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<DIV dir=ltr align=left><SPAN class=351264601-18082006><FONT face=Arial
color=#0000ff size=2>What's the Dial command being used to pass the call to the
Softphones? </FONT></SPAN></DIV><BR>
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<FONT face=Tahoma size=2><B>From:</B> asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <B>On Behalf Of </B>Crazy
Boy<BR><B>Sent:</B> Wednesday, August 16, 2006 3:23 AM<BR><B>To:</B>
radamson@routers.com; Asterisk Users Mailing List - Non-Commercial
Discussion<BR><B>Subject:</B> Re: [asterisk-users] CallerID is not displaying
for my incoming calls<BR></FONT><BR></DIV>
<DIV></DIV>Hi,<BR><BR>As you said, I have changed my configurations. But,
callerid is not displaying. What I have to do? Please tell
me.<BR><BR>Thanks&Regards,<BR>Chandra.<BR><BR><B><I>Rich Adamson
<radamson@routers.com></I></B> wrote:
<BLOCKQUOTE class=replbq
style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: rgb(16,16,255) 2px solid">Crazy
Boy wrote:<BR>> Hi Friends,<BR>> <BR>> We have installed Asterisk
with Digium 04B card (4 FXO ports). Now, I <BR>> have connected my PSTN
line directly to first port. I am making outgoing <BR>> calls and
receiving incoming calls successfully through my Asterisk. The <BR>>
problem is: When I am receiving a call from outside (PSTN), I am not
<BR>> getting the callerid number and getting callerid as "Asterisk" in
my <BR>> softphones (XLite). Here I am giving my configuration
files.<BR>> <BR>> zaptel.conf file contents:<BR>> <BR>> loadzone
= us<BR>> defaultzone=us<BR>> fxsks=1-4<BR>> <BR>> zapata.conf
file contents:<BR>> <BR>> [channels]<BR>> context=incoming<BR>>
signalling=fxs_ks<BR>> busydetect=1<BR>> busycount=7<BR>>
relaxdtmf=yes<BR>> callwaiting=yes<BR>>
callwaitingcallerid=yes<BR>> threewaycalling=yes<BR>>
cancallforward=yes<BR>> echocancelwhenbridged=yes<BR>>
rxgain=0.0<BR>> txgain=0.0<BR>> callerid=asreceived<BR>>
language=en<BR>> usecallerid=yes<BR>> hidecallerid=no<BR>>
echocancel=yes<BR>> transfer=yes<BR>> immediate=no<BR>>
group=1<BR>> callgroup=9<BR>> pickupgroup=9<BR>> channel =>
1<BR><BR>The above entries appear to be reasonable and correct. If you have
not <BR>properly set rxgain and txgain, it "could" impact callerid. If those
<BR>gains are too high/low, asterisk will not properly read the callerid
<BR>data when sent to you.<BR><BR>> extensions.conf file
contents:<BR>> <BR>> [incoming]<BR>> exten => s,1,Answer<BR>>
exten => s,2,SetMusicOnHold(default)<BR>> exten =>
s,3,DigitTimeout,5<BR>> exten => s,4,ResponseTimeout,10<BR>> exten
=> s,5,Background(/tmp/virg2)<BR>> exten => s,6,Goto(s,1)<BR>>
include => leader<BR><BR>> Got event 18 (Ring Begin)...<BR>> Aug 14
14:11:58 WARNING[26744]: pbx.c:5869 pbx_builtin_dtimeout: <BR>>
DigitTimeout is deprecated, please use Set(TIMEOUT(digit)=timeout)
instead.<BR>> Aug 14 14:11:58 WARNING[26744]: pbx.c:5845
pbx_builtin_rtimeout: <BR>> ResponseTimeout is deprecated, please use
Set(TIMEOUT(response)=timeout) <BR>> instead.<BR><BR>The above two
WARNING statements are telling you that either you are <BR>copying those
exten=> statements from someone's old config files, or, <BR>you haven't
read the asterisk documentation. The message is telling you <BR>that your
statement "exten => s,3,DigitTimeout,5" should be replaced <BR>with the
Set(TIMEOUT(digit)=timeout) syntax. Your statements are still <BR>executing
properly today, but the next time you upgrade asterisk code, <BR>they are
likely to fail due to the old syntax not being supported.<BR><BR>Try 'show
function TIMEOUT' from your CLI and read it.<BR><BR>> What I have to do
to display the PSTN caller number on my softphones? <BR>> Please tell me.
Looking forward to your response. Thank
you.<BR><BR>_______________________________________________<BR>--Bandwidth
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