[asterisk-users] Issue with transfer (sip)

Olivier oza_4h07 at yahoo.fr
Fri Sep 17 06:02:45 CDT 2010


2010/9/17 Benoit <maverick at maverick.eu.org>

>
> Hi,
>
> I'm experiencing an issue with asterisk 1.6.2.10 & 12rc1,
> i'm not sure if it's to be expected or not, so here it is:
>
> When transferring call (blind-transfer) using asterisk feature key,
> things are working OK, however when using ZoIPer's transfer key
> (which is implemented with a "Refer-To" SIP message) the call is
> ended and the third party isn't even called.
>
> I've made a little bit of research and it seems to be because of a hangup
> handler in the current context. The handler is used for PRI hangup status
> analysis (and maybe mis-placed) and end with a "Hangup(${CAUSE})".
>
> But this handler isn't triggered when using the feature key.
>
> The question is who's fault is it (mine, zoiper's or asterisk's) ?
>

Asterisk's, it seems (see https://issues.asterisk.org/view.php?id=17999 and
related)


>
> thanks
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>               http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100917/4ed2ab73/attachment.htm 


More information about the asterisk-users mailing list