Hi, I am trying to dial a sip user via his IP:PORT Combination. i am using XYZ as target user which is registered.
Asterisk server IP: 70.118.x.x calling user IP: 117.58.x.x called user IP: 117.58.x.x:5062 First I dialed my registered user in normal way like this, Dial(SIP/XYZ,30,rtT) 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, Dial(SIP/x...@117.58.x.x:5062,30,rtT) during conversation audio was one way just like before (calling party can hear called party but called party can not hear calling). 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. *NORMAL DIAL * INVITE sip:x...@xxxxxxxxxxxx:28614;rinstance=0266b8b94f488588 SIP/2.0 To: <sip:x...@xxxxxxxxxxxx:28614;rinstance=0266b8b94f488588> Contact: <sip:1334225...@xxxxxxxxxxx:5060> *IP DIAL* INVITE sip:x...@xxxxxxxxxxx:28614 SIP/2.0 To: <sip:x...@xxxxxxxxxxxx:28614> Contact: <sip:1334225...@xxxxxxxxxxx:5060> Is there something to be done with "rinstance" ?? 1) how can we assign this parameter when dialing via IP:PORT? 2) what else options do we have if we want to dial using IP:PORT mechanism. waiting for your kind response. Nasir Javaid.
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