[Asterisk-Users] Sip Trunking

Eduardo Goncalves
Mon, 05 Jan 2004 09:24:53 -0000

Hi list,

	I have to connect two asterisk box, in this scenario:

[asterisk1]----sip----[asterisk2]----PSTN

	I must use sip, cos we'll use cisco rtp header-compression to save
bandwidth.

	Could you tell me the best way to send calls from asterisk1 to
asterisk2, since I cannot use IAX trunking?

Thanks in advance
Eduardo