/* * Asterisk -- An open source telephony toolkit. * * Copyright (C) 2013, Digium, Inc. * * Joshua Colp * * See http://www.asterisk.org for more information about * the Asterisk project. Please do not directly contact * any of the maintainers of this project for assistance; * the project provides a web site, mailing lists and IRC * channels for your use. * * This program is free software, distributed under the terms of * the GNU General Public License Version 2. See the LICENSE file * at the top of the source tree. */ /*! \file * * \author Joshua Colp * * \brief PSJIP SIP Channel Driver * * \ingroup channel_drivers */ /*** MODULEINFO pjproject res_pjsip res_pjsip_session core ***/ #include "asterisk.h" #include #include #include #include "asterisk/lock.h" #include "asterisk/channel.h" #include "asterisk/module.h" #include "asterisk/pbx.h" #include "asterisk/rtp_engine.h" #include "asterisk/acl.h" #include "asterisk/callerid.h" #include "asterisk/file.h" #include "asterisk/cli.h" #include "asterisk/app.h" #include "asterisk/musiconhold.h" #include "asterisk/causes.h" #include "asterisk/taskprocessor.h" #include "asterisk/dsp.h" #include "asterisk/stasis_endpoints.h" #include "asterisk/stasis_channels.h" #include "asterisk/indications.h" #include "asterisk/format_cache.h" #include "asterisk/translate.h" #include "asterisk/threadstorage.h" #include "asterisk/features_config.h" #include "asterisk/pickup.h" #include "asterisk/test.h" #include "asterisk/res_pjsip.h" #include "asterisk/res_pjsip_session.h" #include "asterisk/stream.h" #include "pjsip/include/chan_pjsip.h" #include "pjsip/include/dialplan_functions.h" #include "pjsip/include/cli_functions.h" AST_THREADSTORAGE(uniqueid_threadbuf); #define UNIQUEID_BUFSIZE 256 static const char channel_type[] = "PJSIP"; static unsigned int chan_idx; static void chan_pjsip_pvt_dtor(void *obj) { } /* \brief Asterisk core interaction functions */ static struct ast_channel *chan_pjsip_request(const char *type, struct ast_format_cap *cap, const struct ast_assigned_ids *assignedids, const struct ast_channel *requestor, const char *data, int *cause); static struct ast_channel *chan_pjsip_request_with_stream_topology(const char *type, struct ast_stream_topology *topology, const struct ast_assigned_ids *assignedids, const struct ast_channel *requestor, const char *data, int *cause); static int chan_pjsip_sendtext(struct ast_channel *ast, const char *text); static int chan_pjsip_digit_begin(struct ast_channel *ast, char digit); static int chan_pjsip_digit_end(struct ast_channel *ast, char digit, unsigned int duration); static int chan_pjsip_call(struct ast_channel *ast, const char *dest, int timeout); static int chan_pjsip_hangup(struct ast_channel *ast); static int chan_pjsip_answer(struct ast_channel *ast); static struct ast_frame *chan_pjsip_read_stream(struct ast_channel *ast); static int chan_pjsip_write(struct ast_channel *ast, struct ast_frame *f); static int chan_pjsip_write_stream(struct ast_channel *ast, int stream_num, struct ast_frame *f); static int chan_pjsip_indicate(struct ast_channel *ast, int condition, const void *data, size_t datalen); static int chan_pjsip_transfer(struct ast_channel *ast, const char *target); static int chan_pjsip_fixup(struct ast_channel *oldchan, struct ast_channel *newchan); static int chan_pjsip_devicestate(const char *data); static int chan_pjsip_queryoption(struct ast_channel *ast, int option, void *data, int *datalen); static const char *chan_pjsip_get_uniqueid(struct ast_channel *ast); /*! \brief PBX interface structure for channel registration */ struct ast_channel_tech chan_pjsip_tech = { .type = channel_type, .description = "PJSIP Channel Driver", .requester = chan_pjsip_request, .requester_with_stream_topology = chan_pjsip_request_with_stream_topology, .send_text = chan_pjsip_sendtext, .send_digit_begin = chan_pjsip_digit_begin, .send_digit_end = chan_pjsip_digit_end, .call = chan_pjsip_call, .hangup = chan_pjsip_hangup, .answer = chan_pjsip_answer, .read_stream = chan_pjsip_read_stream, .write = chan_pjsip_write, .write_stream = chan_pjsip_write_stream, .exception = chan_pjsip_read_stream, .indicate = chan_pjsip_indicate, .transfer = chan_pjsip_transfer, .fixup = chan_pjsip_fixup, .devicestate = chan_pjsip_devicestate, .queryoption = chan_pjsip_queryoption, .func_channel_read = pjsip_acf_channel_read, .get_pvt_uniqueid = chan_pjsip_get_uniqueid, .properties = AST_CHAN_TP_WANTSJITTER | AST_CHAN_TP_CREATESJITTER }; /*! \brief SIP session interaction functions */ static void chan_pjsip_session_begin(struct ast_sip_session *session); static void chan_pjsip_session_end(struct ast_sip_session *session); static int chan_pjsip_incoming_request(struct ast_sip_session *session, struct pjsip_rx_data *rdata); static void chan_pjsip_incoming_response(struct ast_sip_session *session, struct pjsip_rx_data *rdata); /*! \brief SIP session supplement structure */ static struct ast_sip_session_supplement chan_pjsip_supplement = { .method = "INVITE", .priority = AST_SIP_SUPPLEMENT_PRIORITY_CHANNEL, .session_begin = chan_pjsip_session_begin, .session_end = chan_pjsip_session_end, .incoming_request = chan_pjsip_incoming_request, .incoming_response = chan_pjsip_incoming_response, /* It is important that this supplement runs after media has been negotiated */ .response_priority = AST_SIP_SESSION_AFTER_MEDIA, }; static int chan_pjsip_incoming_ack(struct ast_sip_session *session, struct pjsip_rx_data *rdata); static struct ast_sip_session_supplement chan_pjsip_ack_supplement = { .method = "ACK", .priority = AST_SIP_SUPPLEMENT_PRIORITY_CHANNEL, .incoming_request = chan_pjsip_incoming_ack, }; /*! \brief Function called by RTP engine to get local audio RTP peer */ static enum ast_rtp_glue_result chan_pjsip_get_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance **instance) { struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan); struct ast_sip_endpoint *endpoint; struct ast_datastore *datastore; struct ast_sip_session_media *media; if (!channel || !channel->session) { return AST_RTP_GLUE_RESULT_FORBID; } /* XXX Getting the first RTP instance for direct media related stuff seems just * absolutely wrong. But the native RTP bridge knows no other method than single-stream * for direct media. So this is the best we can do. */ media = channel->session->active_media_state->default_session[AST_MEDIA_TYPE_AUDIO]; if (!media || !media->rtp) { return AST_RTP_GLUE_RESULT_FORBID; } datastore = ast_sip_session_get_datastore(channel->session, "t38"); if (datastore) { ao2_ref(datastore, -1); return AST_RTP_GLUE_RESULT_FORBID; } endpoint = channel->session->endpoint; *instance = media->rtp; ao2_ref(*instance, +1); ast_assert(endpoint != NULL); if (endpoint->media.rtp.encryption != AST_SIP_MEDIA_ENCRYPT_NONE) { return AST_RTP_GLUE_RESULT_FORBID; } if (endpoint->media.direct_media.enabled) { return AST_RTP_GLUE_RESULT_REMOTE; } return AST_RTP_GLUE_RESULT_LOCAL; } /*! \brief Function called by RTP engine to get local video RTP peer */ static enum ast_rtp_glue_result chan_pjsip_get_vrtp_peer(struct ast_channel *chan, struct ast_rtp_instance **instance) { struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan); struct ast_sip_endpoint *endpoint; struct ast_sip_session_media *media; if (!channel || !channel->session) { return AST_RTP_GLUE_RESULT_FORBID; } media = channel->session->active_media_state->default_session[AST_MEDIA_TYPE_VIDEO]; if (!media || !media->rtp) { return AST_RTP_GLUE_RESULT_FORBID; } endpoint = channel->session->endpoint; *instance = media->rtp; ao2_ref(*instance, +1); ast_assert(endpoint != NULL); if (endpoint->media.rtp.encryption != AST_SIP_MEDIA_ENCRYPT_NONE) { return AST_RTP_GLUE_RESULT_FORBID; } return AST_RTP_GLUE_RESULT_LOCAL; } /*! \brief Function called by RTP engine to get peer capabilities */ static void chan_pjsip_get_codec(struct ast_channel *chan, struct ast_format_cap *result) { ast_format_cap_append_from_cap(result, ast_channel_nativeformats(chan), AST_MEDIA_TYPE_UNKNOWN); } /*! \brief Destructor function for \ref transport_info_data */ static void transport_info_destroy(void *obj) { struct transport_info_data *data = obj; ast_free(data); } /*! \brief Datastore used to store local/remote addresses for the * INVITE request that created the PJSIP channel */ static struct ast_datastore_info transport_info = { .type = "chan_pjsip_transport_info", .destroy = transport_info_destroy, }; static struct ast_datastore_info direct_media_mitigation_info = { }; static int direct_media_mitigate_glare(struct ast_sip_session *session) { RAII_VAR(struct ast_datastore *, datastore, NULL, ao2_cleanup); if (session->endpoint->media.direct_media.glare_mitigation == AST_SIP_DIRECT_MEDIA_GLARE_MITIGATION_NONE) { return 0; } datastore = ast_sip_session_get_datastore(session, "direct_media_glare_mitigation"); if (!datastore) { return 0; } /* Removing the datastore ensures we won't try to mitigate glare on subsequent reinvites */ ast_sip_session_remove_datastore(session, "direct_media_glare_mitigation"); if ((session->endpoint->media.direct_media.glare_mitigation == AST_SIP_DIRECT_MEDIA_GLARE_MITIGATION_OUTGOING && session->inv_session->role == PJSIP_ROLE_UAC) || (session->endpoint->media.direct_media.glare_mitigation == AST_SIP_DIRECT_MEDIA_GLARE_MITIGATION_INCOMING && session->inv_session->role == PJSIP_ROLE_UAS)) { return 1; } return 0; } /*! \brief Helper function to find the position for RTCP */ static int rtp_find_rtcp_fd_position(struct ast_sip_session *session, struct ast_rtp_instance *rtp) { int index; for (index = 0; index < AST_VECTOR_SIZE(&session->active_media_state->read_callbacks); ++index) { struct ast_sip_session_media_read_callback_state *callback_state = AST_VECTOR_GET_ADDR(&session->active_media_state->read_callbacks, index); if (callback_state->fd != ast_rtp_instance_fd(rtp, 1)) { continue; } return index; } return -1; } /*! * \pre chan is locked */ static int check_for_rtp_changes(struct ast_channel *chan, struct ast_rtp_instance *rtp, struct ast_sip_session_media *media, struct ast_sip_session *session) { int changed = 0, position = -1; if (media->rtp) { position = rtp_find_rtcp_fd_position(session, media->rtp); } if (rtp) { changed = ast_rtp_instance_get_and_cmp_remote_address(rtp, &media->direct_media_addr); if (media->rtp) { if (position != -1) { ast_channel_set_fd(chan, position + AST_EXTENDED_FDS, -1); } ast_rtp_instance_set_prop(media->rtp, AST_RTP_PROPERTY_RTCP, 0); } } else if (!ast_sockaddr_isnull(&media->direct_media_addr)){ ast_sockaddr_setnull(&media->direct_media_addr); changed = 1; if (media->rtp) { ast_rtp_instance_set_prop(media->rtp, AST_RTP_PROPERTY_RTCP, 1); if (position != -1) { ast_channel_set_fd(chan, position + AST_EXTENDED_FDS, ast_rtp_instance_fd(media->rtp, 1)); } } } return changed; } struct rtp_direct_media_data { struct ast_channel *chan; struct ast_rtp_instance *rtp; struct ast_rtp_instance *vrtp; struct ast_format_cap *cap; struct ast_sip_session *session; }; static void rtp_direct_media_data_destroy(void *data) { struct rtp_direct_media_data *cdata = data; ao2_cleanup(cdata->session); ao2_cleanup(cdata->cap); ao2_cleanup(cdata->vrtp); ao2_cleanup(cdata->rtp); ao2_cleanup(cdata->chan); } static struct rtp_direct_media_data *rtp_direct_media_data_create( struct ast_channel *chan, struct ast_rtp_instance *rtp, struct ast_rtp_instance *vrtp, const struct ast_format_cap *cap, struct ast_sip_session *session) { struct rtp_direct_media_data *cdata = ao2_alloc(sizeof(*cdata), rtp_direct_media_data_destroy); if (!cdata) { return NULL; } cdata->chan = ao2_bump(chan); cdata->rtp = ao2_bump(rtp); cdata->vrtp = ao2_bump(vrtp); cdata->cap = ao2_bump((struct ast_format_cap *)cap); cdata->session = ao2_bump(session); return cdata; } static int send_direct_media_request(void *data) { struct rtp_direct_media_data *cdata = data; struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(cdata->chan); struct ast_sip_session *session; int changed = 0; int res = 0; /* XXX In an ideal world each media stream would be direct, but for now preserve behavior * and connect only the default media sessions for audio and video. */ /* The channel needs to be locked when checking for RTP changes. * Otherwise, we could end up destroying an underlying RTCP structure * at the same time that the channel thread is attempting to read RTCP */ ast_channel_lock(cdata->chan); session = channel->session; if (session->active_media_state->default_session[AST_MEDIA_TYPE_AUDIO]) { changed |= check_for_rtp_changes( cdata->chan, cdata->rtp, session->active_media_state->default_session[AST_MEDIA_TYPE_AUDIO], session); } if (session->active_media_state->default_session[AST_MEDIA_TYPE_VIDEO]) { changed |= check_for_rtp_changes( cdata->chan, cdata->vrtp, session->active_media_state->default_session[AST_MEDIA_TYPE_VIDEO], session); } ast_channel_unlock(cdata->chan); if (direct_media_mitigate_glare(cdata->session)) { ast_debug(4, "Disregarding setting RTP on %s: mitigating re-INVITE glare\n", ast_channel_name(cdata->chan)); ao2_ref(cdata, -1); return 0; } if (cdata->cap && ast_format_cap_count(cdata->cap) && !ast_format_cap_identical(cdata->session->direct_media_cap, cdata->cap)) { ast_format_cap_remove_by_type(cdata->session->direct_media_cap, AST_MEDIA_TYPE_UNKNOWN); ast_format_cap_append_from_cap(cdata->session->direct_media_cap, cdata->cap, AST_MEDIA_TYPE_UNKNOWN); changed = 1; } if (changed) { ast_debug(4, "RTP changed on %s; initiating direct media update\n", ast_channel_name(cdata->chan)); res = ast_sip_session_refresh(cdata->session, NULL, NULL, NULL, cdata->session->endpoint->media.direct_media.method, 1, NULL); } ao2_ref(cdata, -1); return res; } /*! \brief Function called by RTP engine to change where the remote party should send media */ static int chan_pjsip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance *rtp, struct ast_rtp_instance *vrtp, struct ast_rtp_instance *tpeer, const struct ast_format_cap *cap, int nat_active) { struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan); struct ast_sip_session *session = channel->session; struct rtp_direct_media_data *cdata; /* Don't try to do any direct media shenanigans on early bridges */ if ((rtp || vrtp || tpeer) && !ast_channel_is_bridged(chan)) { ast_debug(4, "Disregarding setting RTP on %s: channel is not bridged\n", ast_channel_name(chan)); return 0; } if (nat_active && session->endpoint->media.direct_media.disable_on_nat) { ast_debug(4, "Disregarding setting RTP on %s: NAT is active\n", ast_channel_name(chan)); return 0; } cdata = rtp_direct_media_data_create(chan, rtp, vrtp, cap, session); if (!cdata) { return 0; } if (ast_sip_push_task(session->serializer, send_direct_media_request, cdata)) { ast_log(LOG_ERROR, "Unable to send direct media request for channel %s\n", ast_channel_name(chan)); ao2_ref(cdata, -1); } return 0; } /*! \brief Local glue for interacting with the RTP engine core */ static struct ast_rtp_glue chan_pjsip_rtp_glue = { .type = "PJSIP", .get_rtp_info = chan_pjsip_get_rtp_peer, .get_vrtp_info = chan_pjsip_get_vrtp_peer, .get_codec = chan_pjsip_get_codec, .update_peer = chan_pjsip_set_rtp_peer, }; static void set_channel_on_rtp_instance(const struct ast_sip_session *session, const char *channel_id) { int i; for (i = 0; i < AST_VECTOR_SIZE(&session->active_media_state->sessions); ++i) { struct ast_sip_session_media *session_media; session_media = AST_VECTOR_GET(&session->active_media_state->sessions, i); if (!session_media || !session_media->rtp) { continue; } ast_rtp_instance_set_channel_id(session_media->rtp, channel_id); } } /*! * \brief Determine if a topology is compatible with format capabilities * * This will return true if ANY formats in the topology are compatible with the format * capabilities. * * XXX When supporting true multistream, we will need to be sure to mark which streams from * top1 are compatible with which streams from top2. Then the ones that are not compatible * will need to be marked as "removed" so that they are negotiated as expected. * * \param top Topology * \param cap Format capabilities * \retval 1 The topology has at least one compatible format * \retval 0 The topology has no compatible formats or an error occurred. */ static int compatible_formats_exist(struct ast_stream_topology *top, struct ast_format_cap *cap) { struct ast_format_cap *cap_from_top; int res; cap_from_top = ast_format_cap_from_stream_topology(top); if (!cap_from_top) { return 0; } res = ast_format_cap_iscompatible(cap_from_top, cap); ao2_ref(cap_from_top, -1); return res; } /*! \brief Function called to create a new PJSIP Asterisk channel */ static struct ast_channel *chan_pjsip_new(struct ast_sip_session *session, int state, const char *exten, const char *title, const struct ast_assigned_ids *assignedids, const struct ast_channel *requestor, const char *cid_name) { struct ast_channel *chan; struct ast_format_cap *caps; RAII_VAR(struct chan_pjsip_pvt *, pvt, NULL, ao2_cleanup); struct ast_sip_channel_pvt *channel; struct ast_variable *var; struct ast_stream_topology *topology; if (!(pvt = ao2_alloc_options(sizeof(*pvt), chan_pjsip_pvt_dtor, AO2_ALLOC_OPT_LOCK_NOLOCK))) { return NULL; } chan = ast_channel_alloc_with_endpoint(1, state, S_COR(session->id.number.valid, session->id.number.str, ""), S_COR(session->id.name.valid, session->id.name.str, ""), session->endpoint->accountcode, exten, session->endpoint->context, assignedids, requestor, 0, session->endpoint->persistent, "PJSIP/%s-%08x", ast_sorcery_object_get_id(session->endpoint), (unsigned) ast_atomic_fetchadd_int((int *) &chan_idx, +1)); if (!chan) { return NULL; } ast_channel_tech_set(chan, &chan_pjsip_tech); if (!(channel = ast_sip_channel_pvt_alloc(pvt, session))) { ast_channel_unlock(chan); ast_hangup(chan); return NULL; } ast_channel_tech_pvt_set(chan, channel); if (!ast_stream_topology_get_count(session->pending_media_state->topology) || !compatible_formats_exist(session->pending_media_state->topology, session->endpoint->media.codecs)) { caps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT); if (!caps) { ast_channel_unlock(chan); ast_hangup(chan); return NULL; } ast_format_cap_append_from_cap(caps, session->endpoint->media.codecs, AST_MEDIA_TYPE_UNKNOWN); topology = ast_stream_topology_clone(session->endpoint->media.topology); } else { caps = ast_format_cap_from_stream_topology(session->pending_media_state->topology); topology = ast_stream_topology_clone(session->pending_media_state->topology); } if (!topology || !caps) { ao2_cleanup(caps); ast_stream_topology_free(topology); ast_channel_unlock(chan); ast_hangup(chan); return NULL; } ast_channel_stage_snapshot(chan); ast_channel_nativeformats_set(chan, caps); ast_channel_set_stream_topology(chan, topology); if (!ast_format_cap_empty(caps)) { struct ast_format *fmt; fmt = ast_format_cap_get_best_by_type(caps, AST_MEDIA_TYPE_AUDIO); if (!fmt) { /* Since our capabilities aren't empty, this will succeed */ fmt = ast_format_cap_get_format(caps, 0); } ast_channel_set_writeformat(chan, fmt); ast_channel_set_rawwriteformat(chan, fmt); ast_channel_set_readformat(chan, fmt); ast_channel_set_rawreadformat(chan, fmt); ao2_ref(fmt, -1); } ao2_ref(caps, -1); if (state == AST_STATE_RING) { ast_channel_rings_set(chan, 1); } ast_channel_adsicpe_set(chan, AST_ADSI_UNAVAILABLE); ast_party_id_copy(&ast_channel_caller(chan)->id, &session->id); ast_party_id_copy(&ast_channel_caller(chan)->ani, &session->id); if (!ast_strlen_zero(exten)) { /* Set provided DNID on the new channel. */ ast_channel_dialed(chan)->number.str = ast_strdup(exten); } ast_channel_priority_set(chan, 1); ast_channel_callgroup_set(chan, session->endpoint->pickup.callgroup); ast_channel_pickupgroup_set(chan, session->endpoint->pickup.pickupgroup); ast_channel_named_callgroups_set(chan, session->endpoint->pickup.named_callgroups); ast_channel_named_pickupgroups_set(chan, session->endpoint->pickup.named_pickupgroups); if (!ast_strlen_zero(session->endpoint->language)) { ast_channel_language_set(chan, session->endpoint->language); } if (!ast_strlen_zero(session->endpoint->zone)) { struct ast_tone_zone *zone = ast_get_indication_zone(session->endpoint->zone); if (!zone) { ast_log(LOG_ERROR, "Unknown country code '%s' for tonezone. Check indications.conf for available country codes.\n", session->endpoint->zone); } ast_channel_zone_set(chan, zone); } for (var = session->endpoint->channel_vars; var; var = var->next) { char buf[512]; pbx_builtin_setvar_helper(chan, var->name, ast_get_encoded_str( var->value, buf, sizeof(buf))); } ast_channel_stage_snapshot_done(chan); ast_channel_unlock(chan); set_channel_on_rtp_instance(session, ast_channel_uniqueid(chan)); return chan; } static int answer(void *data) { pj_status_t status = PJ_SUCCESS; pjsip_tx_data *packet = NULL; struct ast_sip_session *session = data; if (session->inv_session->state == PJSIP_INV_STATE_DISCONNECTED) { ast_log(LOG_ERROR, "Session already DISCONNECTED [reason=%d (%s)]\n", session->inv_session->cause, pjsip_get_status_text(session->inv_session->cause)->ptr); #ifdef HAVE_PJSIP_INV_SESSION_REF pjsip_inv_dec_ref(session->inv_session); #endif return 0; } pjsip_dlg_inc_lock(session->inv_session->dlg); if (session->inv_session->invite_tsx) { status = pjsip_inv_answer(session->inv_session, 200, NULL, NULL, &packet); } else { ast_log(LOG_ERROR,"Cannot answer '%s' because there is no associated SIP transaction\n", ast_channel_name(session->channel)); } pjsip_dlg_dec_lock(session->inv_session->dlg); if (status == PJ_SUCCESS && packet) { ast_sip_session_send_response(session, packet); } #ifdef HAVE_PJSIP_INV_SESSION_REF pjsip_inv_dec_ref(session->inv_session); #endif if (status != PJ_SUCCESS) { char err[PJ_ERR_MSG_SIZE]; pj_strerror(status, err, sizeof(err)); ast_log(LOG_WARNING,"Cannot answer '%s': %s\n", ast_channel_name(session->channel), err); /* * Return this value so we can distinguish between this * failure and the threadpool synchronous push failing. */ return -2; } return 0; } /*! \brief Function called by core when we should answer a PJSIP session */ static int chan_pjsip_answer(struct ast_channel *ast) { struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast); struct ast_sip_session *session; int res; if (ast_channel_state(ast) == AST_STATE_UP) { return 0; } ast_setstate(ast, AST_STATE_UP); session = ao2_bump(channel->session); #ifdef HAVE_PJSIP_INV_SESSION_REF if (pjsip_inv_add_ref(session->inv_session) != PJ_SUCCESS) { ast_log(LOG_ERROR, "Can't increase the session reference counter\n"); ao2_ref(session, -1); return -1; } #endif /* the answer task needs to be pushed synchronously otherwise a race condition can occur between this thread and bridging (specifically when native bridging attempts to do direct media) */ ast_channel_unlock(ast); res = ast_sip_push_task_synchronous(session->serializer, answer, session); if (res) { if (res == -1) { ast_log(LOG_ERROR,"Cannot answer '%s': Unable to push answer task to the threadpool.