pyVoIP 1.5.1

Fixed issue tayler6000#2 pointed out by rushant001.

NOTES

@@ -1,3 +1,6 @@
+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.

docs/VoIP.rst

@@ -135,5 +135,5 @@ The VoIPPhone class is used to manage the :ref:`SIPClient` class and create :ref
     This method ends all currently ongoing calls, then stops the :ref:`SIPClient` class
 
   **call**\ (number)
-    Originates a call using PCMU and telephone-event. The *number* argument must be a string, and it returns a :ref:`VoIPCall` class in CallState.DIALING.  You should use a while loop to wait until the CallState is ANSWRED. **NOTE:** In testing with Asterisk 13, calls made this way could not hangup.  This issue may exist on other PBXs as well.
-
+    Originates a call using PCMU and telephone-event. The *number* argument must be a string, and it returns a :ref:`VoIPCall` class in CallState.DIALING.  You should use a while loop to wait until the CallState is ANSWRED.
+

pyVoIP/SIP.py

@@ -626,7 +626,7 @@ def genRegister(self, request):
     regRequest += self.username+"@"+self.myIP+":"+str(self.myPort)
     regRequest += ";transport=UDP>\r\nTo: <sip:"""+self.username+"@"
     regRequest += self.server+";transport=UDP>\r\nFrom: <sip:"+self.username
-    regRequest += "@192.168.0.102;transport=UDP>;tag="+self.genTag()
+    regRequest += "@"+self.server+";transport=UDP>;tag="+self.genTag()
     regRequest += "\r\nCall-ID: "+request.headers['Call-ID']
     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"
@@ -841,7 +841,7 @@ def bye(self, request):
 
   def deregister(self):
     self.recvLock.acquire()
-    fake = SIPMessage(b'SIP/2.0 401 Unauthorized\r\nVia: SIP/2.0/UDP 192.168.0.64:5060;received=192.168.0.64\r\nFrom: <sip:5555555@192.168.0.102;transport=UDP>;tag=b4dbea69\r\nTo: <sip:5555555@192.168.0.102;transport=UDP>;tag=as6845844a\r\nCall-ID: '+self.genCallID().encode('utf8')+b'\r\nCSeq: 25273 REGISTER\r\nServer: Asterisk PBX 16.2.1~dfsg-1+deb10u1\r\nAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE\r\nSupported: replaces, timer\r\nWWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7140386d"\r\nContent-Length: 0\r\n\r\n')
+    fake = SIPMessage(b'SIP/2.0 401 Unauthorized\r\nVia: SIP/2.0/UDP 192.168.0.64:5060;received=192.168.0.64\r\nFrom: <sip:5555555@127.0.0.1;transport=UDP>;tag=b4dbea69\r\nTo: <sip:5555555@127.0.0.1;transport=UDP>;tag=as6845844a\r\nCall-ID: '+self.genCallID().encode('utf8')+b'\r\nCSeq: 25273 REGISTER\r\nServer: Asterisk PBX 16.2.1~dfsg-1+deb10u1\r\nAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE\r\nSupported: replaces, timer\r\nWWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7140386d"\r\nContent-Length: 0\r\n\r\n')
 
     regRequest = self.genRegister(fake).replace('Expires: 300', 'Expires: 0')
 
@@ -870,7 +870,7 @@ def deregister(self):
 
   def register(self):
     self.recvLock.acquire()
-    fake = SIPMessage(b'SIP/2.0 401 Unauthorized\r\nVia: SIP/2.0/UDP 192.168.0.64:5060;received=192.168.0.64\r\nFrom: <sip:5555555@192.168.0.102;transport=UDP>;tag=b4dbea69\r\nTo: <sip:5555555@192.168.0.102;transport=UDP>;tag=as6845844a\r\nCall-ID: '+self.genCallID().encode('utf8')+b'\r\nCSeq: 25273 REGISTER\r\nServer: Asterisk PBX 16.2.1~dfsg-1+deb10u1\r\nAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE\r\nSupported: replaces, timer\r\nWWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7140386d"\r\nContent-Length: 0\r\n\r\n')
+    fake = SIPMessage(b'SIP/2.0 401 Unauthorized\r\nVia: SIP/2.0/UDP 192.168.0.64:5060;received=192.168.0.64\r\nFrom: <sip:5555555@127.0.0.1;transport=UDP>;tag=b4dbea69\r\nTo: <sip:5555555@127.0.0.1;transport=UDP>;tag=as6845844a\r\nCall-ID: '+self.genCallID().encode('utf8')+b'\r\nCSeq: 25273 REGISTER\r\nServer: Asterisk PBX 16.2.1~dfsg-1+deb10u1\r\nAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE\r\nSupported: replaces, timer\r\nWWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7140386d"\r\nContent-Length: 0\r\n\r\n')
 
     regRequest = self.genRegister(fake)
 
@@ -906,4 +906,4 @@ def register(self):
       raise InvalidAccountInfoError("Invalid Username or Password for SIP server "+self.server+':'+str(self.myPort))
 
 
-  
+

pyVoIP/__init__.py

@@ -1,7 +1,7 @@
 
 __all__ = ['SIP', 'RTP', 'VoIP']
 
-version_info = (1, 5, 0)
+version_info = (1, 5, 1)
 
 __version__ = ".".join([str(x) for x in version_info])
 
@@ -19,4 +19,5 @@ def debug(s, e=None):
 SIPCompatibleVersions = ['SIP/2.0']
 
 RTPCompatibleVersions = [2]
-RTPCompatibleCodecs = [PayloadType.PCMU, PayloadType.PCMA, PayloadType.EVENT]
+RTPCompatibleCodecs = [PayloadType.PCMU, PayloadType.PCMA, PayloadType.EVENT]
+

setup.py

@@ -3,7 +3,7 @@
 
 setup(
   name = 'pyVoIP',
-  version = '1.5.0',
+  version = '1.5.1',
   description = 'PyVoIP is a pure python VoIP/SIP/RTP library.',
   author = 'Tayler Porter',
   author_email = '[email protected]',