Phonzo voip / sip provider

Discussion in '3CX Phone System - General' started by opera01, Apr 2, 2009.

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  1. opera01

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    Hi!

    I need some help configuring my 3CX PBX with a provider (http://www.phonzo.com) with theese settings:

    Proxy address: sip.phonzo.com (eller unntaksvis sip.phonzo.com:5060)
    Domain/realm: sip.phonzo.com
    Outbound proxy address: sip.phonzo.com (ikke alle trenger denne)
    Port: 5060
    Codec:
    G711 a-law (høy bitrate - ukomprimert lyd)
    G729 (lav bitrate - komprimert lyd)

    STUN: Phonzo bruker ikke STUN eller ICE server (NO STUN / ICE)

    Incoming phone calls will not work. The 3CX configuration pages shows my provider is Green and "(Idle)". Havent tryed out going calls, because I gonna use this PBX as a digital receptionist. May be some problems with port forwarding...? But is it also some voip / sip provider configuration I need to be aware of?
     
  2. nb

    nb Support Team
    Staff Member 3CX Support

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    Put the logs in verbose mode, from the Advanced section in settings and restart the 3CX Phonesystem service.

    Then wait for the provider to register and make an incoming call.

    Check the server activity log. What do you see? Since it is an unsupported provider check for the following tips : Unidentified source - review invite and adjust source identification. paste the invite here and we will see it.

    Maybe this provider needs some source Identification rules. i will guide you on this but I need to see an invite that fails.
     
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  3. alexeena

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    What are the advantages of VOIP over a normal home telephone system? I haven't got much of an idea what I'm looking at, I need to buy a new phone for the house but I've seen some which are compatible with VOIP but are expensive, is it worth the extra money. Any advice on the whole system would be great.
    ______________________
    market samurai ~ marketsamurai ~ marketsamurai.com
     
  4. nb

    nb Support Team
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    What do you need exactly but?

    You want a pbx?
    1 phone you are going to have only?
    This is for home use right? how many users? These are the questions you need to ask before.
    how much do you currently pay for tel bills
    How much will you pay for a Voip provider + internet Will this option be cheaper?

    Because if this is the scenario I would use a good wireless headset or even DECT phone. Something that you can take around the house.

    If you want a cheap solution - ATA gateway, buy a cheap analogue chordless phone and plug them in, register to pbx and forward calls to the ATA. And you have Chordless telephony around the house too. If you find an ATA with 2 ports (common) you can plug 2 chordless phones and leave one upstairs and the other downstairs example. in this case you create a ring group with RING ALL Option - So when someone rings both phones ring. the options are many.

    Depends what you need - Make some research first - see what others are using - then decide what you want and after you are sure, I will help you giving what in my opinion are good devices. First make some home work and answer the questions above.
     
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  5. opera01

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    Thank you for quick respons! I will paste my log, as soon as I get access to my 3CX later today.
     
  6. opera01

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    Here is my log. It doesnt seems like it register incoming calls..??:

    18:03:25.687 [CM504008]: Fax Service: registered as sip:888@192.168.19.27:5060 with contact sip:888@192.168.19.27:5100;user=phone

    18:02:14.265 [CM504004]: Registration succeeded for: 10000@phonzo

    18:02:13.812 [CM504003]: Sent registration request for 10000@phonzo

    18:02:13.781 Unknown system [CFGManager] tries to connect!

    18:02:13.781 SL: connected NOxxx:0/CFGManager at [NOxxx]/CFGManager

    17:58:50.625 [CM504001]: Ext.800: new contact is registered. Contact(s): [sip:800@127.0.0.1:40600;rinstance=dc176101e954b06d/800]

    17:58:50.515 [CM504001]: Ext.801: new contact is registered. Contact(s): [sip:801@127.0.0.1:40600;rinstance=f16ea8e98203f825/801]

    17:58:50.515 [CM504001]: Ext.999: new contact is registered. Contact(s): [sip:999@127.0.0.1:40600;rinstance=740b26cbf9098d56/999]

    17:58:34.156 [CM504004]: Registration succeeded for: 10000@phonzo

    17:58:32.546 [CM504003]: Sent registration request for 10000@phonzo

    17:58:32.437 IP(s) added:[192.168.19.27]

