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Rameshkumar

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Posts posted by Rameshkumar

  1. Accept Redirect is already enabled in the trunk. Below is the trunk settings.

    #Trunk 2
    aadr: 
    analog: false
    bcp: 
    behind_nat: false
    cid_update: 
    cobusy: 500 Line Unavailable
    codec_lock: false
    codecs: 
    codest: 
    cur: 
    dial_extension: 
    dialplan: 2
    dir: 
    dis: false
    domain: 6
    dtmf: false
    dtmf_mode: 
    earlymedia: false
    expires: 3600
    failover: never
    fraction: 128
    from_source: pai
    from_user: 
    glob: 
    global: false
    hcv: 
    hd: 
    hf: {from}
    hpai: {trunk}
    hppi: 
    hpr: {if clip true}id{fi clip true}
    hrpi: 
    hru: {request-uri}
    ht: {to}
    icid: 
    ignore_18x_sdp: false
    interoffice: false
    minimum: 10
    minor: 
    name: Exchange
    outbound_proxy: sip:173.180.98.133:5060;transport=tcp
    pcap: false
    prack: true
    prefix: 
    redirect: true
    reg_account: 
    reg_display: 
    reg_keep: 
    reg_registrar: 173.180.98.133
    reg_user: 
    remote_party: 
    request_timeout: 
    require: 
    rfcrtp: false
    ring180: false
    rtcpxr: false
    rtp_begin: 
    rtp_end: 
    send_email: 
    sip_port: 
    status: 
    tel: true
    trusted: false
    type: gateway
    use_epid: false
    use_history: false
    use_uuid: false
    user_defined_hdr: 
    uuid: c8521892-9803-4b90-8c09-f8a0b482a62e
    wrtc_dest_name: 
    wrtc_dest_number: 
     

  2. I've integrated almost. Only thing when I am calling to exchange I am getting 302 Moved Temporarily error. I think it is because exchange sending back re-invite on TCP port 5062. Please advise.

     

    SIP/2.0 302 Moved Temporarily
    FROM: "210"<sip:2128513569@test.ramesh.net;user=phone>;tag=34533
    TO: <sip:2128513569@test.ramesh.net;user=phone>;epid=DAF82C1CD2;tag=9a7665d8ef
    CSEQ: 12935 INVITE
    CALL-ID: 6346f572@pbx
    VIA: SIP/2.0/TCP 173.115.23.185:53009;branch=z9hG4bK-12735598d07832221e3aefd2df0f01f9;rport
    CONTACT: <sip:7208@MAIL.vicial.internal:5062;user=phone;transport=Tcp>
    CONTENT-LENGTH: 0
    SERVER: RTCC/5.0.0.0 MSExchangeUM/15.01.0845.039

  3. Hi, I am using Vodia 60.0. I've a question. Does multicast only working on the LAN? I've 3 offices and each office has 10 Yealink Phone. Configuration is same on each phone. When one office send a multicast page it only can be hear into the same office. Other 2 offices can not hear anything.

  4. Hi,

     I am using Vodia 5.3.2 version. I want to send the Extension SIP password to user's via welcome email. How can I achieve it? In "email_welcome.htm" template the variable {ssi htmvar password} uses to send web password. What would be the variable to send SIP password.

     

    Thank You

  5. Hi,

     I am using Vodia version 60. Sometimes call stuck in Active calls while actually it has been disconnected. This cause problem if user is the part of agent group, it considered it busy and can not take calls. How can I kill the process like this without restarting the service?

    Call Stuck.png

  6. Hi All,

     

     I am using Vodia 60.0 version. Before Yealink was manufacturing phone with MAC ID's 001565xxxxxx. These phones were sinking with the PBX in real time. Suppose I make a name change on extension on the PBX it replicate on the phone at the same time. Now Yealink manufacturing with MAC ID's 805EC004xxxx. These phones are not sinking with the PBX as 001565xxxxxx MAC range. On 805EC004xxxx range if I need to change name first I need to "Open account for MAC-based provisioning" then change the name and then it will replicate on phone, If I do without open provisioning it does nothing. With  001565xxxxxx range it works perfect without open provisioning. Please advise.

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