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'auth-id |user-id' für gw1 - gw4

Dieses Thema im Forum "Linksys (VoIP)" wurde erstellt von rolsch, 10 Nov. 2004.

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  1. rolsch

    rolsch Neuer User

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    - soeben vom Sipura-Support bekommen...

    - da bin ich aber mal gespannt...
     
  2. Netview

    Netview IPPF-Promi

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    Beruf:
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    Die Gebete wurden erhört!
     
  3. rolsch

    rolsch Neuer User

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    Hoffentlich funktionierts auch,
    denn der SPA muss sich wohl beim Provider-Wechsel erstmal registrieren
    und ob das so schnell funktioniert, dass ohne Probleme ein abgehender
    Ruf zustande kommt... :?:

    - schade nur, dass ein abgehender Ruf von FXO nach VoIP1 (7-8-3) scheinbar nicht möglich ist...
     
  4. rolsch

    rolsch Neuer User

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    ... der Support hat seine Aussage bestätigt

    Quellenangabe: http://www.sipura.com/Documents/rnote/rn3k-2.0.11g.htm
    ------------------------------------------------------------------------------------------------------
    [/quote]Release Notes for 2.0.11(GWg) -- Sipura Phone Adapter

    SPA-3000 -- 1 Port FXS, 1 Port FXO, 1 Ethernet Interface

    Copyright (C) 2003-2004 Sipura Technology Inc.

    * * * * * * * * * * IMPORTANT * * * * * * * * * * * * * * * * *
    * Use of Proprietary Information and Copyright Notice: *
    * This release note document contains proprietary information *
    * that is to be used only by Sipura Technology customers. *
    * Any unauthorized disclosure, copying, distribution, or use *
    * of this information is prohibited. This restriction includes *
    * ALL Internet based discussion forums, e.g. DSLreports. *
    * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * *

    Bug Fixes
    ===========================================================
    -- since 2.0.11(a)
    1. BUG ID=00230
    When Forward Line 1 to PSTN gateway, SPA does dialog refresh even
    if <VoIP DLG Refresh Intvl> is set to 0

    2. bug ID=231
    If [PSTN Line]<Line Enable> is "no", the
    FXS port relay does not operate properly. If user is calling from
    the FXS port phone to the PSTN while the SPA is powering up, the
    PSTN call will be disconnected when the SPA boots up.
    The problem does not happen if <Line Enable. is "yes" (default)

    3. BUGID=?????
    SPA will treat CWCID ACK digit from the FXS port as digits detected
    at the FXO port and sent AVT tone to the peer for it.


    New Features and Enhancements
    ===========================================================
    -- since 2.0.11(a) --
    1. Added <PSTN Dial Digit Len> option (under [PSTN Line]) to configure
    the on off time of each digit when dialing out to the PSTN line.
    The syntax is <on-time>/, where on-time and off-time are
    specified in seconds with up to 2 deciaml places.
    For example: 0.5/.12
    Default is: .1/.1
    If value is blank, default is used. On off time are limited to
    betwee 0.05s and 3s

    2. Support insertion of pause digits. Any digit that is not in the
    range 0-9*#abcdABCD are treated as a pause digit. We recommend
    to use a ',' (comma) as a pause digit. Each pause digit inserts
    a silence period = <on-time>+ of a digit

    3. Allow [PSTN Line] to use gw1-gw4 to make VoIP calls.
    Hence a PSTN caller can call the SPA-3000, then hop on to
    call one of the configured gw1-gw4, just like Line 1 caller.
    The dial plan syntax is same as for Line 1

    4. Allow mofification of user-id in the From and Contact headers
    when making calls via gw1-gw4, to improve chance of success
    when the gateway (or service) does check these header fields.
    To enable this feature, use the following optional syntax
    in the <Gatway 1-4> parameters: [@]hostname
    where <userid>@ is optional. If @ is not specified, it
    follows the old behavior. Otherwise, the user-id fields in
    the From and Contact headers are changed accordingly.
    If <user-id> = "$usr" (case-insensitive), then the
    corresponding value in <GW1-4 Auth ID> field is used.

    Examples:
    MyAccount@sip.voip_company_a.net:5061
    $usr@company_b.com
    aUserid@some_gateway.com
    fwdnat.pulver.com:5082 (old behavior)

    5. If <NSE Process NSE> is set to "no", the NSE payload type will
    not be included in outbound SDP

    6. Added <DTMF Process AVT> option in [Line 1/2]. If set to "no",
    AVT payload type will not be included in outbound SDP

    7. When sending CAS signal to the phone during CWCID (call-waiting caller-id)
    generation, do not mix in the normal voice audio

    8. Extended the parameter <RTP Packet Size> under [SIP tab] to
    allow specifying a different size for Line 1 an Line 2.
    Syntax: <Line 1 Packet Size>[,]
    If <Line 2 Packet Size> is not given, it will assume the same
    value of <Line 1 Packet Size>. Unit in seconds. Examples:
    .03,.1 ;30 ms for Line 1 and 100 ms for Line 2
    .06 ;60 ms for boht Line 1 and Line 2

    Default is 0.030 (30 ms both Line 1 and Line 2; same as before)
    Notes:
    - If parameter is blank, default value is used
    - If planning to use G723, minimum size is 30ms
    - All size values are limited to between 10 and 160ms
    - All values are rounded to the nearest 10 ms

    9. Added UID1 and UID2 as valid provisioning macro variables, which
    expand to the contents of User_ID[0] and User_ID[1], respectively.

    10. Added two more choices under <DTMF Tx Method>, "InBand+INFO", and
    "AVT+INFO". Selecting either of these methods cause the SPA to
    send SIP INFO message with DTMF events in addition to
    InBand or AVT DTMF signals

    11. Added <DTMF Process INFO> option under Line 1/2, to enable/disable
    playing back of DTMF tone on SIP INFO Messages. Default is "yes".

    12. Relaxed DTMF detection further when waiting for CAS ACK digit from
    the FXS port
     
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