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<DIV>Thanks for the response but still no luck.</DIV>
<DIV>I added those two lines just after the [channel] and updated my dial plan
but the result is the same (there is no CallerID):</DIV>
<DIV><FONT face=Arial size=2>Asterisk Ready.<BR> -- Starting
simple switch on 'Zap/1-1'<BR>Jul 26 00:43:34 NOTICE[8867]: chan_zap.c:5367
ss_thread: Got event 2 (Ring/Answered)...<BR> -- Executing
Wait("Zap/1-1", "2") in new stack<BR> -- Executing
NoOp("Zap/1-1", "") in new stack<BR> -- Executing
SetVar("Zap/1-1", "dnis=100") in new stack<BR>...</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>I hope you have some more ideas,
please?</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>BTW, is there anyone using Asterisk with SBC
Telco in the US?</FONT></DIV>
<DIV><FONT face=Arial size=2>
<DIV><FONT face=Arial size=2>Does your asterisk recognize CallerID?</FONT></DIV>
<DIV></FONT><FONT face=Arial size=2></FONT> </DIV></DIV>
<DIV><FONT face=Arial size=2>Thanks,</FONT></DIV>
<DIV><FONT face=Arial size=2>Boris Zolotarev</FONT></DIV>
<DIV><FONT face=Arial size=2><A
href="mailto:boris.zolotarev@gdspartners.com">boris.zolotarev@gdspartners.com</A></FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV>> I have new TDM04B installed and working fine with Asterisk 1.0.5 built
on RedHat 9.<BR>> <BR>> All is working fine except CallerID that
bothers me big time.<BR>> I have several Panasonic and Sony phones and
CallerID works fine with it (when I plug in the <BR>line into phone instead
into<BR>> Asterisk I get CallerID) but fails with Asterisk.<BR>>
<BR>> I am based in California (San Francisco) and my Telco is SBC (<A
href="">http://www.sbc.com/</A>).<BR>> .<BR>> I would really
appreciate if anyone could take a look below at my zapata.conf and
<BR>extensions.conf and let me know what is<BR>> wrong.<BR>>
<BR>> :: zapata.conf ::<BR>> <BR>> [channels]<BR><BR>Try adding
these right here (right after [channels]. Let's see if<BR>that has any impact on
the problem.
<BR>cidsignalling=bell
<BR>cidstart=ring
<BR><BR>> context=default<BR>> switchtype=national<BR>>
signalling=fxs_ks<BR>> usecallerid=yes<BR>> callerid=asreceived<BR>>
hidecallerid=no<BR>> callwaiting=yes<BR>> usecallingpres=yes<BR>>
callwaitingcallerid=yes<BR>> threewaycalling=yes<BR>> transfer=yes<BR>>
cancallforward=yes<BR>> callreturn=yes<BR>> echocancel=yes<BR>>
echocancelwhenbridged=yes<BR>> echotraining=400<BR>> rxgain=0.0<BR>>
txgain=0.0<BR>> group=1<BR>> callgroup=1<BR>> pickupgroup=1<BR>>
immediate=no<BR>> busydetect=yes<BR>> busycount=7<BR>>
musiconhold=default<BR>> faxdetect=both<BR>> context=zap<BR>>
group=1<BR>> channel => 1-4<BR>> <BR>> <BR>> ::
extensions.con ::<BR>> <BR>> I use this part of the code to trace
Asterisk log and check CallerID and CallerIDName.<BR>> <BR>>
[zap]<BR>> exten => s,1,Wait(2)<BR>> exten => s,2,Answer()<BR>>
exten => s,3,SetVar(dnis=100)<BR>> exten =>
s,4,NoOp,${CALLERID}<BR>> exten => s,5,NoOp,${CALLERIDNAME}<BR><BR>For
incoming Zap calls (from the TDM card), you do not need to "answer"<BR>the call
unless you're going into an IVR. If you are simply ringing<BR>a sip phone, do
something like this:<BR><BR>[zap]<BR>exten =>
s,1,NoOp,${CALLERID}
<BR>exten => s,2,Dial(Sip/1234,15) <BR></DIV>
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