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Google Voice/Gizmo5

Discussion in '3CX Phone System - General' started by kminard, Apr 26, 2010.

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  1. kminard

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    Google now owns Gizmo5. One of the options in Google Vocie is to route inbound Google Voice calls to Gizm5 for a SIP connection. I have successfully built a GIZMO5 SIP connection into 3CX, but calls will not complete. Calls ring into 3CX from Gizmo5, but they have no audio when answered and do not register with Google Voice/Gizmo5 that they have been answered. I have read in the forum others that have the same problem but in all the postings there has never been a reply about what is missing to get 3CX to work with GIZMO5 inbound calls to 3CX. Trixbox and other virtual PBX platforms seem to work fine so this seem a specific problem only for the 3CX platform. There are posts all over the web about this deficiency but never any discussion on a solution.

    Has anyone every gotten this to work and, if so, what is the magic?

    Thanks!
     
  2. leejor

    leejor Well-Known Member

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    The original problem I had with Gizmo/SipPhone.com was that they had barred 3CX from actually registering. After having changed the User Agent, to something they liked, it worked, and continues to work well for me, pretty straight forward options. I don't know if they are using a different server for your particular service.

    I don't know if this is related to the problems you are having..http://www.3cx.com/forums/help-with-sipphone-com-not-registering-3254.html
     
  3. kminard

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    Thanks for the reply. I am not having any issue with registering like the previous post suggest. GIZMO5 (sipphone.com) registers fine and when a call comes in the extensions evne ring, but when answered there is no audio. Also, when using Google Voice to directly call into 3CX via GIZMO5, GIZMO5/Google Voice does not indicate the call has been answered so somethign is wrong with the handshake between 3CX and GIZM5. Reading hte various posts, this seems to be a common problem and no one has ever been able to answer it. Registration completes successfully and 3CX reports the SIP connection is registered and available and GREEN. I am nto using this SIP connection for outbound calls, just inbound from Googel Voice via GIZMO5 (sipphone.com). It woudl be great if someone has this working and would share the magic to get it to work for the rest of us.
     
  4. leejor

    leejor Well-Known Member

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    Maybe you could post the Wireshark of a couple of calls to see what sort of messaging is going on between the two.
     
  5. kminard

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    Any new info out ther on this? Anyone at all have Google Voice routing inbound calls via Gizmo5 to 3CX v8 or v9?
     
  6. carolinainnovative

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    Like leejor said - post a wireshark capture and lets see whats going on under the hood.
     
  7. kminard

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    I certainyl think a wireshark trace is appropraite at some point, but a bigger issue is that Gixmo5 has specifically blocked 3CX connections and no one else. Reading the various posts on this forum, various members have had to "spoof" the user agent field. So, my questions are above the "wireshark" level at this point and include why is the Gizmo5 blocking 3CX PBS connections and with that answer, once the user community has spoofed the "user agent" field, what are the other steps to connect to Gizmo5. I think this one has been tacked repeatedly some I'm looking for an answer from someone who had alreadt pounded thsi one out. It does not look like it specific to me... it looks like many many people have tackled this one, just no one has ever written an update once they got it working on what they did to get there. So, I'm hopipng someone who's been there and tackeled this can post a reply since it seems a common and well explored challenge out there... and even to the point of Gizmo5 blocking anything presenting itself as a 3CX PBX. All other software PBX providers seem in the clear and not hard blocked by Gizmo5.
     
  8. kminard

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    Attached is a Wireshark Trace during an attempt of Google Voice via Gizmo 5 to pass a call to a 3CX PBX and then pass to a Polycom 760 desk set. The call rings on the desk set but when answered there is no call completion by Gizmo 5 or 3CX. The initiating caller continues to hear ringing on the AT&T phoen circuit on the origination side even though the handset has been picked up and answered on the ringing Polycom side. At teh end of the test, the originating side just hung up after hearing continuous ringing. No voice communication is passed between the two ends other than the Polycom does ring when the call is initiated through 3CX. I changed the user Agent to WinGizmo so Gizmo5 would allow the 3CX application to connect. If the user agent presents itself as 3CX, Gizmo5 will not allow the association.
     
  9. kminard

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    Still looking for a qood solution to this problem. The issue still exists and by looking all over the web, seems to be specifically a 3CX issue. All the other software PBX suppliers seem to be supported. Even further, for some reason it seems 3CX is specifically blocked by Gizmo5 so most users have to change the USER AGENT to something else to get Gizmo5 to accept the connection.

