Patton 4112 - Silence on line.

Discussion in '3CX Phone System - General' started by eihli, Mar 2, 2011.

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  1. eihli

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    We have a Patton 4112 set up using the generated config. When a call is placed with a 9 prefix, it is routed through the analog line connected to the Patton. It looks like things are connecting fine in the server activity log, but we don't hear any sound after the call is placed and the number we are dialing never rings.

    I'll post what I have. Let me know what other info is needed to troubleshoot this. Thanks ahead of time for any help.

    Here is the server log:
    Silence....
    17:03:49.102 Currently active calls - 1: [1264]
    17:03:17.100 Currently active calls - 1: [1264]
    17:02:47.080 Currently active calls - 1: [1264]
    17:02:15.079 Currently active calls - 1: [1264]
    17:01:45.077 Currently active calls - 1: [1264]
    17:01:22.312 [CM503007]: Call(1264): Device joined: sip:10003@192.168.1.3:5060
    17:01:22.310 [CM503007]: Call(1264): Device joined: sip:305@64.91.7.121:5060
    17:01:22.310 [CM503022]: Call(1264): Call recording is started, audio file: E:\3CX Recordings\305\[Security Office]_305-93041571_20110302170122(1264).wav
    17:01:22.299 [CM505002]: Gateway:[Patton Backup] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Patton SN4112 JO EUI 00A0BA059D79 R5.6 2010-07-15 H323 SIP FXS FXO M5T SIP Stack/4.0.29.29] PBX contact: [sip:10003@192.168.1.106:5060]
    17:01:22.299 [CM503002]: Call(1264): Alerting sip:10003@192.168.1.3:5060
    17:01:18.300 [CM503025]: Call(1264): Calling PSTNline:13373041571@(Ln.10003@Patton Backup)@[Dev:sip:10003@192.168.1.3:5060]
    17:01:18.271 [CM503004]: Call(1264): Route 1: PSTNline:13373041571@(Ln.10003@Patton Backup)@[Dev:sip:10003@192.168.1.3:5060,Dev:sip:10004@192.168.1.3:5062]
    17:01:18.261 [CM503010]: Making route(s) to <sip:93041571@66.112.95.17>
    17:01:18.259 [CM505001]: Ext.305: Device info: Device Identified: [Man: Linksys;Mod: SPA Series;Rev: General] Capabilities:[reinvite, no-replaces, able-no-sdp, recvonly] UserAgent: [Linksys/SPA942-6.1.5(a)] PBX contact: [sip:305@66.112.95.17:5060]
    17:01:18.250 [CM503001]: Call(1264): Incoming call from Ext.305 to <sip:93041571@66.112.95.17>

    Here is the config:
    #----------------------------------------------------------------#
    # #
    # SN4112/JO/EUI #
    # R5.6 2010-07-15 H323 SIP FXS FXO #
    # 2011-03-02T23:28:36 #
    # SN/00A0BA059D79 #
    # Generated configuration file #
    # #
    #----------------------------------------------------------------#

    cli version 3.20
    clock local default-offset +00:00
    webserver port 80 language en

    system

    ic voice 0
    low-bitrate-codec g729

    profile ppp default

    profile call-progress-tone defaultDialtone
    play 1 1000 350 -13 440 -13

    profile call-progress-tone defaultAlertingtone
    play 1 1000 440 -19 480 -19
    pause 2 3000

    profile call-progress-tone defaultBusytone
    play 1 500 480 -24 620 -24
    pause 2 500

    profile call-progress-tone defaultReleasetone
    play 1 250 480 -24 620 -24
    pause 2 250

    profile call-progress-tone defaultCongestiontone
    play 1 250 480 -24 620 -24
    pause 2 250

    profile tone-set default

    profile voip default
    codec 1 g711alaw64k rx-length 20 tx-length 20
    codec 2 g711ulaw64k rx-length 20 tx-length 20
    fax transmission 1 relay t38-udp

