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PSTN Problems - Cannot make call

Discussion in '3CX Phone System - General' started by Baggins, Feb 12, 2013.

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  1. Baggins

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    Setup - V10 3CX Phone
    Yealink T22P ip phone
    SPA3102 Gateway

    ip of 3CX 192.168.0.10
    ip of SPA 192.168.0.8

    Config - Spa registered ok - Yealink Provisioned ok
    NO VOIP SETUP - Only PSTN line for both incoming & outgoing calls
    -----------------

    I have not been able to make an outgoing call on the PSTN line. I receive a pre-recorded message from the phone company saying 'your call could not be connected, please check the number and try again'. So it appears to be the number sent is incomplete or not compatible with my phone company?

    Also incoming calls once answered continue to ring for around 6-8 rings before the call is established even though the yealink handset has been picked up? Once established the call seems fine.

    I have checked my outbound rules and Im sure its fine, not stripping or pre-pending the number

    I live in Australia and Im wondering if its some regional issue?

    Are my ip addresses correct? I' m unsure if they are correct correct.

    Any help would be greatly appreciated.

    -------------
    16:36:09.537|.\CallCtrl.cpp(28)|Log2||CallCtrl::thread:[CM501007]: *** Started Calls Controller thread ***<br>
    16:36:09.537|.\SLServer.cpp(974)|Log2||MyConnectionListener::ConnectListener:[EC200001]: Configuration server is connected:
    application:eftpos-PC:5485/DBProvider
    local:127.0.0.1:49802
    remote:127.0.0.1:5485<br>
    16:36:09.646|.\CallMgr.cpp(1528)|Log2||LoadLicenceInfo:[CM501010]: License Info: Load Failed - Error licence Licence: 27 System: 2 <br>
    16:36:09.646|.\CallMgr.cpp(412)|Log2|CallManager|CallMgr::Initialize:[CM501001]: Start 3CX PhoneSystem Call Manager<br>
    16:36:09.646|.\CallMgr.cpp(413)|Log2|CallManager|CallMgr::Initialize:[CM501002]: Version: 10.0.23053.0<br>
    16:36:09.646|.\CallMgr.cpp(1135)|Log2|CallManager|CallMgr::findLocalIPs:[CM501006]: Default Local IP address: [10.27.55.12, 192.168.0.10]<br>
    16:36:09.646|.\CM_Stack.cpp(38)|Critical0|SipStack|CallMgr::Stack::Initialize:Failed to obtain short path name for [C:\ProgramData\3CX\Bin\Cert]<br>
    16:36:10.847|.\MCastTransport.cpp(34)|Log5||JoinMCast:Transport [ V4 10.27.55.12:5060 UDP target domain=unspecified mFlowKey=892 ] is joined to SIP multicast group<br>
    16:36:10.847|.\MCastTransport.cpp(34)|Log5||JoinMCast:Transport [ V4 192.168.0.10:5060 UDP target domain=unspecified mFlowKey=924 ] is joined to SIP multicast group<br>
    16:36:10.941|.\StunClient.cpp(356)|Log2|STUN|StunClient::eek:nInitTests:[CM506001]: STUN request to resolve SIP external IP:port mapping is sent to STUN server [ V4 192.198.87.70:3478 UDP target domain=unspecified mFlowKey=0 ] over Transport [ V4 192.168.0.10:5060 UDP target domain=unspecified mFlowKey=0 ]<br>
    16:36:11.284|.\StunClient.cpp(135)|Log2|STUN|StunClient::process:[CM506002]: Resolved SIP external IP:port ([ V4 203.59.229.218:5060 UDP target domain=unspecified mFlowKey=0 ]) on Transport [ V4 192.168.0.10:5060 UDP target domain=unspecified mFlowKey=0 ]<br>
    16:36:11.565|.\SLServer.cpp(970)|Log2||MyConnectionListener::ConnectListener:[EC200006]: Conference server is connected:
    application:eftpos-PC:0/3CXConferenceRoom
    local:127.0.0.1:5482
    remote:127.0.0.1:49805<br>
    16:36:11.658|.\Call.cpp(48)|Log5||??:Currently active calls [none]<br>
    16:36:11.736|.\SLServer.cpp(958)|Log2||MyConnectionListener::ConnectListener:[EC200004]: IVR server is connected:
