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Voice
asterisk
Commits
19da99df
Commit
19da99df
authored
7 years ago
by
Alexei Gradinari
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CHANGES: correct version for a new option 'refer_blind_progress'
Change-Id: If4817d26a8974610827624fb8a4e56d681d6bf97
parent
9f054955
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19da99df
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@@ -21,6 +21,15 @@ app_queue
--- Functionality changes from Asterisk 14.5.0 to Asterisk 14.6.0 ------------
------------------------------------------------------------------------------
res_pjsip
------------------
* A new endpoint option "refer_blind_progress" was added to turn off notifying
the progress details on Blind Transfer. If this option is not set then
the chan_pjsip will send NOTIFY "200 OK" immediately after "202 Accepted".
On default is enabled.
Some SIP phones like Mitel/Aastra or Snom keep the line busy until
receive "200 OK".
res_agi
------------------
* The EAGI() application will now look for a dialplan variable named
...
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@@ -38,15 +47,6 @@ chan_pjsip
--- Functionality changes from Asterisk 14.4.0 to Asterisk 14.5.0 ------------
------------------------------------------------------------------------------
res_pjsip
------------------
* A new endpoint option "refer_blind_progress" was added to turn off notifying
the progress details on Blind Transfer. If this option is not set then
the chan_pjsip will send NOTIFY "200 OK" immediately after "202 Accepted".
On default is enabled.
Some SIP phones like Mitel/Aastra or Snom keep the line busy until
receive "200 OK".
res_rtp_asterisk
------------------
* Added the stun_blacklist option to rtp.conf. Some multihomed servers have
...
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