Handling 302 w/pjsua API

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On 1/8/08, P.J. Cast. <pjcast at yahoo.com> wrote:
>
> I have a question about handling the 302 Moved
> Temporarily Message from an Invite session.
>
> Using the pjsua_call_make_call to place an Invite,
> when the other side is not available, the Sip Proxy is
> sending 302 Moved with the new destination in the
> Contact header field of the Sip message.
>
> Now, with pjsip, I see I can detect this 302 condition
> in the on_call_state callback, when the state is
> PJSIP_INV_STATE_DISCONNECTED and the last_status is
> 302. However, I do I go about continuing the call?


Currently we can only create a new call, using the URI in the Contact header
of 3xx response as the destination of the call. So yes, the Call-ID will
change for the new call. I don't think this is a major issue, CMIIW.


To make this work correctly, I need to be able to get
> another Invite sent out with the new destination (as
> received in the Contact field) using the old CallID
> header field value - since it is part of the same
> transaction as far as I know. I see there is the
> reinvite method in the pjsua API, but it appears just
> to be used for taking a call off hold.
>
> Is there a method I need to call to enable 302
> handling? Or, is there method I can call from the
> callback to continue on with the dialog using the new
> contact... or, do I have to place a completely new
> call with make call method and somehow specify the
> callid from the previous call. I've tried to search
> the mailing list but searching for 302/moved/etc
> didn't yield very good results. And, I couldn't find
> anything in the documentation - if it is there, I'm
> extremely sorry and would love a link to the location
> so I can find future questions :-)


No, this feature is not there, as you can't create a new, initial INVITE
when the previous initial INVITE failed (with pjsip, that is). But do tell
me if you absolutely need this feature, then maybe we can discuss possible
solutions.

cheers,
 -benny


Thanks a bunch.
>
>
>
>
>
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