dialplan_functions.h File Reference

PJSIP dialplan functions header file. More...

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Functions
int	pjsip_acf_channel_read (struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)
	CHANNEL function read callback.
int	pjsip_acf_dial_contacts_read (struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)
	PJSIP_DIAL_CONTACTS function read callback.
int	pjsip_acf_dtmf_mode_read (struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)
	PJSIP_DTMF_MODE function read callback.
int	pjsip_acf_dtmf_mode_write (struct ast_channel *chan, const char *cmd, char *data, const char *value)
	PJSIP_DTMF_MODE function write callback.
int	pjsip_acf_media_offer_read (struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)
	PJSIP_MEDIA_OFFER function read callback.
int	pjsip_acf_media_offer_write (struct ast_channel *chan, const char *cmd, char *data, const char *value)
	PJSIP_MEDIA_OFFER function write callback.
int	pjsip_acf_moh_passthrough_read (struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)
	PJSIP_MOH_PASSTHROUGH function read callback.
int	pjsip_acf_moh_passthrough_write (struct ast_channel *chan, const char *cmd, char *data, const char *value)
	PJSIP_MOH_PASSTHROUGH function write callback.
int	pjsip_acf_parse_uri_read (struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)
	PJSIP_PARSE_URI function read callback.
int	pjsip_acf_session_refresh_write (struct ast_channel *chan, const char *cmd, char *data, const char *value)
	PJSIP_SEND_SESSION_REFRESH function write callback.
int	pjsip_action_hangup (struct mansession *s, const struct message *m)
	Manager action to hang up an incoming PJSIP channel with a SIP response code.
int	pjsip_app_hangup (struct ast_channel *chan, const char *data)
	Hang up an incoming PJSIP channel with a SIP response code.
int	pjsip_transfer_handling_write (struct ast_channel *chan, const char *cmd, char *data, const char *value)
	PJSIP_TRANSFER_HANDLING function write callback.

Detailed Description

PJSIP dialplan functions header file.

Definition in file dialplan_functions.h.

Function Documentation

pjsip_acf_channel_read()

int pjsip_acf_channel_read	(	struct ast_channel *	chan,
		const char *	cmd,
		char *	data,
		char *	buf,
		size_t	len
	)

CHANNEL function read callback.

Parameters

chan	The channel the function is called on
cmd	The name of the function
data	Arguments passed to the function
buf	Out buffer that should be populated with the data
len	Size of the buffer

Return values

0	on success
-1	on failure

Definition at line 356 of file dialplan_functions.c.

{
    struct ast_sip_channel_pvt *channel;
    char *parse = ast_strdupa(data);
    int res = 0;
 
    AST_DECLARE_APP_ARGS(args,
        AST_APP_ARG(param);
        AST_APP_ARG(type);
        AST_APP_ARG(field);
    );
 
    if (!chan) {
        ast_log(LOG_WARNING, "No channel was provided to %s function.\n", cmd);
        return -1;
    }
 
    /* Check for zero arguments */
    if (ast_strlen_zero(parse)) {
        ast_log(LOG_ERROR, "Cannot call %s without arguments\n", cmd);
        return -1;
    }
 
    if (!len) {
        ast_log(LOG_ERROR, "Cannot call %s without an output buffer\n", cmd);
        return -1;
    }
 
    AST_STANDARD_APP_ARGS(args, parse);
 
    ast_channel_lock(chan);
 
    /* Sanity check */
    if (strcmp(ast_channel_tech(chan)->type, "PJSIP")) {
        ast_log(LOG_WARNING, "Cannot call %s on a non-PJSIP channel\n", cmd);
        ast_channel_unlock(chan);
        return 0;
    }
 
    channel = ast_channel_tech_pvt(chan);
    if (!channel) {
        ast_log(LOG_WARNING, "Channel %s has no pvt!\n", ast_channel_name(chan));
        ast_channel_unlock(chan);
        return -1;
    }
 
    if (!channel->session) {
        ast_log(LOG_WARNING, "Channel %s has no session\n", ast_channel_name(chan));
        ast_channel_unlock(chan);
        return -1;
    }
 
    memset(buf, 0, len);
 
