I believe this is what I have, but it still insists on running the transfer from the head office.
Example: Provider --- IAX --- Head Office Provider --- SIP --- Remote Office Provider --- PSTN (Provider is the same * server in all cases) Call comes from PSTN to Head office. Head office transfers to 0xxxx where xxxx is SIP extension according to Provider and 0 is to dial out on the trunk. Call is then connected as follows. PSTN -> Provider -> Head Office -> Provider -> Remote But after it is transferred, I want the resulting route to be: PSTN -> Provider -> Remote Otherwise Head office has 2 times the bandwidth running through it for a call not even going to one of it's own extensions. I had throught that the IAX connection between Provider and Head Office would "pass off" calls that way. Let me know, but thanks for all the help so far. Mike >Instead I'd go for a co-located Asterisk that the remote SIP devices >register with, and then link both * boxes (co-located and central office) >using IAX2 with IAX native transfers enabled. Of course this means that >the office * _only_ talks IAX and that all calls to the remote SIP >clients _always_ go thru the co-located box (with its extra bandwidth). >SER certainly is another way to go (as mentioned before), but in this >specific setup I assume it complicates matters unnecessarily. >Cheers, Philipp -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.8.1 - Release Date: 27/01/2005 _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users