patton 4634 only two simultaneous call allowed

Discussion in '3CX Phone System - General' started by luppazzo, Mar 4, 2010.

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  1. luppazzo

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    I have a patton 4634 3bri and 2 ISDN lines in point-to-point configuration

    when receiving calls I get up to 4 simultaneous calls
    when dialing I am only allowed to do up to 2 simultaneous call

    any ide where to cahnge gateway configuration?

    this is the patton status:
    Domain: 192.168.0.2
    Local Address: 192.168.0.7:5060
    Remote Address: (none):0
    Default Server: 192.168.0.2:0;lr
    Early Proceeding: yes
    Hold using zero IP: yes
    Early Connect: no
    Early Disconnect: no
    Call-Transfer: accept & emit
    Call-Reroute: disabled
    AOC-D Information: accept
    Pull-In Externally Looped Calls: yes
    Address-Complete Indication: Accept: clear
    PhoneContext
    Use Calling Remote-Party-ID: no
    Use Called Remote-Party-ID: yes
    Use Privacy: no

    Outgoing Call Parameter Source
    ------------------------------

    From Header
    User: from call URI or E.164 ()
    Host-Port: from call URI or E.164 :)0)

    To Header
    User: from call URI or E.164 ()
    Host-Port: from interface :)0)

    Request URI
    User: fix (10001)
    Host-Port: from To header :)0)
    Target Parameter: none (Target: , Cause: )

    Calling Remote-Party-ID Header
    Header User: from call URI or E.164 ()
    Header Host-Port: from call URI or E.164 :)0)

    Called Remote-Party-ID Header
    Header User: from call URI or E.164 ()
    Header Host-Port: from interface :)0)

    Identity Header
    Header User: from call URI or E.164 ()
    Header Host-Port: from interface :)0)

    Diversion Header
    Header Host-Port: none :)0)

    Incoming Call Parameter Source
    ------------------------------

    Calling Party
    E.164: from Identity header ()
    URI: from Identity header ((invalid))
    Name: from Identity header ()
    Redirecting E.164: none (Target: , Cause: Unknown)

    Called Party
    E.164: from Request-URL ()
    URI: from Remote-Party-ID header ((invalid))
    Name: from Remote-Party-ID header ()
    Redirecting E.164: none (Target: , Cause: Unknown)

    Play Tones: yes

    VoIP Profile: default
    ---------------------

    Used: by 2 module(s)

    Voice Codecs
    G.711 A-law: Rx=20ms Tx=20ms
    G.711 u-law: Rx=20ms Tx=20ms

    Fax Codecs
    T.38 UDP: Rx=1ms Tx=1ms Relay

    Modem Codecs
    Codec Negotiation: enabled

    Dejitter
    Mode: Adaptive
    Max. Delay: 60ms
    Max. Packet Loss: 4/1000
    Shrink Speed: 1
    Grow Step: 1
    Grow Attenuation: 1
    High Pass Filter: enabled
    Post Filter: enabled

    Fax
    Detection: CED Tone
    T.38 High Speed Redundant Packets: 0
    T.38 Low Speed Redundant Packets: 0
    Max. Bit Rate: 14400bps
    Volume: -9.500dB
    Error Correction: enabled
    HDLC: enabled
    Dejitter Max Delay: 200ms

    Modem
    Max. Bit Rate: 14400
    Volume: -9.500dB
    HDLC: enabled
    Dejitter Max Delay: 200ms

    DTMF
    Relay: enabled
    Mute Encoder: enabled

    RTP
    Payload Type NTE: 101
    Payload Type NSE: 100
    Payload Type Clearmode: 97
    Payload Type Redundancy: 255
    Payload Type G.726-32: 2
    Payload Type G.726-32-AAL2: 2

    BCD
    Avtivation: on-rx-activation
    Timeout: 1000
    Traffic Class: local-voice

    SIP Profile: default
    --------------------

    Used: by 2 module(s)
    Autonomous Transitioning: disabled

    Tone-Set Profile: (none)
    Routing Destination: router (IF_SIP_0-precall-service)
    State: Up
    Allow New Calls: yes
    Allow Incoming Calls: yes
    Wait For Name: no
    Wait For Number: no
    Proceeding Timeout: 4000
    Early Alerting Timeout: 0

    Endpoints
    ---------

    Endpoint: IF_SIP_0-00b69fd8
    Domain: 192.168.0.2
    Local Host: 192.168.0.7
    Local Port: 5060
    Remote Host: (none)
    Remote Port: (default)
    Default Server: 192.168.0.2/0 (loose-router)
    Session-Timer Interval: 0s
    Session-Timer Version: 8
    Call-Transfer Version: 5
    Early Proceeding: yes
    Hold using zero IP: yes
    Early Connect: no
    Early Disconnect: no
    Call-Transfer: accept & emit
    Call-Reroute: disabled
    AOC-D Information: accept
    Pull-In Externally Looped Calls: yes
    Address-Complete Indication: Accept: clear
    Phone Context
    Use Calling Remote-Party-ID: no
    Use Called Remote-Party-ID: yes
    Play Tones: yes
    Tone-Set Profile: (none)
    Call Routing Destination: router (IF_SIP_0-precall-service)
    Active State: Disconnected
    Passive State: Idle

    Endpoint: IF_SIP_0-009297b8
    Domain: 192.168.0.2
    Local Host: 192.168.0.7
    Local Port: 5060
    Remote Host: (none)
    Remote Port: (default)
    Default Server: 192.168.0.2/0 (loose-router)
    Session-Timer Interval: 0s
    Session-Timer Version: 8
    Call-Transfer Version: 5
    Early Proceeding: yes
    Hold using zero IP: yes
    Early Connect: no
    Early Disconnect: no
    Call-Transfer: accept & emit
    Call-Reroute: disabled
    AOC-D Information: accept
    Pull-In Externally Looped Calls: yes
    Address-Complete Indication: Accept: clear
    Phone Context
    Use Calling Remote-Party-ID: no
    Use Called Remote-Party-ID: yes
    Play Tones: yes
    Tone-Set Profile: (none)
    Call Routing Destination: router (IF_SIP_0-precall-service)
    Active State: Connected
    Passive State: Idle
     
  2. mfm

    mfm Active Member

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    Hi,

    On the Management console, select the Patton 4634 Gateway, and in the outbound parameters tab kindly ensure they are set to "DevHostPort", if you are unusure kindly tell me what they are.