    if (!strcmp(args.param, "endpoint")) {
        if (!channel->session->endpoint) {
            res = -1;
        } else {
            ast_copy_string(buf, ast_sorcery_object_get_id(channel->session->endpoint), len);
        }
    } else if (!strcmp(args.param, "contact")) {
        if (channel->session->contact) {
            ast_copy_string(buf, ast_sorcery_object_get_id(channel->session->contact), len);
        }
    } else if (!strcmp(args.param, "aor")) {
        if (channel->session->aor) {
            ast_copy_string(buf, ast_sorcery_object_get_id(channel->session->aor), len);
        }
    } else if (!strcmp(args.param, "pjsip")) {
        res = channel_read_pjsip(chan, channel->session, args.type, buf, len);
    } else if (!strcmp(args.param, "rtp") || !strcmp(args.param, "rtcp")) {
        const struct pjsip_channel_info *info = pjsip_channel_info_get(chan);
        struct ast_rtp_instance *rtp;
        struct ast_sip_session_media *media;
        struct ast_sockaddr no_address = { { 0, } };
        const struct ast_sockaddr *direct_media_addr;
        unsigned int secure;
        unsigned int held;
        int video = !strcmp(S_OR(args.field, "audio"), "video");
 
        if (!info) {
            res = -1;
        } else {
            /* The channel lock keeps these retained instances alive throughout the read. */
            rtp = video ? info->video_rtp : info->audio_rtp;
            media = channel->session->active_media_state
                ? channel->session->active_media_state->default_session[video
                    ? AST_MEDIA_TYPE_VIDEO : AST_MEDIA_TYPE_AUDIO] : NULL;
            direct_media_addr = media ? &media->direct_media_addr : &no_address;
            secure = video ? info->video_secure : info->audio_secure;
            held = video ? info->video_held : info->audio_held;
            if (!strcmp(args.param, "rtp")) {
                res = channel_read_rtp(chan, rtp, direct_media_addr, secure, held,
                    args.type, args.field, buf, len);
            } else {
                res = channel_read_rtcp(chan, rtp, args.type, args.field, buf, len);
            }
        }
    } else {
        res = -1;
    }
    ast_channel_unlock(chan);
    return res;
}

References ast_sip_session::active_media_state, ast_sip_session::aor, args, AST_APP_ARG, ast_channel_lock, ast_channel_name(), ast_channel_tech_pvt(), ast_channel_unlock, ast_copy_string(), AST_DECLARE_APP_ARGS, ast_log, AST_MEDIA_TYPE_AUDIO, AST_MEDIA_TYPE_VIDEO, ast_sorcery_object_get_id(), AST_STANDARD_APP_ARGS, ast_strdupa, ast_strlen_zero(), buf, channel_read_pjsip(), channel_read_rtcp(), channel_read_rtp(), ast_sip_session::contact, ast_sip_session_media_state::default_session, ast_sip_session_media::direct_media_addr, ast_sip_session::endpoint, len(), LOG_ERROR, LOG_WARNING, NULL, pjsip_channel_info_get(), S_OR, ast_sip_channel_pvt::session, and type.

pjsip_acf_dial_contacts_read()

int pjsip_acf_dial_contacts_read	(	struct ast_channel *	chan,
		const char *	cmd,
		char *	data,
		char *	buf,
		size_t	len
	)

PJSIP_DIAL_CONTACTS function read callback.

Parameters

chan	The channel the function is called on
cmd	The name of the function
data	Arguments passed to the function
buf	Out buffer that should be populated with the data
len	Size of the buffer

Return values

0	on success
-1	on failure

Definition at line 461 of file dialplan_functions.c.

{
    RAII_VAR(struct ast_sip_endpoint *, endpoint, NULL, ao2_cleanup);
    RAII_VAR(struct ast_str *, dial, NULL, ast_free_ptr);
    const char *aor_name;
    char *rest;
 
    AST_DECLARE_APP_ARGS(args,
        AST_APP_ARG(endpoint_name);
        AST_APP_ARG(aor_name);
        AST_APP_ARG(request_user);
    );
 