    application:eftpos-PC:0/IVRServer
    local:127.0.0.1:5482
    remote:127.0.0.1:49809<br>
    16:36:11.799|.\SLServer.cpp(966)|Log2||MyConnectionListener::ConnectListener:[EC200005]: Parking Orbit server is connected:
    application:eftpos-PC:0/3CXParkOrbit
    local:127.0.0.1:5482
    remote:127.0.0.1:49810<br>
    16:36:11.912|.\SLServer.cpp(954)|Log2||MyConnectionListener::ConnectListener:[EC200002]: Media server is connected:
    application:eftpos-PC:0/MediaServer
    local:127.0.0.1:5482
    remote:127.0.0.1:49811<br>
    16:36:12.334|.\VExtMgr.cpp(93)|Log2|DataBase|FaxCfg::updateContact:[CM504008]: Fax Service: registered as sip:88@10.27.55.12:5060 with contact sip:88@10.27.55.12:5100;user=phone<br>
    16:36:13.037|.\SLServer.cpp(978)|Error1||MyConnectionListener::ConnectListener:[EC100007]: External application is connected:
    application:eftpos-PC:0/QueueManager
    local:127.0.0.1:5482
    remote:127.0.0.1:49813<br>
    16:36:13.075|.\Extension.cpp(406)|Log2||ExtnCfg::updateContact:[CM504001]: Ext.10: new contact is registered. Contact(s): [sip:10@192.168.0.2:5062/10]<br>
    16:36:13.313|.\Authorization.cpp(180)|Critical0|Authorization|AuthMgr::eek:nAuthFailure:[CM102001]: Authentication failed for SipReq: REGISTER 192.168.0.10 tid=18408109a50f0f3b cseq=REGISTER contact=DialCode@127.0.0.1:5488 / 2 from(wire); Reason: Credentials don't match, check that authorization-ID and password match the ones in extension settings<br>
    16:36:15.915|.\Extension.cpp(406)|Log2||ExtnCfg::updateContact:[CM504001]: Ext.99: new contact is registered. Contact(s): [sip:99@127.0.0.1:40600;rinstance=a190201e014d7407/99]<br>
    16:36:15.925|.\Extension.cpp(406)|Log2||ExtnCfg::updateContact:[CM504001]: Ext.EndCall: new contact is registered. Contact(s): [sip:EndCall@127.0.0.1:40600;rinstance=6b23b4aa465f8301/EndCall]<br>
    16:36:17.562|.\Extension.cpp(406)|Log2||ExtnCfg::updateContact:[CM504001]: Ext.*777: new contact is registered. Contact(s): [sip:*777@127.0.0.1:40000;rinstance=9e3d59c1bc07a80a/*777]<br>
    16:36:17.574|.\Extension.cpp(406)|Log2||ExtnCfg::updateContact:[CM504001]: Ext.*0: new contact is registered. Contact(s): [sip:*0@127.0.0.1:40000;rinstance=7dfbefb5a325696a/*0]<br>
    16:36:17.665|.\Extension.cpp(406)|Log2||ExtnCfg::updateContact:[CM504001]: Ext.*1: new contact is registered. Contact(s): [sip:*1@127.0.0.1:40000;rinstance=36272764bd7e9a87/*1]<br>
    16:36:19.554|.\CallMgr.cpp(1268)|Log2|CallManager|CallMgr::eek:nAddIPs:IP(s) added:[10.27.55.12,192.168.0.10]<br>
    16:36:43.661|.\Call.cpp(48)|Log5||??:Currently active calls [none]<br>
    16:37:15.663|.\Call.cpp(48)|Log5||??:Currently active calls [none]<br>
    16:37:47.665|.\Call.cpp(48)|Log5||??:Currently active calls [none]<br>
    16:38:17.666|.\Call.cpp(48)|Log5||??:Currently active calls [none]<br>
    16:38:49.668|.\Call.cpp(48)|Log5||??:Currently active calls [none]<br>
    16:39:08.634|.\CallLeg.cpp(143)|Log5||CallLeg::eek:nNewCall:[CM500002]: Info on incoming INVITE:
    INVITE sip:96552202@192.168.0.10 SIP/2.0
    Via: SIP/2.0/UDP 192.168.0.2:5062;branch=z9hG4bK1692229894
    Max-Forwards: 70
    Contact: <sip:10@192.168.0.2:5062>
    To: <sip:96552202@192.168.0.10>
    From: "John"<sip:10@192.168.0.10>;tag=1118809000
    Call-ID: 1528819234@192.168.0.2
    CSeq: 2 INVITE
    Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
    Proxy-Authorization: Digest username="10",realm="3CXPhoneSystem",nonce="414d535c07254aac76:8ee75af8f0f897c7f6fcbe99ddab92c4",uri="sip:96552202@192.168.0.10",response="04405c117e5e0adc756f203b1935d353",algorithm=MD5
    Supported: replaces
    User-Agent: Yealink SIP-T22P 7.61.0.80
    Allow-Events: talk, hold, conference, refer, check-sync
    Content-Length: 0

