<span class="Apple-style-span" style="font-family: tahoma; -webkit-border-horizontal-spacing: 2px; -webkit-border-vertical-spacing: 2px; ">Situatation is that operator <a href="http://mangosip.ru">mangosip.ru</a> got on 1 ip many realms. Problem is that asterisk automatically changes host to ip in To: field. So operator send error back. How to force asterisk not to change host to ip? <br>
Settings: <br>register => A**<a href="mailto:1%3Apass@mangosip.ru">1:pass@mangosip.ru</a> - registry OK. But throw callbackextension not working, trying to register by ip and error.<br>[mangosip] <br>secret = pass <br>
defaultuser = A** <br>trunkname = mangosip <br>callerid = <br>hasexten = no <br>hassip = yes <br>hasiax = no <br>host = <a href="http://mangosip.ru">mangosip.ru</a> <br>context = incoming <br>insecure = invite <br>fromuser = A** <br>
fromdomain = <a href="http://mangosip.ru">mangosip.ru</a> <br>type = peer <br>disallow = all <br>allow = alaw <br>nat = no <br>canreinvite = nonat <br>dtmfmode = info <br>Incoming OK<br>with this settings: <br>> sip show registry <br>
Host dnsmgr Username Refresh State Reg.Time <br><a href="http://mangosip.ru:5060">mangosip.ru:5060</a> N A** 285 Registered Wed, 08 Sep 2010 <br>1 SIP registrations. <br>> sip show peers <br>Name/username Host Dyn Forcerport ACL Port Status <br>
mangosip/A** 81.88.80.36 5060 Unmonitored <br>4 sip peers [Monitored: 1 online, 0 offline Unmonitored: 3 online, 0 offline] <br><br>And dialplan: <br>exten => _7495XXXXXXX,1,Dial(SIP/mangosip/${EXTEN}) <br>exten => _7495XXXXXXX,2,HangUp <br>
Getting this error: <br>Transmitting (no NAT) to <a href="http://81.88.80.36:5060">81.88.80.36:5060</a>: <br>ACK sip:7495**@<a href="http://81.88.80.36">81.88.80.36</a> SIP/2.0 <br>Via: SIP/2.0/UDP 62.**:5060;branch=z9hG4bK117d12fa <br>
Max-Forwards: 70 <br>From: "user1" <sip:A**@<a href="http://mangosip.ru">mangosip.ru</a>>;tag=as7372af08 <br>To: <sip:7495**@<a href="http://81.88.80.36">81.88.80.36</a>>;tag=06239f873a4c6ea5e6ca1d6186a625d8.17e2 <br>
Contact: <sip:A**@62.*:5060> <br>Call-ID: 3**@<a href="http://mangosip.ru">mangosip.ru</a> <br>CSeq: 102 ACK <br>User-Agent: Asterisk PBX SVN-trunk-r285456 <br>Content-Length: 0 <br>... <br><--- SIP read from UDP:<a href="http://81.88.80.36:5060">81.88.80.36:5060</a> ---> <br>
<b>SIP/2.0 416 Unsupported URI Scheme</b> <br>Via: SIP/2.0/UDP 62.*:5060;rport=5060;branch=z9hG4bK1001766a <br>To: <sip:7495**@<a href="http://81.88.80.36">81.88.80.36</a>>;tag=410a7c35 <br>From: "user1"<sip:A**@<a href="http://mangosip.ru">mangosip.ru</a>>;tag=as7d4fb06c <br>
Call-ID: 0**@<a href="http://mangosip.ru">mangosip.ru</a> <br>CSeq: 103 INVITE <br>User-Agent: Softswitch2 <br>Content-Length: 0 <br><br>So 416 error because 7495**@<a href="http://81.88.80.36">81.88.80.36</a> host is resolved to ip. How to change 7495**@<a href="http://81.88.80.36">81.88.80.36</a> to 7495**@<a href="http://mangosip.ru">mangosip.ru</a>?</span>