Well, I think sipXbridge uses re-invites to handle a call transfer, and what
is likely happening is they are not re-negotiating media "mid-call", which
means calls will break when MOH is activated too. It sounds like they are
not supporting re-invites, which was discussed on the list just yesterday.
In that discussion the itsp would support REFER which means more limited
success could be achieved via an unmanaged gateway (not using sipxbridge),
but there would still be media issues on certain call types (attended
transfer from polycom handset, as an example).

A call trace would tell a lot.

On Thu, Sep 23, 2010 at 2:41 AM, Todd Hodgen <[email protected]> wrote:

>   Sorry for the cross post, trying to work through an issue since changing
> my email address
>
>
>
> I have an ITSP that has a new platform that differs from its legacy
> platform in that it can only handle Invites that have the SDP.  In my
> testing with them, when a call is answered by the auto attendant, and then
> transferred to an extension, it rings and then hangs the call up.  They call
> this 3PCC, essentially they can only handle Invites with SDP.  This new
> platform supports diffserve marking, which is a great reason to be able to
> move to it, but the lack of support for Invites without SDP is an obstacle
> it seems.
>
>
>
> They claim this is perfectly acceptable in the standards.  Can someone
> confirm if that is the case or not?
>
>
>
> Additionally, there were some fixes being created for ill behaving ITSPs,
> does anyone recall if this was one of them.   It tried these trunks on
> 4.2.1, without success.  Or, has anyone found a workaround for this short of
> an external Session Border Controller?
>
>
>
> Any thoughts or ideas would be appreciated.  I’ll keep reading the archives
> in the mean time.
>
> _______________________________________________
> sipx-dev mailing list
> [email protected]
> List Archive: http://list.sipfoundry.org/archive/sipx-dev/
>



-- 
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