diff --git a/NOTES b/NOTES
deleted file mode 100644
index f4e6dfa..0000000
--- a/NOTES
+++ /dev/null
@@ -1,19 +0,0 @@
-Fixes in v1.5.2:
-[FIXED] Fixed issue #10.
-[FIXED] Fixed documentation typos.
-
-Fixes in v1.5.1:
-Fixed issue #2 pointed out by rushant001.
-
-New in v1.5.0:
-Fixed bug where pyVoIP would accept all codecs proposed by the server even if not compatible.  Will now only accept PCMU, PCMA, and telephone-event.
-Added handling of Native Bridging tested with Asterisk 16 SIP re-invite (External RTP bridge), this seems to still have issues with Asterisk 18, but unsure if it's my hardphone.
-Changed the audio read function in RTP to return b'\x80'*length instead of bytes(length), doing so stops the popping on the client side when no audio is being written.
-Fixed issue with ending phone calls originated by user.
-Added handling of 404 Not Found and 503 Service Unavailable errors.
-Added compatibility with Asterisk PJSIP.
-Fixed bug with multithreaded calling.
-
-Upcoming patches/changes:
-Add support for CANCEL requests.
-Add support for 603 Decline Responses
diff --git a/docs/conf.py b/docs/conf.py
index a6883e5..2e18be0 100644
--- a/docs/conf.py
+++ b/docs/conf.py
@@ -24,7 +24,7 @@
 author = 'Tayler J Porter'
 
 # The full version, including alpha/beta/rc tags
-release = '1.5.2'
+release = '1.5.3'
 
 master_doc = 'index'
 
diff --git a/pyVoIP/__init__.py b/pyVoIP/__init__.py
index d9969a6..bc9990b 100644
--- a/pyVoIP/__init__.py
+++ b/pyVoIP/__init__.py
@@ -1,7 +1,7 @@
 
 __all__ = ['SIP', 'RTP', 'VoIP']
 
-version_info = (1, 5, 2)
+version_info = (1, 5, 3)
 
 __version__ = ".".join([str(x) for x in version_info])
 
diff --git a/setup.py b/setup.py
index 4111e00..2d5dc14 100644
--- a/setup.py
+++ b/setup.py
@@ -7,7 +7,7 @@
 
 setup(
     name='pyVoIP',
-    version='1.5.2',
+    version='1.5.3',
     description='PyVoIP is a pure python VoIP/SIP/RTP library.',
     long_description=long_description,
     long_description_content_type="text/markdown",