\n", ast_channel_name(session->channel)); #ifdef HAVE_PJSIP_INV_SESSION_REF pjsip_inv_dec_ref(session->inv_session); #endif } ao2_ref(session, -1); ast_channel_lock(ast); return -1; } ao2_ref(session, -1); ast_channel_lock(ast); return 0; } /*! \brief Internal helper function called when CNG tone is detected */ static struct ast_frame *chan_pjsip_cng_tone_detected(struct ast_sip_session *session, struct ast_frame *f) { const char *target_context; int exists; int dsp_features; dsp_features = ast_dsp_get_features(session->dsp); dsp_features &= ~DSP_FEATURE_FAX_DETECT; if (dsp_features) { ast_dsp_set_features(session->dsp, dsp_features); } else { ast_dsp_free(session->dsp); session->dsp = NULL; } /* If already executing in the fax extension don't do anything */ if (!strcmp(ast_channel_exten(session->channel), "fax")) { return f; } target_context = S_OR(ast_channel_macrocontext(session->channel), ast_channel_context(session->channel)); /* * We need to unlock the channel here because ast_exists_extension has the * potential to start and stop an autoservice on the channel. Such action * is prone to deadlock if the channel is locked. * * ast_async_goto() has its own restriction on not holding the channel lock. */ ast_channel_unlock(session->channel); ast_frfree(f); f = &ast_null_frame; exists = ast_exists_extension(session->channel, target_context, "fax", 1, S_COR(ast_channel_caller(session->channel)->id.number.valid, ast_channel_caller(session->channel)->id.number.str, NULL)); if (exists) { ast_verb(2, "Redirecting '%s' to fax extension due to CNG detection\n", ast_channel_name(session->channel)); pbx_builtin_setvar_helper(session->channel, "FAXEXTEN", ast_channel_exten(session->channel)); if (ast_async_goto(session->channel, target_context, "fax", 1)) { ast_log(LOG_ERROR, "Failed to async goto '%s' into fax extension in '%s'\n", ast_channel_name(session->channel), target_context); } } else { ast_log(LOG_NOTICE, "FAX CNG detected on '%s' but no fax extension in '%s'\n", ast_channel_name(session->channel), target_context); } ast_channel_lock(session->channel); return f; } /*! * \brief Function called by core to read any waiting frames * * \note The channel is already locked. */ static struct ast_frame *chan_pjsip_read_stream(struct ast_channel *ast) { struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast); struct ast_sip_session *session = channel->session; struct ast_sip_session_media_read_callback_state *callback_state; struct ast_frame *f; int fdno = ast_channel_fdno(ast) - AST_EXTENDED_FDS; if (fdno >= AST_VECTOR_SIZE(&session->active_media_state->read_callbacks)) { return &ast_null_frame; } callback_state = AST_VECTOR_GET_ADDR(&session->active_media_state->read_callbacks, fdno); f = callback_state->read_callback(session, callback_state->session); if (!f) { return f; } if (f->frametype != AST_FRAME_VOICE || callback_state->session != session->active_media_state->default_session[callback_state->session->type]) { return f; } session = channel->session; /* * Asymmetric RTP only has one native format set at a time. * Therefore we need to update the native format to the current * raw read format BEFORE the native format check */ if (!session->endpoint->asymmetric_rtp_codec && ast_format_cmp(ast_channel_rawwriteformat(ast), f->subclass.format) == AST_FORMAT_CMP_NOT_EQUAL) { struct ast_format_cap *caps; /* For maximum compatibility we ensure that the formats match that of the received media */ ast_debug(1, "Oooh, got a frame with format of %s on channel '%s' when we're sending '%s', switching to match\n", ast_format_get_name(f->subclass.format), ast_channel_name(ast), ast_format_get_name(ast_channel_rawwriteformat(ast))); caps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT); if (caps) { ast_format_cap_append_from_cap(caps, ast_channel_nativeformats(ast), AST_MEDIA_TYPE_UNKNOWN); ast_format_cap_remove_by_type(caps, AST_MEDIA_TYPE_AUDIO); ast_format_cap_append(caps, f->subclass.format, 0); ast_channel_nativeformats_set(ast, caps); ao2_ref(caps, -1); } ast_set_write_format_path(ast, ast_channel_writeformat(ast), f->subclass.format); ast_set_read_format_path(ast, ast_channel_readformat(ast), f->subclass.format); if (ast_channel_is_bridged(ast)) { ast_channel_set_unbridged_nolock(ast, 1); } } if (ast_format_cap_iscompatible_format(ast_channel_nativeformats(ast), f->subclass.format) == AST_FORMAT_CMP_NOT_EQUAL) { ast_debug(1, "Oooh, got a frame with format of %s on channel '%s' when it has not been negotiated\n", ast_format_get_name(f->subclass.format), ast_channel_name(ast)); ast_frfree(f); return &ast_null_frame; } if (session->dsp) { int dsp_features; dsp_features = ast_dsp_get_features(session->dsp); if ((dsp_features & DSP_FEATURE_FAX_DETECT) && session->endpoint->faxdetect_timeout && session->endpoint->faxdetect_timeout <= ast_channel_get_up_time(ast)) { dsp_features &= ~DSP_FEATURE_FAX_DETECT; if (dsp_features) { ast_dsp_set_features(session->dsp, dsp_features); } else { ast_dsp_free(session->dsp); session->dsp = NULL; } ast_debug(3, "Channel driver fax CNG detection timeout on %s\n", ast_channel_name(ast)); } } if (session->dsp) { f = ast_dsp_process(ast, session->dsp, f); if (f && (f->frametype == AST_FRAME_DTMF)) { if (f->subclass.integer == 'f') { ast_debug(3, "Channel driver fax CNG detected on %s\n", ast_channel_name(ast)); f = chan_pjsip_cng_tone_detected(session, f); } else { ast_debug(3, "* Detected inband DTMF '%c' on '%s'\n", f->subclass.integer, ast_channel_name(ast)); } } } return f; } static int chan_pjsip_write_stream(struct ast_channel *ast, int stream_num, struct ast_frame *frame) { struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast); struct ast_sip_session *session = channel->session; struct ast_sip_session_media *media = NULL; int res = 0; /* The core provides a guarantee that the stream will exist when we are called if stream_num is provided */ if (stream_num >= 0) { /* What is not guaranteed is that a media session will exist */ if (stream_num < AST_VECTOR_SIZE(&channel->session->active_media_state->sessions)) { media = AST_VECTOR_GET(&channel->session->active_media_state->sessions, stream_num); } } switch (frame->frametype) { case AST_FRAME_VOICE: if (!media) { return 0; } else if (media->type != AST_MEDIA_TYPE_AUDIO) { ast_debug(3, "Channel %s stream %d is of type '%s', not audio!\n", ast_channel_name(ast), stream_num, ast_codec_media_type2str(media->type)); return 0; } else if (media == channel->session->active_media_state->default_session[AST_MEDIA_TYPE_AUDIO] && ast_format_cap_iscompatible_format(ast_channel_nativeformats(ast), frame->subclass.format) == AST_FORMAT_CMP_NOT_EQUAL) { struct ast_str *cap_buf = ast_str_alloca(AST_FORMAT_CAP_NAMES_LEN); struct ast_str *write_transpath = ast_str_alloca(256); struct ast_str *read_transpath = ast_str_alloca(256); ast_log(LOG_WARNING, "Channel %s asked to send %s frame when native formats are %s (rd:%s->%s;%s wr:%s->%s;%s)\n", ast_channel_name(ast), ast_format_get_name(frame->subclass.format), ast_format_cap_get_names(ast_channel_nativeformats(ast), &cap_buf), ast_format_get_name(ast_channel_rawreadformat(ast)), ast_format_get_name(ast_channel_readformat(ast)), ast_translate_path_to_str(ast_channel_readtrans(ast), &read_transpath), ast_format_get_name(ast_channel_writeformat(ast)), ast_format_get_name(ast_channel_rawwriteformat(ast)), ast_translate_path_to_str(ast_channel_writetrans(ast), &write_transpath)); return 0; } else if (media->write_callback) { res = media->write_callback(session, media, frame); } break; case AST_FRAME_VIDEO: if (!media) { return 0; } else if (media->type != AST_MEDIA_TYPE_VIDEO) { ast_debug(3, "Channel %s stream %d is of type '%s', not video!\n", ast_channel_name(ast), stream_num, ast_codec_media_type2str(media->type)); return 0; } else if (media->write_callback) { res = media->write_callback(session, media, frame); } break; case AST_FRAME_MODEM: if (!media) { return 0; } else if (media->type != AST_MEDIA_TYPE_IMAGE) { ast_debug(3, "Channel %s stream %d is of type '%s', not image!\n", ast_channel_name(ast), stream_num, ast_codec_media_type2str(media->type)); return 0; } else if (media->write_callback) { res = media->write_callback(session, media, frame); } break; case AST_FRAME_CNG: break; default: ast_log(LOG_WARNING, "Can't send %u type frames with PJSIP\n", frame->frametype); break; } return res; } static int chan_pjsip_write(struct ast_channel *ast, struct ast_frame *frame) { return chan_pjsip_write_stream(ast, -1, frame); } /*! \brief Function called by core to change the underlying owner channel */ static int chan_pjsip_fixup(struct ast_channel *oldchan, struct ast_channel *newchan) { struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(newchan); if (channel->session->channel != oldchan) { return -1; } /* * The masquerade has suspended the channel's session * serializer so we can safely change it outside of * the serializer thread. */ channel->session->channel = newchan; set_channel_on_rtp_instance(channel->session, ast_channel_uniqueid(newchan)); return 0; } /*! AO2 hash function for on hold UIDs */ static int uid_hold_hash_fn(const void *obj, const int flags) { const char *key = obj; switch (flags & OBJ_SEARCH_MASK) { case OBJ_SEARCH_KEY: break; case OBJ_SEARCH_OBJECT: break; default: /* Hash can only work on something with a full key. */ ast_assert(0); return 0; } return ast_str_hash(key); } /*! AO2 sort function for on hold UIDs */ static int uid_hold_sort_fn(const void *obj_left, const void *obj_right, const int flags) { const char *left = obj_left; const char *right = obj_right; int cmp; switch (flags & OBJ_SEARCH_MASK) { case OBJ_SEARCH_OBJECT: case OBJ_SEARCH_KEY: cmp = strcmp(left, right); break; case OBJ_SEARCH_PARTIAL_KEY: cmp = strncmp(left, right, strlen(right)); break; default: /* Sort can only work on something with a full or partial key. */ ast_assert(0); cmp = 0; break; } return cmp; } static struct ao2_container *pjsip_uids_onhold; /*! * \brief Add a channel ID to the list of PJSIP channels on hold * * \param chan_uid - Unique ID of the channel being put into the hold list * * \retval 0 Channel has been added to or was already in the hold list * \retval -1 Failed to add channel to the hold list */ static int chan_pjsip_add_hold(const char *chan_uid) { RAII_VAR(char *, hold_uid, NULL, ao2_cleanup); hold_uid = ao2_find(pjsip_uids_onhold, chan_uid, OBJ_SEARCH_KEY); if (hold_uid) { /* Device is already on hold. Nothing to do. */ return 0; } /* Device wasn't in hold list already. Create a new one. */ hold_uid = ao2_alloc_options(strlen(chan_uid) + 1, NULL, AO2_ALLOC_OPT_LOCK_NOLOCK); if (!hold_uid) { return -1; } ast_copy_string(hold_uid, chan_uid, strlen(chan_uid) + 1); if (ao2_link(pjsip_uids_onhold, hold_uid) == 0) { return -1; } return 0; } /*! * \brief Remove a channel ID from the list of PJSIP channels on hold * * \param chan_uid - Unique ID of the channel being taken out of the hold list */ static void chan_pjsip_remove_hold(const char *chan_uid) { ao2_find(pjsip_uids_onhold, chan_uid, OBJ_SEARCH_KEY | OBJ_UNLINK | OBJ_NODATA); } /*! * \brief Determine whether a channel ID is in the list of PJSIP channels on hold * * \param chan_uid - Channel being checked * * \retval 0 The channel is not in the hold list * \retval 1 The channel is in the hold list */ static int chan_pjsip_get_hold(const char *chan_uid) { RAII_VAR(char *, hold_uid, NULL, ao2_cleanup); hold_uid = ao2_find(pjsip_uids_onhold, chan_uid, OBJ_SEARCH_KEY); if (!hold_uid) { return 0; } return 1; } /*! \brief Function called to get the device state of an endpoint */ static int chan_pjsip_devicestate(const char *data) { RAII_VAR(struct ast_sip_endpoint *, endpoint, ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "endpoint", data), ao2_cleanup); enum ast_device_state state = AST_DEVICE_UNKNOWN; RAII_VAR(struct ast_endpoint_snapshot *, endpoint_snapshot, NULL, ao2_cleanup); RAII_VAR(struct stasis_cache *, cache, NULL, ao2_cleanup); struct ast_devstate_aggregate aggregate; int num, inuse = 0; if (!endpoint) { return AST_DEVICE_INVALID; } endpoint_snapshot = ast_endpoint_latest_snapshot(ast_endpoint_get_tech(endpoint->persistent), ast_endpoint_get_resource(endpoint->persistent)); if (!endpoint_snapshot) { return AST_DEVICE_INVALID; } if (endpoint_snapshot->state == AST_ENDPOINT_OFFLINE) { state = AST_DEVICE_UNAVAILABLE; } else if (endpoint_snapshot->state == AST_ENDPOINT_ONLINE) { state = AST_DEVICE_NOT_INUSE; } if (!endpoint_snapshot->num_channels || !