i think your problem is how to make the SipGateway send the same
callerId that receives from the PSTN. dont you think? so far i dont
know the SipGateway configuration, sorry.<br>
<br>
best regards<br><br><div><span class="gmail_quote">On 8/25/05, <b class="gmail_sendername">Santiago Vega</b> &lt;<a href="mailto:svega@sisapanama.com">svega@sisapanama.com</a>&gt; wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">











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<p><font face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial;">Hi, asterisk Users, sorry for my bad English… </span></font></p>

<p><font face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial;">im really newbie with this excellent pbx. But I 've a
problem with callerid num when I recive a call from PSTN.</span></font></p>

<p><font face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial;">&nbsp;</span></font></p>

<p><font face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial;">PSTN-&gt; SipGateWay(Welltech3504)-&gt; Asterisk-&gt; BT100</span></font></p>

<p><font face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial;">How can I configure my asterisk to receive the callerid from
callers and not the callerid from the extension of the SipGAteway…</span></font></p>

<p><font face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial;">&nbsp;</span></font></p>

<p><font face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial;">Extension of Gateway (sip.conf)</span></font></p>

<p><font face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial;">[115]</span></font></p>

<p><font face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial;">type=friend&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
; either &quot;friend&quot; (peer+user), &quot;peer&quot; or &quot;user&quot;</span></font></p>

<p><font face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial;">context=sip</span></font></p>

<p><font face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial;">user=115</span></font></p>

<p><font face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial;">host=dynamic</span></font></p>

<p><font face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial;">canreinvite=no</span></font></p>

<p><font face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial;">nat=no&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
; there is not NAT between phone and Asterisk</span></font></p>

<p><font face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial;">disallow=all&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
; need to disallow=all before we can use allow=</span></font></p>

<p><font face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial;">allow=ulaw&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
; Note: In user sections the order of codecs</span></font></p>

<p><font face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial;">allow=alaw</span></font></p>

<p><font face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial;">&nbsp;</span></font></p>

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