[Asterisk-code-review] chan pjsip: add a new function PJSIP DTMF MODE (asterisk[14])
Jenkins2
asteriskteam at digium.com
Thu Jul 27 08:46:46 CDT 2017
Jenkins2 has submitted this change and it was merged. ( https://gerrit.asterisk.org/6087 )
Change subject: chan_pjsip: add a new function PJSIP_DTMF_MODE
......................................................................
chan_pjsip: add a new function PJSIP_DTMF_MODE
This function is a replica of SIPDtmfMode, allowing the DTMF mode of a
PJSIP call to be modified on a per-call basis
ASTERISK-27085 #close
Change-Id: I20eef5da3e5d1d3e58b304416bc79683f87e7612
---
M CHANGES
M channels/chan_pjsip.c
M channels/pjsip/dialplan_functions.c
M channels/pjsip/include/dialplan_functions.h
M include/asterisk/res_pjsip.h
M include/asterisk/res_pjsip_session.h
M res/res_pjsip.c
M res/res_pjsip/pjsip_configuration.c
M res/res_pjsip_sdp_rtp.c
M res/res_pjsip_session.c
10 files changed, 293 insertions(+), 37 deletions(-)
Approvals:
Sean Bright: Looks good to me, but someone else must approve
Joshua Colp: Looks good to me, but someone else must approve; Verified
George Joseph: Looks good to me, approved
Jenkins2: Approved for Submit
diff --git a/CHANGES b/CHANGES
index 5907e13..63422b4 100644
--- a/CHANGES
+++ b/CHANGES
@@ -28,6 +28,9 @@
which sends signals to the application and its descendants directly, or
"process" which sends signals only to the application itself.
+ * New dialplan function PJSIP_DTMF_MODE added to get or change the DTMF mode
+ of a channel on a per-call basis.
+
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 14.5.0 to Asterisk 14.6.0 ------------
------------------------------------------------------------------------------
diff --git a/channels/chan_pjsip.c b/channels/chan_pjsip.c
index 6a11b59..93e74a5 100644
--- a/channels/chan_pjsip.c
+++ b/channels/chan_pjsip.c
@@ -1749,7 +1749,7 @@
struct ast_sip_session_media *media = pvt->media[SIP_MEDIA_AUDIO];
int res = 0;
- switch (channel->session->endpoint->dtmf) {
+ switch (channel->session->dtmf) {
case AST_SIP_DTMF_RFC_4733:
if (!media || !media->rtp) {
return -1;
@@ -1869,7 +1869,7 @@
struct ast_sip_session_media *media = pvt->media[SIP_MEDIA_AUDIO];
int res = 0;
- switch (channel->session->endpoint->dtmf) {
+ switch (channel->session->dtmf) {
case AST_SIP_DTMF_AUTO_INFO:
{
if (!media || !media->rtp) {
@@ -2681,6 +2681,12 @@
.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,
@@ -2722,6 +2728,11 @@
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;
}
@@ -2784,6 +2795,7 @@
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);
@@ -2806,6 +2818,7 @@
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);
diff --git a/channels/pjsip/dialplan_functions.c b/channels/pjsip/dialplan_functions.c
index 01081e9..13d9fab 100644
--- a/channels/pjsip/dialplan_functions.c
+++ b/channels/pjsip/dialplan_functions.c
@@ -68,6 +68,18 @@
<ref type="function">PJSIP_SEND_SESSION_REFRESH</ref>
</see-also>
</function>
+<function name="PJSIP_DTMF_MODE" language="en_US">
+ <synopsis>
+ Get or change the DTMF mode for a SIP call.