(cache = ast_channel_cache())) { return state; } ast_devstate_aggregate_init(&aggregate); ao2_ref(cache, +1); for (num = 0; num < endpoint_snapshot->num_channels; num++) { RAII_VAR(struct stasis_message *, msg, NULL, ao2_cleanup); struct ast_channel_snapshot *snapshot; msg = stasis_cache_get(cache, ast_channel_snapshot_type(), endpoint_snapshot->channel_ids[num]); if (!msg) { continue; } snapshot = stasis_message_data(msg); if (chan_pjsip_get_hold(snapshot->uniqueid)) { ast_devstate_aggregate_add(&aggregate, AST_DEVICE_ONHOLD); } else { ast_devstate_aggregate_add(&aggregate, ast_state_chan2dev(snapshot->state)); } if ((snapshot->state == AST_STATE_UP) || (snapshot->state == AST_STATE_RING) || (snapshot->state == AST_STATE_BUSY)) { inuse++; } } if (endpoint->devicestate_busy_at && (inuse == endpoint->devicestate_busy_at)) { state = AST_DEVICE_BUSY; } else if (ast_devstate_aggregate_result(&aggregate) != AST_DEVICE_INVALID) { state = ast_devstate_aggregate_result(&aggregate); } return state; } /*! \brief Function called to query options on a channel */ static int chan_pjsip_queryoption(struct ast_channel *ast, int option, void *data, int *datalen) { struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast); struct ast_sip_session *session = channel->session; int res = -1; enum ast_t38_state state = T38_STATE_UNAVAILABLE; switch (option) { case AST_OPTION_T38_STATE: if (session->endpoint->media.t38.enabled) { switch (session->t38state) { case T38_LOCAL_REINVITE: case T38_PEER_REINVITE: state = T38_STATE_NEGOTIATING; break; case T38_ENABLED: state = T38_STATE_NEGOTIATED; break; case T38_REJECTED: state = T38_STATE_REJECTED; break; default: state = T38_STATE_UNKNOWN; break; } } *((enum ast_t38_state *) data) = state; res = 0; break; default: break; } return res; } static const char *chan_pjsip_get_uniqueid(struct ast_channel *ast) { struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast); char *uniqueid = ast_threadstorage_get(&uniqueid_threadbuf, UNIQUEID_BUFSIZE); if (!uniqueid) { return ""; } ast_copy_pj_str(uniqueid, &channel->session->inv_session->dlg->call_id->id, UNIQUEID_BUFSIZE); return uniqueid; } struct indicate_data { struct ast_sip_session *session; int condition; int response_code; void *frame_data; size_t datalen; }; static void indicate_data_destroy(void *obj) { struct indicate_data *ind_data = obj; ast_free(ind_data->frame_data); ao2_ref(ind_data->session, -1); } static struct indicate_data *indicate_data_alloc(struct ast_sip_session *session, int condition, int response_code, const void *frame_data, size_t datalen) { struct indicate_data *ind_data = ao2_alloc(sizeof(*ind_data), indicate_data_destroy); if (!ind_data) { return NULL; } ind_data->frame_data = ast_malloc(datalen); if (!ind_data->frame_data) { ao2_ref(ind_data, -1); return NULL; } memcpy(ind_data->frame_data, frame_data, datalen); ind_data->datalen = datalen; ind_data->condition = condition; ind_data->response_code = response_code; ao2_ref(session, +1); ind_data->session = session; return ind_data; } static int indicate(void *data) { pjsip_tx_data *packet = NULL; struct indicate_data *ind_data = data; struct ast_sip_session *session = ind_data->session; int response_code = ind_data->response_code; if ((session->inv_session->state != PJSIP_INV_STATE_DISCONNECTED) && (pjsip_inv_answer(session->inv_session, response_code, NULL, NULL, &packet) == PJ_SUCCESS)) { ast_sip_session_send_response(session, packet); } #ifdef HAVE_PJSIP_INV_SESSION_REF pjsip_inv_dec_ref(session->inv_session); #endif ao2_ref(ind_data, -1); return 0; } /*! \brief Send SIP INFO with video update request */ static int transmit_info_with_vidupdate(void *data) { const char * xml = "\r\n" " \r\n" " \r\n" " \r\n" " \r\n" " \r\n" " \r\n" " \r\n"; const struct ast_sip_body body = { .type = "application", .subtype = "media_control+xml", .body_text = xml }; RAII_VAR(struct ast_sip_session *, session, data, ao2_cleanup); struct pjsip_tx_data *tdata; if (session->inv_session->state == PJSIP_INV_STATE_DISCONNECTED) { ast_log(LOG_ERROR, "Session already DISCONNECTED [reason=%d (%s)]\n", session->inv_session->cause, pjsip_get_status_text(session->inv_session->cause)->ptr); goto failure; } if (ast_sip_create_request("INFO", session->inv_session->dlg, session->endpoint, NULL, NULL, &tdata)) { ast_log(LOG_ERROR, "Could not create text video update INFO request\n"); goto failure; } if (ast_sip_add_body(tdata, &body)) { ast_log(LOG_ERROR, "Could not add body to text video update INFO request\n"); goto failure; } ast_sip_session_send_request(session, tdata); #ifdef HAVE_PJSIP_INV_SESSION_REF pjsip_inv_dec_ref(session->inv_session); #endif return 0; failure: #ifdef HAVE_PJSIP_INV_SESSION_REF pjsip_inv_dec_ref(session->inv_session); #endif return -1; } /*! * \internal * \brief TRUE if a COLP update can be sent to the peer. * \since 13.3.0 * * \param session The session to see if the COLP update is allowed. * * \retval 0 Update is not allowed. * \retval 1 Update is allowed. */ static int is_colp_update_allowed(struct ast_sip_session *session) { struct ast_party_id connected_id; int update_allowed = 0; if (!session->endpoint->id.send_pai && !session->endpoint->id.send_rpid) { return 0; } /* * Check if privacy allows the update. Check while the channel * is locked so we can work with the shallow connected_id copy. */ ast_channel_lock(session->channel); connected_id = ast_channel_connected_effective_id(session->channel); if (connected_id.number.valid && (session->endpoint->id.trust_outbound || (ast_party_id_presentation(&connected_id) & AST_PRES_RESTRICTION) == AST_PRES_ALLOWED)) { update_allowed = 1; } ast_channel_unlock(session->channel); return update_allowed; } /*! \brief Update connected line information */ static int update_connected_line_information(void *data) { struct ast_sip_session *session = data; if (session->inv_session->state == PJSIP_INV_STATE_DISCONNECTED) { ast_log(LOG_ERROR, "Session already DISCONNECTED [reason=%d (%s)]\n", session->inv_session->cause, pjsip_get_status_text(session->inv_session->cause)->ptr); #ifdef HAVE_PJSIP_INV_SESSION_REF pjsip_inv_dec_ref(session->inv_session); #endif ao2_ref(session, -1); return -1; } if (ast_channel_state(session->channel) == AST_STATE_UP || session->inv_session->role == PJSIP_ROLE_UAC) { if (is_colp_update_allowed(session)) { enum ast_sip_session_refresh_method method; int generate_new_sdp; method = session->endpoint->id.refresh_method; if (session->inv_session->options & PJSIP_INV_SUPPORT_UPDATE) { method = AST_SIP_SESSION_REFRESH_METHOD_UPDATE; } /* Only the INVITE method actually needs SDP, UPDATE can do without */ generate_new_sdp = (method == AST_SIP_SESSION_REFRESH_METHOD_INVITE); ast_sip_session_refresh(session, NULL, NULL, NULL, method, generate_new_sdp, NULL); } } else if (session->endpoint->id.rpid_immediate && session->inv_session->state != PJSIP_INV_STATE_DISCONNECTED && is_colp_update_allowed(session)) { int response_code = 0; if (ast_channel_state(session->channel) == AST_STATE_RING) { response_code = !session->endpoint->inband_progress ? 180 : 183; } else if (ast_channel_state(session->channel) == AST_STATE_RINGING) { response_code = 183; } if (response_code) { struct pjsip_tx_data *packet = NULL; if (pjsip_inv_answer(session->inv_session, response_code, NULL, NULL, &packet) == PJ_SUCCESS) { ast_sip_session_send_response(session, packet); } } } #ifdef HAVE_PJSIP_INV_SESSION_REF pjsip_inv_dec_ref(session->inv_session); #endif ao2_ref(session, -1); return 0; } /*! \brief Callback which changes the value of locally held on the media stream */ static void local_hold_set_state(struct ast_sip_session_media *session_media, unsigned int held) { if (session_media) { session_media->locally_held = held; } } /*! \brief Update local hold state and send a re-INVITE with the new SDP */ static int remote_send_hold_refresh(struct ast_sip_session *session, unsigned int held) { AST_VECTOR_CALLBACK_VOID(&session->active_media_state->sessions, local_hold_set_state, held); ast_sip_session_refresh(session, NULL, NULL, NULL, AST_SIP_SESSION_REFRESH_METHOD_INVITE, 1, NULL); ao2_ref(session, -1); return 0; } /*! \brief Update local hold state to be held */ static int remote_send_hold(void *data) { return remote_send_hold_refresh(data, 1); } /*! \brief Update local hold state to be unheld */ static int remote_send_unhold(void *data) { return remote_send_hold_refresh(data, 0); } struct topology_change_refresh_data { struct ast_sip_session *session; struct ast_sip_session_media_state *media_state; }; static void topology_change_refresh_data_free(struct topology_change_refresh_data *refresh_data) { ao2_cleanup(refresh_data->session); ast_sip_session_media_state_free(refresh_data->media_state); ast_free(refresh_data); } static struct topology_change_refresh_data *topology_change_refresh_data_alloc( struct ast_sip_session *session, const struct ast_stream_topology *topology) { struct topology_change_refresh_data *refresh_data; refresh_data = ast_calloc(1, sizeof(*refresh_data)); if (!refresh_data) { return NULL; } refresh_data->session = ao2_bump(session); refresh_data->media_state = ast_sip_session_media_state_alloc(); if (!refresh_data->media_state) { topology_change_refresh_data_free(refresh_data); return NULL; } refresh_data->media_state->topology = ast_stream_topology_clone(topology); if (!refresh_data->media_state->topology) { topology_change_refresh_data_free(refresh_data); return NULL; } return refresh_data; } static int on_topology_change_response(struct ast_sip_session *session, pjsip_rx_data *rdata) { if (PJSIP_IS_STATUS_IN_CLASS(rdata->msg_info.msg->line.status.code, 200)) { /* The topology was changed to something new so give notice to what requested * it so it queries the channel and updates accordingly. */ if (session->channel) { ast_queue_control(session->channel, AST_CONTROL_STREAM_TOPOLOGY_CHANGED); } } else if (300 <= rdata->msg_info.msg->line.status.code) { /* The topology change failed, so drop the current pending media state */ ast_sip_session_media_state_reset(session->pending_media_state); } return 0; } static int send_topology_change_refresh(void *data) { struct topology_change_refresh_data *refresh_data = data; int ret; ret = ast_sip_session_refresh(refresh_data->session, NULL, NULL, on_topology_change_response, AST_SIP_SESSION_REFRESH_METHOD_INVITE, 1, refresh_data->media_state); refresh_data->media_state = NULL; topology_change_refresh_data_free(refresh_data); return ret; } static int handle_topology_request_change(struct ast_sip_session *session, const struct ast_stream_topology *proposed) { struct topology_change_refresh_data *refresh_data; int res; refresh_data = topology_change_refresh_data_alloc(session, proposed); if (!refresh_data) { return -1; } res = ast_sip_push_task(session->serializer, send_topology_change_refresh, refresh_data); if (res) { topology_change_refresh_data_free(refresh_data); } return res; } /*! \brief Function called by core to ask the channel to indicate some sort of condition */ static int chan_pjsip_indicate(struct ast_channel *ast, int condition, const void *data, size_t datalen) { struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast); struct ast_sip_session_media *media; int response_code = 0; int res = 0; char *device_buf; size_t device_buf_size; int i; const struct ast_stream_topology *topology; switch (condition) { case AST_CONTROL_RINGING: if (ast_channel_state(ast) == AST_STATE_RING) { if (channel->session->endpoint->inband_progress) { response_code = 183; res = -1; } else { response_code = 180; } } else { res = -1; } ast_devstate_changed(AST_DEVICE_UNKNOWN, AST_DEVSTATE_CACHABLE, "PJSIP/%s", ast_sorcery_object_get_id(channel->session->endpoint)); break; case AST_CONTROL_BUSY: if (ast_channel_state(ast) != AST_STATE_UP) { response_code = 486; } else { res = -1; } break; case AST_CONTROL_CONGESTION: if (ast_channel_state(ast) != AST_STATE_UP) { response_code = 503; } else { res = -1; } break; case AST_CONTROL_INCOMPLETE: if (ast_channel_state(ast) != AST_STATE_UP) { response_code = 484; } else { res = -1; } break; case AST_CONTROL_PROCEEDING: if (ast_channel_state(ast) != AST_STATE_UP) { response_code = 100; } else { res = -1; } break; case AST_CONTROL_PROGRESS: if (ast_channel_state(ast) != AST_STATE_UP) { response_code = 183; } else { res = -1; } ast_devstate_changed(AST_DEVICE_UNKNOWN, AST_DEVSTATE_CACHABLE, "PJSIP/%s", ast_sorcery_object_get_id(channel->session->endpoint)); break; case AST_CONTROL_VIDUPDATE: for (i = 0; i < AST_VECTOR_SIZE(&channel->session->active_media_state->sessions); ++i) { media = AST_VECTOR_GET(&channel->session->active_media_state->sessions, i); if (!media || media->type != AST_MEDIA_TYPE_VIDEO) { continue; } if (media->rtp) { /* FIXME: Only use this for VP8. Additional work would have to be done to * fully support other video codecs */ if (ast_format_cap_iscompatible_format(ast_channel_nativeformats(ast), ast_format_vp8) != AST_FORMAT_CMP_NOT_EQUAL || ast_format_cap_iscompatible_format(ast_channel_nativeformats(ast), ast_format_vp9) != AST_FORMAT_CMP_NOT_EQUAL || (channel->session->endpoint->media.webrtc && ast_format_cap_iscompatible_format(ast_channel_nativeformats(ast), ast_format_h264) != AST_FORMAT_CMP_NOT_EQUAL)) { /* FIXME Fake RTP write, this will be sent as an RTCP packet. Ideally the * RTP engine would provide a way to externally write/schedule RTCP * packets */ struct ast_frame fr; fr.frametype = AST_FRAME_CONTROL; fr.subclass.integer = AST_CONTROL_VIDUPDATE; res = ast_rtp_instance_write(media->rtp, &fr); } else { ao2_ref(channel->session, +1); #ifdef HAVE_PJSIP_INV_SESSION_REF if (pjsip_inv_add_ref(channel->session->inv_session) != PJ_SUCCESS) { ast_log(LOG_ERROR, "Can't increase the session reference counter\n"); ao2_cleanup(channel->session); } else { #endif if (ast_sip_push_task(channel->session->serializer, transmit_info_with_vidupdate, channel->session)) { ao2_cleanup(channel->session); } #ifdef HAVE_PJSIP_INV_SESSION_REF } #endif } ast_test_suite_event_notify("AST_CONTROL_VIDUPDATE", "Result: Success"); } else { ast_test_suite_event_notify("AST_CONTROL_VIDUPDATE", "Result: Failure"); res = -1; } } /* XXX If there were no video streams, then this should set * res to -1 */ break; case AST_CONTROL_CONNECTED_LINE: ao2_ref(channel->session, +1); #ifdef HAVE_PJSIP_INV_SESSION_REF if (pjsip_inv_add_ref(channel->session->inv_session) != PJ_SUCCESS) { ast_log(LOG_ERROR, "Can't increase the session reference counter\n"); ao2_cleanup(channel->session); return -1; } #endif if (ast_sip_push_task(channel->session->serializer, update_connected_line_information, channel->session)) { #ifdef HAVE_PJSIP_INV_SESSION_REF pjsip_inv_dec_ref(channel->session->inv_session); #endif ao2_cleanup(channel->session); } break; case AST_CONTROL_UPDATE_RTP_PEER: break; case AST_CONTROL_PVT_CAUSE_CODE: res = -1; break; case AST_CONTROL_MASQUERADE_NOTIFY: ast_assert(datalen == sizeof(int)); if (*(int *) data) { /* * Masquerade is beginning: * Wait for session serializer to get suspended. */ ast_channel_unlock(ast); ast_sip_session_suspend(channel->session); ast_channel_lock(ast); } else { /* * Masquerade is complete: * Unsuspend the session serializer. */ ast_sip_session_unsuspend(channel->session); } break; case AST_CONTROL_HOLD: chan_pjsip_add_hold(ast_channel_uniqueid(ast)); device_buf_size = strlen(ast_channel_name(ast)) + 1; device_buf = alloca(device_buf_size); ast_channel_get_device_name(ast, device_buf, device_buf_size); ast_devstate_changed_literal(AST_DEVICE_ONHOLD, 1, device_buf); if (!channel->session->endpoint->moh_passthrough) { ast_moh_start(ast, data, NULL); } else { if (ast_sip_push_task(channel->session->serializer, remote_send_hold, ao2_bump(channel->session))) { ast_log(LOG_WARNING, "Could not queue task to remotely put session '%s' on hold with endpoint '%s'\n", ast_sorcery_object_get_id(channel->session), ast_sorcery_object_get_id(channel->session->endpoint)); ao2_ref(channel->session, -1); } } break; case AST_CONTROL_UNHOLD: chan_pjsip_remove_hold(ast_channel_uniqueid(ast)); device_buf_size = strlen(ast_channel_name(ast)) + 1; device_buf = alloca(device_buf_size); ast_channel_get_device_name(ast, device_buf, device_buf_size); ast_devstate_changed_literal(AST_DEVICE_UNKNOWN, 1, device_buf); if (!channel->session->endpoint->moh_passthrough) { ast_moh_stop(ast); } else { if (ast_sip_push_task(channel->session->serializer, remote_send_unhold, ao2_bump(channel->session))) { ast_log(LOG_WARNING, "Could not queue task to remotely take session '%s' off hold with endpoint '%s'\n", ast_sorcery_object_get_id(channel->session), ast_sorcery_object_get_id(channel->session->endpoint)); ao2_ref(channel->session, -1); } } break; case AST_CONTROL_SRCUPDATE: break; case AST_CONTROL_SRCCHANGE: break; case AST_CONTROL_REDIRECTING: if (ast_channel_state(ast) != AST_STATE_UP) { response_code = 181; } else { res = -1; } break; case AST_CONTROL_T38_PARAMETERS: res = 0; if (channel->session->t38state == T38_PEER_REINVITE) { const struct ast_control_t38_parameters *parameters = data; if (parameters->request_response == AST_T38_REQUEST_PARMS) { res = AST_T38_REQUEST_PARMS; } } break; case AST_CONTROL_STREAM_TOPOLOGY_REQUEST_CHANGE: topology = data; res = handle_topology_request_change(channel->session, topology); break; case AST_CONTROL_STREAM_TOPOLOGY_CHANGED: break; case AST_CONTROL_STREAM_TOPOLOGY_SOURCE_CHANGED: break; case -1: res = -1; break; default: ast_log(LOG_WARNING, "Don't know how to indicate condition %d\n", condition); res = -1; break; } if (response_code) { struct indicate_data *ind_data = indicate_data_alloc(channel->session, condition, response_code, data, datalen); if (!ind_data) { return -1; } #ifdef HAVE_PJSIP_INV_SESSION_REF if (pjsip_inv_add_ref(ind_data->session->inv_session) != PJ_SUCCESS) { ast_log(LOG_ERROR, "Can't increase the session reference counter\n"); ao2_cleanup(ind_data); return -1; } #endif if (ast_sip_push_task(channel->session->serializer, indicate, ind_data)) { ast_log(LOG_NOTICE, "Cannot send response code %d to endpoint %s. Could not queue task properly\n", response_code, ast_sorcery_object_get_id(channel->session->endpoint)); #ifdef HAVE_PJSIP_INV_SESSION_REF pjsip_inv_dec_ref(ind_data->session->inv_session); #endif ao2_cleanup(ind_data); res = -1; } } return res; } struct transfer_data { struct ast_sip_session *session; char *target; }; static void transfer_data_destroy(void *obj) { struct transfer_data *trnf_data = obj; ast_free(trnf_data->target); ao2_cleanup(trnf_data->session); } static struct transfer_data *transfer_data_alloc(struct ast_sip_session *session, const char *target) { struct transfer_data *trnf_data = ao2_alloc(sizeof(*trnf_data), transfer_data_destroy); if (!trnf_data) { return NULL; } if (!(trnf_data->target = ast_strdup(target))) { ao2_ref(trnf_data, -1); return NULL; } ao2_ref(session, +1); trnf_data->session = session; return trnf_data; } static void transfer_redirect(struct ast_sip_session *session, const char *target) { pjsip_tx_data *packet; enum ast_control_transfer message = AST_TRANSFER_SUCCESS; pjsip_contact_hdr *contact; pj_str_t tmp; if (pjsip_inv_end_session(session->inv_session, 302, NULL, &packet) != PJ_SUCCESS || !packet) { ast_log(LOG_WARNING, "Failed to redirect PJSIP session for channel %s\n", ast_channel_name(session->channel)); message = AST_TRANSFER_FAILED; ast_queue_control_data(session->channel, AST_CONTROL_TRANSFER, &message, sizeof(message)); return; } if (!(contact = pjsip_msg_find_hdr(packet->msg, PJSIP_H_CONTACT, NULL))) { contact = pjsip_contact_hdr_create(packet->pool); } pj_strdup2_with_null(packet->pool, &tmp, target); if (!(contact->uri = pjsip_parse_uri(packet->pool, tmp.ptr, tmp.slen, PJSIP_PARSE_URI_AS_NAMEADDR))) { ast_log(LOG_WARNING, "Failed to parse destination URI '%s' for channel %s\n", target, ast_channel_name(session->channel)); message = AST_TRANSFER_FAILED; ast_queue_control_data(session->channel, AST_CONTROL_TRANSFER, &message, sizeof(message)); pjsip_tx_data_dec_ref(packet); return; } pjsip_msg_add_hdr(packet->msg, (pjsip_hdr *) contact); ast_sip_session_send_response(session, packet); ast_queue_control_data(session->channel, AST_CONTROL_TRANSFER, &message, sizeof(message)); } static void transfer_refer(struct ast_sip_session *session, const char *target) { pjsip_evsub *sub; enum ast_control_transfer message = AST_TRANSFER_SUCCESS; pj_str_t tmp; pjsip_tx_data *packet; const char *ref_by_val; char local_info[pj_strlen(&session->inv_session->dlg->local.info_str) + 1]; if (pjsip_xfer_create_uac(session->inv_session->dlg, NULL, &sub) != PJ_SUCCESS) { message = AST_TRANSFER_FAILED; ast_queue_control_data(session->channel, AST_CONTROL_TRANSFER, &message, sizeof(message)); return; } if (pjsip_xfer_initiate(sub, pj_cstr(&tmp, target), &packet) != PJ_SUCCESS) { message = AST_TRANSFER_FAILED; ast_queue_control_data(session->channel, AST_CONTROL_TRANSFER, &message, sizeof(message)); pjsip_evsub_terminate(sub, PJ_FALSE); return; } ref_by_val = pbx_builtin_getvar_helper(session->channel, "SIPREFERREDBYHDR"); if (!ast_strlen_zero(ref_by_val)) { ast_sip_add_header(packet, "Referred-By", ref_by_val); } else { ast_copy_pj_str(local_info, &session->inv_session->dlg->local.info_str, sizeof(local_info)); ast_sip_add_header(packet, "Referred-By", local_info); } pjsip_xfer_send_request(sub, packet); ast_queue_control_data(session->channel, AST_CONTROL_TRANSFER, &message, sizeof(message)); } static int transfer(void *data) { struct transfer_data *trnf_data = data; struct ast_sip_endpoint *endpoint = NULL; struct ast_sip_contact *contact = NULL; const char *target = trnf_data->target; if (trnf_data->session->inv_session->state == PJSIP_INV_STATE_DISCONNECTED) { ast_log(LOG_ERROR, "Session already DISCONNECTED [reason=%d (%s)]\n", trnf_data->session->inv_session->cause, pjsip_get_status_text(trnf_data->session->inv_session->cause)->ptr); } else { /* See if we have an endpoint; if so, use its contact */ endpoint = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "endpoint", target); if (endpoint) { contact = ast_sip_location_retrieve_contact_from_aor_list(endpoint->aors); if (contact && !ast_strlen_zero(contact->uri)) { target = contact->uri; } } if (ast_channel_state(trnf_data->session->channel) == AST_STATE_RING) { transfer_redirect(trnf_data->session, target); } else { transfer_refer(trnf_data->session, target); } } #ifdef HAVE_PJSIP_INV_SESSION_REF pjsip_inv_dec_ref(trnf_data->session->inv_session); #endif ao2_ref(trnf_data, -1); ao2_cleanup(endpoint); ao2_cleanup(contact); return 0; } /*! \brief Function called by core for Asterisk initiated transfer */ static int chan_pjsip_transfer(struct ast_channel *chan, const char *target) { struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan); struct transfer_data *trnf_data = transfer_data_alloc(channel->session, target); if (!trnf_data) { return -1; } #ifdef HAVE_PJSIP_INV_SESSION_REF if (pjsip_inv_add_ref(trnf_data->session->inv_session) != PJ_SUCCESS) { ast_log(LOG_ERROR, "Can't increase the session reference counter\n"); ao2_cleanup(trnf_data); return -1; } #endif if (ast_sip_push_task(channel->session->serializer, transfer, trnf_data)) { ast_log(LOG_WARNING, "Error requesting transfer\n"); #ifdef HAVE_PJSIP_INV_SESSION_REF pjsip_inv_dec_ref(trnf_data->session->inv_session); #endif ao2_cleanup(trnf_data); return -1; } return 0; } /*! \brief Function called by core to start a DTMF digit */ static int chan_pjsip_digit_begin(struct ast_channel *chan, char digit) { struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan); struct ast_sip_session_media *media; int res = 0; media = channel->session->active_media_state->default_session[AST_MEDIA_TYPE_AUDIO]; switch (channel->session->dtmf) { case AST_SIP_DTMF_RFC_4733: if (!media || !media->rtp) { return -1; } ast_rtp_instance_dtmf_begin(media->rtp, digit); break; case AST_SIP_DTMF_AUTO: if (!media || !media->rtp || (ast_rtp_instance_dtmf_mode_get(media->rtp) == AST_RTP_DTMF_MODE_INBAND)) { return -1; } ast_rtp_instance_dtmf_begin(media->rtp, digit); break; case AST_SIP_DTMF_AUTO_INFO: if (!media || !media->rtp || (ast_rtp_instance_dtmf_mode_get(media->rtp) == AST_RTP_DTMF_MODE_NONE)) { return -1; } ast_rtp_instance_dtmf_begin(media->rtp, digit); break; case AST_SIP_DTMF_NONE: break; case AST_SIP_DTMF_INBAND: res = -1; break; default: break; } return res; } struct info_dtmf_data { struct ast_sip_session *session; char digit; unsigned int duration; }; static void info_dtmf_data_destroy(void *obj) { struct info_dtmf_data *dtmf_data = obj; ao2_ref(dtmf_data->session, -1); } static struct info_dtmf_data *info_dtmf_data_alloc(struct ast_sip_session *session, char digit, unsigned int duration) { struct info_dtmf_data *dtmf_data = ao2_alloc(sizeof(*dtmf_data), info_dtmf_data_destroy); if (!dtmf_data) { return NULL; } ao2_ref(session, +1); dtmf_data->session = session; dtmf_data->digit = digit; dtmf_data->duration = duration; return dtmf_data; } static int transmit_info_dtmf(void *data) { RAII_VAR(struct info_dtmf_data *, dtmf_data, data, ao2_cleanup); struct ast_sip_session *session = dtmf_data->session; struct pjsip_tx_data *tdata; RAII_VAR(struct ast_str *, body_text, NULL, ast_free_ptr); struct ast_sip_body body = { .type = "application", .subtype = "dtmf-relay", }; if (session->inv_session->state == PJSIP_INV_STATE_DISCONNECTED) { ast_log(LOG_ERROR, "Session already DISCONNECTED [reason=%d (%s)]\n", session->inv_session->cause, pjsip_get_status_text(session->inv_session->cause)->ptr); goto failure; } if (!(body_text = ast_str_create(32))) { ast_log(LOG_ERROR, "Could not allocate buffer for INFO DTMF.\n"); goto failure; } ast_str_set(&body_text, 0, "Signal=%c\r\nDuration=%u\r\n", dtmf_data->digit, dtmf_data->duration); body.body_text = ast_str_buffer(body_text); if (ast_sip_create_request("INFO", session->inv_session->dlg, session->endpoint, NULL, NULL, &tdata)) { ast_log(LOG_ERROR, "Could not create DTMF INFO request\n"); goto failure; } if (ast_sip_add_body(tdata, &body)) { ast_log(LOG_ERROR, "Could not add body to DTMF INFO request\n"); pjsip_tx_data_dec_ref(tdata); goto failure; } ast_sip_session_send_request(session, tdata); #ifdef HAVE_PJSIP_INV_SESSION_REF pjsip_inv_dec_ref(session->inv_session); #endif return 0; failure: #ifdef HAVE_PJSIP_INV_SESSION_REF pjsip_inv_dec_ref(session->inv_session); #endif return -1; } /*! \brief Function called by core to stop a DTMF digit */ static int chan_pjsip_digit_end(struct ast_channel *ast, char digit, unsigned int duration) { struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast); struct ast_sip_session_media *media; int res = 0; media = channel->session->active_media_state->default_session[AST_MEDIA_TYPE_AUDIO]; switch (channel->session->dtmf) { case AST_SIP_DTMF_AUTO_INFO: { if (!media || !media->rtp) { return -1; } if (ast_rtp_instance_dtmf_mode_get(media->rtp) != AST_RTP_DTMF_MODE_NONE) { ast_debug(3, "Told to send end of digit on Auto-Info channel %s RFC4733 negotiated so using it.\n", ast_channel_name(ast)); ast_rtp_instance_dtmf_end_with_duration(media->rtp, digit, duration); break; } /* If RFC_4733 was not negotiated, fail through to the DTMF_INFO processing */ ast_debug(3, "Told to send end of digit on Auto-Info channel %s RFC4733 NOT negotiated using INFO instead.\n", ast_channel_name(ast)); } case AST_SIP_DTMF_INFO: { struct info_dtmf_data *dtmf_data = info_dtmf_data_alloc(channel->session, digit, duration); if (!dtmf_data) { return -1; } #ifdef HAVE_PJSIP_INV_SESSION_REF if (pjsip_inv_add_ref(dtmf_data->session->inv_session) != PJ_SUCCESS) { ast_log(LOG_ERROR, "Can't increase the session reference counter\n"); ao2_cleanup(dtmf_data); return -1; } #endif if (ast_sip_push_task(channel->session->serializer, transmit_info_dtmf, dtmf_data)) { ast_log(LOG_WARNING, "Error sending DTMF via INFO.\n"); #ifdef HAVE_PJSIP_INV_SESSION_REF pjsip_inv_dec_ref(dtmf_data->session->inv_session); #endif ao2_cleanup(dtmf_data); return -1; } break; } case AST_SIP_DTMF_RFC_4733: if (!media || !media->rtp) { return -1; } ast_rtp_instance_dtmf_end_with_duration(media->rtp, digit, duration); break; case AST_SIP_DTMF_AUTO: if (!media || !media->rtp || (ast_rtp_instance_dtmf_mode_get(media->rtp) == AST_RTP_DTMF_MODE_INBAND)) { return -1; } ast_rtp_instance_dtmf_end_with_duration(media->rtp, digit, duration); break; case AST_SIP_DTMF_NONE: break; case AST_SIP_DTMF_INBAND: res = -1; break; } return res; } static void update_initial_connected_line(struct ast_sip_session *session) { struct ast_party_connected_line connected; /* * Use the channel CALLERID() as the initial connected line data. * The core or a predial handler may have supplied missing values * from the session->endpoint->id.self about who we are calling. */ ast_channel_lock(session->channel); ast_party_id_copy(&session->id, &ast_channel_caller(session->channel)->id); ast_channel_unlock(session->channel); /* Supply initial connected line information if available. */ if (!session->id.number.valid && !session->id.name.valid) { return; } ast_party_connected_line_init(&connected); connected.id = session->id; connected.source = AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER; ast_channel_queue_connected_line_update(session->channel, &connected, NULL); } static int call(void *data) { struct ast_sip_channel_pvt *channel = data; struct ast_sip_session *session = channel->session; pjsip_tx_data *tdata; int res = ast_sip_session_create_invite(session, &tdata); if (res) { ast_set_hangupsource(session->channel, ast_channel_name(session->channel), 0); ast_queue_hangup(session->channel); } else { set_channel_on_rtp_instance(session, ast_channel_uniqueid(session->channel)); update_initial_connected_line(session); ast_sip_session_send_request(session, tdata); } ao2_ref(channel, -1); return res; } /*! \brief Function called by core to actually start calling a remote party */ static int chan_pjsip_call(struct ast_channel *ast, const char *dest, int timeout) { struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast); ao2_ref(channel, +1); if (ast_sip_push_task(channel->session->serializer, call, channel)) { ast_log(LOG_WARNING, "Error attempting to place outbound call to '%s'\n", dest); ao2_cleanup(channel); return -1; } return 0; } /*! \brief Internal function which translates from Asterisk cause codes to SIP response codes */ static int hangup_cause2sip(int cause) { switch (cause) { case AST_CAUSE_UNALLOCATED: /* 1 */ case AST_CAUSE_NO_ROUTE_DESTINATION: /* 3 IAX2: Can't find extension in context */ case AST_CAUSE_NO_ROUTE_TRANSIT_NET: /* 2 */ return 404; case AST_CAUSE_CONGESTION: /* 34 */ case AST_CAUSE_SWITCH_CONGESTION: /* 42 */ return 503; case AST_CAUSE_NO_USER_RESPONSE: /* 18 */ return 408; case AST_CAUSE_NO_ANSWER: /* 19 */ case AST_CAUSE_UNREGISTERED: /* 20 */ return 480; case AST_CAUSE_CALL_REJECTED: /* 21 */ return 403; case AST_CAUSE_NUMBER_CHANGED: /* 22 */ return 410; case AST_CAUSE_NORMAL_UNSPECIFIED: /* 31 */ return 480; case AST_CAUSE_INVALID_NUMBER_FORMAT: return 484; case AST_CAUSE_USER_BUSY: return 486; case AST_CAUSE_FAILURE: return 500; case AST_CAUSE_FACILITY_REJECTED: /* 29 */ return 501; case AST_CAUSE_CHAN_NOT_IMPLEMENTED: return 503; case AST_CAUSE_DESTINATION_OUT_OF_ORDER: return 502; case AST_CAUSE_BEARERCAPABILITY_NOTAVAIL: /* Can't find codec to connect to host */ return 488; case AST_CAUSE_INTERWORKING: /* Unspecified Interworking issues */ return 500; case AST_CAUSE_NOTDEFINED: default: ast_debug(1, "AST hangup cause %d (no match found in PJSIP)\n", cause); return 0; } /* Never reached */ return 0; } struct hangup_data { int cause; struct ast_channel *chan; }; static void hangup_data_destroy(void *obj) { struct hangup_data *h_data = obj; h_data->chan = ast_channel_unref(h_data->chan); } static struct hangup_data *hangup_data_alloc(int cause, struct ast_channel *chan) { struct hangup_data *h_data = ao2_alloc(sizeof(*h_data), hangup_data_destroy); if (!h_data) { return NULL; } h_data->cause = cause; h_data->chan = ast_channel_ref(chan); return h_data; } /*! \brief Clear a channel from a session along with its PVT */ static void clear_session_and_channel(struct ast_sip_session *session, struct ast_channel *ast) { session->channel = NULL; set_channel_on_rtp_instance(session, ""); ast_channel_tech_pvt_set(ast, NULL); } static int hangup(void *data) { struct hangup_data *h_data = data; struct ast_channel *ast = h_data->chan; struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast); struct ast_sip_session *session = channel->session; int cause = h_data->cause; /* * It's possible that session_terminate might cause the session to be destroyed * immediately so we need to keep a reference to it so we can NULL session->channel * afterwards. */ ast_sip_session_terminate(ao2_bump(session), cause); clear_session_and_channel(session, ast); ao2_cleanup(session); ao2_cleanup(channel); ao2_cleanup(h_data); return 0; } /*! \brief Function called by core to hang up a PJSIP session */ static int chan_pjsip_hangup(struct ast_channel *ast) { struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast); int cause; struct hangup_data *h_data; if (!channel || !channel->session) { return -1; } cause = hangup_cause2sip(ast_channel_hangupcause(channel->session->channel)); h_data = hangup_data_alloc(cause, ast); if (!h_data) { goto failure; } if (ast_sip_push_task(channel->session->serializer, hangup, h_data)) { ast_log(LOG_WARNING, "Unable to push hangup task to the threadpool. Expect bad things\n"); goto failure; } return 0; failure: /* Go ahead and do our cleanup of the session and channel even if we're not going * to be able to send our SIP request/response */ clear_session_and_channel(channel->session, ast); ao2_cleanup(channel); ao2_cleanup(h_data); return -1; } struct request_data { struct ast_sip_session *session; struct ast_stream_topology *topology; const char *dest; int cause; }; static int request(void *obj) { struct request_data *req_data = obj; struct ast_sip_session *session = NULL; char *tmp = ast_strdupa(req_data->dest), *endpoint_name = NULL, *request_user = NULL; struct ast_sip_endpoint *endpoint; AST_DECLARE_APP_ARGS(args, AST_APP_ARG(endpoint); AST_APP_ARG(aor); ); if (ast_strlen_zero(tmp)) { ast_log(LOG_ERROR, "Unable to create PJSIP channel with empty destination\n"); req_data->cause = AST_CAUSE_CHANNEL_UNACCEPTABLE; return -1; } AST_NONSTANDARD_APP_ARGS(args, tmp, '/'); if (ast_sip_get_disable_multi_domain()) { /* If a request user has been specified extract it from the endpoint name portion */ if ((endpoint_name = strchr(args.endpoint, '@'))) { request_user = args.endpoint; *endpoint_name++ = '\0'; } else { endpoint_name = args.endpoint; } if (ast_strlen_zero(endpoint_name)) { if (request_user) { ast_log(LOG_ERROR, "Unable to create PJSIP channel with empty endpoint name: %s@\n", request_user); } else { ast_log(LOG_ERROR, "Unable to create PJSIP channel with empty endpoint name\n"); } req_data->cause = AST_CAUSE_CHANNEL_UNACCEPTABLE; return -1; } endpoint = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "endpoint", endpoint_name); if (!endpoint) { ast_log(LOG_ERROR, "Unable to create PJSIP channel - endpoint '%s' was not found\n", endpoint_name); req_data->cause = AST_CAUSE_NO_ROUTE_DESTINATION; return -1; } } else { /* First try to find an exact endpoint match, for single (user) or multi-domain (user@domain) */ endpoint_name = args.endpoint; if (ast_strlen_zero(endpoint_name)) { ast_log(LOG_ERROR, "Unable to create PJSIP channel with empty endpoint name\n"); req_data->cause = AST_CAUSE_CHANNEL_UNACCEPTABLE; return -1; } endpoint = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "endpoint", endpoint_name); if (!endpoint) { /* It seems it's not a multi-domain endpoint or single endpoint exact match, * it's possible that it's a SIP trunk with a specified user (user@trunkname), * so extract the user before @ sign. */ endpoint_name = strchr(args.endpoint, '@'); if (!endpoint_name) { /* * Couldn't find an '@' so it had to be an endpoint * name that doesn't exist. */ ast_log(LOG_ERROR, "Unable to create PJSIP channel - endpoint '%s' was not found\n", args.endpoint); req_data->cause = AST_CAUSE_NO_ROUTE_DESTINATION; return -1; } request_user = args.endpoint; *endpoint_name++ = '\0'; if (ast_strlen_zero(endpoint_name)) { ast_log(LOG_ERROR, "Unable to create PJSIP channel with empty endpoint name: %s@\n", request_user); req_data->cause = AST_CAUSE_CHANNEL_UNACCEPTABLE; return -1; } endpoint = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "endpoint", endpoint_name); if (!endpoint) { ast_log(LOG_ERROR, "Unable to create PJSIP channel - endpoint '%s' was not found\n", endpoint_name); req_data->cause = AST_CAUSE_NO_ROUTE_DESTINATION; return -1; } } } session = ast_sip_session_create_outgoing(endpoint, NULL, args.aor, request_user, req_data->topology); ao2_ref(endpoint, -1); if (!session) { ast_log(LOG_ERROR, "Failed to create outgoing session to endpoint '%s'\n", endpoint_name); req_data->cause = AST_CAUSE_NO_ROUTE_DESTINATION; return -1; } req_data->session = session; return 0; } /*! \brief Function called by core to create a new outgoing PJSIP session */ static struct ast_channel *chan_pjsip_request_with_stream_topology(const char *type, struct ast_stream_topology *topology, const struct ast_assigned_ids *assignedids, const struct ast_channel *requestor, const char *data, int *cause) { struct request_data req_data; RAII_VAR(struct ast_sip_session *, session, NULL, ao2_cleanup); req_data.topology = topology; req_data.dest = data; /* Default failure value in case ast_sip_push_task_synchronous() itself fails. */ req_data.cause = AST_CAUSE_FAILURE; if (ast_sip_push_task_synchronous(NULL, request, &req_data)) { *cause = req_data.cause; return NULL; } session = req_data.session; if (!(session->channel = chan_pjsip_new(session, AST_STATE_DOWN, NULL, NULL, assignedids, requestor, NULL))) { /* Session needs to be terminated prematurely */ return NULL; } return session->channel; } static struct ast_channel *chan_pjsip_request(const char *type, struct ast_format_cap *cap, const struct ast_assigned_ids *assignedids, const struct ast_channel *requestor, const char *data, int *cause) { struct ast_stream_topology *topology; struct ast_channel *chan; topology = ast_stream_topology_create_from_format_cap(cap); if (!topology) { return NULL; } chan = chan_pjsip_request_with_stream_topology(type, topology, assignedids, requestor, data, cause); ast_stream_topology_free(topology); return chan; } struct sendtext_data { struct ast_sip_session *session; char text[0]; }; static void sendtext_data_destroy(void *obj) { struct sendtext_data *data = obj; ao2_ref(data->session, -1); } static struct sendtext_data* sendtext_data_create(struct ast_sip_session *session, const char *text) { int size = strlen(text) + 1; struct sendtext_data *data = ao2_alloc(sizeof(*data)+size, sendtext_data_destroy); if (!data) { return NULL; } data->session = session; ao2_ref(data->session, +1); ast_copy_string(data->text, text, size); return data; } static int sendtext(void *obj) { RAII_VAR(struct sendtext_data *, data, obj, ao2_cleanup); pjsip_tx_data *tdata; const struct ast_sip_body body = { .type = "text", .subtype = "plain", .body_text = data->text }; if (data->session->inv_session->state == PJSIP_INV_STATE_DISCONNECTED) { ast_log(LOG_ERROR, "Session already DISCONNECTED [reason=%d (%s)]\n", data->session->inv_session->cause, pjsip_get_status_text(data->session->inv_session->cause)->ptr); } else { ast_debug(3, "Sending in dialog SIP message\n"); ast_sip_create_request("MESSAGE", data->session->inv_session->dlg, data->session->endpoint, NULL, NULL, &tdata); ast_sip_add_body(tdata, &body); ast_sip_send_request(tdata, data->session->inv_session->dlg, data->session->endpoint, NULL, NULL); } #ifdef HAVE_PJSIP_INV_SESSION_REF pjsip_inv_dec_ref(data->session->inv_session); #endif return 0; } /*! \brief Function called by core to send text on PJSIP session */ static int chan_pjsip_sendtext(struct ast_channel *ast, const char *text) { struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast); struct sendtext_data *data = sendtext_data_create(channel->session, text); if (!data) { return -1; } #ifdef HAVE_PJSIP_INV_SESSION_REF if (pjsip_inv_add_ref(data->session->inv_session) != PJ_SUCCESS) { ast_log(LOG_ERROR, "Can't increase the session reference counter\n"); ao2_ref(data, -1); return -1; } #endif if (ast_sip_push_task(channel->session->serializer, sendtext, data)) { #ifdef HAVE_PJSIP_INV_SESSION_REF pjsip_inv_dec_ref(data->session->inv_session); #endif ao2_ref(data, -1); return -1; } return 0; } /*! \brief Convert SIP hangup causes to Asterisk hangup causes */ static int hangup_sip2cause(int cause) { /* Possible values taken from causes.h */ switch(cause) { case 401: /* Unauthorized */ return AST_CAUSE_CALL_REJECTED; case 403: /* Not found */ return AST_CAUSE_CALL_REJECTED; case 404: /* Not found */ return AST_CAUSE_UNALLOCATED; case 405: /* Method not allowed */ return AST_CAUSE_INTERWORKING; case 407: /* Proxy authentication required */ return AST_CAUSE_CALL_REJECTED; case 408: /* No reaction */ return AST_CAUSE_NO_USER_RESPONSE; case 409: /* Conflict */ return AST_CAUSE_NORMAL_TEMPORARY_FAILURE; case 410: /* Gone */ return AST_CAUSE_NUMBER_CHANGED; case 411: /* Length required */ return AST_CAUSE_INTERWORKING; case 413: /* Request entity too large */ return AST_CAUSE_INTERWORKING; case 414: /* Request URI too large */ return AST_CAUSE_INTERWORKING; case 415: /* Unsupported media type */ return AST_CAUSE_INTERWORKING; case 420: /* Bad extension */ return AST_CAUSE_NO_ROUTE_DESTINATION; case 480: /* No answer */ return AST_CAUSE_NO_ANSWER; case 481: /* No answer */ return AST_CAUSE_INTERWORKING; case 482: /* Loop detected */ return AST_CAUSE_INTERWORKING; case 483: /* Too many hops */ return AST_CAUSE_NO_ANSWER; case 484: /* Address incomplete */ return AST_CAUSE_INVALID_NUMBER_FORMAT; case 485: /* Ambiguous */ return AST_CAUSE_UNALLOCATED; case 486: /* Busy everywhere */ return AST_CAUSE_BUSY; case 487: /* Request terminated */ return AST_CAUSE_INTERWORKING; case 488: /* No codecs approved */ return AST_CAUSE_BEARERCAPABILITY_NOTAVAIL; case 491: /* Request pending */ return AST_CAUSE_INTERWORKING; case 493: /* Undecipherable */ return AST_CAUSE_INTERWORKING; case 500: /* Server internal failure */ return AST_CAUSE_FAILURE; case 501: /* Call rejected */ return AST_CAUSE_FACILITY_REJECTED; case 502: return AST_CAUSE_DESTINATION_OUT_OF_ORDER; case 503: /* Service unavailable */ return AST_CAUSE_CONGESTION; case 504: /* Gateway timeout */ return AST_CAUSE_RECOVERY_ON_TIMER_EXPIRE; case 505: /* SIP version not supported */ return AST_CAUSE_INTERWORKING; case 600: /* Busy everywhere */ return AST_CAUSE_USER_BUSY; case 603: /* Decline */ return AST_CAUSE_CALL_REJECTED; case 604: /* Does not exist anywhere */ return AST_CAUSE_UNALLOCATED; case 606: /* Not acceptable */ return AST_CAUSE_BEARERCAPABILITY_NOTAVAIL; default: if (cause < 500 && cause >= 400) { /* 4xx class error that is unknown - someting wrong with our request */ return AST_CAUSE_INTERWORKING; } else if (cause < 600 && cause >= 500) { /* 5xx class error - problem in the remote end */ return AST_CAUSE_CONGESTION; } else if (cause < 700 && cause >= 600) { /* 6xx - global errors in the 4xx class */ return AST_CAUSE_INTERWORKING; } return AST_CAUSE_NORMAL; } /* Never reached */ return 0; } static void chan_pjsip_session_begin(struct ast_sip_session *session) { RAII_VAR(struct ast_datastore *, datastore, NULL, ao2_cleanup); if (session->endpoint->media.direct_media.glare_mitigation == AST_SIP_DIRECT_MEDIA_GLARE_MITIGATION_NONE) { return; } datastore = ast_sip_session_alloc_datastore(&direct_media_mitigation_info, "direct_media_glare_mitigation"); if (!datastore) { return; } ast_sip_session_add_datastore(session, datastore); } /*! \brief Function called when the session ends */ static void chan_pjsip_session_end(struct ast_sip_session *session) { if (!session->channel) { return; } chan_pjsip_remove_hold(ast_channel_uniqueid(session->channel)); ast_set_hangupsource(session->channel, ast_channel_name(session->channel), 0); if (!ast_channel_hangupcause(session->channel) && session->inv_session) { int cause = hangup_sip2cause(session->inv_session->cause); ast_queue_hangup_with_cause(session->channel, cause); } else { ast_queue_hangup(session->channel); } } /*! \brief Function called when a request is received on the session */ static int chan_pjsip_incoming_request(struct ast_sip_session *session, struct pjsip_rx_data *rdata) { RAII_VAR(struct ast_datastore *, datastore, NULL, ao2_cleanup); struct transport_info_data *transport_data; pjsip_tx_data *packet = NULL; if (session->channel) { return 0; } /* Check for a to-tag to determine if this is a reinvite */ if (rdata->msg_info.to->tag.slen) { /* Weird case. We've received a reinvite but we don't have a channel. The most * typical case for this happening is that a blind transfer fails, and so the * transferer attempts to reinvite himself back into the call. We already got * rid of that channel, and the other side of the call is unrecoverable. * * We treat this as a failure, so our best bet is to just hang this call * up and not create a new channel. Clearing defer_terminate here ensures that * calling ast_sip_session_terminate() can result in a BYE being sent ASAP. */ session->defer_terminate = 0; ast_sip_session_terminate(session, 400); return -1; } datastore = ast_sip_session_alloc_datastore(&transport_info, "transport_info"); if (!datastore) { return -1; } transport_data = ast_calloc(1, sizeof(*transport_data)); if (!transport_data) { return -1; } pj_sockaddr_cp(&transport_data->local_addr, &rdata->tp_info.transport->local_addr); pj_sockaddr_cp(&transport_data->remote_addr, &rdata->pkt_info.src_addr); datastore->data = transport_data; ast_sip_session_add_datastore(session, datastore); if (!