[asterisk-users] SIP URI Dial has one way audio

[Nasir Javaid]

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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