Update:<div><br></div><div>No luck with versions 1.6 and 1.8.7 I had to revert back to 1.4 which worked with no problem.</div><div><br></div><div>Probably if I have some time, I will do more testing with version 1.8.7 to see what the difference is and what changes need to be made for this kind of setup to work in 1.8.7</div>
<div><br></div><div>Joseph<br><br><div class="gmail_quote">On Mon, Aug 6, 2012 at 10:59 AM, Joseph Begumisa <span dir="ltr"><<a href="mailto:j.begumisa@gmail.com" target="_blank">j.begumisa@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Hello,<div><br></div><div>Using asterisk 1.6 as sip client to register with sip provider and terminate calls through them. SIP Provider has provided sip proxy and sip server details. The problem is that the sip server FQDN does not resolve on the internet. So I can only presume that the SIP proxy knows how to reach the sip server. Asterisk 1.6 seems to have a problem with this. This is my config below:</div>
<div><br></div><div>---<---</div><div>[trunk1]</div><div><div>defaultuser=<a href="mailto:xxxxxxxx@sip.provider.com" target="_blank">xxxxxxxx@sip.provider.com</a></div><div>fromuser=xxxxxxxx</div><div>fromdomain=<a href="http://sip.provider.com" target="_blank">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" target="_blank">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" target="_blank">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" target="_blank">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" target="_blank">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. 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. 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><span class="HOEnZb"><font color="#888888"><div><br></div><div>Joseph</div>
</font></span></blockquote></div><br></div>