "SIP attended transfer: Unlocking channel %s\n", targetcall_pvt->owner->name); ast_channel_unlock(targetcall_pvt->owner); } } @@ -14844,8 +14680,7 @@ static int handle_request_refer(struct sip_pvt *p, struct sip_request *req, int if (!p->owner) { /* This is a REFER outside of an existing SIP dialog */ /* We can't handle that, so decline it */ - if (option_debug > 2) - ast_log(LOG_DEBUG, "Call %s: Declined REFER, outside of dialog...\n", p->callid); + ast_debug(3, "Call %s: Declined REFER, outside of dialog...\n", p->callid); transmit_response(p, "603 Declined (No dialog)", req); if (!ast_test_flag(req, SIP_PKT_IGNORE)) { append_history(p, "Xfer", "Refer failed. Outside of dialog."); @@ -14888,14 +14723,14 @@ static int handle_request_refer(struct sip_pvt *p, struct sip_request *req, int case -2: /* Syntax error */ transmit_response(p, "400 Bad Request (Refer-to missing)", req); append_history(p, "Xfer", "Refer failed. Refer-to missing."); - if (ast_test_flag(req, SIP_PKT_DEBUG) && option_debug) - ast_log(LOG_DEBUG, "SIP transfer to black hole can't be handled (no refer-to: )\n"); + if (ast_test_flag(req, SIP_PKT_DEBUG)) + ast_debug(1, "SIP transfer to black hole can't be handled (no refer-to: )\n"); break; case -3: transmit_response(p, "603 Declined (Non sip: uri)", req); append_history(p, "Xfer", "Refer failed. Non SIP uri"); - if (ast_test_flag(req, SIP_PKT_DEBUG) && option_debug) - ast_log(LOG_DEBUG, "SIP transfer to non-SIP uri denied\n"); + if (ast_test_flag(req, SIP_PKT_DEBUG)) + ast_debug(1, "SIP transfer to non-SIP uri denied\n"); break; default: /* Refer-to extension not found, fake a failed transfer */ @@ -14903,8 +14738,8 @@ static int handle_request_refer(struct sip_pvt *p, struct sip_request *req, int append_history(p, "Xfer", "Refer failed. Bad extension."); transmit_notify_with_sipfrag(p, seqno, "404 Not found", TRUE); ast_clear_flag(&p->flags[0], SIP_GOTREFER); - if (ast_test_flag(req, SIP_PKT_DEBUG) && option_debug) - ast_log(LOG_DEBUG, "SIP transfer to bad extension: %s\n", p->refer->refer_to); + if (ast_test_flag(req, SIP_PKT_DEBUG)) + ast_debug(1, "SIP transfer to bad extension: %s\n", p->refer->refer_to); break; } return 0; @@ -14915,14 +14750,14 @@ static int handle_request_refer(struct sip_pvt *p, struct sip_request *req, int /* If we do not support SIP domains, all transfers are local */ if (allow_external_domains && check_sip_domain(p->refer->refer_to_domain, NULL, 0)) { p->refer->localtransfer = 1; - if (sipdebug && option_debug > 2) - ast_log(LOG_DEBUG, "This SIP transfer is local : %s\n", p->refer->refer_to_domain); + if (sipdebug) + ast_debug(3, "This SIP transfer is local : %s\n", p->refer->refer_to_domain); } else if (AST_LIST_EMPTY(&domain_list)) { /* This PBX don't bother with SIP domains, so all transfers are local */ p->refer->localtransfer = 1; } else - if (sipdebug && option_debug > 2) - ast_log(LOG_DEBUG, "This SIP transfer is to a remote SIP extension (remote domain %s)\n", p->refer->refer_to_domain); + if (sipdebug) + ast_debug(3, "This SIP transfer is to a remote SIP extension (remote domain %s)\n", p->refer->refer_to_domain); /* Is this a repeat of a current request? Ignore it */ /* Don't know what else to do right now. */ @@ -14960,15 +14795,15 @@ static int handle_request_refer(struct sip_pvt *p, struct sip_request *req, int /* Find the other part of the bridge (2) - transferee */ current.chan2 = ast_bridged_channel(current.chan1); - if (sipdebug && option_debug > 2) - ast_log(LOG_DEBUG, "SIP %s transfer: Transferer channel %s, transferee channel %s\n", p->refer->attendedtransfer ? "attended" : "blind", current.chan1->name, current.chan2 ? current.chan2->name : "<none>"); + if (sipdebug) + ast_debug(3, "SIP %s transfer: Transferer channel %s, transferee channel %s\n", p->refer->attendedtransfer ? "attended" : "blind", current.chan1->name, current.chan2 ? current.chan2->name : "<none>"); if (!current.chan2 && !p->refer->attendedtransfer) { /* No bridged channel, propably IVR or echo or similar... */ /* Guess we should masquerade or something here */ /* Until we figure it out, refuse transfer of such calls */ - if (sipdebug && option_debug > 2) - ast_log(LOG_DEBUG,"Refused SIP transfer on non-bridged channel.