On 13 Jul 2006 at 18:40, Andy Jaikissoon wrote: > > JP...this may just be it! > > One more question though, which parameter in the SIP_HEADER function do > I modify? > > I´ve looked around on the Net and many others seem not to know what > parameters are modifiable there. Any clue?
Andy, here is something from Ciscos site on SIP responses, perhaps they may help http://www.cisco.com/univercd/cc/td/doc/product/voice/sipsols/big gulp/bgsipcom.htm JP > > Andy Jaikissoon > Senior Switch Administrator > 450Tel Communications Inc. > > Tel: (647) 435-2478 x5001 > Tel: (905) 493-0650 > [EMAIL PROTECTED] > > > > > From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] > Sent: Thursday, July 13, 2006 6:24 PM > To: [email protected] > Subject: Re: [on-asterisk] SIP Forced Release Codes > > > > On 13 Jul 2006 at 18:10, Andy Jaikissoon wrote: > > > When releasing a call, is there a way of specifying which release > you > > would send back to a carrier? I know that you can with a PRI > hooked up > > to the Asterisk box but I have yet to find anything mentioning this > for > > SIP trunks. > > > > In particular, when calls are coming into my system and I generate > a > > message saying "Call Rejected", the party that sent me the call is > > seeing that as a completed call. I want to be able to send back the > > "Call Rejected" message and not have it show as a connected call > because > > officially I won´t be connecting to the far end party beyond my > switch. > > ISUP/ISDN Code 21 would be the one that I want to apply. > > There is a 'SIP_HEADER()' command, so you should be able to put > in place logic to determine if you will in fact ANSWER a call, if not > just return the correct SIP response to the remote provider > > > >
