[FIXED] Fixed issue tayler6000#10.

[FIXED] Fixed documentation typos.

docs/RTP.rst

@@ -30,10 +30,10 @@ RTP.\ **RTPProtocol**
     This means the audio should be sent with pure UDP.  Returns ``'udp'`` when string casted.
 
   RTPProtocol.\ **AVP**
-    This means the audio should be sent with RTP Audio/Video Protocol described in :rfc:3551. Returns ``'RTP/AVP'`` when string casted.
+    This means the audio should be sent with RTP Audio/Video Protocol described in :RFC:`3551`. Returns ``'RTP/AVP'`` when string casted.
 
   RTPProtocol.\ **SAVP**
-    This means the audio should be sent with RTP Secure Audio/Video Protocol described in :rfc:3711. Returns ``'RTP/SAVP'`` when string casted.
+    This means the audio should be sent with RTP Secure Audio/Video Protocol described in :RFC:`3711`. Returns ``'RTP/SAVP'`` when string casted.
 
 .. _transmittype:
 
@@ -226,4 +226,4 @@ The RTPClient is used to send and receive RTP packets and encode/decode the audi
 
   **parseTelephoneEvent**\ (packet)
     This method is called by parsePacket().  It will decode the *packet*'s payload from the telephone-event non-codec to the string representation of the event.  It will then call :ref:`VoIPCall`.dtmfCallback().
-
+

docs/VoIP.rst

@@ -24,7 +24,7 @@ Enums
 VoIP.\ **CallState**
   CallState is an Enum with four attributes.
 
-  CallState.\ **DIALING***
+  CallState.\ **DIALING**
     This CallState is used to describe when a :term:`user` has originated a call to a :term:`client`, but it has yet to be answered.
 
   CallState.\ **RINGING**

docs/conf.py

@@ -20,11 +20,11 @@
 # -- Project information -----------------------------------------------------
 
 project = 'pyVoIP'
-copyright = '2020, Sorta Strange Productions'
+copyright = '2021, Tayler Porter'
 author = 'Tayler J Porter'
 
 # The full version, including alpha/beta/rc tags
-release = '0.5.0 Beta'
+release = '1.5.2'
 
 master_doc = 'index'
 
@@ -58,4 +58,4 @@
 # Add any paths that contain custom static files (such as style sheets) here,
 # relative to this directory. They are copied after the builtin static files,
 # so a file named "default.css" will overwrite the builtin "default.css".
-html_static_path = ['_static']
+html_static_path = ['_static']

pyVoIP/SIP.py

@@ -619,6 +619,7 @@ def genAuthorization(self, request):
   def genRegister(self, request):
     response = self.genAuthorization(request)
     nonce = request.authentication['nonce']
+    realm = request.authentication['realm']
 
     regRequest = "REGISTER sip:"+self.server
     regRequest += " SIP/2.0\r\nVia: SIP/2.0/UDP "+self.myIP+":"+str(self.myPort)
@@ -631,7 +632,7 @@ def genRegister(self, request):
     regRequest += "\r\nCSeq: "+str(self.registerCounter.next())+" REGISTER"
     regRequest +="\r\nExpires: 300\r\nAllow: "+(", ".join(pyVoIP.SIPCompatibleMethods))+"\r\nUser-Agent: pyVoIP """+pyVoIP.__version__+"\r\n"
     regRequest += 'Authorization: Digest username="'+self.username
-    regRequest += '",realm="asterisk",nonce="'+nonce
+    regRequest += '",realm="+realm+",nonce="'+nonce
     regRequest += '",uri="sip:'+self.server
     regRequest += ';transport=UDP",response="'+str(response, 'utf8')
     regRequest += '",algorithm=MD5\r\n'+"Content-Length: 0\r\n\r\n"
@@ -820,8 +821,9 @@ def invite(self, number, ms, sendtype):
     debug("Acknowledged")
     authhash = self.genAuthorization(response)
     nonce = response.authentication['nonce']
+    realm = response.authentication['realm']
     auth = 'Authorization: Digest username="'+self.username
-    auth += '",realm="asterisk",nonce="'+nonce
+    auth += '",realm="+realm+",nonce="'+nonce
     auth += '",uri="sip:'+self.server
     auth += ';transport=UDP",response="'+str(authhash, 'utf8')
     auth += '",algorithm=MD5\r\n'

pyVoIP/__init__.py

@@ -1,7 +1,7 @@
 
 __all__ = ['SIP', 'RTP', 'VoIP']
 
-version_info = (1, 5, 1)
+version_info = (1, 5, 2)
 
 __version__ = ".".join([str(x) for x in version_info])
 

setup.py

@@ -3,7 +3,7 @@
 
 setup(
   name = 'pyVoIP',
-  version = '1.5.1',
+  version = '1.5.2',
   description = 'PyVoIP is a pure python VoIP/SIP/RTP library.',
   author = 'Tayler Porter',
   author_email = '[email protected]',