\n"); + if (sipdebug) + ast_debug(3,"Refused SIP transfer on non-bridged channel.\n"); p->refer->status = REFER_FAILED; append_history(p, "Xfer", "Refer failed. Non-bridged channel."); transmit_response(p, "603 Declined", req); @@ -14976,8 +14811,8 @@ static int handle_request_refer(struct sip_pvt *p, struct sip_request *req, int } if (current.chan2) { - if (sipdebug && option_debug > 3) - ast_log(LOG_DEBUG, "Got SIP transfer, applying to bridged peer '%s'\n", current.chan2->name); + if (sipdebug) + ast_debug(4, "Got SIP transfer, applying to bridged peer '%s'\n", current.chan2->name); ast_queue_control(current.chan1, AST_CONTROL_UNHOLD); } @@ -14989,8 +14824,8 @@ static int handle_request_refer(struct sip_pvt *p, struct sip_request *req, int if ((res = local_attended_transfer(p, ¤t, req, seqno))) return res; /* We're done with the transfer */ /* Fall through for remote transfers that we did not find locally */ - if (sipdebug && option_debug > 3) - ast_log(LOG_DEBUG, "SIP attended transfer: Still not our call - generating INVITE with replaces\n"); + if (sipdebug) + ast_debug(4, "SIP attended transfer: Still not our call - generating INVITE with replaces\n"); /* Fallthrough if we can't find the call leg internally */ } @@ -15004,8 +14839,8 @@ static int handle_request_refer(struct sip_pvt *p, struct sip_request *req, int ast_clear_flag(&p->flags[0], SIP_GOTREFER); p->refer->status = REFER_200OK; append_history(p, "Xfer", "REFER to call parking."); - if (sipdebug && option_debug > 3) - ast_log(LOG_DEBUG, "SIP transfer to parking: trying to park %s. Parked by %s\n", current.chan2->name, current.chan1->name); + if (sipdebug) + ast_debug(4, "SIP transfer to parking: trying to park %s. Parked by %s\n", current.chan2->name, current.chan1->name); sip_park(current.chan2, current.chan1, req, seqno); return res; } @@ -15014,8 +14849,7 @@ static int handle_request_refer(struct sip_pvt *p, struct sip_request *req, int transmit_response(p, "202 Accepted", req); if (current.chan1 && current.chan2) { - if (option_debug > 2) - ast_log(LOG_DEBUG, "chan1->name: %s\n", current.chan1->name); + ast_debug(3, "chan1->name: %s\n", current.chan1->name); pbx_builtin_setvar_helper(current.chan1, "BLINDTRANSFER", current.chan2->name); } if (current.chan2) { @@ -15077,8 +14911,7 @@ static int handle_request_refer(struct sip_pvt *p, struct sip_request *req, int if (!res) { /* Success - we have a new channel */ - if (option_debug > 2) - ast_log(LOG_DEBUG, "%s transfer succeeded. Telling transferer.\n", p->refer->attendedtransfer? "Attended" : "Blind"); + ast_debug(3, "%s transfer succeeded. Telling transferer.\n", p->refer->attendedtransfer? "Attended" : "Blind"); transmit_notify_with_sipfrag(p, seqno, "200 Ok", TRUE); if (p->refer->localtransfer) p->refer->status = REFER_200OK; @@ -15091,8 +14924,7 @@ static int handle_request_refer(struct sip_pvt *p, struct sip_request *req, int res = 0; } else { ast_clear_flag(&p->flags[0], SIP_DEFER_BYE_ON_TRANSFER); /* Don't delay hangup */ - if (option_debug > 2) - ast_log(LOG_DEBUG, "%s transfer failed. Resuming original call.\n", p->refer->attendedtransfer? "Attended" : "Blind"); + ast_debug(3, "%s transfer failed. Resuming original call.