Thank you Mr.Antony Francis for the reply. Actually where to add that wait(1) in the server?. Please reply in detail about this.<div><br></div><div>Regards,</div><div>Aruns</div><div><br></div><div><br><br><div class="gmail_quote">
On Wed, Dec 30, 2009 at 8:48 PM, Anthony Francis - Handy Networks LLC <span dir="ltr">&lt;<a href="mailto:anthony@handynetworks.com">anthony@handynetworks.com</a>&gt;</span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">









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<p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D">You need to wait at least 1 second on an incoming POTS line for CID
info, add a wait(1) as the first step on incoming connections.</span></p>

<p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D"> </span></p>

<p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D">Thank you and have a  nice day,</span></p>

<p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D">Anthony Francis</span></p><div class="im">

<p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D"> </span></p>

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<p class="MsoNormal"><b><span style="font-size:10.0pt">From:</span></b><span style="font-size:10.0pt">
<a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a>
[mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a>] <b>On Behalf Of </b>Arun
Sasidhar<br>
<b>Sent:</b> Wednesday, December 30, 2009 7:56 AM<br>
<b>To:</b> <a href="mailto:asterisk-users@lists.digium.com" target="_blank">asterisk-users@lists.digium.com</a><br>
<b>Subject:</b> [asterisk-users] CID not working.</span></p>

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<p class="MsoNormal"> </p>

</div><p class="MsoNormal" style="margin-bottom:12.0pt">Hi,</p><div><div></div><div class="h5"><br>
<br>
    I am using asterisk 1.4.28 with freepbx and Wildcard TDM410P
card. Everything is working fine except the caller ID of incoming call from
PSTN line. The phone display is showing &quot;Unknown&quot; when there is an
incoming call.<br>
<br>
<b>My log file showing this while an incoming call on PSTN line:</b><br>
tail -f /var/log/asterisk/full<br>
<br>
[Dec 30 06:36:16] DEBUG[2559] dsp.c: dsp busy pattern set to 0,0<br>
[Dec 30 06:36:16] VERBOSE[2715] logger.c:     -- Starting
simple switch on &#39;DAHDI/1-1&#39;<br>
[Dec 30 06:36:16] VERBOSE[2715] logger.c:     -- Executing
[s@from-pstn:1] Set(&quot;DAHDI/1-1&quot;, &quot;__FROM_DID=s&quot;) in new
stack<br>
[Dec 30 06:36:16] VERBOSE[2715] logger.c:     -- Executing
[s@from-pstn:2] Gosub(&quot;DAHDI/1-1&quot;,
&quot;app-blacklist-check|s|1&quot;) in new stack<br>
[Dec 30 06:36:16] VERBOSE[2715] logger.c:     -- Executing
[s@app-blacklist-check:1] LookupBlacklist(&quot;DAHDI/1-1&quot;, &quot;&quot;)
in new stack<br>
[Dec 30 06:36:16] VERBOSE[2715] logger.c:     -- Executing
[s@app-blacklist-check:2] GotoIf(&quot;DAHDI/1-1&quot;,
&quot;0?blacklisted&quot;) in new stack<br>
[Dec 30 06:36:16] VERBOSE[2715] logger.c:     -- Executing
[s@app-blacklist-check:3] Set(&quot;DAHDI/1-1&quot;, &quot;CALLED_BLACKLIST=1&quot;)
in new stack<br>
[Dec 30 06:36:16] VERBOSE[2715] logger.c:     -- Executing
[s@app-blacklist-check:4] Return(&quot;DAHDI/1-1&quot;, &quot;&quot;) in new
stack<br>
[Dec 30 06:36:16] VERBOSE[2715] logger.c:     -- Executing
[s@from-pstn:3] ExecIf(&quot;DAHDI/1-1&quot;, &quot;1 |Set|CALLERID(name)=&quot;)
in new stack<br>
[Dec 30 06:36:16] VERBOSE[2715] logger.c:     -- Executing
[s@from-pstn:4] Set(&quot;DAHDI/1-1&quot;, &quot;FAX_RX=disabled&quot;) in new
stack<br>
[Dec 30 06:36:16] VERBOSE[2715] logger.c:     -- Executing
[s@from-pstn:5] Set(&quot;DAHDI/1-1&quot;, &quot;__CALLINGPRES_SV=allowed_not_screened&quot;)
in new stack<br>
<br>
<br>
<b>My chan_dahdi.conf file is as like this.</b><br>
vim /etc/asterisk/chan_dahdi.conf<br>
<br>
[channels]<br>
language=en<br>
hanguponpolarityswitch=yes<br>
answeronpolarityswitch=yes<br>
busydetect=yes<br>
busycount=3<br>
callprogress=yes<br>
callerid=asreceived<br>
immediate=yes<br>
cidsignalling=dtmf<br>
cidstart=polarity<br>
;cidstart=ring<br>
useincomingcalleridonzaptransfer=yes<br>
;cidsignalling=bell<br>
; include dahdi extensions defined in FreePBX<br>
#include chan_dahdi_additional.conf<br>
<br>
; XTDM20B Port #1,2 plugged into PSTN<br>
;AMPLABEL:Channel %c - Button %n<br>
<br>
Please help me for fixing this issue. I am from India.<br>
<br>
<br>
Regards,<br>
Aruns<br>
<br>
<br>
<br>
<br>
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