    AST_STANDARD_APP_ARGS(args, data);
 
    if (ast_strlen_zero(args.endpoint_name)) {
        ast_log(LOG_WARNING, "An endpoint name must be specified when using the '%s' dialplan function\n", cmd);
        return -1;
    } else if (!(endpoint = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "endpoint", args.endpoint_name))) {
        ast_log(LOG_WARNING, "Specified endpoint '%s' was not found\n", args.endpoint_name);
        return -1;
    }
 
    aor_name = S_OR(args.aor_name, endpoint->aors);
 
    if (ast_strlen_zero(aor_name)) {
        ast_log(LOG_WARNING, "No AOR has been provided and no AORs are configured on endpoint '%s'\n", args.endpoint_name);
        return -1;
    } else if (!(dial = ast_str_create(len))) {
        ast_log(LOG_WARNING, "Could not get enough buffer space for dialing contacts\n");
        return -1;
    } else if (!(rest = ast_strdupa(aor_name))) {
        ast_log(LOG_WARNING, "Could not duplicate provided AORs\n");
        return -1;
    }
 
    while ((aor_name = ast_strip(strsep(&rest, ",")))) {
        RAII_VAR(struct ast_sip_aor *, aor, ast_sip_location_retrieve_aor(aor_name), ao2_cleanup);
        RAII_VAR(struct ao2_container *, contacts, NULL, ao2_cleanup);
        struct ao2_iterator it_contacts;
        struct ast_sip_contact *contact;
 
        if (!aor) {
            /* If the AOR provided is not found skip it, there may be more */
            continue;
        } else if (!(contacts = ast_sip_location_retrieve_aor_contacts_filtered(aor, AST_SIP_CONTACT_FILTER_REACHABLE))) {
            /* No contacts are available, skip it as well */
            continue;
        } else if (!ao2_container_count(contacts)) {
            /* We were given a container but no contacts are in it... */
            continue;
        }
 
        it_contacts = ao2_iterator_init(contacts, 0);
        for (; (contact = ao2_iterator_next(&it_contacts)); ao2_ref(contact, -1)) {
            ast_str_append(&dial, -1, "PJSIP/");
 
            if (!ast_strlen_zero(args.request_user)) {
                ast_str_append(&dial, -1, "%s@", args.request_user);
            }
            ast_str_append(&dial, -1, "%s/%s&", args.endpoint_name, contact->uri);
        }
        ao2_iterator_destroy(&it_contacts);
    }
 
    /* Trim the '&' at the end off */
    ast_str_truncate(dial, ast_str_strlen(dial) - 1);
 
    ast_copy_string(buf, ast_str_buffer(dial), len);
 
    return 0;
}

References ao2_cleanup, ao2_container_count(), ao2_iterator_destroy(), ao2_iterator_init(), ao2_iterator_next, ao2_ref, ast_sip_contact::aor, args, AST_APP_ARG, ast_copy_string(), AST_DECLARE_APP_ARGS, ast_free_ptr(), ast_log, AST_SIP_CONTACT_FILTER_REACHABLE, ast_sip_get_sorcery(), ast_sip_location_retrieve_aor(), ast_sip_location_retrieve_aor_contacts_filtered(), ast_sorcery_retrieve_by_id(), AST_STANDARD_APP_ARGS, ast_str_append(), ast_str_buffer(), ast_str_create, ast_str_strlen(), ast_str_truncate(), ast_strdupa, ast_strip(), ast_strlen_zero(), buf, len(), LOG_WARNING, NULL, RAII_VAR, S_OR, strsep(), and ast_sip_contact::uri.

pjsip_acf_dtmf_mode_read()

int pjsip_acf_dtmf_mode_read	(	struct ast_channel *	chan,
		const char *	cmd,
		char *	data,
		char *	buf,
		size_t	len
	)

PJSIP_DTMF_MODE function read callback.

Parameters

chan	The channel the function is called on
cmd	The name of the function
data	Arguments passed to the function
buf	Out buffer that should be populated with the data
len	Size of the buffer

Return values

0	on success
-1	on failure

Definition at line 929 of file dialplan_functions.c.

{
    struct ast_sip_channel_pvt *channel;
 
    if (!chan) {
        ast_log(LOG_WARNING, "No channel was provided to %s function.\n", cmd);
        return -1;
    }
 
    ast_channel_lock(chan);
    if (strcmp(ast_channel_tech(chan)->type, "PJSIP")) {
        ast_log(LOG_WARNING, "Cannot call %s on a non-PJSIP channel\n", cmd);
        ast_channel_unlock(chan);
        return -1;
    }
 
    channel = ast_channel_tech_pvt(chan);
 
    if (ast_sip_dtmf_to_str(channel->session->dtmf, buf, len) < 0) {
        ast_log(LOG_WARNING, "Unknown DTMF mode %d on PJSIP channel %s\n", channel->session->dtmf, ast_channel_name(chan));
        ast_channel_unlock(chan);
        return -1;
    }
 
    ast_channel_unlock(chan);
    return 0;
}

References ast_channel_lock, ast_channel_name(), ast_channel_tech_pvt(), ast_channel_unlock, ast_log, ast_sip_dtmf_to_str(), buf, ast_sip_session::dtmf, len(), LOG_WARNING, ast_sip_channel_pvt::session, and type.

pjsip_acf_dtmf_mode_write()

int pjsip_acf_dtmf_mode_write	(	struct ast_channel *	chan,
		const char *	cmd,
		char *	data,
		const char *	value
	)

PJSIP_DTMF_MODE function write callback.