    <br>
    16:39:08.650|.\CallCtrl.cpp(346)|Log2||CallCtrl::eek:nIncomingCall:[CM503001]: Call(1): Incoming call from Ext.10 to <sip:96552202@192.168.0.10><br>
    16:39:08.652|.\Extension.cpp(1407)|Log3||Extension::printEndpointInfo:[CM505001]: Ext.10: Device info: Device Identified: [Man: Yealink;Mod: T22;Rev: General] Capabilities:[reinvite, replaces, unable-no-sdp, no-recvonly] UserAgent: [Yealink SIP-T22P 7.61.0.80] PBX contact: [sip:10@192.168.0.10:5060]<br>
    16:39:08.652|.\CallLeg.cpp(1194)|Log5||CallLeg::setRemoteSdp:Remote SDP is set for legC:1.1<br>
    16:39:08.657|.\SLServer.cpp(861)|Log5|MediaServer|MediaServerReporting::SetRemoteParty:[MS210000] C:1.1:Offer received. RTP connection: 192.168.0.2:11782(11783)<br>
    16:39:08.663|.\CallCtrl.cpp(529)|Log3||CallCtrl::eek:nSelectRouteReq:[CM503010]: Making route(s) to <sip:96552202@192.168.0.10><br>
    16:39:08.664|.\CallCtrl.cpp(708)|Log2||CallCtrl::eek:nSelectRouteReq:[CM503004]: Call(1): Route 1: PSTNline:96552202@(Ln.10000@LINKSYS)@[Dev:sip:10000@192.168.0.8:5062]<br>
    16:39:08.664|.\CallCtrl.cpp(708)|Log2||CallCtrl::eek:nSelectRouteReq:[CM503004]: Call(1): Route 2: PSTNline:96552202@(Ln.10000@LINKSYS)@[Dev:sip:10000@192.168.0.8:5062]<br>
    16:39:08.664|.\CallCtrl.cpp(708)|Log2||CallCtrl::eek:nSelectRouteReq:[CM503004]: Call(1): Route 3: PSTNline:96552202@(Ln.10000@LINKSYS)@[Dev:sip:10000@192.168.0.8:5062]<br>
    16:39:08.688|.\SLServer.cpp(861)|Log5|MediaServer|MediaServerReporting::SetRemoteParty:[MS210006] C:1.2:Offer provided. Connection(by pass mode): 192.168.0.2:11782(11783)<br>
    16:39:08.689|.\Target.cpp(441)|Log2||Target::makeOneInvite:[CM503025]: Call(1): Calling PSTNline:96552202@(Ln.10000@LINKSYS)@[Dev:sip:10000@192.168.0.8:5062]<br>
    16:39:08.766|.\CallLeg.cpp(315)|Log3||CallLeg::eek:nAnswer:[CM503002]: Call(1): Alerting sip:10000@192.168.0.8:5062<br>
    16:39:08.766|.\Line.cpp(1452)|Log2||Line::printEndpointInfo:[CM505002]: Gateway:[LINKSYS] Device info: Device Identified: [Man: Linksys;Mod: SPA Series;Rev: General] Capabilities:[reinvite, no-replaces, able-no-sdp, recvonly] UserAgent: [Linksys/SPA3102-5.2.13(GW002)] PBX contact: [sip:10000@192.168.0.10:5060]<br>
    16:39:08.767|.\CallLeg.cpp(1194)|Log5||CallLeg::setRemoteSdp:Remote SDP is set for legC:1.2<br>
    16:39:08.769|.\SLServer.cpp(861)|Log5|MediaServer|MediaServerReporting::SetRemoteParty:[MS210001] C:1.2:Answer received. RTP connection[unsecure]: 192.168.0.8:16426(16427)<br>
    16:39:08.772|.\SLServer.cpp(861)|Log5|MediaServer|MediaServerReporting::SetRemoteParty:[MS210007] C:1.1:Answer provided. Connection(by pass mode): 192.168.0.8:16426(16427)<br>
    16:39:08.774|.\CallCtrl.cpp(885)|Log2||CallCtrl::eek:nLegConnected:[CM503007]: Call(1): Device joined: sip:10@192.168.0.2:5062<br>
    16:39:08.776|.\CallCtrl.cpp(885)|Log2||CallCtrl::eek:nLegConnected:[CM503007]: Call(1): Device joined: sip:10000@192.168.0.8:5062<br>
    16:39:08.888|.\CallLeg.cpp(594)|Log5||CallLeg::eek:nConfirmed:Session 46 of leg C:1.1 is confirmed<br>
    16:39:16.339|.\Call.cpp(1396)|Log2||Call::Terminate:[CM503008]: Call(1): Call is terminated<br>
    16:39:21.670|.\Call.cpp(48)|Log5||??:Currently active calls [none]<br>
    16:39:51.672|.\Call.cpp(48)|Log5||??:Currently active calls [none]<br>
    16:40:23.673|.\Call.cpp(48)|Log5||??:Currently active calls [none]<br>
    16:40:55.675|.\Call.cpp(48)|Log5||??:Currently active calls [none]<br>
    16:41:25.677|.\Call.cpp(48)|Log5||??:Currently active calls [none]<br>
     

