Remote extensions

Discussion in '3CX Phone System - General' started by buster1075, Jun 26, 2011.

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  1. buster1075

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    Hi,

    I am trying to set up 5 remote extensions for a new 3cx phone system we have installed and I am struggling a bit to get them to work correctly.

    At the moment I have them all set up at home but they will be sent out to various remote users. They are registering fine on the phone system and if you ring the extensions they do actually ring, the problem is when you answer them there is no audio in either direction. Also strangely once you hang up it often rings the called extension again, If you make a call out from the extension to an external number (I have been using my mobile) the call connects and has audio in both directions.

    I must be missing something somewhere, I have read a lot of guides etc... but there is nothing to say I should port forward anything on my home router, is this the case? If you do have to port forward how would that work when I have more than one extension on the same network.

    Please let me know if you can help or if you need any more information from me.

    Thanks
     
  2. nbailey

    nbailey Member

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    UDP 9000-9049 for RTP traffic, 5060 TCP and disable SIP alg and SIP helper you should be fine. if you are using multiple remote extensions you need to be careful with TCP 5060. SIP proxy manager is the best method for remote extensions because you don't need to worry about NAT issues.

    Thanks,
    Nate
     
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  3. eagle2

    eagle2 Well-Known Member

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    Hi,

    most probably the problem is with NAT at remote location – most (inexpensive) routers will not handle correctly voice traffic between extensions in such setup (peer-to-peer). If your router at remote location is SIP ALG capable, most probably everything will be fine. In any case no port forwarding is necessary at your home router.

    There are several approaches to the problem:

    1. Try changing the router (with higher model), or establish a VPN tunnel with the 3CX server site, or use 3CX proxy server at the remote site -- in your case this may be not appropriate, nevertheless this will solve the problem.

    2. Try selecting on 'PBX delivers audio' in extension settings in 3CX server for each of the remote extensions. This will force voice traffic (RTP packets) to be routed through the PBX, instead of being routed peer-to-peer.

    3. Try NOT using STUN at remote extensions, instead of this select 'ALLOWSOURCEASOUTBOUND' parameter set to '1' in Custom parameters of 3CX server. This will overcome problems with most NAT implementations (your case is typical).
    See more about this parameter in the following post (towards to end): http://www.3cx.com/forums/no-incoming-calls-19487.html

    Hopefully any of the above three methods will solve the problem.

    Regards.
     
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  4. leejor

    leejor Well-Known Member

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    What works remotely (settings), may not work locally. As you may have left the sets at the default port of 5060 and enabled STUN in anticipation of remote operation, those things may be causing issues.
     
  5. buster1075

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    The server is in an office elsewhere, that is why the phones are at home and not in the office. I am trying to replicate the remote users set up for testing.
     
  6. buster1075

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    Hi,

    Thank you for your response, the router I have at home is a BT Voyager 2110 so is pretty basic. I have had a look at the other options but they do not work either. The 'ALLOWSOURCEASOUTBOUND' parameter was set to 1 already so i disabled stun on the phone but no joy. I also set the 'PBX delivers audio' option but that did not work either, do you have to restart services or wait any time after making a change?

    Thanks again
     
  7. eagle2

    eagle2 Well-Known Member

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    Hi,

    Restarting remote router and phones may be useful. If you like you may restart also 3CX system services and the router in front.

    Stick to point 1 then -- replace the router, or use local proxy server (3cx) or VPN tunnel (by other means). If you are going to use the phones in various locations you may not need to do anything (most probably they will be working fine there).

    You are running the latest Version 10 SP1.1, yeah ?
    Sorry, no other idea.

    Regards,
    Orlin.
     
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  8. leejor

    leejor Well-Known Member

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    Sorry, I didn't pay enough attention the first time I read your post to realize that you were trying to use all of the extensions remotely, at one site currently.

    I have seen "no audio" issues at a remote location, as the router (can depend on the brand/model) becomes confused as to where to send the audio packets when more than one extension is in use, not using the proxy server, all behind one router.

    You can try giving each a unique port number, but that may still not solve your issue, as it's the voice packets not getting through. If all setting are correct, you should have no problems if each set is installed, by itself, behind different routers.
     
