How to rewrite CID name + number?

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Asterisk Users 5 Comments

Hello

I use the Linksys 3102 to connect Asterisk to a POTS line, and XLite
on XP as an SIP client:

http://img694.imageshack.us/img694/1421/3102asteriskxlitecid.png

The problem is that by default, Asterisk doesn’t rewrite the CID name
+ number in incoming calls, so that XLite displays whatever name I
used in the 3102 and the extension the 3102 uses to register with
Asterisk.

How can I tell Asterisk to rewrite the CID name + number so that XLite
displays the actual caller’s information instead?

Since the 3102 does display the real caller information in its Voice >
Info page, I assume Asterisk is getting this information and could
rewrite the CID information before ringing XLite, but I don’t know how
to do this.

Thank you.

5 thoughts on - How to rewrite CID name + number?

  • Here how I changed my information calling an xlite client from a polycom
    501.
    Sipuser = xlite
    144 = polycom
    Exten => 145,1,set(CALLERID(num)=5551212)
    Exten => 145,n,set(CALLERID(name)=JOES POOL HALL)
    Exten => 145,n,Dial(SIP/sipuser,20,m)

  • On Mon, 6 Dec 2010 10:15:34 -0600, “Danny Nicholas”
    wrote:

    Thanks Danny, but what I need, is to get the real caller ID as sent by
    the telco and available in the Linksys. By default, Asterisk send the
    Linksys name and extension instead of the original caller ID.

    I’m using 1.4.4, but don’t see anything displayed while in the console
    and calling into the 3102:

    ====
    exten => 6011,1,NoOp(${CALLERID(all)})
    exten => 6011,n,Dial(SIP/6011)
    ====

    Any idea why Asterisk shows nothing, and how to retrieve the original
    CID information?

    Thank you.

  • On Mon, 06 Dec 2010 20:03:03 +0100, Gilles
    wrote:

    Sorry about that, I forgot that the console had to be started in
    verbose mode for NoOp() to display data:

    ip04*CLI>

  • On Mon, 6 Dec 2010 13:39:33 -0600, “Danny Nicholas”
    wrote:

    Thanks Danny, and sorry for the trouble: I was paying so much
    attention to the wealth of locale/regional settings in the Linksys…
    that I didn’t think about checking the default sip.conf:

    ===========
    [6013]
    type=friend
    context=default
    host=dynamic
    secret=6013
    canreinvite=no
    ;Argh!
    callerid=6013
    disallow=all
    allow=ulaw,alaw
    qualify=yes
    ===========

    Commenting out the “callerid=6013” line and reloading Asterisk did the
    trick 🙂

    Thanks again.