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/XYZ at 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:XYZ at xxxxxxxxxxxx:28614;rinstance=0266b8b94f488588 SIP/2.0
To: <sip:XYZ at xxxxxxxxxxxx:28614;rinstance=0266b8b94f488588>
Contact: <sip:1334225544 at xxxxxxxxxxx:5060>
*IP DIAL*
INVITE sip:XYZ at xxxxxxxxxxx:28614 SIP/2.0
To: <sip:XYZ at xxxxxxxxxxxx:28614>
Contact: <sip:1334225544 at 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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