You really should consider looking at a existing channel driver a good starting point would be to check out the ast_channel_tech and ast_channel_tech_info structures and callbacks.<br><br><br><div><span class="gmail_quote">
On 3/29/07, <b class="gmail_sendername">Muazzam Arslan Bhatti</b> <<a href="mailto:muazzamarslan@com-cept.com">muazzamarslan@com-cept.com</a>> wrote:</span><blockquote class="gmail_quote" style="margin-top: 0; margin-right: 0; margin-bottom: 0; margin-left: 0; margin-left: 0.80ex; border-left-color: #cccccc; border-left-width: 1px; border-left-style: solid; padding-left: 1ex">
<div bgcolor="#ffffff"><div><font face="Arial" size="2">hi,i am new to asterisk usage.i want to write a custom SIP channel in asterisk.where should i start.can anyone explain or point to some tutorial ?.</font></div><div>
</div><div><font face="Arial" size="2">Regards,</font></div><div><font face="Arial" size="2">Arslan</font></div><div> </div></div><br>_______________________________________________<br>--Bandwidth and Colocation provided by
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