\n", p->refer->attendedtransfer? "Attended" : "Blind"); append_history(p, "Xfer", "Refer failed."); /* Failure of some kind */ p->refer->status = REFER_FAILED; @@ -15114,8 +14946,7 @@ static int handle_request_cancel(struct sip_pvt *p, struct sip_request *req) if (p->owner && p->owner->_state == AST_STATE_UP) { /* This call is up, cancel is ignored, we need a bye */ transmit_response(p, "200 OK", req); - if (option_debug) - ast_log(LOG_DEBUG, "Got CANCEL on an answered call. Ignoring... \n"); + ast_debug(1, "Got CANCEL on an answered call. Ignoring... \n"); return 0; } @@ -15217,8 +15048,8 @@ static int handle_request_bye(struct sip_pvt *p, struct sip_request *req) p->invitestate = INV_TERMINATED; copy_request(&p->initreq, req); - if (sipdebug && option_debug) - ast_log(LOG_DEBUG, "Initializing initreq for method %s - callid %s\n", sip_methods[req->method].text, p->callid); + if (sipdebug) + ast_debug(1, "Initializing initreq for method %s - callid %s\n", sip_methods[req->method].text, p->callid); check_via(p, req); sip_alreadygone(p); @@ -15274,12 +15105,10 @@ static int handle_request_bye(struct sip_pvt *p, struct sip_request *req) } } else if (p->owner) { ast_queue_hangup(p->owner); - if (option_debug > 2) - ast_log(LOG_DEBUG, "Received bye, issuing owner hangup\n"); + ast_debug(3, "Received bye, issuing owner hangup\n"); } else { sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT); - if (option_debug > 2) - ast_log(LOG_DEBUG, "Received bye, no owner, selfdestruct soon.\n"); + ast_debug(3, "Received bye, no owner, selfdestruct soon.\n"); } transmit_response(p, "200 OK", req); @@ -15317,16 +15146,13 @@ static int handle_request_subscribe(struct sip_pvt *p, struct sip_request *req, /* For transfers, this could happen, but since we haven't seen it happening, let us just refuse this */ transmit_response(p, "403 Forbidden (within dialog)", req); /* Do not destroy session, since we will break the call if we do */ - if (option_debug) - ast_log(LOG_DEBUG, "Got a subscription within the context of another call, can't handle that - %s (Method %s)\n", p->callid, sip_methods[p->initreq.method].text); + ast_debug(1, "Got a subscription within the context of another call, can't handle that - %s (Method %s)\n", p->callid, sip_methods[p->initreq.method].text); return 0; } else if (ast_test_flag(req, SIP_PKT_DEBUG)) { - if (option_debug) { - if (resubscribe) - ast_log(LOG_DEBUG, "Got a re-subscribe on existing subscription %s\n", p->callid); - else - ast_log(LOG_DEBUG, "Got a new subscription %s (possibly with auth)\n", p->callid); - } + if (resubscribe) + ast_debug(1, "Got a re-subscribe on existing subscription %s\n", p->callid); + else + ast_debug(1, "Got a new subscription %s (possibly with auth)\n", p->callid); } } @@ -15345,8 +15171,8 @@ static int handle_request_subscribe(struct sip_pvt *p, struct sip_request *req, ast_verbose("Creating new subscription\n"); copy_request(&p->initreq, req); - if (option_debug > 3 && sipdebug) - ast_log(LOG_DEBUG, "Initializing initreq for method %s - callid %s\n", sip_methods[req->method].text, p->callid); + if (sipdebug) + ast_debug(4, "Initializing initreq for method %s - callid %s\n", sip_methods[req->method].text, p->callid); check_via(p, req); } else if (ast_test_flag(req, SIP_PKT_DEBUG) && ast_test_flag(req, SIP_PKT_IGNORE)) ast_verbose("Ignoring this SUBSCRIBE request\n"); @@ -15354,8 +15180,7 @@ static int handle_request_subscribe(struct sip_pvt *p, struct sip_request *req, /* Find parameters to Event: header value and remove them for now */ if (ast_strlen_zero(eventheader)) { transmit_response(p, "489 Bad Event", req); - if (option_debug > 1) - ast_log(LOG_DEBUG, "Received SIP subscribe for unknown event package: <none>\n"); + ast_debug(2, "Received SIP subscribe for unknown event package: <none>\n"); ast_set_flag(&p->flags[0], SIP_NEEDDESTROY); return 0; } @@ -15468,8 +15293,7 @@ static int handle_request_subscribe(struct sip_pvt *p, struct sip_request *req, if (!ast_strlen_zero(accept) && strcmp(accept, "application/simple-message-summary")) { /* Format requested that we do not support */ transmit_response(p, "406 Not Acceptable", req); - if (option_debug > 1) - ast_log(LOG_DEBUG, "Received SIP mailbox subscription for unknown format: %s\n", accept); + ast_debug(2, "Received SIP mailbox subscription for unknown format: %s\n", accept); ast_set_flag(&p->flags[0], SIP_NEEDDESTROY); if (authpeer) unref_peer(authpeer); @@ -15503,8 +15327,7 @@ static int handle_request_subscribe(struct sip_pvt *p, struct sip_request *req, /* Do not release authpeer here */ } else { /* At this point, Asterisk does not understand the specified event */ transmit_response(p, "489 Bad Event", req); - if (option_debug > 1) - ast_log(LOG_DEBUG, "Received SIP subscribe for unknown event package: %s\n", event); + ast_debug(2, "Received SIP subscribe for unknown event package: %s\n", event); ast_set_flag(&p->flags[0], SIP_NEEDDESTROY); if (authpeer) unref_peer(authpeer); @@ -15526,11 +15349,11 @@ static int handle_request_subscribe(struct sip_pvt *p, struct sip_request *req, if (p->expiry < min_expiry && p->expiry > 0) p->expiry = min_expiry; - if (sipdebug || option_debug > 1) { + if (sipdebug) { if (p->subscribed == MWI_NOTIFICATION && p->relatedpeer) - ast_log(LOG_DEBUG, "Adding subscription for mailbox notification - peer %s Mailbox %s\n", p->relatedpeer->name, p->relatedpeer->mailbox); + ast_debug(2, "Adding subscription for mailbox notification - peer %s Mailbox %s\n", p->relatedpeer->name, p->relatedpeer->mailbox); else - ast_log(LOG_DEBUG, "Adding subscription for extension %s context %s for peer %s\n", p->exten, p->context, p->username); + ast_debug(2, "Adding subscription for extension %s context %s for peer %s\n", p->exten, p->context, p->username); } if (p->autokillid > -1) sip_cancel_destroy(p); /* Remove subscription expiry for renewals */ @@ -15600,8 +15423,8 @@ static int handle_request_register(struct sip_pvt *p, struct sip_request *req, s /* Use this as the basis */ copy_request(&p->initreq, req); - if (option_debug > 3 && sipdebug) - ast_log(LOG_DEBUG, "Initializing initreq for method %s - callid %s\n", sip_methods[req->method].text, p->callid); + if (sipdebug) + ast_debug(4, "Initializing initreq for method %s - callid %s\n", sip_methods[req->method].text, p->callid); check_via(p, req); if ((res = register_verify(p, sin, req, e)) < 0) { const char *reason; @@ -15697,8 +15520,7 @@ static int handle_request(struct sip_pvt *p, struct sip_request *req, struct soc within an existing dialog */ /* Response to our request -- Do some sanity checks */ if (p->ocseq < seqno) { - if (option_debug) - ast_log(LOG_DEBUG, "Ignoring out of order response %d (expecting %d)\n", seqno, p->ocseq); + ast_debug(1, "Ignoring out of order response %d (expecting %d)\n", seqno, p->ocseq); return -1; } else if (p->ocseq != seqno) { /* ignore means "don't do anything with it" but still have to @@ -15728,12 +15550,10 @@ static int handle_request(struct sip_pvt *p, struct sip_request *req, struct soc */ p->method = req->method; /* Find out which SIP method they are using */ - if (option_debug > 3) - ast_log(LOG_DEBUG, "**** Received %s (%d) - Command in SIP %s\n", sip_methods[p->method].text, sip_methods[p->method].id, cmd); + ast_debug(4, "**** Received %s (%d) - Command in SIP %s\n", sip_methods[p->method].text, sip_methods[p->method].id, cmd); if (p->icseq && (p->icseq > seqno)) { - if (option_debug) - ast_log(LOG_DEBUG, "Ignoring too old SIP packet packet %d (expecting >= %d)\n", seqno, p->icseq); + ast_debug(1, "Ignoring too old SIP packet packet %d (expecting >= %d)\n", seqno, p->icseq); if (req->method != SIP_ACK) transmit_response(p, "503 Server error", req); /* We must respond according to RFC 3261 sec 12.2 */ return -1; @@ -15745,8 +15565,7 @@ static int handle_request(struct sip_pvt *p, struct sip_request *req, struct soc respond appropriately. We do this if we receive a repeat of the last sequence number */ ast_set_flag(req, SIP_PKT_IGNORE); - if (option_debug > 2) - ast_log(LOG_DEBUG, "Ignoring SIP message because of retransmit (%s Seqno %d, ours %d)\n", sip_methods[p->method].text, p->icseq, seqno); + ast_debug(3, "Ignoring SIP message because of retransmit (%s Seqno %d, ours %d)\n", sip_methods[p->method].text, p->icseq, seqno); } if (seqno >= p->icseq) @@ -15778,8 +15597,7 @@ static int handle_request(struct sip_pvt *p, struct sip_request *req, struct soc transmit_response(p, "481 Call/Transaction Does Not Exist", req); sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT); } else { - if (option_debug) - ast_log(LOG_DEBUG, "Got ACK for unknown dialog... strange.