[asterisk-dev] Problem using the sip_header-function

Watkins, Bradley Bradley.Watkins at compuware.com
Fri May 9 08:50:45 CDT 2008


Michael Hirschbichler wrote:
> Hmm, as I am quite new to Asterisk, I do not really understand this 
> timing problem:
> In the extensions.conf, the lines look this way:
> exten => 1226,1,NoOp(First Via: ${SIP_HEADER(VIA,1)} Second Via: 
> ${SIP_HEADER(VIA,2)} Third Via: ${SIP_HEADER(VIA,3)})
> exten => 1226,n,agi,dosomething.pl|${SIP_HEADER(VIA,1)}
> exten => 1226,n,agi,dosomething.pl|${SIP_HEADER(VIA,2)}
> exten => 1226,n,agi,dosomething.pl|${SIP_HEADER(Via,3)}
> so, in line 1, I still have the Vias available, in the 
> following lines, 
> they are gone?

That's pretty much the size of it, yes.

I forget the precise problem, and it's been too long since I perused the
code, so I can't explain it with adequate specificity.  But that is how
it works.

If my approach works with the way your dialplan code is written, I'd say
that's your best bet.  If not, then.... ???  I don't know.

Sorry I can't be of more help.

- Brad

More information about the asterisk-dev mailing list