Caller ID with A1200P

Discussion in 'Gateways' started by BEYOND REALITY, Nov 15, 2010.

  1. BEYOND REALITY

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    Hi,

    I've installed OpenVox A1200P 2 FXO, 10 FXS
    for the testing purpose, I've installed PSTN to FXO and installed three extensions
    1 SIP ext. using Yealink phone
    1 SIP ext. using X-lite shofphone
    1 ZAP ext. using noraml analogh phone.

    It works fine with me when trying to call from one ext. to another.
    It works fine when trying to call outside world.
    It receive the outside calls but ti shows "Unkown caller"

    I connnect the analoug phone to the PSTN directly and when I make a call to the phone line I was able to see the caller ID.

    What changes I've to do to activate the caller ID through Elstix and openvox A1200P.

    Thanks
     
  2. lisa.gao

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    Hi,
    1) Please ask your telecom operator to confirm the cidsignalling and cidstart.
    Then modify /etc/asterisk/chan_dahdi.conf like this:
    usecallerid=yes
    hidecallerid=no
    callerid=asreceived
    cidsignalling=
    cidstart=

    2)Please use this dialplan to have a test.
    [from-pstn]
    exten=>s,1,NoOp(${CALLERID(ALL)})
    exten=>s,n,Hangup()
     
  3. BEYOND REALITY

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    Hi,

    Regarding the telecom, I've tested the analog phone directly with the line and it shows the callerID.

    and when I called the telecom, and they know nothing about cidstart or cid signalling

    do you have any values that I can try?

    I didn't understand this

    2)Please use this dialplan to have a test.
    [from-pstn]
    exten=>s,1,NoOp(${CALLERID(ALL)})
    exten=>s,n,Hangup()

    ========

    I've also one issue is that when I call the phone from outside, it takes two rings until the phone begin ringing and after I hang up they will continue ringing for two addiational rings, that's mean it delaying the rings for two rings "around 10 seconds"

    http://www.elastix.org/en/component/kun ... calls.html
     
  4. lisa.gao

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    Hi,
    Generally,cidstart should be ring or polarity,cidsignalling should be dtmf,bell or v23.
    For the dialplan,it's a inbound route.
    Please try this one. You should add it to the bottom of /etc/asterisk/extensions.conf
    [from-opvx]
    exten=>s,1,NoOp(${CALLERID(ALL)})
    exten=>s,n,Hangup()

    Edit dahdi-channels.conf,set context of the channel which you want to test.

    ; Span 1: OPVXA1200/0 "OpenVox A1200P Board 1" (MASTER)
    ;;; line="1 OPVXA1200/0/0"
    signalling=fxs_ks
    callerid=asreceived
    group=0
    context=from-opvx
    channel => 1
    context=default

    [from-opvx] in extensions.conf and context in dahdi-channels.conf should be the same.
     
  5. BEYOND REALITY

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    Hi,

    I've tried all combination of cidsignalling and cidstart, one of them the phone didn't ring when I put it which is
    cidsinalling=dtmf
    cidstart=ring

    also, when I tried to change context=from-pstn to context=from-opvx also the phoen didn't ring.

    I still not able to get the callerID

    Any more suggestion?

    Also, I want to add that there is two rings delay until the internal phone begin ringing
     
  6. lisa.gao

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    Hi,
    Please delete the inbound route in web page.Then follow the steps:

    1) You should add it to the bottom of /etc/asterisk/extensions.conf
    [from-opvx]
    exten=>s,1,NoOp(${CALLERID(ALL)})
    exten=>s,n,Hangup()

    2) Edit dahdi-channels.conf,set context of the channel which you want to test.

    ; Span 1: OPVXA1200/0 "OpenVox A1200P Board 1" (MASTER)
    ;;; line="1 OPVXA1200/0/0"
    signalling=fxs_ks
    callerid=asreceived
    group=0
    context=from-opvx
    channel => 1
    context=default

    [from-opvx] in extensions.conf and context in dahdi-channels.conf should be the same.

    If you don't know how to set it. You can send SSH and FXO number to my email: lisa.gao@openvox.cn
    I can check for you.
     

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