\n"); + ast_debug(1, "Got ACK for unknown dialog... strange.\n"); } return res; } @@ -15884,8 +15702,8 @@ static int sipsock_read(int *id, int fd, short events, void *ignore) ast_log(LOG_WARNING, "Recv error: %s\n", strerror(errno)); return 1; } - if (option_debug && res == sizeof(req.data)) { - ast_log(LOG_DEBUG, "Received packet exceeds buffer. Data is possibly lost\n"); + if (res == sizeof(req.data)) { + ast_debug(1, "Received packet exceeds buffer. Data is possibly lost\n"); req.data[sizeof(req.data) - 1] = '\0'; } else req.data[res] = '\0'; @@ -15913,8 +15731,7 @@ static int sipsock_read(int *id, int fd, short events, void *ignore) /* Find the active SIP dialog or create a new one */ p = find_call(&req, &sin, req.method); /* returns p locked */ if (p == NULL) { - if (option_debug) - ast_log(LOG_DEBUG, "Invalid SIP message - rejected , no callid, len %d\n", req.len); + ast_debug(1, "Invalid SIP message - rejected , no callid, len %d\n", req.len); ast_mutex_unlock(&netlock); return 1; } @@ -15922,8 +15739,7 @@ static int sipsock_read(int *id, int fd, short events, void *ignore) /* becaues this is deadlock-prone, we need to try and unlock if failed */ if (!p->owner || !ast_channel_trylock(p->owner)) break; /* locking succeeded */ - if (option_debug) - ast_log(LOG_DEBUG, "Failed to grab owner channel lock, trying again. (SIP call %s)\n", p->callid); + ast_debug(1, "Failed to grab owner channel lock, trying again. (SIP call %s)\n", p->callid); sip_pvt_unlock(p); ast_mutex_unlock(&netlock); /* Sleep for a very short amount of time */ @@ -15947,8 +15763,7 @@ static int sipsock_read(int *id, int fd, short events, void *ignore) nounlock = 0; if (handle_request(p, &req, &sin, &recount, &nounlock) == -1) { /* Request failed */ - if (option_debug) - ast_log(LOG_DEBUG, "SIP message could not be handled, bad request: %-70.70s\n", p->callid[0] ? p->callid : "<no callid>"); + ast_debug(1, "SIP message could not be handled, bad request: %-70.70s\n", p->callid[0] ? p->callid : "<no callid>"); } if (p->owner && !nounlock) @@ -16153,13 +15968,13 @@ restartsearch: if ((res < 0) || (res > 1000)) res = 1000; res = ast_io_wait(io, res); - if (option_debug && res > 20) - ast_log(LOG_DEBUG, "chan_sip: ast_io_wait ran %d all at once\n", res); + if (res > 20) + ast_debug(1, "chan_sip: ast_io_wait ran %d all at once\n", res); ast_mutex_lock(&monlock); if (res >= 0) { res = ast_sched_runq(sched); - if (option_debug && res >= 20) - ast_log(LOG_DEBUG, "chan_sip: ast_sched_runq ran %d all at once\n", res); + if (res >= 20) + ast_debug(1, "chan_sip: ast_sched_runq ran %d all at once\n", res); } ast_mutex_unlock(&monlock); } @@ -16331,8 +16146,7 @@ static int sip_devicestate(void *data) if ((tmp = strchr(host, '@'))) host = tmp + 1; - if (option_debug > 2) - ast_log(LOG_DEBUG, "Checking device state for peer %s\n", host); + ast_debug(3, "Checking device state for peer %s\n", host); if ((p = find_peer(host, NULL, 1))) { if (p->addr.sin_addr.s_addr || p->defaddr.sin_addr.s_addr) { @@ -16400,8 +16214,7 @@ static struct ast_channel *sip_request_call(const char *type, int format, void * *cause = AST_CAUSE_BEARERCAPABILITY_NOTAVAIL; /* Can't find codec to connect to host */ return NULL; } - if (option_debug) - ast_log(LOG_DEBUG, "Asked to create a SIP channel with formats: %s\n", ast_getformatname_multiple(tmp, sizeof(tmp), oldformat)); + ast_debug(1, "Asked to create a SIP channel with formats: %s\n", ast_getformatname_multiple(tmp, sizeof(tmp), oldformat)); if (!