Hi, sorry for joining the discussion so lately. I'd like to ask you to check http://bugs.digium.com/view.php?id=14810. The patch tries to address the issue using channel-variables to propagate the hangup-cause to the calling channel.
Best regards, Marcus On Fri, Jan 23, 2009 at 3:08 PM, Johansson Olle E <[email protected]> wrote: > > 21 jan 2009 kl. 11.49 skrev Klaus Darilion: > > > Hi Olle! > > > > Currently we have the problem that due to > > SIP<->hangupcause<->SIP<->hangupcause.... conversions the original > > hangupcause gets lost in a chain of Asterisk servers using SIP. > > > > In chan_sip there is already code for adding the X-Asterisk-Hangupcode > > header. What about reading this header on the receiving side for > > setting > > the hangupcause instead of doing SIP->hangupcause mapping ? > In this case we could do that, but there has to be an option to enable > it > since it will change the behaviour in existing networks. > > Good idea! > /O > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Dipl.-Inf. (FH) Marcus Hunger - [email protected] Telefon: +49 (0)211-63 55 55-61 Telefax: +49 (0)211-63 55 55-22 sipgate GmbH - Gladbacher Str. 74 - 40219 Düsseldorf HRB Düsseldorf 39841 - Geschäftsführer: Thilo Salmon, Tim Mois Steuernummer: 106 / 5724 / 7147, Umsatzsteuer-ID: DE219349391 www.sipgate.de - www.sipgate.at - www.sipgate.co.uk
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