  9. buster1075

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    Hi,

    I have pasted the part of the log below that seems to be the call that is not working if that sheds any more light.

    Code:
    08:51:13.878  [CM503008]: Call(1): Call is terminated
    08:51:03.956  [CM503025]: Call(1): Calling Ext:Ext.227@[Dev:sip:227@87.112.121.xxx:5062]
    08:51:03.954  [MS210002] C:1.2:Offer provided. Connection(transcoding mode): 87.127.16.xxx:9002(9003)
    08:51:03.802  [CM503004]: Call(1): Route 1: Ext:Ext.227@[Dev:sip:227@87.112.121.xxx:5062]
    08:51:03.798  [CM503010]: Making route(s) to <sip:227@192.168.1.xxx:5060>
    08:51:03.791  [MS210000] C:1.1:Offer received. RTP connection: 93.95.124.xxx:36366(36367)
    08:51:03.784  Remote SDP is set for legC:1.1
    08:51:03.784  [CM505003]: Provider:[Voiceflex] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [VoiceFlex] PBX contact: [sip:84418255@87.127.16.xxx:5060]
    08:51:03.782  [CM503001]: Call(1): Incoming call from 07709xxxxxx@(Ln.10000@Voiceflex) to <sip:227@192.168.1.xxx:5060>
    08:51:03.637  [CM503012]: Inbound any hours rule (00441592xxxxxx) for 10000 forwards to DN:227
    08:51:03.635  Looking for inbound target: called=00441592xxxxxx; caller=07709xxxxxx
    08:51:03.631  [CM500002]: Info on incoming INVITE:
      INVITE sip:00441592xxxxxx@87.127.16.xxx:5060 SIP/2.0
      Via: SIP/2.0/UDP 93.95.124.xxx:5060;branch=z9hG4bK7b9f8928;rport=5060
      Max-Forwards: 70
      Contact: <sip:07709xxxxxx@93.95.124.xxx>
      To: <sip:00441592xxxxxx@87.127.16.xxx:5060>
      From: "07709xxxxxx"<sip:07709xxxxxx@93.95.124.xxx>;tag=as1ff93614
      Call-ID: 3a4f23d27a8ca6d03c2567bd2d15ed47@93.95.124.xxx
      CSeq: 102 INVITE
      Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
      Date: Mon, 27 Jun 2011 07:51:09 GMT
      Supported: replaces
      User-Agent: VoiceFlex
      Content-Length: 0
    
     
  10. eagle2

    eagle2 Well-Known Member

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    Hi,

    I bought last year an inexpensive router -- Netgear WGR614 V9 (under € 50.-) which is handling your case correctly -- I have at home 4 SIP phones registered to remote 3CX server (at the office) and they can talk to each other, transfer calls, etc.

    Before that I tried different things like different SIP ports, different RTP ranges, port forwarding, etc. and I managed to get some results, but as whole I didn't succeeded entirely. Calls to/from external destinations worked always (RTP path is through the PBX). You can see what's happening and where you loose your RTP packets (i.e. voice) with wireshark or other network analyzer tool.

    I have a customer having a Cisco 871 at the remote site (and on the main site). The same operation (with several remote extensions) is working fine at the remote site, nevertheless we switched now to tunnel model (Cisco-to-Cisco GRE). The main difference is both routers are SIP ALG capable and handle correctly internal peer-to-peer traffic at remote sites, nevertheless the Netgear's documentation doesn't include a single word on the topic.

    Also it is not necessary to use STUN server in remote extensions configuration, if they are behind a SIP ALG NAT/router (it is translating correctly the SIP messages as well). Otherwise use 'ALLOWSOURCEASOUTBOUND' with V10 SP1.1 (helps a lot).

    Regards,
    Orlin.
     
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  11. buster1075

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    Hi,

    I actually managed to get this working by accident :). I plugged one of the phones I was testing in to our network at work which has a different phone system on it so it provisioned it for that system which included updating the firmware. Once this was done I had to set the phone up again as a remote extension on the 3CX. Once I had done this that phone was working, I then connected all the others to our phone system in the same way to update the firmware and all are now working.

    I do still also have the send audio through PBX ticked as well.

    Thank you all for your comments and responses.
     
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