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Using 3CX 100% externally

Discussion in '3CX Phone System - General' started by RyanWVU, Jan 9, 2011.

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  1. RyanWVU

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    Hi guys, I'm new to 3CX and the whole IP PBX world, so please bear with me. What I would like to do is to use 3CX mainly as an answering service for my team. I would like to set it up so that if a person calls my phone number, 3CX will pick up the call and prompt the user to enter an extension and then once the user enters the extension it will ring my mobile phone but then if it's not answered within so many rings (or seconds) it transfers the user to my voicemail. Currently what I have set up is SIP-to-Skype (with a Skype online number) and Exchange 2010 integration. I am currently using Exchange 2010 for my auto attendant and I have set up the system according to the following page: http://www.3cx.com/blog/docs/exchange-server-configuration/. I do not have any physical phones connected to the network and plan on using the 3CX system to ring our mobile phones and for voicemail.

    My current issue is that when I call my Skype online number, it rings once then disconnects the call. This happens when I call from my mobile number stored in Active Directory and when I call as a blocked number. Here is my current setup:

    Host Machine: Windows Server 2008 R2 Datacenter, AD Domain Controller, DNS Server, IIS Server, Exchange 2010, SharePoint 2010

    Virtual Machine: Windows Server 2008 R2 Datacenter, IIS Server, Lync 2010 Server, 3CX Server (w/SIP-to-Skype Gateway)

    I should also note that the host machine does not have a sound card installed, but after some reading most people say that sound cards are not necessary.

    Any help would be GREATLY appreciated, as I have been racking my brain for a couple days on this issue.
     
  2. RyanWVU

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    Here is a print out of my Server Activity Log. I cleared it, then called my online number; these are the results...

    20:57:59.619 [CM503008]: Call(9): Call is terminated
    20:57:53.253 [CM503007]: Call(9): Device joined: sip:100@10.0.1.7:5060;transport=TCP
    20:57:53.248 [CM503007]: Call(9): Device joined: sip:10000@127.0.0.1:6060;rinstance=a7d1974714b070e1
    20:57:53.244 [CM503002]: Call(9): Alerting sip:10.0.1.7:5060;transport=TCP
    20:57:52.675 [CM503025]: Call(9): Calling @[Dev:sip:10.0.1.7:5060;transport=TCP]
    20:57:52.649 [CM503005]: Call(9): Forwarding: @[Dev:sip:10.0.1.7:5060;transport=TCP]
    20:57:52.648 [CM503016]: Call(9): Attempt to reach <sip:100@10.0.1.8:5060> failed. Reason: Not Registered
    20:57:52.647 [CM503017]: Call(9): Target is not registered: Ext:Ext.100
    20:57:52.646 [CM503010]: Making route(s) to <sip:100@10.0.1.8:5060>
    20:57:52.644 [CM505002]: Gateway:[Skype] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CXSkypeGateway 2.0.9919.0] PBX contact: [sip:10000@127.0.0.1:5060]
    20:57:52.628 [CM503001]: Call(9): Incoming call from 0+19087987735@(Ln.10000@Skype) to <sip:100@10.0.1.8:5060>
    20:57:52.623 [CM503012]: Inbound out-of-office hours rule (unnamed) for 10000 forwards to DN:100
    20:53:52.147 Currently active calls - 1: [8]
    20:53:46.791 [CM503007]: Call(8): Device joined: sip:10000@127.0.0.1:6060;rinstance=a7d1974714b070e1
    20:53:46.230 [CM503025]: Call(8): Calling @[Dev:sip:10.0.1.7:5060;transport=TCP]
    20:53:46.198 [CM503016]: Call(8): Attempt to reach <sip:100@10.0.1.8:5060> failed. Reason: Not Registered
    20:53:46.196 [CM503010]: Making route(s) to <sip:100@10.0.1.8:5060>
    20:53:46.176 [CM503001]: Call(8): Incoming call from 0+19087987735@(Ln.10000@Skype) to <sip:100@10.0.1.8:5060>
    20:49:46.820 [CM506001]: STUN request to resolve SIP external IP:port mapping is sent to STUN server 96.9.132.83:3478 over Transport 10.0.1.8:5060
    20:37:51.488 [CM503007]: Call(7): Device joined: sip:100@10.0.1.7:5060;transport=TCP
    20:37:51.478 [CM503002]: Call(7): Alerting sip:10.0.1.7:5060;transport=TCP
    20:37:50.877 [CM503005]: Call(7): Forwarding: @[Dev:sip:10.0.1.7:5060;transport=TCP]
    20:37:50.876 [CM503017]: Call(7): Target is not registered: Ext:Ext.100
    20:37:50.873 [CM505002]: Gateway:[Skype] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CXSkypeGateway 2.0.9919.0] PBX contact: [sip:10000@127.0.0.1:5060]
     