+ </synopsis>
+ <syntax>
+ </syntax>
+ <description>
+ <para>When read, returns the current DTMF mode</para>
+ <para>When written, sets the current DTMF mode</para>
+ <para>This function uses the same DTMF mode naming as the dtmf_mode configuration option</para>
+ </description>
+</function>
<function name="PJSIP_SEND_SESSION_REFRESH" language="en_US">
<synopsis>
W/O: Initiate a session refresh via an UPDATE or re-INVITE on an established media session
@@ -440,6 +452,7 @@
#include "asterisk/app.h"
#include "asterisk/channel.h"
#include "asterisk/format.h"
+#include "asterisk/dsp.h"
#include "asterisk/pbx.h"
#include "asterisk/res_pjsip.h"
#include "asterisk/res_pjsip_session.h"
@@ -1039,6 +1052,34 @@
return ast_sip_push_task_synchronous(channel->session->serializer, media_offer_write_av, &mdata);
}
+int pjsip_acf_dtmf_mode_read(struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)
+{
+ 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;
+}
+
struct refresh_data {
struct ast_sip_session *session;
enum ast_sip_session_refresh_method method;
@@ -1067,6 +1108,117 @@
return 0;
}
+static int dtmf_mode_refresh_cb(void *obj)
+{
+ struct refresh_data *data = obj;
+
+ if (data->session->inv_session->state == PJSIP_INV_STATE_CONFIRMED) {
+ ast_debug(3, "Changing DTMF mode on channel %s after OFFER/ANSER completion. Sending session refresh\n", ast_channel_name(data->session->channel));
+
+ ast_sip_session_refresh(data->session, NULL, NULL,
+ sip_session_response_cb, data->method, 1);
+ }
+
+ return 0;
+}
+
+int pjsip_acf_dtmf_mode_write(struct ast_channel *chan, const char *cmd, char *data, const char *value)
+{
+ struct ast_sip_channel_pvt *channel;
+ struct chan_pjsip_pvt *pjsip_pvt;
+ 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;
+
+ pjsip_pvt = channel->pvt;
+ if (pjsip_pvt->media[SIP_MEDIA_AUDIO] && (pjsip_pvt->media[SIP_MEDIA_AUDIO])->rtp) {
+ if (channel->session->dtmf == AST_SIP_DTMF_RFC_4733) {
+ ast_rtp_instance_set_prop((pjsip_pvt->media[SIP_MEDIA_AUDIO])->rtp, AST_RTP_PROPERTY_DTMF, 1);
+ ast_rtp_instance_dtmf_mode_set((pjsip_pvt->media[SIP_MEDIA_AUDIO])->rtp, AST_RTP_DTMF_MODE_RFC2833);
+ } else if (channel->session->dtmf == AST_SIP_DTMF_INFO) {
+ ast_rtp_instance_set_prop((pjsip_pvt->media[SIP_MEDIA_AUDIO])->rtp, AST_RTP_PROPERTY_DTMF, 0);
+ ast_rtp_instance_dtmf_mode_set((pjsip_pvt->media[SIP_MEDIA_AUDIO])->rtp, AST_RTP_DTMF_MODE_NONE);
+ } else if (channel->session->dtmf == AST_SIP_DTMF_INBAND) {
+ ast_rtp_instance_set_prop((pjsip_pvt->media[SIP_MEDIA_AUDIO])->rtp, AST_RTP_PROPERTY_DTMF, 0);
+ ast_rtp_instance_dtmf_mode_set((pjsip_pvt->media[SIP_MEDIA_AUDIO])->rtp, AST_RTP_DTMF_MODE_INBAND);
+ } else if (channel->session->dtmf == AST_SIP_DTMF_NONE) {
+ ast_rtp_instance_set_prop((pjsip_pvt->media[SIP_MEDIA_AUDIO])->rtp, AST_RTP_PROPERTY_DTMF, 0);
+ ast_rtp_instance_dtmf_mode_set((pjsip_pvt->media[SIP_MEDIA_AUDIO])->rtp, AST_RTP_DTMF_MODE_NONE);
+ } else if (channel->session->dtmf == AST_SIP_DTMF_AUTO) {
+ if (ast_rtp_instance_dtmf_mode_get((pjsip_pvt->media[SIP_MEDIA_AUDIO])->rtp) != AST_RTP_DTMF_MODE_RFC2833) {
+ /* no RFC4733 negotiated, enable inband */
+ ast_rtp_instance_dtmf_mode_set((pjsip_pvt->media[SIP_MEDIA_AUDIO])->rtp, AST_RTP_DTMF_MODE_INBAND);
+ }
+ } else if (channel->session->dtmf == AST_SIP_DTMF_AUTO_INFO) {
+ ast_rtp_instance_set_prop((pjsip_pvt->media[SIP_MEDIA_AUDIO])->rtp, AST_RTP_PROPERTY_DTMF, 0);
+ if (ast_rtp_instance_dtmf_mode_get((pjsip_pvt->media[SIP_MEDIA_AUDIO])->rtp) == AST_RTP_DTMF_MODE_INBAND) {
+ /* if inband, switch to INFO */
+ ast_rtp_instance_dtmf_mode_set((pjsip_pvt->media[SIP_MEDIA_AUDIO])->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_synchronous(channel->session->serializer, dtmf_mode_refresh_cb, &rdata);
+}
+
static int refresh_write_cb(void *obj)
{
struct refresh_data *data = obj;
diff --git a/channels/pjsip/include/dialplan_functions.h b/channels/pjsip/include/dialplan_functions.h
index 8b80bfa..731e91d 100644
--- a/channels/pjsip/include/dialplan_functions.h
+++ b/channels/pjsip/include/dialplan_functions.h
@@ -48,6 +48,31 @@
int pjsip_acf_media_offer_write(struct ast_channel *chan, const char *cmd, char *data, const char *value);
/*!
