I posted a similar message to asterisk-users a few days ago and no-one replied, and this looks like a bug, so I'm both sending another message to asterisk-users and reporting it here.<br><br>In my setup, sip calls coming in through a proxy with a
sip.conf entry
set to "autocreatepeer=yes" and context="proxy" get placed into context
proxy in the dial plan. That is expected.<br><br>However, if the
username in the From: address exists in the sippeers table, it gets
challenged for the password and on success is dropped into context
"default," even if the sip domain is not being served by asterisk.
<br><br>Here is my sip.conf:<br><br>[general]<br>Autocreatepeer = no<br>context=default<br>domain = <a href="http://b.com/" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">b.com</a><br>domain = <a href="http://sip.b.com/" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
sip.b.com</a><br>
realm=<a href="http://b.com/" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">b.com</a><br>bindport=5060<br>bindaddr=<a href="http://4.2.2.2/" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
4.2.2.2</a><br>allow=g729,ulaw,alaw,speex,gsm<br>dtmfmode=rfc2833<br>rtcachefriends=yes<br>;bindaddr=<a href="http://0.0.0.0/" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">0.0.0.0
</a><br>srvlookup=yes<br>rtpkeepalive=1000<br>rtupdate=yes<br>port=5060<br>defaultexpirey=3600<br>tos=0x18<br>insecure=no<br><br>[ser]<br>type=peer<br>context=ser<br>host=<a href="http://4.2.2.3/" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
4.2.2.3</a><br>canreinvite=yes
<br>inseure=very<br>autocreatepeer=yes<br>