<p>Kevin Harwell has uploaded this change for <strong>review</strong>.</p><p><a href="https://gerrit.asterisk.org/c/asterisk/+/14142">View Change</a></p><pre style="font-family: monospace,monospace; white-space: pre-wrap;">chan_pjsip: digit_begin - constant DTMF tone if RTP is not setup yet<br><br>If chan_pjsip is configured for DTMF_RFC_4733, and the core triggers a digit<br>begin before media, or rtp has been setup then it's possible the outgoing<br>channel will hear a constant DTMF tone upon answering.<br><br>This happens because when there is no media, or rtp chan_pjsip notifies the<br>core to initiate inband DTMF. However, upon digit end if media, and rtp become<br>available then chan_pjsip does not notify the core to stop inband DTMF. Thus<br>the tone continues playing.<br><br>This patch makes it so chan_pjsip only notifies the core to start inband DTMF<br>in only the required cases. Now if there is no media, or rtp availabe upon<br>digit begin chan_pjsip does nothing, but tells the core it handled it.<br><br>ASTERISK-28817 #close<br><br>Change-Id: I0dbea9fff444a2595fb18c64b89653e90d2f6eb5<br>---<br>M channels/chan_pjsip.c<br>1 file changed, 18 insertions(+), 15 deletions(-)<br><br></pre><pre style="font-family: monospace,monospace; white-space: pre-wrap;">git pull ssh://gerrit.asterisk.org:29418/asterisk refs/changes/42/14142/1</pre><pre style="font-family: monospace,monospace; white-space: pre-wrap;"><span>diff --git a/channels/chan_pjsip.c b/channels/chan_pjsip.c</span><br><span>index 33c023d..79e7c6e 100644</span><br><span>--- a/channels/chan_pjsip.c</span><br><span>+++ b/channels/chan_pjsip.c</span><br><span>@@ -1816,18 +1816,21 @@</span><br><span> struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);</span><br><span> struct chan_pjsip_pvt *pvt = channel->pvt;</span><br><span> struct ast_sip_session_media *media = pvt->media[SIP_MEDIA_AUDIO];</span><br><span style="color: hsl(0, 100%, 40%);">- int res = 0;</span><br><span> </span><br><span> switch (channel->session->dtmf) {</span><br><span> case AST_SIP_DTMF_RFC_4733:</span><br><span> if (!media || !media->rtp) {</span><br><span style="color: hsl(0, 100%, 40%);">- return -1;</span><br><span style="color: hsl(120, 100%, 40%);">+ return 0;</span><br><span> }</span><br><span> </span><br><span> ast_rtp_instance_dtmf_begin(media->rtp, digit);</span><br><span> break;</span><br><span> case AST_SIP_DTMF_AUTO:</span><br><span style="color: hsl(0, 100%, 40%);">- if (!media || !media->rtp || (ast_rtp_instance_dtmf_mode_get(media->rtp) == AST_RTP_DTMF_MODE_INBAND)) {</span><br><span style="color: hsl(120, 100%, 40%);">+ if (!media || !media->rtp) {</span><br><span style="color: hsl(120, 100%, 40%);">+ return 0;</span><br><span style="color: hsl(120, 100%, 40%);">+ }</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+ if (ast_rtp_instance_dtmf_mode_get(media->rtp) == AST_RTP_DTMF_MODE_INBAND) {</span><br><span> return -1;</span><br><span> }</span><br><span> </span><br><span>@@ -1842,13 +1845,12 @@</span><br><span> case AST_SIP_DTMF_NONE:</span><br><span> break;</span><br><span> case AST_SIP_DTMF_INBAND:</span><br><span style="color: hsl(0, 100%, 40%);">- res = -1;</span><br><span style="color: hsl(0, 100%, 40%);">- break;</span><br><span style="color: hsl(120, 100%, 40%);">+ return -1;</span><br><span> default:</span><br><span> break;</span><br><span> }</span><br><span> </span><br><span style="color: hsl(0, 100%, 40%);">- return res;</span><br><span style="color: hsl(120, 100%, 40%);">+ return 0;</span><br><span> }</span><br><span> </span><br><span> struct info_dtmf_data {</span><br><span>@@ -1936,7 +1938,6 @@</span><br><span> struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);</span><br><span> struct chan_pjsip_pvt *pvt = channel->pvt;</span><br><span> struct ast_sip_session_media *media = pvt->media[SIP_MEDIA_AUDIO];</span><br><span style="color: hsl(0, 100%, 40%);">- int res = 0;</span><br><span> </span><br><span> if (!channel || !channel->session) {</span><br><span> /* This happens when the channel is hungup while a DTMF digit is playing. See ASTERISK-28086 */</span><br><span>@@ -1948,8 +1949,9 @@</span><br><span> case AST_SIP_DTMF_AUTO_INFO:</span><br><span> {</span><br><span> if (!media || !media->rtp) {</span><br><span style="color: hsl(0, 100%, 40%);">- return -1;</span><br><span style="color: hsl(120, 100%, 40%);">+ return 0;</span><br><span> }</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span> if (ast_rtp_instance_dtmf_mode_get(media->rtp) != AST_RTP_DTMF_MODE_NONE) {</span><br><span> ast_debug(3, "Told to send end of digit on Auto-Info channel %s RFC4733 negotiated so using it.\n", ast_channel_name(ast));</span><br><span> ast_rtp_instance_dtmf_end_with_duration(media->rtp, digit, duration);</span><br><span>@@ -1987,28 +1989,29 @@</span><br><span> }</span><br><span> case AST_SIP_DTMF_RFC_4733:</span><br><span> if (!media || !media->rtp) {</span><br><span style="color: hsl(0, 100%, 40%);">- return -1;</span><br><span style="color: hsl(120, 100%, 40%);">+ return 0;</span><br><span> }</span><br><span> </span><br><span> ast_rtp_instance_dtmf_end_with_duration(media->rtp, digit, duration);</span><br><span> break;</span><br><span> case AST_SIP_DTMF_AUTO:</span><br><span style="color: hsl(0, 100%, 40%);">- if (!media || !media->rtp || (ast_rtp_instance_dtmf_mode_get(media->rtp) == AST_RTP_DTMF_MODE_INBAND)) {</span><br><span style="color: hsl(120, 100%, 40%);">+ if (!media || !media->rtp) {</span><br><span style="color: hsl(120, 100%, 40%);">+ return 0;</span><br><span style="color: hsl(120, 100%, 40%);">+ }</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+ if (ast_rtp_instance_dtmf_mode_get(media->rtp) == AST_RTP_DTMF_MODE_INBAND) {</span><br><span> return -1;</span><br><span> }</span><br><span> </span><br><span> ast_rtp_instance_dtmf_end_with_duration(media->rtp, digit, duration);</span><br><span> break;</span><br><span style="color: hsl(0, 100%, 40%);">-</span><br><span style="color: hsl(0, 100%, 40%);">-</span><br><span> case AST_SIP_DTMF_NONE:</span><br><span> break;</span><br><span> case AST_SIP_DTMF_INBAND:</span><br><span style="color: hsl(0, 100%, 40%);">- res = -1;</span><br><span style="color: hsl(0, 100%, 40%);">- break;</span><br><span style="color: hsl(120, 100%, 40%);">+ return -1;</span><br><span> }</span><br><span> </span><br><span style="color: hsl(0, 100%, 40%);">- return res;</span><br><span style="color: hsl(120, 100%, 40%);">+ return 0;</span><br><span> }</span><br><span> </span><br><span> static void update_initial_connected_line(struct ast_sip_session *session)</span><br><span></span><br></pre><p>To view, visit <a href="https://gerrit.asterisk.org/c/asterisk/+/14142">change 14142</a>. 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<div style="display:none"> Gerrit-Project: asterisk </div>
<div style="display:none"> Gerrit-Branch: 13 </div>
<div style="display:none"> Gerrit-Change-Id: I0dbea9fff444a2595fb18c64b89653e90d2f6eb5 </div>
<div style="display:none"> Gerrit-Change-Number: 14142 </div>
<div style="display:none"> Gerrit-PatchSet: 1 </div>
<div style="display:none"> Gerrit-Owner: Kevin Harwell <kharwell@digium.com> </div>
<div style="display:none"> Gerrit-MessageType: newchange </div>