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Outside clients with 3CX & Linksys SPA 3102

Discussion in '3CX Phone System - General' started by optix, Jan 18, 2009.

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  1. optix

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    Hi all,

    I have a following setup:

    Internet line coming to Cisco router (public, static IP)
    3CX box behind NAT (IP - 10.10.10.202), tcp ports 5060, 5090; udp ports 5060,9000-9015 forwarded to the machine. Current ver 6.1.0
    Linksys SPA 3102 voice gateway (IP - 10.10.10.137 connected to WAN port) with an outside POTS line connected

    I've tried all possible combinations that I could think of in order to get outside clients (also behind NAT) to make calls but without much success. Registration passes with STUN enabled on the phones, but in best case when making call only one side can be heard.

    I could go on about the things I've tried, but before I go into that, I'm just wondering if this can be generally achieved?
    Unfortunately I cannot assign public address to any of the machines, but I can forward any port. Local calls to outside POTS network and vice versa works fine.

    If this can be done, can some one point me to the correct configuration, or to the nature of problem?
    Greatly appreciated!
     
  2. discovery1

    discovery1 Member

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    Is there any reason you are not using the latest version of 3CX?
    Can you give us the make and model of the phones you are using externally?

    Have you tried using the 3CX VOIP client to test with as well?

    Can you set the server logging to verbose and make some calls and post the results here so we can have a look at the call progress?
     
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  3. optix

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    There is, I wasn't able to get some of the features I needed to work on version 7. Irrelevant to this issue. But while version 7 was installed, it still wasn't working.

    Outside clients are Nokia E51's, using default SIP client (with advanced SIP VoIP settings - to enable STUN).
    I have tried with 3CX VoIP client, but it only works in tunnel mode.

    I'm pasting the current log file, when outside client registers and tries to make a call.
    I'll reboot in verbose mode in a minute.

    Code:
    21:35:43.939  	MediaServerReporting::RTPReceiver  	[MS105000] C:29.1: No RTP packets were received:remoteAddr=212.200.240.5:49152,extAddr=89.216.209.3:9006,localAddr=89.216.209.3:9006
    21:35:28.423 	ExtnCfg::updateContact 	[CM504002]: Ext.103: a contact is unregistered. Contact(s): []
    21:35:11.735 	Call::Terminate 	[CM503008]: Call(29): Call is terminated
    21:35:11.735 	Call::Terminate 	[CM503008]: Call(29): Call is terminated
    21:35:11.735 	Call::Terminate 	[CM503008]: Call(29): Call is terminated
    21:35:11.735 	InviteADS::onAckNotReceived 	[CM503019]: Call(29): ACK is not received
    21:34:39.689 	CallCtrl::onLegConnected 	[CM503007]: Call(29): Device joined: sip:10000@10.10.10.137:5061
    21:34:39.689 	CallCtrl::onLegConnected 	[CM503007]: Call(29): Device joined: sip:103@212.200.240.5:1070;transport=UDP
    21:34:39.673 	Line::printEndpointInfo 	[CM505002]: Gateway:[Telekom] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Linksys/SPA3102-5.1.7(GW)] Transport: [sip:10.10.10.202:5060]
    21:34:39.673 	CallCtrl::onAnsweredCall 	[CM503002]: Call(29): Alerting sip:10000@10.10.10.137:5061
    21:34:39.579 	CallCtrl::onSelectRouteReq 	[CM503004]: Call(29): Calling: PSTNline:3060606@(Ln.10000@Telekom)@[Dev:sip:10000@10.10.10.137:5061]
    21:34:39.579 	CallCtrl::onSelectRouteReq 	[CM503010]: Making route(s) to [sip:03060606@optix.dyndns.org;user=phone]
    21:34:39.579 	Extension::printEndpointInfo 	[CM505001]: Ext.103: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Nokia RM-244 200.34.36] Transport: [sip:10.10.10.202:5060]
    21:34:39.579 	CallCtrl::onIncomingCall 	[CM503001]: Call(29): Incoming call from Ext.103 to [sip:03060606@optix.dyndns.org;user=phone]
    21:34:17.579 	ExtnCfg::updateContact 	[CM504001]: Ext.103: new contact is registered. Contact(s): [sip:103@212.200.240.5:1070;transport=UDP/103]
    
