<div>hello all,</div>
<div> i have a problem, and i'm tearing my hair out...any assistance is appreciated. I am trying to redirect from SER to Asterisk, both on the same machine. In 1.09 I didnt need to set up a peer for SER, just autocreatepeer=yes, and rewritehostport from SER as below, and asterisk accepted the requests without a problem. When I updated to
1.23 requests from SER to asterisk die quietly, no matter how verbose my asterisk is. It's as if the requests dont exist at all.</div>
<div> </div>
<div> My setup is as follows: asterisk and SER on the same box, SER running on 5060 and asterisk on 5070.</div>
<div> All i want is a simple redirect from SER to asterisk, in ser.cfg thusly:</div>
<div> </div>
<div>
<p> if (uri == "<a href="mailto:sip:151@mydomain.com">sip:151@mydomain.com</a>")<br> {<br> log(1, "Forwarding to Voicemail\n");<br> rewritehostport("myIP:5070");
<br> route(1);<br> break;</p>
<p> }<br></p>
<p>and in SIP.conf (this is what i have after some hours of trying, but it doesnt seem to be helping):</p>
<p>bindaddr=myIP</p>
<p>bindport=5070<br>disallow=all ; Disallow all codecs<br>allow=ulaw<br>allow=alaw<br>allow=ilbc<br>allow=gsm<br>dtmfmode=rfc2833<br>autocreatepeer=yes<br>insecure=port,invite</p>
<p>[SER]<br>type=friend<br>host=myIP<br>fromdomain=myDomain<br>context=mycontext<br>canreinvite=no<br>insecure=very<br></p>
<p>if anyone can help i'd me most grateful. I originally thought it would be as simple as changing "port" to "bindport" in sip.conf. Oh, how wrong i was.</p>
<p> </p>
<p>thanks,</p>
<p> yair<br></p>
<p> </p></div>