    profile pstn default

    profile sip default
    no autonomous-transitioning

    profile aaa default
    method 1 local
    method 2 none

    context ip router

    interface IF_IP_LAN
    ipaddress 192.168.1.3 255.255.255.0
    tcp adjust-mss rx mtu
    tcp adjust-mss tx mtu

    context cs switch
    digit-collection timeout 2

    interface sip IF_SIP_0
    bind context sip-gateway GW_SIP_0
    route call dest-interface IF_FXO_0
    remote 192.168.1.106 5060
    early-connect
    early-disconnect
    address-translation outgoing-call request-uri user-part fix 10003 host-part to-header target-param none

    interface sip IF_SIP_1
    bind context sip-gateway GW_SIP_1
    route call dest-interface IF_FXO_1
    remote 192.168.1.106 5060
    early-connect
    early-disconnect
    address-translation outgoing-call request-uri user-part fix 10004 host-part to-header target-param none

    interface fxo IF_FXO_0
    route call dest-interface IF_SIP_0
    loop-break-duration min 60 max 1000
    disconnect-signal loop-break
    disconnect-signal busy-tone
    ring-number on-caller-id
    dial-after timeout 1
    mute-dialing

    interface fxo IF_FXO_1
    route call dest-interface IF_SIP_1
    loop-break-duration min 60 max 1000
    disconnect-signal loop-break
    disconnect-signal busy-tone
    ring-number on-caller-id
    dial-after timeout 1
    mute-dialing

    context cs switch
    no shutdown

    authentication-service AS_ALL_LINES
    realm 1 3CXPhoneSystem
    username 10003 password xG9fBlsMRfgO35Hi6/G3jA== encrypted
    username 10004 password fzoqYA6NNosfsq6Eef+QDw== encrypted

    location-service LS_10003
    domain 1 192.168.1.106

    identity-group default

    authentication outbound
    authenticate 1 authentication-service AS_ALL_LINES username 10003

    identity 10003

    authentication outbound
    authenticate 1 authentication-service AS_ALL_LINES

    registration outbound
    registrar 192.168.1.106 5060
    lifetime 300
    register auto

    location-service LS_10004
    domain 1 192.168.1.106

    identity-group default

    authentication outbound
    authenticate 1 authentication-service AS_ALL_LINES username 10004

    identity 10004

    authentication outbound
    authenticate 1 authentication-service AS_ALL_LINES

    registration outbound
    registrar 192.168.1.106 5060
    lifetime 300
    register auto

    context sip-gateway GW_SIP_0

    interface LAN
    bind interface IF_IP_LAN context router port 5060

    context sip-gateway GW_SIP_0
    bind location-service LS_10003
    no shutdown

    context sip-gateway GW_SIP_1

    interface LAN
    bind interface IF_IP_LAN context router port 5062

    context sip-gateway GW_SIP_1
    bind location-service LS_10004
    no shutdown

    port ethernet 0 0
    medium 10 half
    encapsulation ip
    bind interface IF_IP_LAN router
    no shutdown

    port fxo 0 0
    use profile fxo us
    encapsulation cc-fxo
    bind interface IF_FXO_0 switch
    no shutdown

    port fxo 0 1
    use profile fxo us
    encapsulation cc-fxo
    bind interface IF_FXO_1 switch
    no shutdown
     
  2. ccomley

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    Did you ever resolve this?

    I'm setting up a 4112 today.

    1) I got no dialling in or out - then I found the Patton was showing the line as "down" with the supplied calble. Swapped to a workign cable nicked form the back of a phone and it was fine.

    But

    2) Now I can dial in or out - but when the call connects, no audio passes. If this was a SIP call I'd suspect STUN settings or similar, but the Patton is on the same LAN segment as the 3CX...
     
  3. ccomley

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    Just realised I never came back to this,

    We *did* solve it - we realised there was no Default Gateway setting in the Patton and it wasn't getting one from the downloaded config. I realise it doesn't matter for 90% of situations but in our case the phones themselves are NOT on the same LAN range as the PBX and the Patton. Add a gateway to give the Patton a route to the phones and Bingo - good as you could ask for.
     
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