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Outbound call using PSTN 3102

Discussion in '3CX Phone System - General' started by leosa, Nov 7, 2008.

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  1. leosa

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    Hi,

    Im having problems with outbound calls throught PSTN on a SPA3102 (as I browsed the forum, the boring SPA...).

    Our setup includes two lines, one VoIP and one PSTN. We created the outbound rules this way:
    - Numbers starting with "0" are routed on VoIP;
    - Starting with "9" throught the PSTN.

    Below is a dump from server status to help the experts assist me. As we can see, the call is routed correctly, the rules are respected but still indicates "User not found".

    What am I missing?

    Best regards,
    Leo

    ---- Dump ----

    18:26:07.237 Call::Terminate [CM503008]: Call(98): Call is terminated
    18:26:07.177 Call::RouteFailed [CM503015]: Call(98): Attempt to reach "932244000"[sip:932244000@10.0.1.198] failed. Reason: Not Found
    18:26:07.177 CallLeg::eek:nFailure [CM503003]: Call(98): Call to sip:32244000@10.0.1.197:5060 has failed; Cause: 404 Not Found; from IP:10.0.1.197:5060
    18:26:07.047 CallCtrl::eek:nSelectRouteReq [CM503004]: Call(98): Calling: PSTNline:32244000@(Ln.10001@PSTN)@[Dev:sip:10001@10.0.1.197:5061]
    18:26:07.007 CallCtrl::eek:nSelectRouteReq [CM503010]: Making route(s) to "932244000"[sip:932244000@10.0.1.198]
    18:26:06.997 Extension::printEndpointInfo [CM505001]: Ext.10: Device info: Device Identified: [Man: Counterpath;Mod: eyeBeam;Rev: General] Capabilities:[reinvite, no-replaces, unable-no-sdp, recvonly] UserAgent: [eyeBeam release 1009r stamp 38169] Transport: [sip:10.0.1.198:5060]
    18:26:06.987 CallCtrl::eek:nIncomingCall [CM503001]: Call(98): Incoming call from Ext.10 to "932244000"[sip:932244000@10.0.1.198]

    Update: I did what is suggested by renaissance on topic "SPA 3102 Outgoing" and the error changed to:

    19:11:28.209 Call::Terminate [CM503008]: Call(8): Call is terminated
    19:11:28.189 Call::RouteFailed [CM503015]: Call(8): Attempt to reach "932244000"[sip:932244000@10.0.1.198] failed. Reason: Not Registered
    19:11:28.189 Call::RouteFailed [CM503015]: Call(8): Attempt to reach "932244000"[sip:932244000@10.0.1.198] failed. Reason: Not Registered
    19:11:28.189 CallCtrl::eek:nSelectRouteReq [CM503016]: Call(8): Target is not registered: PSTNline:32244000 dialed on (AnyLine@PSTN)
    19:11:28.189 Line::Line [CM303003]: There are no available outbound lines on gateway PSTN at this time.
    19:11:28.129 CallCtrl::eek:nSelectRouteReq [CM503010]: Making route(s) to "932244000"[sip:932244000@10.0.1.198]
    19:11:28.119 Extension::printEndpointInfo [CM505001]: Ext.10: Device info: Device Identified: [Man: Counterpath;Mod: eyeBeam;Rev: General] Capabilities:[reinvite, no-replaces, unable-no-sdp, recvonly] UserAgent: [eyeBeam release 1009r stamp 38169] Transport: [sip:10.0.1.198:5060]
     
  2. galal202

    galal202 New Member

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    Hello And welcome,
    Change pstn port from 5060 to 5061 on server and spa3102
    if it does not work reset to factory set and reconfig it again as the following

    Hard ware Version must be updated to 5.1.7(GW)
    From linksys site

    dial **** to enter configuration menu
    dial 73738# to reset to factory set
    dial 1 to confirm
    Hangup (will restart)

    unplug pstn line now
    dial **** to enter configuration menu
    dial 110# to know your web IP

    to enable web control
    dial 7932# option 1 then 1 to save
    plug pstn line now

    click admin/advanced
    Click ROUTER/WAN SETUP

    Internet Connection Settings
    Change Connection Type: to static ip


    Static IP Settings
    : set your network parameters Static IP:,NetMask:,Gateway

    set dns provided from your ISP
    set ntp to ntp server (option)

    in pstn LINE

    SIP Settings
    leave SIP Port:5061 as it

    Proxy and Registration
    proxy:your server ip
    register expires:300

    Subscriber Information
    set user id and password

    Audio Configuration
    Preferred Codec: g729

    Dial Plans
    Dial Plan 8: (S0<:xxxxx>)
    which xxxxx= 3cx virtual number


    PSTN-To-VoIP Gateway Setup
    PSTN Ring Thru Line 1:No
    PSTN CID Number Prefix:eek:ption
    PSTN CID For VoIP CID: yes
    PSTN Caller Default DP: 8


    FXO Timer Values (sec)
    PSTN Answer Delay:From 4 to 6

    PSTN Disconnect Detection
    Detect PSTN Long Silence
    PSTN Long Silence Duration:300

    International Control
    Line-In-Use Voltage: 10
    SPA To PSTN Gain: 15
    PSTN To SPA Gain: 15


    In LINE1

    Proxy and Registration
    proxy
    register expires:300

    Subscriber Information
    set user id and password

    Audio Configuration
    Preferred Codec: g729

    Option to connect from remote to 3cx server

    In VOICE/SIp
    NAT Support Parameters

    STUN Enable: yes
    STUN Server: stun.fwdnet.net

    In pstn line
    Nat settings
    NAT Mapping Enable: yes

    In Line 1
    Nat settings
    NAT Mapping Enable: yes
     
  3. leosa

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    Hi galal202,

    My friend, thank you very much!

    I followed your instructions and now we're able to dial with both outbound rules.

    3CX is a great piece of software and its community is a huge part of it.

    Thanks again,
    Leo
     
  4. galal202

    galal202 New Member

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    Hello Leo,
    You're welcome
     
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