Hi Users<br>I'm new to Asterisk PBX.<br>Mainly i'm using the openser for call routing and Asterisk as PBX and Voicemail generating.<br><br>let see my secnario ---><br><br>UAC --------------> ser -------->Asterisk(for voice mail only and extension and PBX Purposes....
<br><span style="font-weight: bold;"><span style="font-weight: bold;"> SER system ip is <a href="http://192.168.2.75:5060">192.168.2.75:5060</a><br>Asterisk is in <a href="http://192.168.2.76:5060">192.168.2.76:5060</a><br>
<br>When i start the asterisk server by typing " asterisk -vvvvc "<br><br><span style="color: rgb(255, 0, 0);">[chan_sip.so] => (Session Initiation Protocol (SIP))</span><br style="color: rgb(255, 0, 0);">
<span style="color: rgb(255, 0, 0);"> == Parsing '/etc/asterisk/sip.conf': Found</span><br style="color: rgb(255, 0, 0);"><span style="color: rgb(255, 0, 0);">Aug 30 19:41:05 WARNING[6694]: acl.c:244 ast_get_ip_or_srv: Unable to lookup '
<a href="http://192.168.2.75:5060">192.168.2.75:5060</a>'</span><br style="color: rgb(255, 0, 0);"><span style="color: rgb(255, 0, 0);">Aug 30 19:41:11 WARNING[6694]: acl.c:244 ast_get_ip_or_srv: Unable to lookup '<a href="http://192.168.2.75:5060">
192.168.2.75:5060</a>'</span><br style="color: rgb(255, 0, 0);"><span style="color: rgb(255, 0, 0);"> == SIP Listening on <a href="http://0.0.0.0:5060">0.0.0.0:5060</a></span><br style="color: rgb(255, 0, 0);"><span style="color: rgb(255, 0, 0);">
== Using TOS bits 0</span><br style="color: rgb(255, 0, 0);"><span style="font-weight: bold;"><br>please help in this.....<br><br></span></span></span> <br>-- <br>Thanks and Regards with cheers<br>Sunkara Ravi Prakash (Voip Developer)
<br>Hyperion Technology<br>Kondapur, Hi-tech city,<br>Hyderabad.<br><a href="http://www.hyperion-tech.com">www.hyperion-tech.com</a><br>+91-9985077535