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author | Jenkins2 <jenkins2@gerrit.asterisk.org> | 2017-06-01 09:48:48 -0500 |
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committer | Gerrit Code Review <gerrit2@gerrit.digium.api> | 2017-06-01 09:48:48 -0500 |
commit | 3e8eea0325d18ef4b29c45349cddbf13552b4d86 (patch) | |
tree | 5e4caee9e058512aa97d54d7945ab573cd08ab55 /configs/samples | |
parent | a76b473c491f14786d120c7bba127ee53eb380d0 (diff) | |
parent | 6af2dd34afc2c20bdabd07bc3836821690db4c86 (diff) |
Merge "res_pjsip: New endpoint option "refer_blind_progress"" into 13
Diffstat (limited to 'configs/samples')
-rw-r--r-- | configs/samples/pjsip.conf.sample | 4 |
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] |