<div><span class="gmail_quote">2008/4/25, Kevin P. Fleming <<a href="mailto:kpfleming@digium.com">kpfleming@digium.com</a>>:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Andreas Brodmann wrote:<br> > After chasing a problem I looked at the SIP code (1.4.19)<br> > with a colleague and as far as we understood it, overlapped<br> > dialing on sip trunks for outgoing calls is not supported (yet?).<br>
<br> <br>I have never heard of 'overlapped dialing' on SIP. SIP is always<br> en-block dialing.<br> <br> Can you provide any references for what you are talking about?<br> <br> --<br> Kevin P. Fleming<br> Director of Software Technologies<br>
Digium, Inc. - "The Genuine Asterisk Experience" (TM)</blockquote><div><br>Kevin,<br><br>in sip.conf you can set 'allowoverlap=yes'<br><br>If a phone is configured accordingly (early dial) it will send an INVITE request<br>
for each key a user presses. e.g. 1@asterisk, 11@asterisk and then 111@asterisk.<br>For each incomplete INVITE asterisk will return 484 "Number incomplete" until the<br>client sends a number which is complete (e.g. matches a pattern).<br>
<br>This works fine until a client tries to call a number that asterisk reaches via a sip trunk to<br>another pbx (or carrier), whereas the length of the number is unknown:<br>// e.g. International Calls<br>e.g. exten => 000!,1,Dial(SIP/${EXTEN:1}@carrier,120)<br>
<br>in this case, asterisk will send an INVITE to the carrier after the first 3 zeros. The answer<br>from the carrier will be 484 (number incomplete). Instead of forwarding this response to the<br>phone asterisk will end the call -> congestion.<br>
<br>---<br>Andreas Brodmann<br><br><br></div></div>