<div>Hi all,</div>
<div>I am having a strange problem. I am using my asterisk server AST1 to register with another asterisk server AST2 using 2 accounts (2 register commands in sip.conf). I have made 2 local users in AST1, and configured my dialplan in such a way that both local accounts on AST1 use different trunks to send the call to AST2 server. These 2 different trunks are for 2 accounts i have registered on AST1.</div>
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<div>line1 ---> trunk1(ON AST1) ---> AST2</div>
<div>line2 ---> trunk2(ON AST1) ---> AST2</div>
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<div>These 2 trunks are to differentiate that the call is coming from one of the 2 registered accounts on AST1.</div>
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<div>The problem is, my AST2 server cannot differentiate between 2 accounts. It always dumps the cdr at the end of every call against only one of the 2 registered accounts (acc2 even if im dialing from acc1) on AST1 i.e. the call always goes out using account-2 even if i dial from accout-1. Here is my sip.conf</div>
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<div>TRUNKS</div>
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<div>[acc1]<br>username=acc1<br>type=friend<br>secret=123<br>qualify=yes<br>port=9060<br>nat=yes<br>insecure=port,invite<br>host=ip-of-my-AST2<br>dtmfmode=rfc2833<br>canreinvite=no<br>disallow=all<br>allow=ulaw<br>allow=alaw<br>
allow=gsm</div>
<div><br clear="all">[acc2]<br>username=acc2<br>type=friend<br>secret=123<br>qualify=yes<br>port=9060<br>nat=yes<br>insecure=port,invite<br>host=ip-of-my-AST2<br>dtmfmode=rfc2833<br>canreinvite=no<br>disallow=all<br>allow=ulaw<br>
allow=alaw<br>allow=gsm</div>
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<div>REGSITRATION</div>
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<div>register => acc1:123@ip-of-my-AST2:9060<br>register => acc2:123@ip-of-my-AST2:9060</div>
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<div>local lines on AST1 use trunk acc1 and acc2 to throw calls to my AST2.</div>
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<div>But it seems AST2 does not recognise that calls are coming from 2 different accounts.</div>
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<div>How can i make asterisk realize it?</div>
<div><br>-- <br>Best Regards<br>Rizwan Hisham<br>Software Engineer<br>Axvoice Inc.<br><a href="http://www.axvoice.com">www.axvoice.com</a> </div>