    17:58:31.187 [CM504001]: Ext.*1: new contact is registered. Contact(s): [sip:*1@127.0.0.1:40000;rinstance=8103df22eeb53797/*1]

    17:58:31.171 [CM504001]: Ext.*0: new contact is registered. Contact(s): [sip:*0@127.0.0.1:40000;rinstance=96455f67237f94fb/*0]

    17:58:25.546 [CM504008]: Fax Service: registered as sip:888@192.168.19.27:5060 with contact sip:888@192.168.19.27:5100;user=phone

    17:58:25.234 [CM504004]: Registration succeeded for: 10000@phonzo

    17:58:24.937 [CM504003]: Sent registration request for 10000@phonzo

    17:58:23.937 SL: connected NOxxx:0/3CXConferenceRoom at [NOxxx]/3CXConferenceRoom

    17:58:23.812 SL: connected NOxxx:0/IVRServer at [NOxxx]/IVRServer

    17:58:23.640 SL: connected NOxxx:0/3CXParkOrbit at [NOxxx]/3CXParkOrbit

    17:58:23.312 [CM192000] Media Server is connected

    17:58:23.312 SL: connected NOxxx:0/MediaServer at [NOxxx]/MediaServer

    17:58:22.921 [CM506005]: Public IP=80.123.123.123 is used for WAN communications through local interface with IP=192.168.19.27

    17:58:22.921 [CM501006]: Default Local IP address: [192.168.19.27]

    17:58:22.390 [CM501002]: Version: 7.1.6274.0

    17:58:22.390 [CM501001]: Start 3CX PhoneSystem Call Manager

    17:58:22.390 [CM501007]: *** Started Calls Controller thread ***

    17:58:22.203 Unknown system [DBProvider] tries to connect!

    17:58:22.203 SL: connected NOxxx:5485/DBProvider at [NOxxx]/DBProvider
    17:58:22.187 [CM501010]: License Info: Load Failed
     
  7. nb

    nb Support Team
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    Hi

    Are you sure you are in verbose mode? Put the logs in verbose mode and Make an incoming call.

    Also make a firewall check and send me the output.

    If you do this and after you make a call you see nothing, it means that the call is not coming to you. Make sure that it is not registered somewhere else as well. if it is constantly registering somewhere else, the call will go there.

    I can get this information if you provide me with a wireshark capture of a registration.

    Can you make an outgoing call?

    Also I will need full logs not just this - But first you have to reach 3CX. You have to see an invite coming in one way or another.
     
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  8. opera01

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    hmm... I dont understand. I have it in verbose mode. And I did reset the servers...

    Ive uploaded the logs.. Is it some more places to find logs than under "Server activity log"?

    Ive tested outgoing calls, and it did work.

    I didnt thought I needed a Inbound rule, as long as the setting on voip providers where configured where calls should be routed...
     

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  9. nb

    nb Support Team
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    Good we are making progress - now it is in verbose mode

    Because I can see the invite

    Look at this

    20:55:15.531 [CM503015]: Call(12): Attempt to reach <sip:0@192.168.11.27:5060> failed. Reason: Forbidden

    20:55:15.531 [CM503003]: Call(12): Call to sip:sip.phonzo.com:5060 has failed; Cause: 403 Rejected: no CLD; from IP:83.233.248.50:5060

    the provider is sending forbidden. The reason is that there is no number sent. you see that 0 in the above line? 0@192.168.11.27

    In fact the invite is generated like this 0@etc etc

    the provider can do nothing with this because the number is sent incorrectly.

    Go to help agt the top in the management console and click on Generate support info - Send this by email bec it contains personal config info.
    nb@3cx.com

    I will check your config.
     
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  10. opera01

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    Mail is sent.
    Grateful for all help!
     