Parameters

chan	The channel the function is called on
cmd	The name of the function
data	Arguments passed to the function
value	Value to be set by the function

Return values

0	on success
-1	on failure

Definition at line 1030 of file dialplan_functions.c.

{
    struct ast_sip_channel_pvt *channel;
    struct ast_sip_session_media *media;
    int dsp_features = 0;
    int dtmf = -1;
    struct refresh_data rdata = {
            .method = AST_SIP_SESSION_REFRESH_METHOD_INVITE,
        };
 
    if (!chan) {
        ast_log(LOG_WARNING, "No channel was provided to %s function.\n", cmd);
        return -1;
    }
 
    ast_channel_lock(chan);
    if (strcmp(ast_channel_tech(chan)->type, "PJSIP")) {
        ast_log(LOG_WARNING, "Cannot call %s on a non-PJSIP channel\n", cmd);
        ast_channel_unlock(chan);
        return -1;
    }
 
    channel = ast_channel_tech_pvt(chan);
    rdata.session = channel->session;
 
    dtmf = ast_sip_str_to_dtmf(value);
 
    if (dtmf == -1) {
        ast_log(LOG_WARNING, "Cannot set DTMF mode to '%s' on channel '%s' as value is invalid.\n", value,
            ast_channel_name(chan));
        ast_channel_unlock(chan);
        return -1;
    }
 
    if (channel->session->dtmf == dtmf) {
        /* DTMF mode unchanged, nothing to do! */
        ast_channel_unlock(chan);
        return 0;
    }
 
    channel->session->dtmf = dtmf;
 
    media = channel->session->active_media_state->default_session[AST_MEDIA_TYPE_AUDIO];
 
    if (media && media->rtp) {
        if (channel->session->dtmf == AST_SIP_DTMF_RFC_4733) {
            ast_rtp_instance_set_prop(media->rtp, AST_RTP_PROPERTY_DTMF, 1);
            ast_rtp_instance_dtmf_mode_set(media->rtp, AST_RTP_DTMF_MODE_RFC2833);
        } else if (channel->session->dtmf == AST_SIP_DTMF_INFO) {
            ast_rtp_instance_set_prop(media->rtp, AST_RTP_PROPERTY_DTMF, 0);
            ast_rtp_instance_dtmf_mode_set(media->rtp, AST_RTP_DTMF_MODE_NONE);
        } else if (channel->session->dtmf == AST_SIP_DTMF_INBAND) {
            ast_rtp_instance_set_prop(media->rtp, AST_RTP_PROPERTY_DTMF, 0);
            ast_rtp_instance_dtmf_mode_set(media->rtp, AST_RTP_DTMF_MODE_INBAND);
        } else if (channel->session->dtmf == AST_SIP_DTMF_NONE) {
            ast_rtp_instance_set_prop(media->rtp, AST_RTP_PROPERTY_DTMF, 0);
            ast_rtp_instance_dtmf_mode_set(media->rtp, AST_RTP_DTMF_MODE_NONE);
        } else if (channel->session->dtmf == AST_SIP_DTMF_AUTO) {
            if (ast_rtp_instance_dtmf_mode_get(media->rtp) != AST_RTP_DTMF_MODE_RFC2833) {
                /* no RFC4733 negotiated, enable inband */
                ast_rtp_instance_dtmf_mode_set(media->rtp, AST_RTP_DTMF_MODE_INBAND);
            }
        } else if (channel->session->dtmf == AST_SIP_DTMF_AUTO_INFO) {
            ast_rtp_instance_set_prop(media->rtp, AST_RTP_PROPERTY_DTMF, 0);
            if (ast_rtp_instance_dtmf_mode_get(media->rtp) == AST_RTP_DTMF_MODE_INBAND) {
                /* if inband, switch to INFO */
                ast_rtp_instance_dtmf_mode_set(media->rtp, AST_RTP_DTMF_MODE_NONE);
            }
        }
    }
 