(p = sip_alloc(NULL, NULL, 0, SIP_INVITE))) { ast_log(LOG_ERROR, "Unable to build sip pvt data for '%s' (Out of memory or socket error)\n", dest); @@ -16432,8 +16245,7 @@ static struct ast_channel *sip_request_call(const char *type, int format, void * if (create_addr(p, host)) { *cause = AST_CAUSE_UNREGISTERED; - if (option_debug > 2) - ast_log(LOG_DEBUG, "Cant create SIP call - target device not registred\n"); + ast_debug(3, "Cant create SIP call - target device not registred\n"); sip_destroy(p); return NULL; } @@ -16631,7 +16443,7 @@ static int add_sip_domain(const char *domain, const enum domain_mode mode, const AST_LIST_UNLOCK(&domain_list); if (sipdebug) - ast_log(LOG_DEBUG, "Added local SIP domain '%s'\n", domain); + ast_debug(1, "Added local SIP domain '%s'\n", domain); return 1; } @@ -16681,8 +16493,7 @@ static struct sip_auth *add_realm_authentication(struct sip_auth *authlist, char if (ast_strlen_zero(configuration)) return authlist; - if (option_debug) - ast_log(LOG_DEBUG, "Auth config :: %s\n", configuration); + ast_debug(1, "Auth config :: %s\n", configuration); ast_copy_string(authcopy, configuration, sizeof(authcopy)); stringp = authcopy; @@ -16984,8 +16795,7 @@ static struct sip_peer *build_peer(const char *name, struct ast_variable *v, str if (realtime) { rpeerobjs++; - if (option_debug > 2) - ast_log(LOG_DEBUG,"-REALTIME- peer built. Name: %s. Peer objects: %d\n", name, rpeerobjs); + ast_debug(3,"-REALTIME- peer built. Name: %s. Peer objects: %d\n", name, rpeerobjs); } else speerobjs++; ASTOBJ_INIT(peer); @@ -17193,8 +17003,7 @@ static struct sip_peer *build_peer(const char *name, struct ast_variable *v, str if ((nowtime - regseconds) > 0) { destroy_association(peer); memset(&peer->addr, 0, sizeof(peer->addr)); - if (option_debug) - ast_log(LOG_DEBUG, "Bah, we're expired (%d/%d/%d)!\n", (int)(nowtime - regseconds), (int)regseconds, (int)nowtime); + ast_debug(1, "Bah, we're expired (%d/%d/%d)!\n", (int)(nowtime - regseconds), (int)regseconds, (int)nowtime); } } ast_copy_flags(&peer->flags[0], &peerflags[0], mask[0].flags); @@ -17380,8 +17189,7 @@ static int reload_config(enum channelreloadreason reason) ast_copy_string(global_realm, v->value, sizeof(global_realm)); } else if (!strcasecmp(v->name, "useragent")) { ast_copy_string(global_useragent, v->value, sizeof(global_useragent)); - if (option_debug) - ast_log(LOG_DEBUG, "Setting SIP channel User-Agent Name to %s\n", global_useragent); + ast_debug(1, "Setting SIP channel User-Agent Name to %s\n", global_useragent); } else if (!strcasecmp(v->name, "allowtransfer")) { global_allowtransfer = ast_true(v->value) ? TRANSFER_OPENFORALL : TRANSFER_CLOSED; } else if (!strcasecmp(v->name, "rtcachefriends")) { @@ -17563,8 +17371,8 @@ static int reload_config(enum channelreloadreason reason) if (context) *context++ = '\0'; - if (option_debug && ast_strlen_zero(context)) - ast_log(LOG_DEBUG, "No context specified at line %d for domain '%s'\n", v->lineno, domain); + if (ast_strlen_zero(context)) + ast_debug(1, "No context specified at line %d for domain '%s'\n", v->lineno, domain); if (ast_strlen_zero(domain)) ast_log(LOG_WARNING, "Empty domain specified at line %d\n", v->lineno); else @@ -17837,14 +17645,10 @@ static int sip_set_udptl_peer(struct ast_channel *chan, struct ast_udptl *udptl) memset(&p->udptlredirip, 0, sizeof(p->udptlredirip)); if (!ast_test_flag(&p->flags[0], SIP_GOTREFER)) { if (!p->pendinginvite) { - if (option_debug > 2) { - ast_log(LOG_DEBUG, "Sending reinvite on SIP '%s' - It's UDPTL soon redirected to IP %s:%d\n", p->callid, ast_inet_ntoa(udptl ? p->udptlredirip.sin_addr : p->ourip), udptl ? ntohs(p->udptlredirip.sin_port) : 0); - } + ast_debug(3, "Sending reinvite on SIP '%s' - It's UDPTL soon redirected to IP %s:%d\n", p->callid, ast_inet_ntoa(udptl ? p->udptlredirip.sin_addr : p->ourip), udptl ? ntohs(p->udptlredirip.sin_port) : 0); transmit_reinvite_with_sdp(p, TRUE); } else if (!ast_test_flag(&p->flags[0], SIP_PENDINGBYE)) { - if (option_debug > 2) { - ast_log(LOG_DEBUG, "Deferring reinvite on SIP '%s' - It's UDPTL will be redirected to IP %s:%d\n", p->callid, ast_inet_ntoa(udptl ? p->udptlredirip.sin_addr : p->ourip), udptl ? ntohs(p->udptlredirip.sin_port) : 0); - } + ast_debug(3, "Deferring reinvite on SIP '%s' - It's UDPTL will be redirected to IP %s:%d\n", p->callid, ast_inet_ntoa(udptl ? p->udptlredirip.sin_addr : p->ourip), udptl ? ntohs(p->udptlredirip.sin_port) : 0); ast_set_flag(&p->flags[0], SIP_NEEDREINVITE); } } @@ -17893,20 +17697,16 @@ static int sip_handle_t38_reinvite(struct ast_channel *chan, struct sip_pvt *pvt } if (!ast_test_flag(&p->flags[0], SIP_GOTREFER)) { if (!p->pendinginvite) { - if (option_debug > 2) { - if (flag) - ast_log(LOG_DEBUG, "Sending reinvite on SIP '%s' - It's UDPTL soon redirected to IP %s:%d\n", p->callid, ast_inet_ntoa(p->udptlredirip.sin_addr), ntohs(p->udptlredirip.sin_port)); - else - ast_log(LOG_DEBUG, "Sending reinvite on SIP '%s' - It's UDPTL soon redirected to us (IP %s)\n", p->callid, ast_inet_ntoa(p->ourip)); - } + if (flag) + ast_debug(3, "Sending reinvite on SIP '%s' - It's UDPTL soon redirected to IP %s:%d\n", p->callid, ast_inet_ntoa(p->udptlredirip.sin_addr), ntohs(p->udptlredirip.sin_port)); + else + ast_debug(3, "Sending reinvite on SIP '%s' - It's UDPTL soon redirected to us (IP %s)\n", p->callid, ast_inet_ntoa(p->ourip)); transmit_reinvite_with_sdp(p, TRUE); } else if (!ast_test_flag(&p->flags[0], SIP_PENDINGBYE)) { - if (option_debug > 2) { - if (flag) - ast_log(LOG_DEBUG, "Deferring reinvite on SIP '%s' - It's UDPTL will be redirected to IP %s:%d\n", p->callid, ast_inet_ntoa(p->udptlredirip.sin_addr), ntohs(p->udptlredirip.sin_port)); - else - ast_log(LOG_DEBUG, "Deferring reinvite on SIP '%s' - It's UDPTL will be redirected to us (IP %s)\n", p->callid, ast_inet_ntoa(p->ourip)); - } + if (flag) + ast_debug(3, "Deferring reinvite on SIP '%s' - It's UDPTL will be redirected to IP %s:%d\n", p->callid, ast_inet_ntoa(p->udptlredirip.sin_addr), ntohs(p->udptlredirip.sin_port)); + else + ast_debug(3, "Deferring reinvite on SIP '%s' - It's UDPTL will be redirected to us (IP %s)\n", p->callid, ast_inet_ntoa(p->ourip)); ast_set_flag(&p->flags[0], SIP_NEEDREINVITE); } } @@ -17921,18 +17721,14 @@ static int sip_handle_t38_reinvite(struct ast_channel *chan, struct sip_pvt *pvt } else { memset(&p->udptlredirip, 0, sizeof(p->udptlredirip)); } - if (option_debug > 2) { - if (flag) - ast_log(LOG_DEBUG, "Responding 200 OK on SIP '%s' - It's UDPTL soon redirected to IP %s:%d\n", p->callid, ast_inet_ntoa(p->udptlredirip.sin_addr), ntohs(p->udptlredirip.sin_port)); - else - ast_log(LOG_DEBUG, "Responding 200 OK on SIP '%s' - It's UDPTL soon redirected to us (IP %s)\n", p->callid, ast_inet_ntoa(p->ourip)); - } + if (flag) + ast_debug(3, "Responding 200 OK on SIP '%s' - It's UDPTL soon redirected to IP %s:%d\n", p->callid, ast_inet_ntoa(p->udptlredirip.sin_addr), ntohs(p->udptlredirip.sin_port)); + else + ast_debug(3, "Responding 200 OK on SIP '%s' - It's UDPTL soon redirected to us (IP %s)\n", p->callid, ast_inet_ntoa(p->ourip)); pvt->t38.state = T38_ENABLED; p->t38.state = T38_ENABLED; - if (option_debug > 1) { - ast_log(LOG_DEBUG, "T38 changed state to %d on channel %s\n", pvt->t38.state, pvt->owner ? pvt->owner->name : "<none>"); - ast_log(LOG_DEBUG, "T38 changed state to %d on channel %s\n", p->t38.state, chan ? chan->name : "<none>"); - } + ast_debug(2, "T38 changed state to %d on channel %s\n", pvt->t38.state, pvt->owner ? pvt->owner->name : "<none>"); + ast_debug(2, "T38 changed state to %d on channel %s\n", p->t38.state, chan ? chan->name : "<none>"); transmit_response_with_t38_sdp(p, "200 OK", &p->initreq, XMIT_CRITICAL); p->lastrtprx = p->lastrtptx = time(NULL); sip_pvt_unlock(p); @@ -18080,17 +17876,12 @@ static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struc if (chan->_state != AST_STATE_UP) { /* We are in early