+ * \brief PJSIP_DTMF_MODE function read callback
+ * \param chan The channel the function is called on
+ * \param cmd The name of the function
+ * \param data Arguments passed to the function
+ * \param buf Out buffer that should be populated with the data
+ * \param len Size of the buffer
+ *
+ * \retval 0 on success
+ * \retval -1 on failure
+ */
+int pjsip_acf_dtmf_mode_read(struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len);
+
+/*!
+ * \brief PJSIP_DTMF_MODE function write callback
+ * \param chan The channel the function is called on
+ * \param cmd The name of the function
+ * \param data Arguments passed to the function
+ * \param value Value to be set by the function
+ *
+ * \retval 0 on success
+ * \retval -1 on failure
+ */
+int pjsip_acf_dtmf_mode_write(struct ast_channel *chan, const char *cmd, char *data, const char *value);
+
+/*!
* \brief PJSIP_MEDIA_OFFER function read callback
* \param chan The channel the function is called on
* \param cmd The name of the function
diff --git a/include/asterisk/res_pjsip.h b/include/asterisk/res_pjsip.h
index 9b42527..a674313 100644
--- a/include/asterisk/res_pjsip.h
+++ b/include/asterisk/res_pjsip.h
@@ -2859,4 +2859,31 @@
int ast_sip_dlg_set_transport(const struct ast_sip_endpoint *endpoint, pjsip_dialog *dlg,
pjsip_tpselector *selector);
+/*!
+ * \brief Convert the DTMF mode enum value into a string
+ * \since 13.18.0
+ *
+ * \param dtmf the dtmf mode
+ * \param buf Buffer to receive dtmf mode string
+ * \param buf_len Buffer length
+ *
+ * \retval 0 Success
+ * \retval -1 Failure
+ *
+ */
+int ast_sip_dtmf_to_str(const enum ast_sip_dtmf_mode dtmf,
+ char *buf, size_t buf_len);
+
+/*!
+ * \brief Convert the DTMF mode name into an enum
+ * \since 13.18.0
+ *
+ * \param dtmf_mode dtmf mode as a string
+ *
+ * \retval >= 0 The enum value
+ * \retval -1 Failure
+ *
+ */
+int ast_sip_str_to_dtmf(const char *dtmf_mode);
+
#endif /* _RES_PJSIP_H */
diff --git a/include/asterisk/res_pjsip_session.h b/include/asterisk/res_pjsip_session.h
index e2a9066..ca331cb 100644
--- a/include/asterisk/res_pjsip_session.h
+++ b/include/asterisk/res_pjsip_session.h
@@ -159,6 +159,8 @@
unsigned int defer_end:1;
/*! Session end (remote hangup) requested while termination deferred */
unsigned int ended_while_deferred:1;
+ /*! DTMF mode to use with this session, from endpoint but can change */
+ enum ast_sip_dtmf_mode dtmf;
};
typedef int (*ast_sip_session_request_creation_cb)(struct ast_sip_session *session, pjsip_tx_data *tdata);
diff --git a/res/res_pjsip.c b/res/res_pjsip.c
index 4fa28a1..e2d2c0a 100644
--- a/res/res_pjsip.c
+++ b/res/res_pjsip.c
@@ -4431,6 +4431,56 @@
return NULL;
}
+int ast_sip_dtmf_to_str(const enum ast_sip_dtmf_mode dtmf,
+ char *buf, size_t buf_len)
+{
+ switch (dtmf) {
+ case AST_SIP_DTMF_NONE:
+ ast_copy_string(buf, "none", buf_len);
+ break;
+ case AST_SIP_DTMF_RFC_4733:
+ ast_copy_string(buf, "rfc4733", buf_len);
+ break;
+ case AST_SIP_DTMF_INBAND:
+ ast_copy_string(buf, "inband", buf_len);
+ break;
+ case AST_SIP_DTMF_INFO:
+ ast_copy_string(buf, "info", buf_len);
+ break;
+ case AST_SIP_DTMF_AUTO:
+ ast_copy_string(buf, "auto", buf_len);
+ break;
+ case AST_SIP_DTMF_AUTO_INFO:
+ ast_copy_string(buf, "auto_info", buf_len);
+ break;
+ default:
+ buf[0] = '\0';
+ return -1;
+ }
+ return 0;
+}
+
+int ast_sip_str_to_dtmf(const char * dtmf_mode)
+{
+ int result = -1;
+
+ if (!strcasecmp(dtmf_mode, "info")) {
+ result = AST_SIP_DTMF_INFO;
+ } else if (!strcasecmp(dtmf_mode, "rfc4733")) {
+ result = AST_SIP_DTMF_RFC_4733;
+ } else if (!strcasecmp(dtmf_mode, "inband")) {
+ result = AST_SIP_DTMF_INBAND;
+ } else if (!strcasecmp(dtmf_mode, "none")) {
+ result = AST_SIP_DTMF_NONE;
+ } else if (!strcasecmp(dtmf_mode, "auto")) {
+ result = AST_SIP_DTMF_AUTO;
+ } else if (!strcasecmp(dtmf_mode, "auto_info")) {
+ result = AST_SIP_DTMF_AUTO_INFO;
+ }
+
+ return result;
+}
+
/*!
* \brief Set name and number information on an identity header.
*
diff --git a/res/res_pjsip/pjsip_configuration.c b/res/res_pjsip/pjsip_configuration.c
index 4e501e6..fb15db0 100644
--- a/res/res_pjsip/pjsip_configuration.c
+++ b/res/res_pjsip/pjsip_configuration.c
@@ -365,47 +365,29 @@
static int dtmf_handler(const struct aco_option *opt, struct ast_variable *var, void *obj)
{
struct ast_sip_endpoint *endpoint = obj;
+ enum ast_sip_dtmf_mode dtmf = ast_sip_str_to_dtmf(var->value);
- if (!strcasecmp(var->value, "rfc4733")) {
- endpoint->dtmf = AST_SIP_DTMF_RFC_4733;
- } else if (!strcasecmp(var->value, "inband")) {
- endpoint->dtmf = AST_SIP_DTMF_INBAND;
- } else if (!strcasecmp(var->value, "auto_info")) {
- endpoint->dtmf = AST_SIP_DTMF_AUTO_INFO;
- } else if (!strcasecmp(var->value, "info")) {
- endpoint->dtmf = AST_SIP_DTMF_INFO;
- } else if (!strcasecmp(var->value, "auto")) {
- endpoint->dtmf = AST_SIP_DTMF_AUTO;
- } else if (!strcasecmp(var->value, "none")) {
- endpoint->dtmf = AST_SIP_DTMF_NONE;
- } else {
+ if (dtmf == -1) {
return -1;
}
+ endpoint->dtmf = dtmf;
return 0;
}
static int dtmf_to_str(const void *obj, const intptr_t *args, char **buf)
{
const struct ast_sip_endpoint *endpoint = obj;
+ char dtmf_str[20];
+ int result = -1;
- switch (endpoint->dtmf) {
- case AST_SIP_DTMF_RFC_4733 :
- *buf = "rfc4733"; break;
- case AST_SIP_DTMF_INBAND :
- *buf = "inband"; break;
- case AST_SIP_DTMF_INFO :
- *buf = "info"; break;
- case AST_SIP_DTMF_AUTO :
- *buf = "auto"; break;
- case AST_SIP_DTMF_AUTO_INFO :
- *buf = "auto_info";
- break;
- default:
- *buf = "none";
+ result = ast_sip_dtmf_to_str(endpoint->dtmf, dtmf_str, sizeof(dtmf_str));
+
+ if (result == 0) {
+ *buf = ast_strdup(dtmf_str);
+ } else {
+ *buf = ast_strdup("none");
}
-
- *buf = ast_strdup(*buf);
return 0;
}
diff --git a/res/res_pjsip_sdp_rtp.c b/res/res_pjsip_sdp_rtp.c
index 6b29e79..5fff767 100644
--- a/res/res_pjsip_sdp_rtp.c
+++ b/res/res_pjsip_sdp_rtp.c
@@ -247,10 +247,10 @@
ice->stop(session_media->rtp);
}
