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authorJenkins2 <jenkins2@gerrit.asterisk.org>2017-06-01 10:05:53 -0500
committerGerrit Code Review <gerrit2@gerrit.digium.api>2017-06-01 10:05:53 -0500
commit48d047ad5a6cbd24368a003027c5a9c5cc43ef26 (patch)
tree8472f53e74055380032ae378a2384845f8120058 /configs
parent77f7c1b98f1b293f66ab4042758e138be9a5e613 (diff)
parent808f2998085b78225addfd73289fdc6c473b7fa1 (diff)
Merge "res_pjsip: New endpoint option "refer_blind_progress""
Diffstat (limited to 'configs')
-rw-r--r--configs/samples/pjsip.conf.sample4
1 files changed, 4 insertions, 0 deletions
diff --git a/configs/samples/pjsip.conf.sample b/configs/samples/pjsip.conf.sample
index 612649ec5..a992ff8b6 100644
--- a/configs/samples/pjsip.conf.sample
+++ b/configs/samples/pjsip.conf.sample
@@ -770,6 +770,10 @@
; default is no.
;asymmetric_rtp_codec= ; Allow the sending and receiving codec to differ and
; not be automatically matched (default: "no")
+;refer_blind_progress= ; Whether to notifies all the progress details on blind
+ ; transfer (default: "yes"). The value "no" is useful
+ ; for some SIP phones (Mitel/Aastra, Snom) which expect
+ ; a sip/frag "200 OK" after REFER has been accepted.
;==========================AUTH SECTION OPTIONS=========================
;[auth]