state */ if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY)) append_history(p, "ExtInv", "Initial invite sent with remote bridge proposal."); - if (option_debug) - ast_log(LOG_DEBUG, "Early remote bridge setting SIP '%s' - Sending media to %s\n", p->callid, ast_inet_ntoa(rtp ? p->redirip.sin_addr : p->ourip)); + ast_debug(1, "Early remote bridge setting SIP '%s' - Sending media to %s\n", p->callid, ast_inet_ntoa(rtp ? p->redirip.sin_addr : p->ourip)); } else if (!p->pendinginvite) { /* We are up, and have no outstanding invite */ - if (option_debug > 2) { - ast_log(LOG_DEBUG, "Sending reinvite on SIP '%s' - It's audio soon redirected to IP %s\n", p->callid, ast_inet_ntoa(rtp ? p->redirip.sin_addr : p->ourip)); - } + ast_debug(3, "Sending reinvite on SIP '%s' - It's audio soon redirected to IP %s\n", p->callid, ast_inet_ntoa(rtp ? p->redirip.sin_addr : p->ourip)); transmit_reinvite_with_sdp(p, FALSE); } else if (!ast_test_flag(&p->flags[0], SIP_PENDINGBYE)) { - if (option_debug > 2) { - ast_log(LOG_DEBUG, "Deferring reinvite on SIP '%s' - It's audio will be redirected to IP %s\n", p->callid, ast_inet_ntoa(rtp ? p->redirip.sin_addr : p->ourip)); - } + ast_debug(3, "Deferring reinvite on SIP '%s' - It's audio will be redirected to IP %s\n", p->callid, ast_inet_ntoa(rtp ? p->redirip.sin_addr : p->ourip)); /* We have a pending Invite. Send re-invite when we're done with the invite */ ast_set_flag(&p->flags[0], SIP_NEEDREINVITE); } @@ -18197,7 +17988,7 @@ static int sip_addheader(struct ast_channel *chan, void *data) if (ok) { pbx_builtin_setvar_helper (chan, varbuf, inbuf); if (sipdebug) - ast_log(LOG_DEBUG,"SIP Header added \"%s\" as %s\n", inbuf, varbuf); + ast_debug(1,"SIP Header added \"%s\" as %s\n", inbuf, varbuf); } else { ast_log(LOG_WARNING, "Too many SIP headers added, max 50\n"); } @@ -18318,8 +18109,7 @@ static void sip_send_all_registers(void) /*! \brief Reload module */ static int sip_do_reload(enum channelreloadreason reason) { - if (option_debug > 3) - ast_log(LOG_DEBUG, "--------------- SIP reload started\n"); + ast_debug(4, "--------------- SIP reload started\n"); clear_realm_authentication(authl); clear_sip_domains(); @@ -18330,8 +18120,7 @@ static int sip_do_reload(enum channelreloadreason reason) ASTOBJ_CONTAINER_TRAVERSE(®l, 1, do { ASTOBJ_RDLOCK(iterator); if (iterator->call) { - if (option_debug > 2) - ast_log(LOG_DEBUG, "Destroying active SIP dialog for registry %s@%s\n", iterator->username, iterator->hostname); + ast_debug(3, "Destroying active SIP dialog for registry %s@%s\n", iterator->username, iterator->hostname); /* This will also remove references to the registry */ sip_destroy(iterator->call); } @@ -18341,18 +18130,15 @@ static int sip_do_reload(enum channelreloadreason reason) /* Then, actually destroy users and registry */ ASTOBJ_CONTAINER_DESTROYALL(&userl, sip_destroy_user); - if (option_debug > 3) - ast_log(LOG_DEBUG, "--------------- Done destroying user list\n"); + ast_debug(4, "--------------- Done destroying user list\n"); ASTOBJ_CONTAINER_DESTROYALL(®l, sip_registry_destroy); - if (option_debug > 3) - ast_log(LOG_DEBUG, "--------------- Done destroying registry list\n"); + ast_debug(4, "--------------- Done destroying registry list\n"); ASTOBJ_CONTAINER_MARKALL(&peerl); reload_config(reason); /* Prune peers who still are supposed to be deleted */ ASTOBJ_CONTAINER_PRUNE_MARKED(&peerl, sip_destroy_peer); - if (option_debug > 3) - ast_log(LOG_DEBUG, "--------------- Done destroying pruned peers\n"); + ast_debug(4, "--------------- Done destroying pruned peers\n"); /* Send qualify (OPTIONS) to all peers */ sip_poke_all_peers(); @@ -18360,8 +18146,7 @@ static int sip_do_reload(enum channelreloadreason reason) /* Register with all services */ sip_send_all_registers(); - if (option_debug > 3) - ast_log(LOG_DEBUG, "--------------- SIP reload done\n"); + ast_debug(4, "--------------- SIP reload done\n"); return 0; } |