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"><<a href="mailto:anthony@handynetworks.com">anthony@handynetworks.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
<div lang="EN-US" link="blue" vlink="purple">
<div>
<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>
<div style="border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in">
<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>
</div>
<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 "Unknown" 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 'DAHDI/1-1'<br>
[Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing
[s@from-pstn:1] Set("DAHDI/1-1", "__FROM_DID=s") in new
stack<br>
[Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing
[s@from-pstn:2] Gosub("DAHDI/1-1",
"app-blacklist-check|s|1") in new stack<br>
[Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing
[s@app-blacklist-check:1] LookupBlacklist("DAHDI/1-1", "")
in new stack<br>
[Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing
[s@app-blacklist-check:2] GotoIf("DAHDI/1-1",
"0?blacklisted") in new stack<br>
[Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing
[s@app-blacklist-check:3] Set("DAHDI/1-1", "CALLED_BLACKLIST=1")
in new stack<br>
[Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing
[s@app-blacklist-check:4] Return("DAHDI/1-1", "") in new
stack<br>
[Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing
[s@from-pstn:3] ExecIf("DAHDI/1-1", "1 |Set|CALLERID(name)=")
in new stack<br>
[Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing
[s@from-pstn:4] Set("DAHDI/1-1", "FAX_RX=disabled") in new
stack<br>
[Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing
[s@from-pstn:5] Set("DAHDI/1-1", "__CALLINGPRES_SV=allowed_not_screened")
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>
</div></div><p></p>
</div>
</div>
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System Administrator.<br>Cabot Solutions<br><a href="http://www.cabotsolutions.com">www.cabotsolutions.com</a><br>
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