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Can't place calls using SipDroid

Discussion in '3CX Phone System - General' started by BenYL, Jan 25, 2010.

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  1. BenYL

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    Has anyone had luck using the free 3CX phone system with SipDroid, a VoIP client program for Android phones? In general it works great for me, but for some reason I cannot place calls. The 3CX server does not recognize the extensions when I type them in. I am calling over WiFi only.

    Thanks!
    Ben
     
  2. mylove4life

    mylove4life New Member

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    Same here, I can get it to work sometimes tho...
     
  3. Nick Galea

    Nick Galea Site Admin

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    We are interested to help troubleshoot this. Can you post some logs? I.e when you are not able to call can you check the logs of 3CX Phone System and post them here? Make sure that you can still reach the phone system server when that happens...
     
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  4. BenYL

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    Thanks! Simply nothing shows up on the Server Activity Log when I try to place a call to another extension. The extensions I have setup are things like '111' and '222'. I'll try to install WireShark when I can to see what packets are going back and forth. SipDroid is the most popular SIP client on Android phones, so it would be good to get this working.
     
  5. Nick Galea

    Nick Galea Site Admin

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    Hi Ben

    Thanks for the quick reply. Yes wireshark would be great. If you prefer to email me the capture this is also possible at n ng at 3cx.com

    However, if the server status logs show nothing it might be that we are not receiving any incoming traffic at all, in which case it would point to a problem with the client or the network connection and then there would be little we can do. But lets check out the logs and we will see.

    If there is something we can do we will do it. An open standard mobile voip client and platform is something we have been waiting for for a long time! (Apple with appstore is limited and windows mobile does not always encourage SIP because of operators)

    So lets hope we can get it to work. I assume this is the latest Google Android platform? Please let us know versions if possible.
     
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  6. jelrod

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    I am having the same problem the sip client receives calls with no problem and even after enabling verbose logging on the server it doesn't show the outbound call as being initiated any help will be great here.

    Thanks,

    Jeremy
     
  7. archie

    archie Well-Known Member
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    It would be much easier to help if we had some wireshark captures or verbose logs.
    According to some I saw on other forums in internet I can definitely say that SipDroid doesn't follow SIP standards. It specifies 127.0.0.1 as a Contact IP. Imagine, what happens if SIP server will send a request to that IP? The same problem is in SDP. Though we have several workarounds already implemented, we can not guarantee correct interoperability until correct IPs are provided.
     
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  8. anto67b

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    Hi, does anyone find out why we can't place call with sipdroid?
    I'm using nexus one vers 2.1update1, sipdroid 1.2.4 beta.
    i can receive calls, but when i try to place calls nothing happens. even if i call another internal phone.
    pls see the log from 3cx, it seems there is some authentication problems..ps i took of the part after @ (my domain name..)
    14:42:02.861 [CM302001]: Authorization system can not identify source of: SipReq: SUBSCRIBE 102@
    14:41:58.885 [CM302001]: Authorization system can not identify source of: SipReq: SUBSCRIBE 102@
    14:41:22.475 [CM302001]: Authorization system can not identify source of: SipReq: SUBSCRIBE 102@
    14:41:21.714 [CM500002]: Unidentified incoming call. Review INVITE and adjust source identification:
    INVITE sip:100@ SIP/2.0
    Via: SIP/2.0/UDP 192.168.2.70:39959;rport=39959;branch=z9hG4bK83410
    Max-Forwards: 70
    Contact: <sip:102@192.168.2.70:39959>
    To: <sip:100@>
    From: <sip:102@>;tag=z9hG4bK50255206
    Call-ID: 565904817776@192.168.2.70
    CSeq: 1 INVITE
    Expires: 3600
    User-Agent: Sipdroid/1.2.4 beta/Nexus One
    Content-Length: 0
     
  9. Nick Galea

    Nick Galea Site Admin

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    Try switching on the 'PBXdeliversaudio' option for that Extension in the Other tab, Extension capabilities.

    This will proxy all RTP via the PBX and will resolve as many SIP error made by the phone as possible.
     
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