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authorJenkins2 <jenkins2@gerrit.asterisk.org>2017-06-01 09:48:48 -0500
committerGerrit Code Review <gerrit2@gerrit.digium.api>2017-06-01 09:48:48 -0500
commit3e8eea0325d18ef4b29c45349cddbf13552b4d86 (patch)
tree5e4caee9e058512aa97d54d7945ab573cd08ab55 /configs
parenta76b473c491f14786d120c7bba127ee53eb380d0 (diff)
parent6af2dd34afc2c20bdabd07bc3836821690db4c86 (diff)
Merge "res_pjsip: New endpoint option "refer_blind_progress"" into 13
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 5ef1c97f7..494a32a62 100644
--- a/configs/samples/pjsip.conf.sample
+++ b/configs/samples/pjsip.conf.sample
@@ -764,6 +764,10 @@
; header to this value (default: "")
;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]