    if (channel->session->dsp) {
        dsp_features = ast_dsp_get_features(channel->session->dsp);
    }
    if (channel->session->dtmf == AST_SIP_DTMF_INBAND ||
        channel->session->dtmf == AST_SIP_DTMF_AUTO) {
        dsp_features |= DSP_FEATURE_DIGIT_DETECT;
    } else {
        dsp_features &= ~DSP_FEATURE_DIGIT_DETECT;
    }
    if (dsp_features) {
        if (!channel->session->dsp) {
            if (!(channel->session->dsp = ast_dsp_new())) {
                ast_channel_unlock(chan);
                return 0;
            }
        }
        ast_dsp_set_features(channel->session->dsp, dsp_features);
    } else if (channel->session->dsp) {
        ast_dsp_free(channel->session->dsp);
        channel->session->dsp = NULL;
    }
 
    ast_channel_unlock(chan);
 
    return ast_sip_push_task_wait_serializer(channel->session->serializer, dtmf_mode_refresh_cb, &rdata);
}

References ast_sip_session::active_media_state, ast_channel_lock, ast_channel_name(), ast_channel_tech_pvt(), ast_channel_unlock, ast_dsp_free(), ast_dsp_get_features(), ast_dsp_new(), ast_dsp_set_features(), ast_log, AST_MEDIA_TYPE_AUDIO, AST_RTP_DTMF_MODE_INBAND, AST_RTP_DTMF_MODE_NONE, AST_RTP_DTMF_MODE_RFC2833, ast_rtp_instance_dtmf_mode_get(), ast_rtp_instance_dtmf_mode_set(), ast_rtp_instance_set_prop(), AST_RTP_PROPERTY_DTMF, AST_SIP_DTMF_AUTO, AST_SIP_DTMF_AUTO_INFO, AST_SIP_DTMF_INBAND, AST_SIP_DTMF_INFO, AST_SIP_DTMF_NONE, AST_SIP_DTMF_RFC_4733, ast_sip_push_task_wait_serializer, AST_SIP_SESSION_REFRESH_METHOD_INVITE, ast_sip_str_to_dtmf(), ast_sip_session_media_state::default_session, ast_sip_session::dsp, DSP_FEATURE_DIGIT_DETECT, ast_sip_session::dtmf, dtmf_mode_refresh_cb(), LOG_WARNING, refresh_data::method, NULL, ast_sip_session_media::rtp, ast_sip_session::serializer, refresh_data::session, ast_sip_channel_pvt::session, type, and value.

pjsip_acf_media_offer_read()

int pjsip_acf_media_offer_read	(	struct ast_channel *	chan,
		const char *	cmd,
		char *	data,
		char *	buf,
		size_t	len
	)

PJSIP_MEDIA_OFFER function read callback.

Parameters

chan	The channel the function is called on
cmd	The name of the function
data	Arguments passed to the function
buf	Out buffer that should be populated with the data
len	Size of the buffer

Return values

0	on success
-1	on failure

Definition at line 872 of file dialplan_functions.c.

{
    struct ast_sip_channel_pvt *channel;
 
    if (!chan) {
        ast_log(LOG_WARNING, "No channel was provided to %s function.\n", cmd);
        return -1;
    }
 
    if (strcmp(ast_channel_tech(chan)->type, "PJSIP")) {
        ast_log(LOG_WARNING, "Cannot call %s on a non-PJSIP channel\n", cmd);
        return -1;
    }
 
    channel = ast_channel_tech_pvt(chan);
 
    if (!strcmp(data, "audio")) {
        return media_offer_read_av(channel->session, buf, len, AST_MEDIA_TYPE_AUDIO);
    } else if (!strcmp(data, "video")) {
        return media_offer_read_av(channel->session, buf, len, AST_MEDIA_TYPE_VIDEO);
    } else {
        /* Ensure that the buffer is empty */
        buf[0] = '\0';
    }
 
    return 0;
}

References ast_channel_tech_pvt(), ast_log, AST_MEDIA_TYPE_AUDIO, AST_MEDIA_TYPE_VIDEO, buf, len(), LOG_WARNING, media_offer_read_av(), ast_sip_channel_pvt::session, and type.

pjsip_acf_media_offer_write()

int pjsip_acf_media_offer_write	(	struct ast_channel *	chan,
		const char *	cmd,
		char *	data,
		const char *	value
	)

PJSIP_MEDIA_OFFER function write callback.

