Jan 22, 2008 Good Evening, I have searched the forum and can see that some people have managed to get SIP TAPI to work with 3CX. I have downloaded SIP TAPI and. SIPTAPI is a SIP based call-control client which can be used to initiate phone calls. SIPTAPI is not a full SIP client. SIP TAPI Service Provider Configure. Apr 01, 2016 I need a work around for Windows 8 not being able to add siptapi.tsp files. When I attempt this then restart and go to phone dialer advanced to find it, it. Xtelsio TAPI Driver for Asterisk™ supports dialing and hanging up and informs your TAPI application about incoming and outgoing calls including the caller and.

Good Evening, I have searched the forum and can see that some people have managed to get SIP TAPI to work with 3CX. I have downloaded SIP TAPI and installed/configured it, when I make a call my SIP phone rings however when I pick up the phone the call disconnects. Below are the logs. 20:52:15.296 Call::Terminate [CM503008]: Call(7): Call is terminated 20:52:15.296 Call::Terminate [CM503008]: Call(7): Call is terminated 20:52:15.156 Call::RouteFailed [CM503014]: Call(7): Attempt to reach [sip:07852oooooo@192.168.16.2] failed. Reason: Not Found 20:52:15.109 CallCtrl: nSelectRouteReq [CM503013]: Call(7): No known route to target: [sip:07852oooooo@192.168.16.2] 20:52:13.984 CallCtrl: nLegConnected [CM503007]: Call(7): Device joined: sip:211@192.168.

20:52:13.984 CallCtrl: nLegConnected [CM503007]: Call(7): Device joined: sip:211@192.168. 20:52:10.593 CallCtrl: nSelectRouteReq [CM503004]: Call(7): Calling: Shared:211@[Dev:sip:211@192.168., Dev:sip:211@192.168.] 20:52:10.546 CallCtrl: nIncomingCall [CM503001]: Call(7): Incoming call from Ext.211 to [sip:211@192.168.16. How To Edit A Patch Mw2. 2] number blanked out for obvious reasons. I am using free version of 3CX ver 5 If anyone knows how to configure SIP TAPI I would appreciate any help. Hi, I cannot turn off my firewall, it is built in to my router but i did put the 3CX server into a DMZ and still no effect. Not sure how to change codecs phones always seem to use G711u. Here is a snippet from the logs when the call is made. 19:42:27.250 Call::Terminate [CM503008]: Call(23): Call is terminated 19:42:01.921 Line: rintEndpointInfo [CM505003]: Provider:[SIPgate] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [] Transport: [sip:192.168.16.2:5060] 19:41:54.671 CallCtrl: nSelectRouteReq [CM503004]: Call(23): Calling: VoIPline:10000@[Dev:sip:2988075@sipgate.co.uk:5060, Dev:sip:2988075@sipgate.co.uk:5060, Dev:sip:2988075@sipgate.co.uk:5060] 19:41:53.390 CallCtrl: nLegConnected [CM503007]: Call(23): Device joined: sip:211@192.168.

19:41:53.375 CallCtrl: nLegConnected [CM503007]: Call(23): Device joined: sip:211@192.168. 19:41:50.703 CallCtrl: nSelectRouteReq [CM503004]: Call(23): Calling: Shared:211@[Dev:sip:211@192.168., Dev:sip:211@192.168.] 19:41:50.656 CallCtrl: nIncomingCall [CM503001]: Call(23): Incoming call from Ext.211 to [sip:211@192.168.16.2] Hope this helps. Do you use v5 build 3790? If not try backup and then uninstall and install latest an restore or try to install over. If not there was a problem with transfer and no audio. Build version 5.0.3790 8 January 2008 New Features Music on hold when transferring from Digital receptionist.