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PortSIP PBX for Unified Communications

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  2. Hello, We'd like to use a Freeswitch server as SIP communication server connecting to our mobile client apps directly. Regarding to your web site Freeswitch is a supported server component to your VOIP SDK. In our test environment we faced the following problem. After the successful connection between the two mobile clients and the call initialized the call is disconnected after a certain timeout. After examining the Freeswitch output and the Freeswitch documentation the cause was revealed: Aduio connection initialization failure. Freeswitch needs a large open port range (16384-32768 UDP). With the server located in a protected NAT environment forwarding this range to one server is not a suitable solution. Could you provide us a safe and working configuration example of that setup? Currently the only goal is to let two clients from the public internet to connect and speak with each other. (We also see that Freeswitch has NAT traversal options using a STUN server, however configuring a STUN server is not straightforward to us at the moment. If this is the only way: Could you help us with information on STUN configuration?) Thanks in advance! Regards, Kristof
  3. Hello Joe, thank you for your help.. you were correct.. the port was 5061 and then it all started to work !! Kind Regards, Tom
  4. SIPSample can choose the Transport to TLS. Please check your TLS port is correct. Please give me anSIP account on your server, i will use test it.
  5. The sample application doesn't have any TLS options at all. Just a Username, Password, SiIPServer SIPServerPort and then 3 buttons saying "Online" "Offline" and "Quit" is there a newer sample project ?
  6. If you run our sample projct and test with TLS, does it work fine ?
  7. You should check your server, the callback events depend on the messages which received from server.
  8. VoIP SDK for Android: I can connect (and make calls) using UDP and non encrypted. When trying to register to a server that supports TLS ( 3cx one.. so i changed the user agent) In the logfile i don't see the transport=TLS on the REGISTER sip packet (from the PortSip logfiles on the Android Device) and the registration fails, in a way that nothing is returned because the server enforces TLS. REGISTER sip:tomschuring.3cx.com.au SIP/2.0 Via: SIP/2.0/ ;branch=z9hG4bK-524287-1---909aab6838f68476;rport Max-Forwards: 70 Contact: <sip:0001>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-000000000000>" To: "0001"<sip:0001@tomschuring.3cx.com.au> From: "0001"<sip:0001@tomschuring.3cx.com.au>;tag=ff043c28 Call-ID: QtMyUk_cOQMsiRoZBRn1wQ.. CSeq: 1 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, REGISTER, SUBSCRIBE, INFO, PUBLISH Supported: replaces, answermode, eventlist, outbound, path User-Agent: Mti Voip Android Allow-Events: hold, talk, conference Content-Length: 0 doing the same via zoiper (which works) (logfile from the 3cx logging): REGISTER sip:tomschuring.3cx.com.au;transport=TLS SIP/2.0 Via: SIP/2.0/TLS 192.168.2.43:64452;branch=z9hG4bK-524287-1---2515daffa981f4f2;rport=43265;received=58.96.100.67 Max-Forwards: 70 Contact: <sip:0001@58.96.100.67:43265;transport=TLS;rinstance=50884b26a10338eb> To: <sip:0001@tomschuring.3cx.com.au;transport=TLS> From: <sip:0001@tomschuring.3cx.com.au;transport=TLS>;tag=91720029 Call-ID: W0DXHb3dRylXW3Lr-pV-YA.. CSeq: 2 REGISTER Expires: 60 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Proxy-Authorization: Digest username="nt6rfTwEKY",realm="3CXPhoneSystem",nonce="414d53595e3a14de29:36416f59986eecbd253ea1b31aef35ea",uri="sip:tomschuring.3cx.com.au;transport=TLS",response="1c48ab0ab1c43ea42a01f047108baeb4",algorithm=MD5 User-Agent: Zoiper rv2.10.6.2 Allow-Events: presence, kpml, talk Content-Length: 0 To try to switch on TLS i Initialize like : instance.mEngine.initialize(TRANSPORT_TYPE.TRANSPORT_TLS, "0.0.0.0", 5060, PORTSIP_LOG_LEVEL.PORTSIP_LOG_DEBUG, GetExternalFilesDir(null).AbsolutePath, 1, "Mti Voip Android", 0, 0, "", "", false); instance.mEngine.setSrtpPolicy(SRTP_POLICY.SRTP_POLICY_FORCE); instance.mEngine.setLicenseKey("REDACTED"); instance.mEngine.enable3GppTags(false); but the REGISTER isn't using the transport=TLS ? What else do i need to do to get it to work ?
  9. Hi, in our case the onInviteFailure was not called, only the two events described above. It's a bit strange anyway why the onInviteConnected was called in the first place when an explicit reject was answered?
  10. Hi, there also has the onInviteFailure callback event.
  11. Hello, We're just about to use PortSIP SDK in our new mobile applications (Android and iOS). We faced the following problem in our POC phase: We cannot identify the positive/negative answer from the called client in the caller app. In our caller application (and in the downloadable portsip client app as well) the same onInviteAnswered and onInviteConnected events are triggered on answering and rejecting the call. The only difference in the two scenarios is a tiny difference in the sipMessage: On "answered" case the \"Outbound Call\" string value appears in the sip message in a certain place. When the called number was "offline": application/sdp\r\n appears. The reject and the hangup cannot be identified from each other. Could you please help me with some details of the correct call-handling? Thanks in advance! Regards, Kristof
  12. I have forward this requirement to our R & D, we will add this feature in future version. Thanks for feedback.
  13. Yes, it's already support the bluetooth device.
  14. Yes, current it supports auto provision with our PortSIP PBX
  15. Is it possible do auto provisioning the PortSIP app?
  16. Is it possible answer a call of PortSip APP using the Bluetooth device?
  17. Hello, It is possible to enable the keypad during the early media (pre-answer call). In our PBX the user must to interact with IVR system during this phase.
  18. Hi, please follow this guide: https://www.portsip.com/knowledge-base/portsp-pbx-high-availability/
  19. How to build redundancy with docker ?
  20. I am working as VM Ware Engineer at Fieldengineer.com which is an On-Demand Marketplace which connects businesses and freelance field engineers in telecom industry.
     The Fastest Growing Start-up with a Global Workforce in 175 Countries. We have currently 40,000 engineers signed up worldwide.

  21. Hi James, the v12 for Linux new version was fixed this bug. Please follow below steps to update: Your currently data will be keep, don't forgot replace the IP to your own IP.
  22. Thank you Hormiga2020. If your server only has the public IP but no private IP, you can simply install the PBX on a public network, then enter the public IP in the PBX setup wizard step1 for both Private IPv4 and Public IPv4, then it will works. BR
  23. Hello, Thank you for the new version of PortSIP. It is a big step ahead. I want use PortSIP with an public IP only. So I need register the ip-phones via public network and the outgoing traffic via SIP-trunk at the same public ip-address. a. can I use the internal SBC as service-based firewall for security issues with individual ruleset? b. in which way I can configure the RTP-traffic of the connected phones and the SIP-trunks? The documentation shows in chapter 16. LAN-based configurations only. No WAN-WAN-config for public phones and SIP-trunks was shown. Is a howto for such architecture available? Regards hormiga2020
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