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-rw-r--r--res/res_pjsip.c16
1 files changed, 16 insertions, 0 deletions
diff --git a/res/res_pjsip.c b/res/res_pjsip.c
index fefbff446..4de074597 100644
--- a/res/res_pjsip.c
+++ b/res/res_pjsip.c
@@ -798,6 +798,22 @@
a hole open in order to allow for media to arrive at Asterisk.
</para></description>
</configOption>
+ <configOption name="rtp_timeout" default="0">
+ <synopsis>Maximum number of seconds without receiving RTP (while off hold) before terminating call.</synopsis>
+ <description><para>
+ This option configures the number of seconds without RTP (while off hold) before
+ considering a channel as dead. When the number of seconds is reached the underlying
+ channel is hung up. By default this option is set to 0, which means do not check.
+ </para></description>
+ </configOption>
+ <configOption name="rtp_timeout_hold" default="0">
+ <synopsis>Maximum number of seconds without receiving RTP (while on hold) before terminating call.</synopsis>
+ <description><para>
+ This option configures the number of seconds without RTP (while on hold) before
+ considering a channel as dead. When the number of seconds is reached the underlying
+ channel is hung up. By default this option is set to 0, which means do not check.
+ </para></description>
+ </configOption>
</configObject>
<configObject name="auth">
<synopsis>Authentication type</synopsis>