Parameters

chan	The channel the function is called on
cmd	The name of the function
data	Arguments passed to the function
value	Value to be set by the function

Return values

0	on success
-1	on failure

Definition at line 900 of file dialplan_functions.c.

{
    struct ast_sip_channel_pvt *channel;
    struct media_offer_data mdata = {
        .value = value
    };
 
    if (!chan) {
        ast_log(LOG_WARNING, "No channel was provided to %s function.\n", cmd);
        return -1;
    }
 
    if (strcmp(ast_channel_tech(chan)->type, "PJSIP")) {
        ast_log(LOG_WARNING, "Cannot call %s on a non-PJSIP channel\n", cmd);
        return -1;
    }
 
    channel = ast_channel_tech_pvt(chan);
    mdata.session = channel->session;
 
    if (!strcmp(data, "audio")) {
        mdata.media_type = AST_MEDIA_TYPE_AUDIO;
    } else if (!strcmp(data, "video")) {
        mdata.media_type = AST_MEDIA_TYPE_VIDEO;
    }
 
    return ast_sip_push_task_wait_serializer(channel->session->serializer, media_offer_write_av, &mdata);
}

References ast_channel_tech_pvt(), ast_log, AST_MEDIA_TYPE_AUDIO, AST_MEDIA_TYPE_VIDEO, ast_sip_push_task_wait_serializer, LOG_WARNING, media_offer_write_av(), media_offer_data::media_type, ast_sip_session::serializer, media_offer_data::session, ast_sip_channel_pvt::session, type, media_offer_data::value, and value.

pjsip_acf_moh_passthrough_read()

int pjsip_acf_moh_passthrough_read	(	struct ast_channel *	chan,
		const char *	cmd,
		char *	data,
		char *	buf,
		size_t	len
	)

PJSIP_MOH_PASSTHROUGH function read callback.

Parameters

chan	The channel the function is called on
cmd	The name of the function
data	Arguments passed to the function
buf	Out buffer that should be populated with the data
len	Size of the buffer

Return values

0	on success
-1	on failure

Definition at line 957 of file dialplan_functions.c.

{
    struct ast_sip_channel_pvt *channel;
 
    if (!chan) {
        ast_log(LOG_WARNING, "No channel was provided to %s function.\n", cmd);
        return -1;
    }
 
    if (len < 3) {
        ast_log(LOG_WARNING, "%s: buffer too small\n", cmd);
        return -1;
    }
 
    ast_channel_lock(chan);
    if (strcmp(ast_channel_tech(chan)->type, "PJSIP")) {
        ast_log(LOG_WARNING, "Cannot call %s on a non-PJSIP channel\n", cmd);
        ast_channel_unlock(chan);
        return -1;
    }
 
    channel = ast_channel_tech_pvt(chan);
    strncpy(buf, AST_YESNO(channel->session->moh_passthrough), len);
 
    ast_channel_unlock(chan);
    return 0;
}

References ast_channel_lock, ast_channel_tech_pvt(), ast_channel_unlock, ast_log, AST_YESNO, buf, len(), LOG_WARNING, ast_sip_session::moh_passthrough, ast_sip_channel_pvt::session, and type.

pjsip_acf_moh_passthrough_write()

int pjsip_acf_moh_passthrough_write	(	struct ast_channel *	chan,
		const char *	cmd,
		char *	data,
		const char *	value
	)

PJSIP_MOH_PASSTHROUGH function write callback.

Parameters

chan	The channel the function is called on
cmd	The name of the function
data	Arguments passed to the function
value	Value to be set by the function

Return values

0	on success
-1	on failure

Definition at line 1128 of file dialplan_functions.c.

{
    struct ast_sip_channel_pvt *channel;
 
    if (!chan) {
        ast_log(LOG_WARNING, "No channel was provided to %s function.\n", cmd);
        return -1;
    }
 
    ast_channel_lock(chan);
    if (strcmp(ast_channel_tech(chan)->type, "PJSIP")) {
        ast_log(LOG_WARNING, "Cannot call %s on a non-PJSIP channel\n", cmd);
        ast_channel_unlock(chan);
        return -1;
    }
 
    channel = ast_channel_tech_pvt(chan);
    channel->session->moh_passthrough = ast_true(value);
 
    ast_channel_unlock(chan);
 
    return 0;
}

References ast_channel_lock, ast_channel_tech_pvt(), ast_channel_unlock, ast_log, ast_true(), LOG_WARNING, ast_sip_session::moh_passthrough, ast_sip_channel_pvt::session, type, and value.

pjsip_acf_parse_uri_read()

int pjsip_acf_parse_uri_read	(	struct ast_channel *	chan,
		const char *	cmd,
		char *	data,
		char *	buf,
		size_t	len
	)

PJSIP_PARSE_URI function read callback.

