Well,
that is great, but I'm not a good programmer, so I would need some
furher details. Probably I will need to edit the file chan_sip.c and
then recompile Asterisk.
Is it true??
Would you please advice me?
Thanks in advance.
Tomas
Olle E. Johansson napsal(a):
Tomáš Komárek wrote:
Hello,
I've got such a problem. I'm configuring Asterisk as a backup server, if
call to the first one fails.
My problem is, that the redirection from the sending machine work so,
that in the INVITE line of the INVITE message is the presentation number
of the Asterisk server and in the To line is the real called number.
So I need to setup Asterisk so, that it will ignore the number in the
INVITE line and takes care about the To line.
In CVS head you can reach the To: header with the SIP_HEADER function.
/Olle
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