Unistim crashes on Elastix 2.0 Asterisk 1.6.2.13

Discussion in 'General' started by palillo, Mar 24, 2011.

  1. palillo

    Joined:
    Aug 27, 2008
    Messages:
    23
    Likes Received:
    0
    Hi,

    I have installed a few weeks ago an Elastix 2.0 server on IBM x3200.

    The thing here is that we are using SIP phones and Unistim (Nortel Phones).

    I managed to make Nortel phones to work properly, however, if they try to transfer a call using the telephone button to a SIP extension, Asterisk dies!!

    I guess chan_unistim is broken with this functionality.

    Here is how I define unistim extensions, if you see anything wrong, plese let me know.

    unistim.conf
    Code:
    [general]
    port=5000                    ; UDP port
    
    [nortel3843]                     ; name of the device
    device=00140D0038ED
    maintext0="Fatima Rivera"  ; default = "Welcome", 24 characters max
    callerid="Fatima Rivera" <3843>
    context=from-internal           ; context, default="default"
    callgroup=1
    pickupgroup=1
    language=es
    qualify=yes
    
    mailbox=3843@device                ; Specify the mailbox number. Used by Message Waiting Indication
    linelabel="3843"         ; Softkey label for the next line=> entry, 9 char max.
    rtp_port=10000              ; RTP port used by the phone, default = 10000. RTCP = rtp_port+1
    rtp_method=3                ; If you don't have sound, you can try 1, 2 or 3, default = 0
    status_method=0             ; If you don't see status text, try 1, default = 0
    titledefault=TimeZone Americas/Caracas       ; default = "TimeZone (your time zone)". 12 characters max
    extension=line              ; Add an extension into the dialplan. Only valid in context specified previously.
                                ; none=don't add (default), ask=prompt user, line=use the line number
    dateformat=1                ; 0 = month/day, 1 (default) = day/month
    timeformat=0                ; 0 = 0:00am ; 1 (default) = 0h00, 2 = 0:00
    contrast=8                  ; define the contrast of the LCD. From 0 to 15. Default = 8
    country=us                  ; country (ccTLD) for dial tone frequency. See README, default = us
    ringvolume=3                ; ring volume : 0,1,2,3, can be overrided by Dial(), default = 2
    ring                 ; ring style : 0 to 7, can be overrided by Dial(),default=3
    callhistory=1               ; 0 = disable, 1 = enable call history, default = 1
    line => 3843                 ; Only one line by device is currently supported.
                                 ; Beware ! only bookmark and softkey entries are allowed after line=>
    bookmark=Buzon Voz@*97@54     ; 54 shows a mailbox icon. See #define FAV_ICON_ for other values (32 to 63)
    
    If you find something or know how to avoid this, I will greatly appreciate your comments.

    Best regards.
    Andres.
     
  2. fmvillares

    Joined:
    Sep 8, 2007
    Messages:
    1,785
    Likes Received:
    0
    as unistim is a privative protocol reverse engineered to make this module it certainly would have errors...u need to seek the changelog in newer asterisk versions to see if the bug is fixed...and recompile it into elastix
     
  3. palillo

    Joined:
    Aug 27, 2008
    Messages:
    23
    Likes Received:
    0
    Thanks Fernando,

    I will do just that. If I find something, I will post it...

    Saludos.
     
  4. sir.gerard

    Joined:
    Jan 7, 2011
    Messages:
    4
    Likes Received:
    0
    Hi, I did read your post.. and i'm interesting about what solutions you found to fix that issue, already i used the 2004ip phone in my Elastix 1.6, but i'm going to migrate to version 2.0.3,

    could you tellme your comments please?

    Thanks in advance...


    palillo escribió:
     
  5. palillo

    Joined:
    Aug 27, 2008
    Messages:
    23
    Likes Received:
    0
    Hi,

    My customer is still using the original version. When they transfer calls, Asterisk crashes and restarts. We suggested customer to avoid using the transfer function in Unistim phones and recommended to replace with SIP terminals.

    Unfortunately, customer does not want to invest money in debugging and research so we gave up in finding a solution. This is the only customer I have with Unistim devices.

    I read somewhere that the Unistim code is being updated and has a new mantainer,it would desirable to install an Asterisk latest build 1.8 or newer and give it a try.

    I guess the latest Elastix distro has the unistim buggy code found on 1.6.x branches...

    I hope you have luck, If I find something, I will post immediately.

    Regards.
     
  6. sir.gerard

    Joined:
    Jan 7, 2011
    Messages:
    4
    Likes Received:
    0
    Thanks for your reply, i hope that unistim debbugin will release soon. if I know about some solution i will lwt you know. Regards.
     

Share This Page