  11. nb

    nb Support Team
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    Do the following:

    1) Enable Re-invites and replaces on Phonzo
    2) Click on the phonzo provider and go to advanced - Since you have stun disabled you have to help prepare the bed for phonzo a bit.:p Therefore to do this: tell the pbx what to put in contact -> Go to the section - "Which IP to use in contact" - and there "specify Specified IP" and put your public ip address there.
    3) Then click on the phonzo in the tree and put max simultaneous calls to more than 1 (this provider can give you more than 1 sim call) Put this to 10. A standard provider supports 5-10. You cant go wrong with this setting.
    4) that DID you put there is not needed. You dont need a DID that is the same as the main number. Delete it.

    With these changes you will get incoming and outgoing calls. With Audio both ways. (unless your router is giving problems) However I can say that I got your configuration working with success. So if you follow this and you still get 1 way audio or no calls, we can concentrate on the router.

    Incoming calls will show caller ID of caller - leave everything as is.

    Outgoing caller id will show after the SECOND call only.
    try it - call your mobile - you should see CALL. Dont answer - REJECT. the provider will cal you again - this time he will show the outgoing CID. probably because of Early media I am not sure. - I think he is connecting too early and sending CID only after the first ring. In fact after you reject the first time, you will hear the phone making the call ringing. Happens in Provider scenarios. Try to see if you can tweak something related to CID in your provider webpage account. Example Send CID at call setup.

    however audio will be ok with the first call - no problem.

    Disconnections work, I had audio both ways and once the call is disconnected, the bye is sent and received. So you will not have any stuck calls. Both sides work ok. I made around 5 calls in total to test.

    Try this and let me know.
     
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  12. opera01

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    Grate! It is working know. Thank you for all help! This is price worthy! :)
    Suddenly I understand what the STUN server was for, so I used stun.3cx.com. Because I actually have dynamic IP.
    I now have to test it on the location, where it shall be installed, and where it is a firewall. I believe it should work.

    Step 2 is to get the digital receptionist to transfer calls to a external number. How do I manage that?
    I tried to set up a constant forwarding on a unused extention (102)... and then let the digital receptionist choose the unused external number 102 when you dial e.g. 5 i the digital receptionist menu.
    But I didn't manage to get it work.

    Step 3 will be to let people call in on a analog line (throug a PSTN device), then meet the digital receptionist, and then get the possibility to dial eg. 1, and then getting forwarded to an external number via eg. the voip provider.

    Any tips?

    (Another thing is that, the digital receptionist will not end the call after 60 sec. Even though I have set Timeout 60 and then "End call"... hmmm.. maybe it is in minute?? :p )
     
  13. nb

    nb Support Team
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    Step 2 -> forward calls to extension 102 and make rule on 102 to forward all calls to external number - include outbound rule

    (Another thing is that, the digital receptionist will not end the call after 60 sec. Even though I have set Timeout 60 and then "End call"... hmmm.. maybe it is in minute?? )

    Is is in seconds - but it will start counting 60 seconds EXCLUSING the time the prompt takes

    Therefore if the prompt is 30 seconds long, the call will end 30+60 = after 90 sec
     
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  14. opera01

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    Hmm... did not work. Dont know if I understood it correctly. What ive done:

    - Made an extention (102) and made a forward rule, that forward all calls all hours all types to a external number. You have to dial 0 for outgoing calls, so Ive tried with and without 0 in front of external number.
    - The outbond rule includes call from extentions 100-999. Digital receptionist has 801 (and another 800). Only 1 route; phonzo. Calls start with 0. Have also set strip 1 digit.

    It will not work.

    When I put 0 in front of external number the receptionis says; call transfer failed. Without 0 im losing the connection, right after she says, your call will be transferred.
     
  15. nb

    nb Support Team
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    try the following

    dial 999

    log in

    press 9 for options, press 3 to dial an outbound call and dial the call from there.

    If this doesnt work make a wireshark capture.
     
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  16. opera01

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    hmmm... which ports needs to be opened on my firewall for my IP PBX with this sip provider? I cant manage to receive incoming calls...! Outgoing is no problemos.

    I've tried the most.
    Ive now opened:
    5060 TCP/UDP (NOT static port mapping?)
    9000-9049 UDP
    16384-16482 UDP


    I've tried the configuration with a direct connection trough mobile broadband, with no firewall/router. That worked! But through my broadband line with firewall/router-modem it will not work...
     
  17. opera01

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    Ok. I sorted it out. I changed to a new modem.
     
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