    I've changed my user agent to a differnet PBX name and it now registers and stays registered. If a call comes in via Google Voice through Gizmo5, the calling party on teh PSTN side hears ringing. On the 3CX side, the call comes through and rings as shown below:

    ______________
    14:44:06.820 [MS105000] C:12.1: No RTP packets were received:remoteAddr=198.65.166.131:47706,extAddr=99.115.135.49:9008,localAddr=99.115.135.49:9008
    14:44:06.455 [CM503008]: Call(12): Call is terminated
    14:43:56.486 [CM503003]: Call(12): Call to sip:105@connect.sargon.com has failed; Cause: 487 Request Terminated; from IP:172.16.16.27:53705
    14:43:56.483 [CM503003]: Call(12): Call to sip:106@connect.sargon.com has failed; Cause: 487 Request Terminated; from IP:172.16.16.19:65404
    14:43:56.316 [CM503003]: Call(12): Call to sip:103@172.16.16.2 has failed; Cause: 487 Request Terminated; from IP:172.16.16.22:5060
    14:43:56.309 [CM503003]: Call(12): Call to sip:101@172.16.16.2 has failed; Cause: 487 Request Terminated; from IP:172.16.16.22:5060
    14:43:56.306 [CM503003]: Call(12): Call to sip:102@172.16.16.2 has failed; Cause: 487 Request Cancelled; from IP:172.16.16.23:49414
    14:43:56.256 [CM503007]: Call(12): Device joined: sip:100@172.16.16.34:5060
    14:43:56.254 [CM503007]: Call(12): Device joined: sip:17474928211@proxy01.sipphone.com:5060
    14:43:54.700 [CM503025]: Call(12): Calling RingAll800:100Ext.100101Ext.101103Ext.103102Ext.102105Ext.105106Ext.106107Ext.107@[Dev:sip:106@172.16.16.19:65404;transport=UDP]
    14:43:54.696 [CM503025]: Call(12): Calling RingAll800:100Ext.100101Ext.101103Ext.103102Ext.102105Ext.105106Ext.106107Ext.107@[Dev:sip:105@172.16.16.27:53705]
    14:43:54.692 [CM503025]: Call(12): Calling RingAll800:100Ext.100101Ext.101103Ext.103102Ext.102105Ext.105106Ext.106107Ext.107@[Dev:sip:102@172.16.16.23:5060;transport=udp]
    14:43:54.688 [CM503025]: Call(12): Calling RingAll800:100Ext.100101Ext.101103Ext.103102Ext.102105Ext.105106Ext.106107Ext.107@[Dev:sip:103@172.16.16.22:5060;line=65366]
    14:43:54.684 [CM503025]: Call(12): Calling RingAll800:100Ext.100101Ext.101103Ext.103102Ext.102105Ext.105106Ext.106107Ext.107@[Dev:sip:101@172.16.16.22:5060;line=3108]
    14:43:54.680 [CM503025]: Call(12): Calling RingAll800:100Ext.100101Ext.101103Ext.103102Ext.102105Ext.105106Ext.106107Ext.107@[Dev:sip:100@172.16.16.34:5060]
    14:43:54.650 [CM503004]: Call(12): Route 1: RingAll800:100Ext.100101Ext.101103Ext.103102Ext.102105Ext.105106Ext.106107Ext.107@[Dev:sip:100@172.16.16.34:5060,Dev:sip:101@172.16.16.22:5060;line=3108,Dev:sip:103@172.16.16.22:5060;line=65366,Dev:sip:102@172.16.16.23:5060;transport=udp,Dev:sip:105@172.16.16.27:53705,Dev:sip:106@172.16.16.19:65404;transport=UDP]
    14:43:54.602 [CM505003]: Provider:[Gizmo5] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [YATE/3.0.0] PBX contact: [sip:17474928211@99.115.135.49:5060]
    14:43:54.600 [CM503001]: Call(12): Incoming call from +14152927014@(Ln.10004@Gizmo5) to <sip:800@172.16.16.2:5060>
    14:43:54.379 [CM503012]: Inbound out-of-office hours rule (unnamed) for 10004 forwards to DN:800
    14:43:51.996 [CM506001]: STUN request to resolve SIP external IP:port mapping is sent to STUN server 96.9.132.83:3478 over Transport 172.16.16.2:5060
    14:43:25.257 [CM504004]: Registration succeeded for: 10004@Gizmo5
    14:43:25.064 [CM504003]: Sent registration request for 10004@Gizmo5
    _____________________

    at teh receiving side, the phone rings and when answered, the call is dead.. no soudn travels in either direction and the calling party on the PSTN contiues to hear ringing during this entire procedure, including when the handset on the 3CX side is lifted. The calling party on the PSTN side must eventually hang up to terminate the call.

    A wireshark of the this entire loop is posted in the previous forum entry.

    Any ideas... if I could get this working then anyone using Google Voice coudl accept the forward/screened calls into and through the 3CX server via SIP the entire way. As it is today the calls must go from Googel Voice (SIP) out to the PSTN and then back to 3Cx via the PSTN. What a waste when the call originated as SIP initially.

    Hope someone has a clue. I am using the template from the 3CX for Gizmo5 OpenSky from the library of 3rd party suppliers. I have associates that use 3CX on both various version of V8 and V9 and all have hte same issues I am having so it must be something commong to 3CX or some sensitivity to a common 3CX setting I'm guessing.
     

    Attached Files:

  10. djvpuls

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    The problem still exsists on the 3CX version 5... I like the 3CX phone and don't want to switch (It works great for outgoing calls), but I'd like to be able to answer my gizmo5 calls that are forwarded by google voice. Please fix this problem..
     
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