Yes, many people have had this problem.<br>
Check the mailing list archives... I think the newest code has the fix.<br>
Workaround for older versions is to Answer before Dial, but you may
still need the 'r' option to Dial as ringing may stop for the caller
after about 10 seconds.<br><br><div><span class="gmail_quote">On 10/27/05, <b class="gmail_sendername">OTR Comm</b> <<a href="mailto:otrcomm@isp-systems.net">otrcomm@isp-systems.net</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Hello all,<br><br>I have a problem calling into asterisk on a PRI going out to a SIP phone<br>(PRI -> SIP). The calling party does not hear ringing and after about five<br>seconds gets an *All circuits are busy* recording. However, the called SIP
<br>phone does ring, and if the called party answers the phone within a few<br>seconds, the call stays in service.<br><br>CLI messages:<br><br>...<br> -- Accepting call from '9288532045' to '6023432727' on channel 0/23,
<br>span 1<br> -- Executing Dial("Zap/23-1", "SIP/102|20|rt") in new stack<br> -- Called 102<br>!! Don't know how to add an IE High-layer Compatibility (125)<br>!! Unable to add IE 'High-layer Compatibility'
<br> -- SIP/102-935d is ringing<br> -- Channel 0/23, span 1 got hangup request<br> == Spawn extension (incoming, 6023432727, 1) exited non-zero on 'Zap/23-1'<br> -- Hungup 'Zap/23-1'<br>...<br><br>NOTE: There is no problem calling from SIP phone out (SIP -> PRI).
<br><br><br>Any body ever have this problem?<br><br>Thanks,<br>Murrah<br><br>_______________________________________________<br>--Bandwidth and Colocation sponsored by <a href="http://Easynews.com">Easynews.com</a> --<br>
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