- if (session->endpoint->dtmf == AST_SIP_DTMF_RFC_4733 || session->endpoint->dtmf == AST_SIP_DTMF_AUTO || session->endpoint->dtmf == AST_SIP_DTMF_AUTO_INFO) {
+ if (session->dtmf == AST_SIP_DTMF_RFC_4733 || session->dtmf == AST_SIP_DTMF_AUTO || session->dtmf == AST_SIP_DTMF_AUTO_INFO) {
ast_rtp_instance_dtmf_mode_set(session_media->rtp, AST_RTP_DTMF_MODE_RFC2833);
ast_rtp_instance_set_prop(session_media->rtp, AST_RTP_PROPERTY_DTMF, 1);
- } else if (session->endpoint->dtmf == AST_SIP_DTMF_INBAND) {
+ } else if (session->dtmf == AST_SIP_DTMF_INBAND) {
ast_rtp_instance_dtmf_mode_set(session_media->rtp, AST_RTP_DTMF_MODE_INBAND);
}
@@ -333,11 +333,11 @@
}
}
}
- if (!tel_event && (session->endpoint->dtmf == AST_SIP_DTMF_AUTO)) {
+ if (!tel_event && (session->dtmf == AST_SIP_DTMF_AUTO)) {
ast_rtp_instance_dtmf_mode_set(session_media->rtp, AST_RTP_DTMF_MODE_INBAND);
}
- if (session->endpoint->dtmf == AST_SIP_DTMF_AUTO_INFO) {
+ if (session->dtmf == AST_SIP_DTMF_AUTO_INFO) {
if (tel_event) {
ast_rtp_instance_dtmf_mode_set(session_media->rtp, AST_RTP_DTMF_MODE_RFC2833);
} else {
@@ -450,7 +450,7 @@
ast_set_write_format(session->channel, ast_channel_writeformat(session->channel));
}
- if ( ((session->endpoint->dtmf == AST_SIP_DTMF_AUTO) || (session->endpoint->dtmf == AST_SIP_DTMF_AUTO_INFO) )
+ if ( ((session->dtmf == AST_SIP_DTMF_AUTO) || (session->dtmf == AST_SIP_DTMF_AUTO_INFO) )
&& (ast_rtp_instance_dtmf_mode_get(session_media->rtp) == AST_RTP_DTMF_MODE_RFC2833)
&& (session->dsp)) {
dsp_features = ast_dsp_get_features(session->dsp);
@@ -1178,7 +1178,7 @@
pj_str_t stmp;
pjmedia_sdp_attr *attr;
int index = 0;
- int noncodec = (session->endpoint->dtmf == AST_SIP_DTMF_RFC_4733 || session->endpoint->dtmf == AST_SIP_DTMF_AUTO || session->endpoint->dtmf == AST_SIP_DTMF_AUTO_INFO) ? AST_RTP_DTMF : 0;
+ int noncodec = (session->dtmf == AST_SIP_DTMF_RFC_4733 || session->dtmf == AST_SIP_DTMF_AUTO || session->dtmf == AST_SIP_DTMF_AUTO_INFO) ? AST_RTP_DTMF : 0;
int min_packet_size = 0, max_packet_size = 0;
int rtp_code;
RAII_VAR(struct ast_format_cap *, caps, NULL, ao2_cleanup);
diff --git a/res/res_pjsip_session.c b/res/res_pjsip_session.c
index aedf3f6..aed047b 100644
--- a/res/res_pjsip_session.c
+++ b/res/res_pjsip_session.c
@@ -1481,6 +1481,8 @@
session->contact = ao2_bump(contact);
session->inv_session = inv_session;
+ session->dtmf = endpoint->dtmf;
+
if (add_supplements(session)) {
/* Release the ref held by session->inv_session */
ao2_ref(session, -1);
--
To view, visit https://gerrit.asterisk.org/6087
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Gerrit-Project: asterisk
Gerrit-Branch: 14
Gerrit-MessageType: merged
Gerrit-Change-Id: I20eef5da3e5d1d3e58b304416bc79683f87e7612
Gerrit-Change-Number: 6087
Gerrit-PatchSet: 1
Gerrit-Owner: Torrey Searle <tsearle at gmail.com>
Gerrit-Reviewer: George Joseph <gjoseph at digium.com>
Gerrit-Reviewer: Jenkins2
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Sean Bright <sean.bright at gmail.com>
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