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  2. lneblett

    lneblett Well-Known Member

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    You have a number of issues going on.

    Check the license status of your 3CX install, the log indicates that
    ||LoadLicenceInfo:[CM501010]: License Info: Load Failed - Error licence Licence: 27 System: 2 <br>

    There were some other failures showing in the logs as well. I would take care of the 3CX issues first as it may affect all phones and such sooner or later.

    Have you checked to ensure that the 3102 is optimized for your region? BY this I mean the various settings for the physical line and progress tones? Look in the blog for the base 3102 set-up for FXO and then consult your manual or perhaps the Linksys site for the particulars on how to set the line variables for your area.
     
  3. leejor

    leejor Well-Known Member

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    Is 96552202 a number that your PSTN provider will accept, because that is what you are sending. If it isn't, then you may want to review your outbound rules, and how they deal with the deletion of dialled digits. I see that the last called PSTN number shows as 10000, in the Info tab. Very odd, as it should show the 96552202.

    Ah, ha...under Voip to PSTN Gateway set-up the VoIP Caller Default DP should not be 8, that, is causing the 10000 to be sent out on the PSTN line.

    As far as incoming calls. I'm not sure if you have the option of caller ID on the PSTN phone line, but I seem to recall that in Australia it is very similar to that used in North America, sent sometime after the first ring. If that is the case, you'd have to increase the PSTN Answer delay to about 3 seconds, and you might try deleting the * in the PSTN Caller ID pattern. I've never found that to be necessary and I know that it has caused other to have problems in the past (both settings are under the PSTN Tab on the 3102).

    I also see that, in the WAN section of the 3102, you have the Gateway set as 192.168.0.10 (the IP of the 3CX server). this should be set as the IP of your router (Gateway), not the SIP server.

    There are also a few other changes you might want to make later, especially if you plan on using the FXS port. Get the FXO port working first.
     
  4. Baggins

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    Thanks Leejor, I'll try your suggestions and get back with an update.

    If the DP should not be 8 - Should I use say 1? and if so what should I put in 1?
     
  5. leejor

    leejor Well-Known Member

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    All other dialplans (Line 1 tab) are left at the factory default, and are not required to do any manipulation for outgoing calls, so just change it back to 1,( which is also the default). DP 8 should only be used to send calls from the incoming PSTN (Line 1) to 3CX.
     
  6. Baggins

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    Ok sorry for the late reply, I have made those changes Leejor, however now getting an server error?

    Please see logs and new screen grabs. Ive added the router grabs also.

    This is getting frustrating, would I be advised to upgrade to version 11?

    Your help is much appreciated

    Thanks.
     

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  7. leejor

    leejor Well-Known Member

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    You can if you want , but it should not be necessary.

    It looks to be registering, but be sure that, in the trunk settings of 3CX, you have the correct IP and port datafilled for the 3102. You may also want to set a Primary DNS server, usually the IP of your router will suffice. Other than that, I can't see anything out of place. You are using the Internet (blue) RJ-45 socket to connect to your LAN? You don't want to use the yellow one unless you make some additional changes, otherwise you end up using the internal router option of the 3102. I'll go over the settings tomorrow morning again.
     
