<div dir="ltr">Any answer? =(<br><br><br></div><div class="gmail_extra"><br><br><div class="gmail_quote">2014-02-06 Alexandr Usov <span dir="ltr"><<a href="mailto:blessendor@gmail.com" target="_blank">blessendor@gmail.com</a>></span>:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Integration - works.<br>Problem - dialing peer to peer via Kamailio OK but with missing VARs and extension number, on dialing/transferring.<br>
Maybe you know other way to configure Asterisk dialplan for users, registered on kamailio and alowing dial as SIP/UserNumber insted SIP/<a href="mailto:UserNumber@kamailio.host.name" target="_blank">UserNumber@kamailio.host.name</a><br>
<div><br><br>What we have.<br><br><br>My Kamailio config:<br><b><a href="http://pastebin.com/p7YxsFaw" target="_blank">http://pastebin.com/p7YxsFaw</a></b><br><br><br></div>Asterisk's user (peer) - registered on kamailio:<br>
<br><div>
<div>[3]<br>host=192.168.144.212<br>qualify=yes<br>dtmfmode=rfc2833<br>canreinvite=no<br>context=local-routing<br>host=dynamic<br>type=friend<br>directmedia=no<br>nat=no<br>qualify=yes<br>disallow=all<br>allow=ulaw<br>allow=alaw<br>
allow=g729<br>call-limit=2<br>limitonpeers=yes<br>callcounter=yes<br>callerid=Usov Mob <3><br><br><br>### Asterisk queue members:<br>;syntax: member => interface,[,penalty][,membername][,state_interface][,ringinuse]<br>
member=SIP/<a href="mailto:1@sip.cloudpbx.com.ua" target="_blank">1@sip.cloudpbx.com.ua</a>,1,1001,SIP/1,no<br>member=SIP/<a href="mailto:3@sip.cloudpbx.com.ua" target="_blank">3@sip.cloudpbx.com.ua</a>,1,1003,SIP/1,no<br>
<br><br></div><div>### Asterisk dialing local peer:<br>
...<br>exten => _X,n,Dial(SIP/${<a href="mailto:EXTEN%7D@sip.cloudpbx.com.ua" target="_blank">EXTEN}@sip.cloudpbx.com.ua</a>,12,tT)<br></div><div>...</div><br><div><br></div><div>### Asterisk attended transfer from 1 to 9 exten:<br>
<b><br>
<a href="http://pastebin.com/k9H4vMgx" target="_blank">http://pastebin.com/k9H4vMgx</a></b><br><br></div>Problem #1: <br>Peer 9 receive clid as <a href="mailto:asterisk@sip.cloudpbx.com.ua" target="_blank">asterisk@sip.cloudpbx.com.ua</a> <br>
<br>Need #1: <br>
<a href="mailto:1@sip.cloudpbx.com.ua" target="_blank">1@sip.cloudpbx.com.ua</a>.<br></div><div><div><br>Problem #2: <br>TRANSFERERNAME=SIP/sip.cloudpbx.com.ua-000000b5 <br><br>Need #2:<br>TRANSFERERNAME=SIP/1@sip.cloudpbx.com.ua-000000b5<br>
</div><div><br><br><br>### Asterisk blind transfer dump log:<br><br><b><a href="http://pastebin.com/NXqXingR" target="_blank">http://pastebin.com/NXqXingR</a></b><br><br></div><div>Problem #3: <br>BLINDTRANSFER=SIP/sip.cloudpbx.com.ua-000000ba<br>
<br>Need #3: <br>BLINDTRANSFER=SIP/9@sip.cloudpbx.com.ua-000000ba <br><br><br></div><div>Main problem it is missed peer number - SIP/<b><u>peer_number</u></b>@<a href="http://sip.cloudpbx.com.ua" target="_blank">sip.cloudpbx.com.ua</a>. And <a href="mailto:asterisk@sip.cloudpbx.com.ua" target="_blank">asterisk@sip.cloudpbx.com.ua</a> instead <a href="mailto:1@sip.cloudpbx.com.ua" target="_blank">1@sip.cloudpbx.com.ua</a> on attended transfer.<br>
</div><div><br>Asterisk sip.conf domain settings:<br>realm=<a href="http://sip.cloudpbx.com.ua" target="_blank">sip.cloudpbx.com.ua</a> <br>fromdomain=<a href="http://sip.cloudpbx.com.ua" target="_blank">sip.cloudpbx.com.ua</a> <br>
</div><div>;domain=<a href="http://sip.cloudpbx.com.ua" target="_blank">sip.cloudpbx.com.ua</a> ;; temporary not uses becaouse not accepting GOIP sim ports registration on asterisk with IP-address of sip proxy instead domain <a href="http://sip.cloudpbx.com.ua" target="_blank">sip.cloudpbx.com.ua</a>; and it's not helps to change <a href="mailto:asterisk@sip.cloudpbx.com.ua" target="_blank">asterisk@sip.cloudpbx.com.ua</a> clid<br>
<br><br></div><div>Any help from all - wellcome!<br></div><div>P.S. domain <a href="http://sip.cloudpbx.com.ua" target="_blank">sip.cloudpbx.com.ua</a> not exist <br></div><div><br></div><div><br></div></div></div>
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