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authorrussell <russell@f38db490-d61c-443f-a65b-d21fe96a405b>2009-12-09 23:13:28 +0000
committerrussell <russell@f38db490-d61c-443f-a65b-d21fe96a405b>2009-12-09 23:13:28 +0000
commit7d3a32a707d6bbf70aff9498e23a2270f7d67782 (patch)
tree2f4b719287f09415ad9a7b9acaea0df7b13095dd /CHANGES
parent4a9521f75838881b2ee65a3276df018754aa2111 (diff)
Fix up the faxdetect entry in CHANGES.
This feature was listed as a 1.6.2 feature, even though it's in all 1.6.X versions. The description of the feature was also no longer accurate. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@234008 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'CHANGES')
-rw-r--r--CHANGES7
1 files changed, 4 insertions, 3 deletions
diff --git a/CHANGES b/CHANGES
index 0b8e705c3..731e8f26b 100644
--- a/CHANGES
+++ b/CHANGES
@@ -376,9 +376,6 @@ SIP Changes
remote services. For backwards compatibility, "secret" still has the
same function as before, but now you can configure both a remote secret and a
local secret for mutual authentication.
- * Added a new 'faxdetect=yes|no' configuration option to sip.conf. When this
- option is enabled, a SIP channel will go to the fax extension (if it exists)
- after T38 is negotiated. This option is disabled by default.
* If the channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is set,
the sound will be played to the target of an attended transfer
* Added two new configuration options, "qualifygap" and "qualifypeers", which allow
@@ -887,6 +884,10 @@ CLI Changes
SIP changes
-----------
+ * Added a new 'faxdetect=yes|no' configuration option to sip.conf. When this
+ option is enabled, Asterisk will watch for a CNG tone in the incoming audio
+ for a received call. If it is detected, the channel will jump to the
+ 'fax' extension in the dialplan.
* Improved NAT and STUN support.
chan_sip now can use port numbers in bindaddr, externip and externhost
options, as well as contact a STUN server to detect its external address