Parameters

chan	The channel the function is called on
cmd	The name of the function
data	Arguments passed to the function
buf	Out buffer that should be populated with the data
len	Size of the buffer

Return values

0	on success
-1	on failure

Definition at line 675 of file dialplan_functions.c.

{
    struct parse_uri_args func_args = { 0, };
    int reading_uri_from_var;
 
    AST_DECLARE_APP_ARGS(args,
        AST_APP_ARG(uri_str);
        AST_APP_ARG(type);
    );
 
    AST_STANDARD_APP_ARGS(args, data);
 
    reading_uri_from_var = !strcasecmp(cmd, "PJSIP_PARSE_URI_FROM");
 
    if (reading_uri_from_var) {
        const char *var;
 
        if (ast_strlen_zero(args.uri_str)) {
            ast_log(LOG_WARNING, "The name of a variable containing a URI must be specified when using the '%s' dialplan function\n", cmd);
            return -1;
        }
 
        ast_channel_lock(chan);
        if ((var = pbx_builtin_getvar_helper(chan, args.uri_str))) {
            args.uri_str = ast_strdupa(var);
        }
        ast_channel_unlock(chan);
    }
 
    if (ast_strlen_zero(args.uri_str)) {
        if (reading_uri_from_var) {
            ast_log(LOG_WARNING, "The variable provided to the '%s' dialplan function must contain a URI\n", cmd);
        } else {
            ast_log(LOG_WARNING, "A URI must be specified when using the '%s' dialplan function\n", cmd);
        }
        return -1;
    }
 
    if (ast_strlen_zero(args.type)) {
        ast_log(LOG_WARNING, "A type part of the URI must be specified when using the '%s' dialplan function\n", cmd);
        return -1;
    }
 
    memset(buf, 0, buflen);
 
    func_args.uri = args.uri_str;
    func_args.type = args.type;
    func_args.buf = buf;
    func_args.buflen = buflen;
    if (ast_sip_push_task_wait_serializer(NULL, parse_uri_cb, &func_args)) {
        ast_log(LOG_WARNING, "Unable to parse URI: failed to push task\n");
        return -1;
    }
 
    return func_args.ret;
}

References args, AST_APP_ARG, ast_channel_lock, ast_channel_unlock, AST_DECLARE_APP_ARGS, ast_log, ast_sip_push_task_wait_serializer, AST_STANDARD_APP_ARGS, ast_strdupa, ast_strlen_zero(), buf, parse_uri_args::buflen, func_args(), LOG_WARNING, NULL, parse_uri_cb(), pbx_builtin_getvar_helper(), type, and var.

pjsip_acf_session_refresh_write()

int pjsip_acf_session_refresh_write	(	struct ast_channel *	chan,
		const char *	cmd,
		char *	data,
		const char *	value
	)

PJSIP_SEND_SESSION_REFRESH function write callback.

Parameters

chan	The channel the function is called on
cmd	the Name of the function
data	Arguments passed to the function
value	Value to be set by the function

Return values

0	on success
-1	on failure

Definition at line 1171 of file dialplan_functions.c.

{
    struct ast_sip_channel_pvt *channel;
    struct refresh_data rdata = {
        .method = AST_SIP_SESSION_REFRESH_METHOD_INVITE,
    };
 
    if (!chan) {
        ast_log(LOG_WARNING, "No channel was provided to %s function.\n", cmd);
        return -1;
    }
 
    if (ast_channel_state(chan) != AST_STATE_UP) {
        ast_log(LOG_WARNING, "'%s' not allowed on unanswered channel '%s'.\n", cmd, ast_channel_name(chan));
        return -1;
    }
 
    if (strcmp(ast_channel_tech(chan)->type, "PJSIP")) {
        ast_log(LOG_WARNING, "Cannot call %s on a non-PJSIP channel\n", cmd);
        return -1;
    }
 
    channel = ast_channel_tech_pvt(chan);
    rdata.session = channel->session;
 
    if (!strcmp(value, "invite")) {
        rdata.method = AST_SIP_SESSION_REFRESH_METHOD_INVITE;
    } else if (!strcmp(value, "update")) {
        rdata.method = AST_SIP_SESSION_REFRESH_METHOD_UPDATE;
    }
 
    return ast_sip_push_task_wait_serializer(channel->session->serializer, refresh_write_cb, &rdata);
}

