<div style="font-family:Arial;font-size:14px"><p><span style="font-family: monospace;">Hi,</span></p>
<p><span style="font-family: monospace;">Did order and setup a SIP trunk to a Swedish ITSP named Tele2. No problem to get outgoing calls to work but i have some problems with incoming.</span></p>
<p><span style="font-family: monospace;">Did set "srvlookup=yes" in sip.conf. "Sending" all outgoing calls to "sip-corporate.tele2.se" which is either sip-corporate1.tele2.se (130.244.190.42) or </span><span style="font-family: monospace;">sip-corporate1.tele2.se </span><span style="font-family: monospace;">(130.244.190.46).</span></p>
<p><span style="font-family: monospace;">If i do a "sip show peer Tele2", I see that Asterisk has chosen one of them: ToHost : sip-corporate.tele2.se<br /> Addr->IP : 130.244.190.46 Port 5060<br /></span></p>
<p><span style="font-family: monospace;">Now my problems starts, when Tele2 sends a call to my Asterisk, the call can come frome any of those two ip-adresses. If it comes from </span><span style="font-family: monospace;">130.244.190.46 everything if fine, but if it comes from </span><span style="font-family: monospace;">130.244.190.42: "[Mar 2 08:46:03] NOTICE[1372]: chan_sip.c:19167 handle_request_invite: Failed to authenticate!"</span></p>
<p><span style="font-family: monospace;">I thought </span><span style="font-family: monospace;">"srvlookup=yes" should take care about that, but then i read a little bit more and found: "Note: Asterisk only uses the first host in SRV records". :(</span></p>
<p><span style="font-family: monospace;">Can anyone plz give me some hint howto solve my problem?</span></p>
<p><span style="font-family: monospace;">Regards,</span></p>
<p><span style="font-family: monospace;">Magnus<br /></span></p></div>