(session->channel = chan_pjsip_new(session, AST_STATE_RING, session->exten, NULL, NULL, NULL, NULL))) { if (pjsip_inv_end_session(session->inv_session, 503, NULL, &packet) == PJ_SUCCESS && packet) { ast_sip_session_send_response(session, packet); } ast_log(LOG_ERROR, "Failed to allocate new PJSIP channel on incoming SIP INVITE\n"); return -1; } /* channel gets created on incoming request, but we wait to call start so other supplements have a chance to run */ return 0; } static int call_pickup_incoming_request(struct ast_sip_session *session, pjsip_rx_data *rdata) { struct ast_features_pickup_config *pickup_cfg; struct ast_channel *chan; /* Check for a to-tag to determine if this is a reinvite */ if (rdata->msg_info.to->tag.slen) { /* We don't care about reinvites */ return 0; } pickup_cfg = ast_get_chan_features_pickup_config(session->channel); if (!pickup_cfg) { ast_log(LOG_ERROR, "Unable to retrieve pickup configuration options. Unable to detect call pickup extension.\n"); return 0; } if (strcmp(session->exten, pickup_cfg->pickupexten)) { ao2_ref(pickup_cfg, -1); return 0; } ao2_ref(pickup_cfg, -1); /* We can't directly use session->channel because the pickup operation will cause a masquerade to occur, * changing the channel pointer in session to a different channel. To ensure we work on the right channel * we store a pointer locally before we begin and keep a reference so it remains valid no matter what. */ chan = ast_channel_ref(session->channel); if (ast_pickup_call(chan)) { ast_channel_hangupcause_set(chan, AST_CAUSE_CALL_REJECTED); } else { ast_channel_hangupcause_set(chan, AST_CAUSE_NORMAL_CLEARING); } /* A hangup always occurs because the pickup operation will have either failed resulting in the call * needing to be hung up OR the pickup operation was a success and the channel we now have is actually * the channel that was replaced, which should be hung up since it is literally in limbo not connected * to anything at all. */ ast_hangup(chan); ast_channel_unref(chan); return 1; } static struct ast_sip_session_supplement call_pickup_supplement = { .method = "INVITE", .priority = AST_SIP_SUPPLEMENT_PRIORITY_LAST - 1, .incoming_request = call_pickup_incoming_request, }; static int pbx_start_incoming_request(struct ast_sip_session *session, pjsip_rx_data *rdata) { int res; /* Check for a to-tag to determine if this is a reinvite */ if (rdata->msg_info.to->tag.slen) { /* We don't care about reinvites */ return 0; } res = ast_pbx_start(session->channel); switch (res) { case AST_PBX_FAILED: ast_log(LOG_WARNING, "Failed to start PBX ;(\n"); ast_channel_hangupcause_set(session->channel, AST_CAUSE_SWITCH_CONGESTION); ast_hangup(session->channel); break; case AST_PBX_CALL_LIMIT: ast_log(LOG_WARNING, "Failed to start PBX (call limit reached) \n"); ast_channel_hangupcause_set(session->channel, AST_CAUSE_SWITCH_CONGESTION); ast_hangup(session->channel); break; case AST_PBX_SUCCESS: default: break; } ast_debug(3, "Started PBX on new PJSIP channel %s\n", ast_channel_name(session->channel)); return (res == AST_PBX_SUCCESS) ? 0 : -1; } static struct ast_sip_session_supplement pbx_start_supplement = { .method = "INVITE", .priority = AST_SIP_SUPPLEMENT_PRIORITY_LAST, .incoming_request = pbx_start_incoming_request, }; /*! \brief Function called when a response is received on the session */ static void chan_pjsip_incoming_response(struct ast_sip_session *session, struct pjsip_rx_data *rdata) { struct pjsip_status_line status = rdata->msg_info.msg->line.status; struct ast_control_pvt_cause_code *cause_code; int data_size = sizeof(*cause_code); if (!session->channel) { return; } /* Build and send the tech-specific cause information */ /* size of the string making up the cause code is "SIP " number + " " + reason length */ data_size += 4 + 4 + pj_strlen(&status.reason); cause_code = ast_alloca(data_size); memset(cause_code, 0, data_size); ast_copy_string(cause_code->chan_name, ast_channel_name(session->channel), AST_CHANNEL_NAME); snprintf(cause_code->code, data_size - sizeof(*cause_code) + 1, "SIP %d %.*s", status.code, (int) pj_strlen(&status.reason), pj_strbuf(&status.reason)); cause_code->ast_cause = hangup_sip2cause(status.code); ast_queue_control_data(session->channel, AST_CONTROL_PVT_CAUSE_CODE, cause_code, data_size); ast_channel_hangupcause_hash_set(session->channel, cause_code, data_size); switch (status.code) { case 180: ast_queue_control(session->channel, AST_CONTROL_RINGING); ast_channel_lock(session->channel); if (ast_channel_state(session->channel) != AST_STATE_UP) { ast_setstate(session->channel, AST_STATE_RINGING); } ast_channel_unlock(session->channel); break; case 183: ast_queue_control(session->channel, AST_CONTROL_PROGRESS); break; case 200: ast_queue_control(session->channel, AST_CONTROL_ANSWER); break; default: break; } } static int chan_pjsip_incoming_ack(struct ast_sip_session *session, struct pjsip_rx_data *rdata) { if (rdata->msg_info.msg->line.req.method.id == PJSIP_ACK_METHOD) { if (session->endpoint->media.direct_media.enabled && session->channel) { ast_queue_control(session->channel, AST_CONTROL_SRCCHANGE); } } return 0; } static int update_devstate(void *obj, void *arg, int flags) { ast_devstate_changed(AST_DEVICE_UNKNOWN, AST_DEVSTATE_CACHABLE, "PJSIP/%s", ast_sorcery_object_get_id(obj)); return 0; } static struct ast_custom_function chan_pjsip_dial_contacts_function = { .name = "PJSIP_DIAL_CONTACTS", .read = pjsip_acf_dial_contacts_read, }; static struct ast_custom_function media_offer_function = { .name = "PJSIP_MEDIA_OFFER", .read = pjsip_acf_media_offer_read, .write = pjsip_acf_media_offer_write }; static struct ast_custom_function dtmf_mode_function = { .name = "PJSIP_DTMF_MODE", .read = pjsip_acf_dtmf_mode_read, .write = pjsip_acf_dtmf_mode_write }; static struct ast_custom_function session_refresh_function = { .name = "PJSIP_SEND_SESSION_REFRESH", .write = pjsip_acf_session_refresh_write, }; /*! * \brief Load the module * * Module loading including tests for configuration or dependencies. * This function can return AST_MODULE_LOAD_FAILURE, AST_MODULE_LOAD_DECLINE, * or AST_MODULE_LOAD_SUCCESS. If a dependency or environment variable fails * tests return AST_MODULE_LOAD_FAILURE. If the module can not load the * configuration file or other non-critical problem return * AST_MODULE_LOAD_DECLINE. On success return AST_MODULE_LOAD_SUCCESS. */ static int load_module(void) { struct ao2_container *endpoints; if (!(chan_pjsip_tech.capabilities = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT))) { return AST_MODULE_LOAD_DECLINE; } ast_format_cap_append_by_type(chan_pjsip_tech.capabilities, AST_MEDIA_TYPE_AUDIO); ast_rtp_glue_register(&chan_pjsip_rtp_glue); if (ast_channel_register(&chan_pjsip_tech)) { ast_log(LOG_ERROR, "Unable to register channel class %s\n", channel_type); goto end; } if (ast_custom_function_register(&chan_pjsip_dial_contacts_function)) { ast_log(LOG_ERROR, "Unable to register PJSIP_DIAL_CONTACTS dialplan function\n"); goto end; } if (ast_custom_function_register(&media_offer_function)) { ast_log(LOG_WARNING, "Unable to register PJSIP_MEDIA_OFFER dialplan function\n"); goto end; } if (ast_custom_function_register(&dtmf_mode_function)) { ast_log(LOG_WARNING, "Unable to register PJSIP_DTMF_MODE dialplan function\n"); goto end; } if (ast_custom_function_register(&session_refresh_function)) { ast_log(LOG_WARNING, "Unable to register PJSIP_SEND_SESSION_REFRESH dialplan function\n"); goto end; } ast_sip_session_register_supplement(&chan_pjsip_supplement); if (!(pjsip_uids_onhold = ao2_container_alloc_hash(AO2_ALLOC_OPT_LOCK_RWLOCK, AO2_CONTAINER_ALLOC_OPT_DUPS_REJECT, 37, uid_hold_hash_fn, uid_hold_sort_fn, NULL))) { ast_log(LOG_ERROR, "Unable to create held channels container\n"); goto end; } ast_sip_session_register_supplement(&call_pickup_supplement); ast_sip_session_register_supplement(&pbx_start_supplement); ast_sip_session_register_supplement(&chan_pjsip_ack_supplement); if (pjsip_channel_cli_register()) { ast_log(LOG_ERROR, "Unable to register PJSIP Channel CLI\n"); ast_sip_session_unregister_supplement(&chan_pjsip_ack_supplement); ast_sip_session_unregister_supplement(&pbx_start_supplement); ast_sip_session_unregister_supplement(&chan_pjsip_supplement); ast_sip_session_unregister_supplement(&call_pickup_supplement); goto end; } /* since endpoints are loaded before the channel driver their device states get set to 'invalid', so they need to be updated */ if ((endpoints = ast_sip_get_endpoints())) { ao2_callback(endpoints, OBJ_NODATA, update_devstate, NULL); ao2_ref(endpoints, -1); } return 0; end: ao2_cleanup(pjsip_uids_onhold); pjsip_uids_onhold = NULL; ast_custom_function_unregister(&dtmf_mode_function); ast_custom_function_unregister(&media_offer_function); ast_custom_function_unregister(&chan_pjsip_dial_contacts_function); ast_custom_function_unregister(&session_refresh_function); ast_channel_unregister(&chan_pjsip_tech); ast_rtp_glue_unregister(&chan_pjsip_rtp_glue); return AST_MODULE_LOAD_DECLINE; } /*! \brief Unload the PJSIP channel from Asterisk */ static int unload_module(void) { ao2_cleanup(pjsip_uids_onhold); pjsip_uids_onhold = NULL; pjsip_channel_cli_unregister(); ast_sip_session_unregister_supplement(&chan_pjsip_supplement); ast_sip_session_unregister_supplement(&pbx_start_supplement); ast_sip_session_unregister_supplement(&chan_pjsip_ack_supplement); ast_sip_session_unregister_supplement(&call_pickup_supplement); ast_custom_function_unregister(&dtmf_mode_function); ast_custom_function_unregister(&media_offer_function); ast_custom_function_unregister(&chan_pjsip_dial_contacts_function); ast_custom_function_unregister(&session_refresh_function); ast_channel_unregister(&chan_pjsip_tech); ao2_ref(chan_pjsip_tech.capabilities, -1); ast_rtp_glue_unregister(&chan_pjsip_rtp_glue); return 0; } AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "PJSIP Channel Driver", .support_level = AST_MODULE_SUPPORT_CORE, .load = load_module, .unload = unload_module, .load_pri = AST_MODPRI_CHANNEL_DRIVER, .requires = "res_pjsip,res_pjsip_session", );