<html><head><meta http-equiv="content-type" content="text/html; charset=utf-8"></head><body dir="auto">I don’t know the answer, but thought of this.. <a href="http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html/asterisk-book.html">http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html/asterisk-book.html</a><br><br><div id="AppleMailSignature"><div>Thanks,</div><div>Bob</div><div>K6ECM</div><div>73</div><div><br></div></div><div><br>On Dec 1, 2017, at 10:02 AM, Michael Lee Lockwood <<a href="mailto:michael@lockwood.us.com">michael@lockwood.us.com</a>> wrote:<br><br></div><blockquote type="cite"><div><div dir="ltr"><div class="gmail_default" style="font-family:trebuchet ms,sans-serif">I'm hoping one of the Asterisk gurus out there can help me with this.</div><div class="gmail_default" style="font-family:trebuchet ms,sans-serif"><br></div><div class="gmail_default" style="font-family:trebuchet ms,sans-serif">I have Asterisk running on a private node with app_rpt (DIAL). That node acts as a hub for other private nodes to connect to via IAX. Everything works as it should, with regards to audio being routed to all linked nodes.</div><div class="gmail_default" style="font-family:trebuchet ms,sans-serif"><br></div><div class="gmail_default" style="font-family:trebuchet ms,sans-serif">Here's what I'm trying to accomplish:</div><div class="gmail_default" style="font-family:trebuchet ms,sans-serif"><br></div><div class="gmail_default" style="font-family:trebuchet ms,sans-serif">1. Hub node registers to a remote SIP server (no problem, I have that covered).</div><div class="gmail_default" style="font-family:trebuchet ms,sans-serif">2. Once SIP registration occurs, hub node needs to automatically "dial" a 4 digit extension.</div><div class="gmail_default" style="font-family:trebuchet ms,sans-serif">3. The remote SIP server automatically answers and the audio is routed to a SIP end appliance (no problem, that part is done by remote SIP server)</div><div class="gmail_default" style="font-family:trebuchet ms,sans-serif">4. Once the SIP call on the Hub node is "connected" to the remote SIP server and the call is active, join that SIP call in to the existing linked nodes audio bridge (DAHDI) so that node audio is passed to/from the SIP call. </div><div class="gmail_default" style="font-family:trebuchet ms,sans-serif"><br></div><div class="gmail_default" style="font-family:trebuchet ms,sans-serif">So, first off is this possible? Secondly, who among this group can point me in the direction to accomplish this?</div><div class="gmail_default" style="font-family:trebuchet ms,sans-serif"><br></div><div class="gmail_default" style="font-family:trebuchet ms,sans-serif">Thanks in advance!</div><div class="gmail_signature" data-smartmail="gmail_signature"><div dir="ltr"><div><div dir="ltr"><div dir="ltr"><div dir="ltr"><div><div style="font-size:small"></div></div></div></div></div></div></div></div>
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