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-rw-r--r--res/res_pjsip.c23
1 files changed, 22 insertions, 1 deletions
diff --git a/res/res_pjsip.c b/res/res_pjsip.c
index 1d57a072b..316a9472e 100644
--- a/res/res_pjsip.c
+++ b/res/res_pjsip.c
@@ -199,7 +199,7 @@
<para>This setting allows to choose the DTMF mode for endpoint communication.</para>
<enumlist>
<enum name="rfc4733">
- <para>DTMF is sent out of band of the main audio stream.This
+ <para>DTMF is sent out of band of the main audio stream. This
supercedes the older <emphasis>RFC-2833</emphasis> used within
the older <literal>chan_sip</literal>.</para>
</enum>
@@ -316,6 +316,27 @@
<configOption name="send_rpid" default="no">
<synopsis>Send the Remote-Party-ID header</synopsis>
</configOption>
+ <configOption name="rpid_immediate" default="no">
+ <synopsis>Immediately send connected line updates on unanswered incoming calls.</synopsis>
+ <description>
+ <para>When enabled, immediately send <emphasis>180 Ringing</emphasis>
+ or <emphasis>183 Progress</emphasis> response messages to the
+ caller if the connected line information is updated before
+ the call is answered. This can send a <emphasis>180 Ringing</emphasis>
+ response before the call has even reached the far end. The
+ caller can start hearing ringback before the far end even gets
+ the call. Many phones tend to grab the first connected line
+ information and refuse to update the display if it changes. The
+ first information is not likely to be correct if the call
+ goes to an endpoint not under the control of this Asterisk
+ box.</para>
+ <para>When disabled, a connected line update must wait for
+ another reason to send a message with the connected line
+ information to the caller before the call is answered. You can
+ trigger the sending of the information by using an appropriate
+ dialplan application such as <emphasis>Ringing</emphasis>.</para>
+ </description>
+ </configOption>
<configOption name="timers_min_se" default="90">
<synopsis>Minimum session timers expiration period</synopsis>
<description><para>