Hi,<br><br>I am trying to dial a sip user via his IP:PORT Combination. i am using XYZ as target user which is registered.<br><br>Asterisk server IP: 70.118.x.x<br><br>calling user IP: 117.58.x.x<br><br>called user IP: 117.58.x.x:5062<br>
<br>First I dialed my registered user in normal way like this,<br>
<br>Dial(SIP/XYZ,30,rtT)<br><br>and during conversation audio was OK in
both ways. Then I dialed the registered user via it's ip and port to
which it was registered. like this,<br><br>Dial(SIP/XYZ@117.58.x.x:5062,30,rtT)<br>
<br>during conversation audio was one way just like before (calling
party can hear called party but called party can not hear calling).<br><br>after
taking debug trace of both methods what I found was that a SIP HEADER
parameter "rinstance" was missing in "to" and "INVITE" header fields
when dialing via IP:PORT. I think this parameter is assigned
automatically by asterisk.<br>
<br><u>NORMAL DIAL </u><br>INVITE sip:XYZ@xxxxxxxxxxxx:28614;rinstance=0266b8b94f488588 SIP/2.0<br>To: <sip:XYZ@xxxxxxxxxxxx:28614;rinstance=0266b8b94f488588><br>Contact: <sip:1334225544@xxxxxxxxxxx:5060><br>
<br>
<u>IP DIAL</u><br>INVITE sip:XYZ@xxxxxxxxxxx:28614 SIP/2.0<br>To: <sip:XYZ@xxxxxxxxxxxx:28614><br>
Contact: <sip:1334225544@xxxxxxxxxxx:5060><br>
<br>Is there something to be done with "rinstance" ??<br><br>1) how can we assign this parameter when dialing via IP:PORT?<br>2) what else options do we have if we want to dial using IP:PORT mechanism.<br>
<br> waiting for your kind response.<br><br>Nasir Javaid.