Hi JP / All,

Took at look at the website and I know that SIP release 603 Decline is what
I want to use.  

Now:

exten => s, 1, set(SIP_HEADER(XXXXXXXXX) = something)

the xxxx part...need know what to put there...i've only seen a few mentioned
here and there...like "To" and "From".  Which one for the release code?

Andy Jaikissoon

-----Original Message-----
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 
Sent: Thursday, July 13, 2006 6:47 PM
To: [email protected]
Subject: RE: [on-asterisk] SIP Forced Release Codes


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
> 
> 
> 
> 



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