Hello,<div><br></div><div>Using asterisk 1.6 as sip client to register with sip provider and terminate calls through them. Â SIP Provider has provided sip proxy and sip server details. Â The problem is that the sip server FQDN does not resolve on the internet. Â So I can only presume that the SIP proxy knows how to reach the sip server. Â Asterisk 1.6 seems to have a problem with this. Â This is my config below:</div>
<div><br></div><div>---<---</div><div>[trunk1]</div><div><div>defaultuser=<a href="mailto:xxxxxxxx@sip.provider.com">xxxxxxxx@sip.provider.com</a></div><div>fromuser=xxxxxxxx</div><div>fromdomain=<a href="http://sip.provider.com">sip.provider.com</a></div>
<div>type=peer</div><div>secret=aaaaaaaaa</div><div>outboundproxy=10.10.10.10 ;(replaced actual ip)</div><div>nat=no</div><div>host=<a href="http://sip.provider.com">sip.provider.com</a></div><div>dtmfmode=auto</div><div>
disallow=all</div><div>context=from-internal</div><div>canreinvite=no</div><div>allow=g729</div><div>trustrpid=yes</div><div>sendrpid=yes</div></div><div><br></div><div><br></div><div>register => xxxxxxxx@sip.provider.com:a<a href="http://aaaaaaaa@10.10.10.10:5060">aaaaaaaa@10.10.10.10:5060</a></div>
<div><br></div><div>--->---</div><div><br></div><div>With the above config, I can register with the providers sip proxy, however, the error below is observed in the logs concerning the host when I try to make a call:</div>
<div><br></div><div>--->---</div><div><div>[2012-08-02 23:37:31] WARNING[26155] acl.c: Unable to lookup '<a href="http://sip.provider.com">sip.provider.com</a>'</div><div>[2012-08-02 23:37:31] ERROR[26155] chan_sip.c: srvlookup failed for host: <a href="http://sip.provider.com">sip.provider.com</a>, on peer trunk1, removing peer</div>
</div><div>---<---</div><div><br></div><div>I have done some research on this issue but not been able to find anything conclusive on why this would happen. Â I tested the sip details provided with a different sip client (actually an IP phone) and was able to register and send / receive calls with no problem. Â The problem just seems to be somewhere in my asterisk client configuration or a known bug with the version of asterisk I am using for this.</div>
<div><br></div><div>Any pointers?</div><div><br></div><div>Thanks.</div><div><br></div><div>Joseph</div>