    Here it is in verbose
    Code:
    21:46:32.562	Call::Terminate	[CM503008]: Call(2): Call is terminated
    21:46:32.562	Call::Terminate	[CM503008]: Call(2): Call is terminated
    21:46:32.562	Call::Terminate	[CM503008]: Call(2): Call is terminated
    21:46:32.562	InviteADS::onAckNotReceived	[CM503019]: Call(2): ACK is not received
    21:46:00.453	CallCtrl::onLegConnected	[CM503007]: Call(2): Device joined: sip:10000@10.10.10.137:5061
    21:46:00.453	CallCtrl::onLegConnected	[CM503007]: Call(2): Device joined: sip:103@212.200.240.5:1027;transport=UDP
    21:46:00.453	MediaServerReporting::SetRemoteParty	[MS210003] C:2.1:Answer provided. Connection(transcoding mode):89.216.209.3:9002(9003)
    21:46:00.437	MediaServerReporting::SetRemoteParty	[MS210001] C:2.2:Answer received. RTP connection: 10.10.10.137:16414(16415)
    21:46:00.421	Line::printEndpointInfo	[CM505002]: Gateway:[Telekom] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Linksys/SPA3102-5.1.7(GW)] Transport: [sip:10.10.10.202:5060]
    21:46:00.421	CallLeg::setRemoteSdp	Remote SDP is set for legC:2.2
    21:46:00.421	CallCtrl::onAnsweredCall	[CM503002]: Call(2): Alerting sip:10000@10.10.10.137:5061
    21:46:00.296	MediaServerReporting::SetRemoteParty	[MS210002] C:2.2:Offer provided. Connection(transcoding mode): 10.10.10.202:7000(7001)
    21:46:00.265	CallCtrl::onSelectRouteReq	[CM503004]: Call(2): Calling: PSTNline:3060606@(Ln.10000@Telekom)@[Dev:sip:10000@10.10.10.137:5061]
    21:46:00.250	CallCtrl::onSelectRouteReq	[CM503010]: Making route(s) to [sip:03060606@optix.dyndns.org;user=phone]
    21:46:00.250	MediaServerReporting::SetRemoteParty	[MS210000] C:2.1:Offer received. RTP connection: 212.200.240.5:49152(49153)
    21:46:00.250	CallLeg::setRemoteSdp	Remote SDP is set for legC:2.1
    21:46:00.250	Extension::printEndpointInfo	[CM505001]: Ext.103: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Nokia RM-244 200.34.36] Transport: [sip:10.10.10.202:5060]
    21:46:00.250	CallCtrl::onIncomingCall	[CM503001]: Call(2): Incoming call from Ext.103 to [sip:03060606@optix.dyndns.org;user=phone]
    21:46:00.046	CallLeg::onNewCall	[CM500002]: Info on incoming INVITE:
    INVITE sip:03060606@optix.dyndns.org;user=phone SIP/2.0
    Via: SIP/2.0/UDP 212.200.240.5:1030;branch=z9hG4bKein7glk8vvoq7nufa581fij;rport=1028
    Max-Forwards: 70
    Route: [sip:optix.dyndns.org;lr;transport=UDP]
    Contact: [sip:103@212.200.240.5:1027;transport=UDP]
    To: [sip:03060606@optix.dyndns.org;user=phone]
    From: [sip:103@optix.dyndns.org];tag=lrf52k53nhhc732d069i
    Call-ID: dw5hk3GroId5Vg6j9UJHOmc0-JknuF
    CSeq: 903 INVITE
    Expires: 120
    Accept: application/sdp
    Allow: INVITE, ACK, BYE, CANCEL, REFER, NOTIFY, OPTIONS, PRACK
    Proxy-Authorization: Digest realm="3CXPhoneSystem",nonce="12876785159:7167c575f5f543fcf729f243fda62569",algorithm=MD5,username="103",uri="sip:03060606@optix.dyndns.org;user=phone",response="1635a582a8f5c67b1d356c1b39d786d4"
    Supported: 100rel, sec-agree
    User-Agent: Nokia RM-244 200.34.36
    Privacy: none
    P-Preferred-Identity: sip:103@optix.dyndns.org
    Content-Length: 0
    
    21:45:48.843	ExtnCfg::updateContact	[CM504001]: Ext.103: new contact is registered. Contact(s): [sip:103@212.200.240.5:1027;transport=UDP/103]
    
     
  4. discovery1

    discovery1 Member

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    So for clarification - you are using Nokia E51's remotely to make calls through 3CX and out through a Linksys SPA3102 via it's PSTN line?
    I also see that you are using dynamic DNS for the remote ends - ext 103 is the remote extension?

    I take it you have followed the setup for the E65 as the E51's aren't supported
    http: //www.3cx.com/sip-phones/nokia-e65/ - NO LONGER AVAILABLE

    And set up the SPA3102 as per the guide
    http: //www.3cx.com/voip-gateways/linksys-3102/ - NO LONGER AVAILABLE

    Can an extension on the same LAN as 3CX make a call through the SPA3102?
    Can you check the firmware version in the SPA3102 to see if it is the same or newer than the version in the guide?

    This is the problem
    Here is some documentation on how to analyse a SIP call setup which may help - the section you want is halfway down the page
    http: //wiki.3cx.com/Home/technical-documentation-dash/analyzing-call-setup - NO LONGER AVAILABLE
     
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  5. optix

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    Thanks for replying.

    All correct.
    Dynamic DNS is just resolving to my IP 89.216.209.3 I've set up Local SIP domain on 3CX to be that domain. However, I've tried with plain IP address (both local and public) with same result on making calls from outside.
    What would be the correct setting here, considering that machine is behind NAT?

    Yes. Almost same settings are used when device is in the same LAN (only difference is the IP address) and they work.

    Yes.

    Yes, all is working fine when clients are in the same LAN.
    Firmware is the same.


    Has anyone successfully configured this to work?
     
  6. discovery1

    discovery1 Member

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    How do the E51's connect to 3CX when they are in the same LAN, Wireless?
    If it works using wireless then it could be either a firewall issue at the remote end or possibly a latency issue across the mobile phone network.

    As far as i know you are the first person to try this configuration. Packet loss and jitter across a 3G mobile network would possibly render the whole thing unworkable and cause the issues you are seeing. I did try doing this a couple of years ago from the mobile phone (E65) to a 3CX extension on the LAN and the quality was very average. Can you successfully call from a remote E51 to an extension on the LAN and get good call quality?

    Try enabling "PBX delivers Audio" for the remote extensions.
     
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  7. optix

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    Yes, WiFi. They also use WiFi when outside, I don't use GSM/UMTS network for this.
    There's also no firewall anywhere, except for NAT.

    I can, apart from some small delay, overall quality is very acceptable.
    I have a feeling that problem lies somewhere on the SPA3102 device, but I don't see any logically related setting.

    Will do.

    ^ I just checked, option was enabled for that extension (103).
     
  8. discovery1

    discovery1 Member

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    Check the 3102 settings in the Voice > PSTN Line section - make sure the preferred codec is something like 711u and the option "use pref codec only" is set to no

    Obviously the "VOIP to PSTN" section needs to be enabled
     
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