  8. Baggins

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    Thanks for your reply.

    The ip address of the machine running 3cx is in the range of 127.x.x.x whereas the router is 192.168.x.x - if you take a look at the logs all the extensions for voicemail 99 etc are using the 127.x.x.x range but extension 10, which is my ip phone is 192.168.x.x range. Also I see in my router GUI under connected devices list the 3cx server computer (127.x.x.x) does not show. its as if the router cannot see it? i run some other business networking stuff on this machine and the people who set it up also setup the ip address for this machine to use the 127.x.x.x with the gateway setting of 192.168.0.1 which is my router. is this a possible cause of my problems? i'm not a guru on networks.

    I do recall having an option of 192.168.0.10 or 127.x.x.x on a drop down for the 3cx server when I first installed version 10 on this machine. Perhaps I should have used the 127.x.x.x ip address instead of the 192.168.0.10? Would this matter?

    I have opened all the correct ports and the fireawall checker is a pass on all tests.

    I'll try the DNS setting. I have the port settings and ip of the SPA set correctly.

    Thanks again, will update but may be a couple of days as Im not on-site.
     
  9. leejor

    leejor Well-Known Member

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    If you are talking about the 127.0.0.1 that shows up in the log when 3CX is starting up, then that is not the actual IP of the 3CX server, it is the local host https://en.wikipedia.org/wiki/Localhost .

    Your network appears to be 192.168.0.XXX. From your settings, the PC running 3CX would appear to be 192.168.0.10 (proxy server in the PSTN tab).
    The firewall/port forwarding should have no affect on calls to and from the 3102 as communication between 3CX and the 3102 is internal and would not be passing out to the internet. Just be sure that the IP of the gateway 192.168.0.8 and port 5062 are in the 3CX trunk settings.

    Check your dialplans in the PSTN tab... the only one that should be 8 is the PSTN Default caller DP...change the Line 1 VoIP caller DP back to 1.
     
  10. Baggins

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    ok sorry for the delay, I havent had a chance to get back to you guys.

    I have tried all the suggestions (thanks). The DNS did not help. So I fully uninstalled 3cx ver 10 and did a fresh ver 11 install.

    I configured everything from scratch. provisioned 1 phone (ext 10 - Yealink)

    Configured Spa3102 - registered fine.

    Created outgoing rule to use ext (10000)

    Unfortunately still no luck. I cannot get the system to make any outgoing call on the PSTN line. Incoming still takes forever to activate the call even though the handset is off hook it continues ringing.

    I enabled 'keep alives' in global settings

    I have attached screen shots. Please help with any suggestions. Im ready to give up on 3CX.
     

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  11. leejor

    leejor Well-Known Member

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    A few settings on the 3102 are not what i would use, but i don't think that they are causing major problems. It does not appear that a "factory reset" was done on the gateway before you began configuring it for use on 3CX. It is a good idea to do this as it resets everything back to the defaults, and gives you a clean start.

    The time server is being accessed, another good sign.

    PSTN to SPA Gain is currently set for 3, the default is 0. I find it best to leave it at 0 unless there is a valid reason to change it. Increasing gain can lead to echo issues.

    Registration Expires, does not need to be as often as 360 (seconds). You are forcing it to re-register more often that 3CX requires (1800 seconds), it's not going to really hurt, but is unnecessary traffic on the network.

    I assume you have no immediate plans to use the LAN port on the 3102, it should to be set to Bridge if the internal router is not going to be used. that would allow you to daisy-chain another device, saving a port on your router.

    Even though you are not currently using line 1, the Line 1 VoIP Caller DP is currently set to 8, Dialplan 8 should only be used for the PSTN dialplan, to "route" incoming calls to 3CX.

    The PSTN portion is registered and Voltage is detected on the PSTN phone line, again, a good sign.

    You outbound calling problems may be a result of your outbound rules, there are no digits specified. You are assuming that all calls made by ext. 10 will find their way out on the gateway. You really need to be more specific about what digits you are dialling and perhaps the length. See... http://www.3cx.com/blog/voip-howto/outbound-rules-a-complete-example/

    There might be a similar issue with inbound calls. You don't show how you have the inbound routing set.

    The 3CX logs are your friend, and can quickly point to misconfigurations.
     
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