Re: [asterisk-dev] SOCKET

2006-07-12 Thread Marc Haisenko
), the second is a Internet socket for Datagrams, also known as UDP. Russel already explained what these are for. C'ya, Marc -- Marc Haisenko Comdasys AG Rüdesheimer Straße 7 D-80686 München Tel:   +49 (0)89 - 548 43 33 0 Fax:   +49 (0)89 - 548 43 33 29 e-mail: [EMAIL PROTECTED] http

Re: [asterisk-dev] how to export symbols from app_ and chan_ objects?

2006-02-22 Thread Marc Haisenko
On Wednesday 22 February 2006 02:52, Tilghman Lesher wrote: On Tuesday 21 February 2006 04:20, Marc Haisenko wrote: On Monday 20 February 2006 23:43, Tilghman Lesher wrote: Why not simply write your app directly into chan_zap.c and use the corresponding ast_register_application

[asterisk-dev] Setting the context in a SIP channel

2006-01-24 Thread Marc Haisenko
? Why not use the ones from the owning channel ? C'ya, Marc -- Marc Haisenko Comdasys AG Rüdesheimer Straße 7 D-80686 München Tel:   +49 (0)89 - 548 43 33 0 Fax:   +49 (0)89 - 548 43 33 29 e-mail: [EMAIL PROTECTED] http://www.comdasys.com

[Asterisk-Dev] [PATCH] Fix bug in handle_request_info

2006-01-13 Thread Marc Haisenko
Hi folks, I spotted a bug in handle_request_info: in an if condition the code assumes to receive NULL on error, while in fact it receives an empty string. The attached trivial patch fixes this. Patch is done against chan_sip.c from r8023. C'ya, Marc -- Marc Haisenko Comdasys AG

Re: [Asterisk-Dev] [PATCH] Fix bug in handle_request_info

2006-01-13 Thread Marc Haisenko
On Friday 13 January 2006 18:29, BJ Weschke wrote: Patched. Thank you! In the future, please also check out http://bugs.digium.com/ for bug reports and patch posting so we've got a better cyber-papertrail of these types of reports. ACK. C'ya, Marc -- Marc Haisenko Comdasys AG

Re: [Asterisk-Dev] Problem with hanging up a SIP channel

2005-11-25 Thread Marc Haisenko
tests showed ast_softhangup works on the original channel. But thanks for the hint anyway :-) C'ya, Marc -- Marc Haisenko Comdasys AG Rüdesheimer Straße 7 D-80686 München Tel:   +49 (0)89 - 548 43 33 0 Fax:   +49 (0)89 - 548 43 33 29 e-mail: [EMAIL PROTECTED] http://www.comdasys.com

[Asterisk-Dev] Problem with hanging up a SIP channel

2005-11-23 Thread Marc Haisenko
is sent ;-) I guess I probably don't understand how to properly create/handle a channel. Could someone explain how I should handle such a channel ? Or have I hit a bug ? :-) Thanks a lot, Marc -- Marc Haisenko Linux Solutions Be O.K. service group GmbH Rüdesheimer Straße 7 D-80686 München