  3. sigma1

    sigma1 Active Member

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    We did the exact same thing... simple solution, get a budget PC (Win XP /7/2003) with a sound card and make it your sis>SIP gateway (skype). Was not worth investing a lot of time into it.
     
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  4. RyanWVU

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    Well the host machine has a sound card, however, I currently have it disabled since I have no use for it. Is a sound card actually required? Are you suggesting the issue is the absence of a sound card or that I am running 3CX on a virtual machine?
     
  5. sigma1

    sigma1 Active Member

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    We could not get it to work on a VM, RDP sound drivers or not, whatever. It is very easy to get it to work on a host with a sound card that wasting time is not worth it. Plus we were not thrilled to have Skype on a server class machine.
     
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  6. RyanWVU

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    I'm really running 3CX in a semi-production state and I don't have the option to add another machine or expand the network. I have read that many people have successfully set up 3CX in a virtualized environment, even with Win 2008. I know there must be a fix for this issue without the need for a separate machine.
     
  7. sigma1

    sigma1 Active Member

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    There is no doubt that 3CX runs in a virtual environment, we do have several undered VMs to attest to that but the Skype Gateway (Sis>SIP) has been a bit of a challenge. 3CX works falwlessly however...
     
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  8. RyanWVU

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    Yeah, I do know that 3CX is working flawlessly as well as Exchange integration as I can connect to 3CX remotely using the softphone. I am unable to make outbound calls and inbound calls are disconnected, however I just realized I don't have an inbound call rule set up but I didn't think that was necessary when auto-attendant is enabled.
     
  9. sigma1

    sigma1 Active Member

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    You always need inbound rules, you need to tell 3CX what to do once the call comes in... likewise you need an outbound rule for anything other than your extensions and system extensions
     
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  10. RyanWVU

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    I apologize, I am an idiot. I realized that I needed to set the trunk rule for the Skype gateway to connect to the digital receptionist instead of the operator since I don't have the operator phone registered. So now when I call, I don't get disconnected, I just get silence after the first ring. I now know that the issue is not the SIP-to-Skype gateway because I am able to ring my extension and answer the call when I set the trunk settings to dial my extension instead of the digital receptionist.

    I think I can safely say that my issue is now sound related. I believe 3CX is working as well as the Skype gateway, it's just that sound isn't being transferred either way. When I press a button nothing happens and when I talk on the softphone you can't hear it on the other end. I have enabled the sound card on the server, however since 3CX is in a virtualized environment the sound card doesn't transfer to the virtualized environment. Also, I have read that many people run 3CX on servers without sound cards, so I'm not even sure that having the sound card will fix my problem.

    I do have an outbound call rule in effect, however any number I enter in the 3CX softphone results in a "Not Available." The call rule is simply a prefix of 9, so when I dial 9-XXX-XXX-XXXX or 9-1-XXX-XXX-XXXX I still get "Not Available."

    EDIT: I should point out that my statement that sound is not working either ways is not completely accurate. On the softphone I am able to dial the auto attendant as well as voicemail and I can hear the prompts perfectly, I am just unable to dial anything. When I call from a physical outside phone (cellular phone) I cannot hear any prompts and I am unable to enter any digits.