References ast_channel_name(), ast_channel_tech_pvt(), ast_log, ast_sip_push_task_wait_serializer, AST_SIP_SESSION_REFRESH_METHOD_INVITE, AST_SIP_SESSION_REFRESH_METHOD_UPDATE, AST_STATE_UP, LOG_WARNING, refresh_data::method, refresh_write_cb(), ast_sip_session::serializer, refresh_data::session, ast_sip_channel_pvt::session, type, and value.

pjsip_action_hangup()

int pjsip_action_hangup	(	struct mansession *	s,
		const struct message *	m
	)

Manager action to hang up an incoming PJSIP channel with a SIP response code.

Parameters

s	session
m	message

Return values

0	on success
-1	on failure

Manager action to hang up an incoming PJSIP channel with a SIP response code.

Definition at line 1317 of file dialplan_functions.c.

{
    return ast_manager_hangup_helper(s, m,
        pjsip_app_hangup_handler, response_code_validator);
}

References ast_manager_hangup_helper(), pjsip_app_hangup_handler(), and response_code_validator().

Referenced by load_module().

pjsip_app_hangup()

int pjsip_app_hangup	(	struct ast_channel *	chan,
		const char *	data
	)

Hang up an incoming PJSIP channel with a SIP response code.

Parameters

chan	The channel the function is called on
data	SIP response code or name

Return values

0	on success
-1	on failure

Hang up an incoming PJSIP channel with a SIP response code.

Definition at line 1297 of file dialplan_functions.c.

{
    int response_code;
    const char *tag = ast_channel_name(chan);
 
    if (ast_strlen_zero(data)) {
        ast_log(LOG_WARNING, "%s: Missing response code parameter\n", tag);
        return -1;
    }
 
    response_code = response_code_validator(tag, data);
 
    pjsip_app_hangup_handler(chan, response_code);
 
    return -1;
}

References ast_channel_name(), ast_log, ast_strlen_zero(), hangup_data::chan, LOG_WARNING, pjsip_app_hangup_handler(), hangup_data::response_code, and response_code_validator().

Referenced by load_module().

pjsip_transfer_handling_write()

int pjsip_transfer_handling_write	(	struct ast_channel *	chan,
		const char *	cmd,
		char *	data,
		const char *	value
	)

PJSIP_TRANSFER_HANDLING function write callback.

Parameters

chan	The channel the function is called on
cmd	the Name of the function
data	Arguments passed to the function
value	Value to be set by the function

Return values

0	on success
-1	on failure

Definition at line 1323 of file dialplan_functions.c.

{
    struct ast_sip_channel_pvt *channel;
    int ret = 0;
 
    if (!chan) {
        ast_log(LOG_WARNING, "No channel was provided to %s function.\n", cmd);
        return -1;
    }
 
    ast_channel_lock(chan);
    if (strcmp(ast_channel_tech(chan)->type, "PJSIP")) {
        ast_log(LOG_WARNING, "Cannot call %s on a non-PJSIP channel %s\n", cmd, ast_channel_name(chan));
        ast_channel_unlock(chan);
        return -1;
    }
 
    channel = ast_channel_tech_pvt(chan);
 
    if (ast_strlen_zero(value) || !strcmp(value, "core")) {
        channel->session->transferhandling_ari = 0;
    } else if (!strcmp(value, "ari-only")) {
        channel->session->transferhandling_ari = 1;
    } else {
        ast_log(AST_LOG_WARNING, "Cannot set unknown transfer handling '%s' on channel '%s', transfer handling will remain unchanged.",
            value, ast_channel_name(chan));
        ret = -1;
    }
 
    ast_channel_unlock(chan);
    return ret;
}

References ast_channel_lock, ast_channel_name(), ast_channel_tech_pvt(), ast_channel_unlock, ast_log, AST_LOG_WARNING, ast_strlen_zero(), LOG_WARNING, ast_sip_channel_pvt::session, ast_sip_session::transferhandling_ari, type, and value.