    RequestLineURI : Host Part
    To : Host Part

    Thanks,
     
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  3. luppazzo

    Joined:
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    thank you for your prompt reply

    I have on the config
    Sip fileds:
    Request Line URI : Host Part "DevHostPort" source address/port
    To : Host Part "DevHostPort" source address/port

    Here's the config file for further check; can you see any other possible issue?
    Another technician suggested the problem was in the pp pmp configuration of ISDN line but I doublechecked with telco and they are all correct.


    #########################################################
    #
    # For SN-4634/3BIS/UI 3-port BRI
    #
    # Author: Kevin Attard Compagno
    # Date: 2008-02-20
    # Contact: kevin@3cx.com
    #
    #########################################################

    webserver port 80 language en

    system

    ic voice 0
    low-bitrate-codec g729

    system
    clock-source 0 0

    profile ppp default

    profile call-progress-tone defaultDialtone
    flush-play-list
    play 1 200 425 0
    pause 2 200
    play 1 600 425 0
    pause 1 1000

    profile call-progress-tone defaultAlertingtone
    flush-play-list
    play 1 1000 425 -7
    pause 2 4000

    profile call-progress-tone defaultBusytone
    flush-play-list
    play 1 200 425 -7
    pause 2 200

    profile call-progress-tone defaultReleasetone
    flush-play-list
    play 1 250 425 -7
    pause 2 250

    profile call-progress-tone defaultCongestiontone
    flush-play-list
    play 1 250 425 -7
    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_WAN
    ipaddress dhcp
    tcp adjust-mss rx mtu
    tcp adjust-mss tx mtu

    interface IF_IP_LAN
    ipaddress 192.168.0.7 255.255.255.0
    tcp adjust-mss rx mtu
    tcp adjust-mss tx mtu

    context cs switch

    mapping-table calling-pi to calling-e164 MP_REM_CLIR
    map restricted to ""

    routing-table called-e164 RT_ISDN_TO_SIP_0
    route T2 dest-interface IF_SIP_0 MP_REM_CLIR

    routing-table called-e164 RT_ISDN_TO_SIP_1
    route T2 dest-interface IF_SIP_1 MP_REM_CLIR

    interface isdn IF_ISDN_0
    route call dest-table RT_ISDN_TO_SIP_0

    interface isdn IF_ISDN_1
    route call dest-table RT_ISDN_TO_SIP_1

    interface sip IF_SIP_0
    bind gateway GW_SIP_0
    %%earlymedia%%
    service default
    route call dest-interface IF_ISDN_0
    remote-party-id called-party
    address-translation outgoing-call request-uri user-part fix 10001 host-part to-header target-param none
    address-translation incoming-call called-e164 request-uri

    interface sip IF_SIP_1
    bind gateway GW_SIP_1
    %%earlymedia%%
    service default
    route call dest-interface IF_ISDN_1
    remote-party-id called-party
    address-translation outgoing-call request-uri user-part fix 10002 host-part to-header target-param none
    address-translation incoming-call called-e164 request-uri

    context cs switch
    no shutdown

    gateway sip GW_SIP_0
    call-signaling-port 5060
    bind interface IF_IP_LAN router

    service default
    domain 192.168.0.2
    defaultserver manual 192.168.0.2 loose-router
    registration-lifetime 300
    registration manual 192.168.0.2
    user 10001 authenticate name 10001 password 10001 default register

    gateway sip GW_SIP_0
    no shutdown

    gateway sip GW_SIP_1
    call-signaling-port 5062
    bind interface IF_IP_LAN router

    service default
    domain 192.168.0.2
    defaultserver manual 192.168.0.2 loose-router
    registration-lifetime 300
    registration manual 192.168.0.2
    user 10002 authenticate name 10002 password 10002 default register

    gateway sip GW_SIP_1
    no shutdown

    port ethernet 0 0
    medium auto
    encapsulation ip
    bind interface IF_IP_WAN router
    no shutdown

    port ethernet 0 1
    medium auto
    encapsulation ip
    bind interface IF_IP_LAN router
    no shutdown

    port bri 0 0
    clock auto
    encapsulation q921

    q921
    protocol pp
    uni-side auto
    encapsulation q931

    q931
    protocol dss1
    uni-side user
    encapsulation cc-isdn
    bind interface IF_ISDN_0 switch

    port bri 0 0
    no shutdown

    port bri 0 1
    clock auto
    encapsulation q921

    q921
    protocol pp
    uni-side auto
    encapsulation q931

    q931
    protocol dss1
    uni-side user
    encapsulation cc-isdn
    bind interface IF_ISDN_1 switch

    port bri 0 1
    no shutdown

    port bri 0 2
    clock auto
    encapsulation q921

    q921
    protocol pp
    uni-side auto
    encapsulation q931

    q931
    protocol dss1
    uni-side net

    port bri 0 2
    shutdown
     
  4. luppazzo

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    can anyone reply to the prevous post?

    thank you
     
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