    Here are the latest logs:

    16:10:04.939 [CM503008]: Call(9): Call is terminated
    16:10:00.201 Currently active calls - 1: [9]
    16:09:27.850 Currently active calls - 1: [9]
    16:09:26.952 [CM503007]: Call(9): Device joined: sip:800@10.0.1.7:5060;transport=TCP
    16:09:26.947 [CM503007]: Call(9): Device joined: sip:10000@127.0.0.1:6060;rinstance=9fcb18e6c2386189
    16:09:26.941 [CM503002]: Call(9): Alerting sip:10.0.1.7:5060;transport=TCP
    16:09:26.359 [CM503025]: Call(9): Calling @[Dev:sip:10.0.1.7:5060;transport=TCP]
    16:09:26.328 [CM503004]: Call(9): Route 1: @[Dev:sip:10.0.1.7:5060;transport=TCP]
    16:09:26.327 [CM503010]: Making route(s) to <sip:800@10.0.1.8:5060>
    16:09:26.326 [CM505002]: Gateway:[Skype] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CXSkypeGateway 2.0.9919.0] PBX contact: [sip:10000@127.0.0.1:5060]
    16:09:26.306 [CM503001]: Call(9): Incoming call from 0+19087987735@(Ln.10000@Skype) to <sip:800@10.0.1.8:5060>
    16:09:26.302 [CM503012]: Inbound out-of-office hours rule (unnamed) for 10000 forwards to DN:800
    16:04:03.783 [CM503016]: Call(8): Attempt to reach <sip:99087987735@69.125.252.116:5060> failed. Reason: Temporarily Unavailable
    16:04:03.777 [CM505002]: Gateway:[Skype] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CXSkypeGateway 2.0.9919.0] PBX contact: [sip:10000@127.0.0.1:5060]
    16:04:03.580 [CM503025]: Call(8): Calling PSTNline:19087987735@(Ln.10000@Skype)@[Dev:sip:10000@127.0.0.1:6060;rinstance=9fcb18e6c2386189]
    16:04:03.532 [CM503010]: Making route(s) to <sip:99087987735@69.125.252.116:5060>
    16:04:03.525 [CM503001]: Call(8): Incoming call from Ext.200 to <sip:99087987735@69.125.252.116:5060>
    16:04:03.525 [MS201000] Use STUN server 'stun.3cx.com:3478'
    16:04:03.448 [MS201000] Use STUN server 'stun3.3cx.com:3478'
    16:04:03.419 [MS201000] Use STUN server 'stun2.3cx.com:3478'
    16:00:01.828 [CM506001]: STUN request to resolve SIP external IP:port mapping is sent to STUN server 96.9.132.83:3478 over Transport 10.0.1.8:5060
    15:58:05.500 [CM503008]: Call(7): Call is terminated
    15:57:47.083 [CM503007]: Call(7): Device joined: sip:200@67.165.92.97:35778
    15:57:46.183 [CM503025]: Call(7): Calling @[Dev:sip:10.0.1.7:5060;transport=TCP]
     
  11. RyanWVU

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    Here's the newest revelation: I enabled tunneling in the 3CX softphone and now I am able to hear prompts and speak to the voice prompts as well as press keys and navigate through the system. I am still unable to make outbound calls from the softphone; it looks like a prefix is being added to the outgoing sip number which may be causing the call to fail, but I am unsure. I am also still unable to hear the prompts if I call from an outside device (cell phone) or press digits or speak to the prompts.
     
  12. RyanWVU

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    SOLVED!

    For the no sound issue, all it took was installing a virtual sound card driver. This fixed all sound issues. As for my outbound call issue, it turned out that I was not using the correct prefix when dialing out for Skype. I added the + (or 00) as a prepend in the outbound rule and it works flawlessly now!
     
  13. Discovery Technology

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    The Skype gateway requires a sound card to be installed on the 3CX server itself, or you'll have no end of audio issues.

    The separate gateway PC concept with a cheap sound card would also work, hoever it is understandable that this would be a major issue for a data centre installation - we faced a similar scenario ourselves recently with a customer that is distributed across several countries and all work remotely to the phone system.

    A virtual sound card driver would work well if your VM environment supports it, which it sounds like your environment does. If this is an issue for a future installation, you can setup a Skype SIP trunk instead and treat the skype calls as you would any other VoIP provider, as opposed to the custom setup that Skype normally offers.

    More info on this service can be found at: http://www.skype.com/intl/en/business/sip/overview

    I believe you can order it from the following link: http://www.skype.com/intl/en/business/sip/get-it-now/
     
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  14. RyanWVU

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    Yeah, I thought about getting the Skype SIP service but I really wanted to try to reduce costs as much as possible as the system is meant for an organization and all costs must come out of pocket.
     
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