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some problems with sip and phone

  • This topic has 1 reply, 1 voice, and was last updated 19 years ago by Omer.
Viewing 2 posts - 1 through 2 (of 2 total)
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  • #29242
    widi
    Guest

    hi folks! i have some problems:

    first of all my config:

    my system is wired with a dlink router, i am using redhat and normal asterisk!
    my phones are aastra 480i.

    now my problems:
    when i try to connect my phone with asterisk, i always get this msg:

    “notice[1903]: chan_sip.c:8802 handle_request_register: registratiopm from ‘1234 ‘ failed for ‘192.168.0.121’”

    my asterisk server ip is: 192.168.0.136
    phone ip: 192.168.0.121

    i am using a tdm400p

    zapata.conf
    ——————————————————————–
    [channels]

    busydetect=1

    busycount=7

    relaxdtmf=yes

    callwaiting=yes

    callwaitingcallerid=yes
    threewaycalling=yes
    transfer=yes
    cancallforward=yes
    usecallerid=yes
    echocancel=yes
    echocancelwhenbridged=yes
    rxgain=0.0
    txgain=0.0
    group=1
    pickupgroup=1-4
    immediate=no
    context=bell
    signalling=fxs_ks
    callerid=asreceived
    channel=2
    context=sip
    group=2
    signalling=fxs_ks
    callerid=”Phone 1″
    channel=3

    ————————–

    sip.conf

    ————————————-

    [phone1]
    type=friend
    host=dynamic
    defaultip=192.168.0.121
    secret=1234
    dtmfmode=rfc2833
    context=sip
    callerid=”Phone 1″

    ———————-

    extensions.conf

    ————————–

    [general]

    [sip]
    exten => 1234,1,Dial(SIP/phone1,20)

    ——————————-

    my aastra.cfg (for the telephone, using tftp)

    ——————————-

    sip registration periso: 300
    sip rtp port: 3000
    sip registrar ip: 192.169.0.136
    sip registrar port: 5060
    sip digit time out: 4
    time server disabled: 1

    sip line1 auth name: 1234
    sip line1 password: 1234
    sip line1 user name: 1234
    sip line1 display name: 1234
    sip line1 screen name: 1234

    ———————–

    when i start asterisk and type in “sip show peers” i get

    :

    name/username/status host dyn nat acl mask port
    phone1/ /unmonitored unspecified d 255.255.255.255 0

    would be nice if anyone hase some awnsers for me and can help me

    thanks

    widi

    #29243
    Omer
    Guest

    Widi,

    you dont have “username” in your sip.conf …..I’m not that much good in asterisk but I think you should use username and then secret to register your phone with asterisk

    thanks

    Omer
    manducks@hotmail.com

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